From cfowler at outpostsentinel.com Fri Feb 2 16:53:15 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:41 2007 Subject: [Aaug] Soft Phone Message-ID: <1170453195.19758.343.camel@shuttle.linxdev.com> I've got my PCI FXO board and will install it and load Trixbox. I want to get it up fast to see how it works. Since I can't wait to order an ATA maybe I could use a softphone. Which ones work well with Linux and Windows. I'll be testing with both platforms. Thanks, Chris From cfowler at outpostsentinel.com Fri Feb 2 17:17:44 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:41 2007 Subject: [Aaug] Soft Phone In-Reply-To: <45C3B3B0.2090009@insightsys.com> References: <1170453195.19758.343.camel@shuttle.linxdev.com> <45C3B3B0.2090009@insightsys.com> Message-ID: <1170454664.19758.349.camel@shuttle.linxdev.com> I like that. Now all I need is a headset and microphone. I found an old microphone. Does Fry's sell a ear piece with mike that has a mic jack and speaker jack? If I do this right I will not have to lug around an ATA (Packet8.net) and analog phone when I travel. On Fri, 2007-02-02 at 16:57 -0500, Scott Plante wrote: > We like x-lite. > http://www.xten.com/index.php?menu=Products&smenu=xlite > > > Christopher Fowler wrote: > > I've got my PCI FXO board and will install it and load Trixbox. I want > > to get it up fast to see how it works. Since I can't wait to order an > > ATA maybe I could use a softphone. Which ones work well with Linux and > > Windows. I'll be testing with both platforms. > > > > Thanks, > > Chris > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > From steve.odom at verso.com Fri Feb 2 17:39:56 2007 From: steve.odom at verso.com (Odom, Steve) Date: Thu May 17 00:41:41 2007 Subject: [Aaug] Soft Phone Message-ID: <9103FBDAC462C140A84247F8AF65D5F706196EB9@ATL01-MXCCL01.verso.com> I use the X-Light sip soft phone, I know others that like the Iax2 soft phone IdeFisk http://www.asteriskguru.com/idefisk/ Check it out, as you will have less NAT problems. Steve -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Friday, February 02, 2007 5:18 PM To: Scott Plante; Aaug@atlaug.com Subject: Re: [Aaug] Soft Phone I like that. Now all I need is a headset and microphone. I found an old microphone. Does Fry's sell a ear piece with mike that has a mic jack and speaker jack? If I do this right I will not have to lug around an ATA (Packet8.net) and analog phone when I travel. On Fri, 2007-02-02 at 16:57 -0500, Scott Plante wrote: > We like x-lite. > http://www.xten.com/index.php?menu=Products&smenu=xlite > > > Christopher Fowler wrote: > > I've got my PCI FXO board and will install it and load Trixbox. I want > > to get it up fast to see how it works. Since I can't wait to order an > > ATA maybe I could use a softphone. Which ones work well with Linux and > > Windows. I'll be testing with both platforms. > > > > Thanks, > > Chris > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From splante at insightsys.com Fri Feb 2 17:47:14 2007 From: splante at insightsys.com (Scott Plante) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Soft Phone In-Reply-To: <1170454664.19758.349.camel@shuttle.linxdev.com> References: <1170453195.19758.343.camel@shuttle.linxdev.com> <45C3B3B0.2090009@insightsys.com> <1170454664.19758.349.camel@shuttle.linxdev.com> Message-ID: <45C3BF72.1090706@insightsys.com> I like the Logitech USB headset w/ mic. I've been meaning to get a Polycom Communicator: http://www.polycom.com/products_services/1,,pw-34-14992-14993,00.html They're supposed to be for Skype, but people say they work just fine with any softphone. There's one button ("call"?) that'll only work with the Skype client. It just look like a usb sound device (speaker & mic) to the computer. We're very happy with the speakerphone on our Polycom IP phones. I've never found a computer mic I've liked that's not attached to my head, so hopefully this would be as good as the phones. Scott Christopher Fowler wrote: > I like that. Now all I need is a headset and microphone. > I found an old microphone. > > Does Fry's sell a ear piece with mike that has a mic jack and speaker > jack? If I do this right I will not have to lug around an ATA > (Packet8.net) and analog phone when I travel. > > > > On Fri, 2007-02-02 at 16:57 -0500, Scott Plante wrote: > >> We like x-lite. >> http://www.xten.com/index.php?menu=Products&smenu=xlite >> >> >> Christopher Fowler wrote: >> >>> I've got my PCI FXO board and will install it and load Trixbox. I want >>> to get it up fast to see how it works. Since I can't wait to order an >>> ATA maybe I could use a softphone. Which ones work well with Linux and >>> Windows. I'll be testing with both platforms. >>> >>> Thanks, >>> Chris >>> >>> _______________________________________________ >>> Aaug mailing list >>> Aaug@atlaug.com >>> http://lists.atlaug.com/mailman/listinfo/aaug >>> >>> >>> > > > -- Scott Plante, CTO Insight Systems, Inc. (+1) 404 873 0058 x104 splante@insightsys.com http://zyross.com From cfowler at outpostsentinel.com Fri Feb 2 18:22:25 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Trixbox Message-ID: <1170458545.19758.351.camel@shuttle.linxdev.com> I've got it installed but no softphone can connect. I've followed the quick start http://buford.linxdev.com/wrong.png http://buford.linxdev.com/wrong2.png Anyone know what I forgot to do? From cfowler at outpostsentinel.com Fri Feb 2 18:23:57 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Trixbox web Message-ID: <1170458637.19758.353.camel@shuttle.linxdev.com> http://buford.linxdev.com:8000/ I'm using the default. Just loaded an hour ago and still no phones working :) From cfowler at outpostsentinel.com Fri Feb 2 19:01:56 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Trixbox web In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F706196EDF@ATL01-MXCCL01.verso.com> References: <9103FBDAC462C140A84247F8AF65D5F706196EDF@ATL01-MXCCL01.verso.com> Message-ID: <1170460916.19758.359.camel@shuttle.linxdev.com> On Fri, 2007-02-02 at 18:39 -0500, Odom, Steve wrote: > Have you set up any extensions? I've got 3 in there now and have gotten the xten phone on Windows to connect. On linux I'm having an issue getting it to connect. Here is the Linux config screen shot http://buford.linxdev.com/linux.png Feb 2 19:00:33 NOTICE[2316] chan_sip.c: Registration from '200 ' failed for '192.168.1.115' - Username/auth name mismatch eb 2 19:07:05 NOTICE[2316] chan_sip.c: Registration from '200 ' failed for '192.168.1.115' - Username/auth name mismatch Feb 2 19:07:20 DEBUG[2316] chan_sip.c: Auto destroying call '055CB5957E6C119EBA24F6858C90B993@192.168.1.102' Feb 2 19:07:27 NOTICE[2316] chan_sip.c: Registration from '200 ' failed for '192.168.1.115' - Username/auth name mismatch From cfowler at outpostsentinel.com Fri Feb 2 19:16:18 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Trixbox web In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F706196EDF@ATL01-MXCCL01.verso.com> References: <9103FBDAC462C140A84247F8AF65D5F706196EDF@ATL01-MXCCL01.verso.com> Message-ID: <1170461778.19758.361.camel@shuttle.linxdev.com> Here are the instructions I followed http://www.trixbox.org/modules/smartsection/item.php?itemid=4 Maybe I did something stupid? On Fri, 2007-02-02 at 18:39 -0500, Odom, Steve wrote: > Have you set up any extensions? > Steve > > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf > Of Christopher Fowler > Sent: Friday, February 02, 2007 6:24 PM > To: Aaug@atlaug.com > Subject: [Aaug] Trixbox web > > http://buford.linxdev.com:8000/ > > I'm using the default. Just loaded an hour ago and still no phones > working :) > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Fri Feb 2 20:31:40 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Trixbox web In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F706196EDF@ATL01-MXCCL01.verso.com> References: <9103FBDAC462C140A84247F8AF65D5F706196EDF@ATL01-MXCCL01.verso.com> Message-ID: <1170466300.19758.366.camel@shuttle.linxdev.com> On Fri, 2007-02-02 at 18:39 -0500, Odom, Steve wrote: > Have you set up any extensions? > Steve I've basically given up so I'm not sure why my soft phones can't do anything. I'm doing this all on a local lan. What I've done now is basically proxy arped this virtual machine into our datacenter since I can't take it there physically. It now has a public IP address of 209.168.246.237 I've not changed the default user/pass for the web interface so if anyone here can get their sip phones working please try. I can't get mine. From cfowler at outpostsentinel.com Fri Feb 2 23:14:19 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider Message-ID: <1170476060.19758.433.camel@shuttle.linxdev.com> http://les.net/products/product_iptermnorth.php There is an idea. Instead of pluging in my Packet8 ATA why not just go with a provider that can allow my PBX to connect via IAX2 or SIP? I pay Packet8 $20/mon. Are there any providers that are good that will allow me to use either an ATA or Asterisk? Then that line would be my real- world line for the PBX. From cfowler at outpostsentinel.com Sat Feb 3 09:38:01 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Bluetooth head set Message-ID: <1170513481.19758.442.camel@shuttle.linxdev.com> I need to get a head set for my desktop and it appears I can not get my mic to work. I know the mic is good because I can use it on other computers. My gues is that either my FC3 installed is fubar or my mic jack is toast. A fresh install of Ubuntu may fix this but it wants me to format / and I can't do that. I can't move me stuff either. One idea I have is to buy a USB -> BT adapter and a head set http://tinyurl.com/2b3bwl Or maybe a USB head set like this. http://tinyurl.com/yud3bm I'm using Linux 99.999999% of the time so it has to work under Linux. From jcf at primeharbor.com Sat Feb 3 10:26:01 2007 From: jcf at primeharbor.com (Chris Farris) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <1170476060.19758.433.camel@shuttle.linxdev.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> Message-ID: <45C4A989.9040702@primeharbor.com> ViaTalk is a BYOD (Bring your own device) provider I'm happy with. Its a bit expensive ($200/yr). Depending on your needs you could look at getting a DID with a free service like SIPPhone. I think I paid $4/mo for a number with them. Both would allow you to register your Asterisk box via SIP to their network. ViaTalk could probably also be configured to allow outbound calling. The SIPPhone solution would only support inbound calling. Chris Christopher Fowler wrote: > http://les.net/products/product_iptermnorth.php > > There is an idea. Instead of pluging in my Packet8 ATA why not just go > with a provider that can allow my PBX to connect via IAX2 or SIP? I pay > Packet8 $20/mon. Are there any providers that are good that will allow > me to use either an ATA or Asterisk? Then that line would be my real- > world line for the PBX. > From cfowler at outpostsentinel.com Sat Feb 3 10:55:42 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <45C4A989.9040702@primeharbor.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4A989.9040702@primeharbor.com> Message-ID: <1170518142.19758.445.camel@shuttle.linxdev.com> Last night around 1am I got voipjet working for outbound. Seems to work great. I would need a DID eventually. I could use voipjet for my outbound and my FXO for a single inbound? For now. On Sat, 2007-02-03 at 10:26 -0500, Chris Farris wrote: > ViaTalk is a BYOD (Bring your own device) provider I'm happy with. Its a > bit expensive ($200/yr). > > Depending on your needs you could look at getting a DID with a free > service like SIPPhone. I think I paid $4/mo for a number with them. > > Both would allow you to register your Asterisk box via SIP to their > network. > > ViaTalk could probably also be configured to allow outbound calling. The > SIPPhone solution would only support inbound calling. > > Chris > > Christopher Fowler wrote: > > http://les.net/products/product_iptermnorth.php > > > > There is an idea. Instead of pluging in my Packet8 ATA why not just go > > with a provider that can allow my PBX to connect via IAX2 or SIP? I pay > > Packet8 $20/mon. Are there any providers that are good that will allow > > me to use either an ATA or Asterisk? Then that line would be my real- > > world line for the PBX. > > From oobx at itmonger.com Sat Feb 3 11:01:57 2007 From: oobx at itmonger.com (Brian Whigham) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <1170476060.19758.433.camel@shuttle.linxdev.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> Message-ID: <45C4B1F5.5030307@itmonger.com> > Are there any providers that are good that will allow >>me to use either an ATA or Asterisk? Then that line would be my real- >>world line for the PBX. Chris, That's the idea behind protocols. Asterisk speaks several, including MGCP, H.323, SIP, IAX, & IAX2. If the provider speaks that protocol, you can connect Asterisk directly to it. There are ATAs that speak IAX2 or SIP (probably requiring a firmware change). And, as you found, you may have a different provider for outbound calls vs. inbound calls. Look for DID to find a direct-inward-dial provider. You'll find a local phone number for as little as $2.92/mo. The lowest price I've found is SIPphone.com (aka Gizmo Project). Conveniently, I've also found them to be one of the most reliable and cheapest termination providers. They're $0.01/min. outbound. Inbound minutes are free. Since I use my cell phone most of the time, I'd only get my money's worth out of Packet8 if I used 1,709 minutes/mo. I've been using my pre-paid $10 to Gizmo for months now. I use other providers in case of failure. I've found Voxee and voipjet to be cheap. But, voipjet recently did something with requiring volume. And Voxee has been unreliable for me. There are also some free DID providers, though voice quality hasn't impressed me with them. Look for < 50ms latency to the provider to avoid excessive delay. Down the road, you might also be interested in having several incoming numbers and termination providers. There's a functionality called least-cost-routing (LCR) which looks up the destination and chooses the cheapest provider dynamically. http://voip-info.org has a page with the cheapest ATAs, DIDs, and termination providers. see how fun VOIP is? Brian From anthony at cosgrove.cc Sat Feb 3 11:04:30 2007 From: anthony at cosgrove.cc (Anthony Cosgrove) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <45C4B1F5.5030307@itmonger.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> Message-ID: <45C4B28E.4090500@cosgrove.cc> There is also voicepulse connect $11/mo for one DID that comes with 4 channels in and 4 channels out. http://connect.voicepulse.com - they are BYOD and LD rates can be .01 once you verify your account. Anthony Itaki Networks Brian Whigham wrote: >> Are there any providers that are good that will allow >> >>> me to use either an ATA or Asterisk? Then that line would be my real- >>> world line for the PBX. >> > > Chris, > > That's the idea behind protocols. Asterisk speaks several, including > MGCP, H.323, SIP, IAX, & IAX2. If the provider speaks that protocol, > you can connect Asterisk directly to it. There are ATAs that speak IAX2 > or SIP (probably requiring a firmware change). > > And, as you found, you may have a different provider for outbound calls > vs. inbound calls. Look for DID to find a direct-inward-dial provider. > You'll find a local phone number for as little as $2.92/mo. The lowest > price I've found is SIPphone.com (aka Gizmo Project). Conveniently, > I've also found them to be one of the most reliable and cheapest > termination providers. They're $0.01/min. outbound. Inbound minutes > are free. Since I use my cell phone most of the time, I'd only get my > money's worth out of Packet8 if I used 1,709 minutes/mo. I've been > using my pre-paid $10 to Gizmo for months now. I use other providers in > case of failure. I've found Voxee and voipjet to be cheap. But, voipjet > recently did something with requiring volume. And Voxee has been > unreliable for me. There are also some free DID providers, though voice > quality hasn't impressed me with them. Look for < 50ms latency to the > provider to avoid excessive delay. > > Down the road, you might also be interested in having several incoming > numbers and termination providers. There's a functionality called > least-cost-routing (LCR) which looks up the destination and chooses the > cheapest provider dynamically. > > http://voip-info.org has a page with the cheapest ATAs, DIDs, and > termination providers. > > see how fun VOIP is? > > Brian > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Sat Feb 3 11:34:01 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <45C4B1F5.5030307@itmonger.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> Message-ID: <1170520442.19758.447.camel@shuttle.linxdev.com> On Sat, 2007-02-03 at 11:01 -0500, Brian Whigham wrote: > > Down the road, you might also be interested in having several incoming > numbers and termination providers. There's a functionality called > least-cost-routing (LCR) which looks up the destination and chooses > the > cheapest provider dynamically. Hopefully in a few months. Maybe SIPphone.com can give me that capability. The PBX will be on 10mpbs pipe to the Internet in Quality Service's Data Center. From oobx at itmonger.com Sat Feb 3 11:40:10 2007 From: oobx at itmonger.com (Brian Whigham) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <1170520472.19758.449.camel@shuttle.linxdev.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> <45C4B28E.4090500@cosgrove.cc> <1170520472.19758.449.camel@shuttle.linxdev.com> Message-ID: <45C4BAEA.707@itmonger.com> Christopher Fowler wrote: >On Sat, 2007-02-03 at 11:04 -0500, Anthony Cosgrove wrote: > > >>BYOD >> where you 'bring your own device'. They don't provide one like Vonage/packet 8 do. You'll buy something like the Handytone 286, the IAXY, or the SPA-3002. Your astersisk box would be considered a "device". bw From oobx at itmonger.com Sat Feb 3 11:41:13 2007 From: oobx at itmonger.com (Brian Whigham) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] reply to group Message-ID: <45C4BB29.4000308@itmonger.com> Anyone else prefer to have replies go to the list? brian From dustin at vecsector.com Sat Feb 3 11:50:37 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <45C4B28E.4090500@cosgrove.cc> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> <45C4B28E.4090500@cosgrove.cc> Message-ID: <45C4BD5D.6020902@vecsector.com> I've used TelIAX for about 2.5 years now as my main only telco for my business. They've always been responsive and reliable - and very good about alerting everyone about maintenance and outages. They also give you 'beta' proxies to test their upcoming servers, so you can address and potential problems. For playing around with outbound calling - go for the cheapest one, but for business-quality uptime, go for reliability - don't get caught up with a $.01/min versus $.02/min provider. Even at 1000 minutes - that's only $10 and having your phones down sure does mean more than $10 to me in the end. ;-) Anyway - my $.02 cents... Dustin Wildes VecSector, LLC Anthony Cosgrove wrote: > There is also voicepulse connect $11/mo for one DID that comes with 4 > channels in and 4 channels out. http://connect.voicepulse.com - they > are BYOD and LD rates can be .01 once you verify your account. > > Anthony > Itaki Networks > > Brian Whigham wrote: > >>> Are there any providers that are good that will allow >>> >>>> me to use either an ATA or Asterisk? Then that line would be my real- >>>> world line for the PBX. >>> >> >> Chris, >> >> That's the idea behind protocols. Asterisk speaks several, including >> MGCP, H.323, SIP, IAX, & IAX2. If the provider speaks that protocol, >> you can connect Asterisk directly to it. There are ATAs that speak IAX2 >> or SIP (probably requiring a firmware change). >> >> And, as you found, you may have a different provider for outbound calls >> vs. inbound calls. Look for DID to find a direct-inward-dial provider. >> You'll find a local phone number for as little as $2.92/mo. The lowest >> price I've found is SIPphone.com (aka Gizmo Project). Conveniently, >> I've also found them to be one of the most reliable and cheapest >> termination providers. They're $0.01/min. outbound. Inbound minutes >> are free. Since I use my cell phone most of the time, I'd only get my >> money's worth out of Packet8 if I used 1,709 minutes/mo. I've been >> using my pre-paid $10 to Gizmo for months now. I use other providers in >> case of failure. I've found Voxee and voipjet to be cheap. But, voipjet >> recently did something with requiring volume. And Voxee has been >> unreliable for me. There are also some free DID providers, though voice >> quality hasn't impressed me with them. Look for < 50ms latency to the >> provider to avoid excessive delay. >> >> Down the road, you might also be interested in having several incoming >> numbers and termination providers. There's a functionality called >> least-cost-routing (LCR) which looks up the destination and chooses the >> cheapest provider dynamically. >> >> http://voip-info.org has a page with the cheapest ATAs, DIDs, and >> termination providers. >> >> see how fun VOIP is? >> >> Brian >> _______________________________________________ >> Aaug mailing list >> Aaug@atlaug.com >> http://lists.atlaug.com/mailman/listinfo/aaug > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From dustin at vecsector.com Sat Feb 3 11:51:33 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] reply to group In-Reply-To: <45C4BB29.4000308@itmonger.com> References: <45C4BB29.4000308@itmonger.com> Message-ID: <45C4BD95.1010308@vecsector.com> I vote 'yes' - we can always delete, but the mailing list is designed to share info, IMO. Brian Whigham wrote: > Anyone else prefer to have replies go to the list? > > brian > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Sat Feb 3 12:14:49 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <45C4BD5D.6020902@vecsector.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> <45C4B28E.4090500@cosgrove.cc> <45C4BD5D.6020902@vecsector.com> Message-ID: <1170522889.19758.452.camel@shuttle.linxdev.com> On Sat, 2007-02-03 at 11:50 -0500, Dustin Wildes wrote: > For playing around with outbound calling - go for the cheapest one, > but > for business-quality uptime, go for reliability - don't get caught up > with a $.01/min versus $.02/min provider. Even at 1000 minutes - > that's > only $10 and having your phones down sure does mean more than $10 to > me > in the end. ;-) > $10 v $20 I do not care about. What I need is one DID with multiple inbounds. So people more than one person can call us at a time. I do not want dead air and I do not want a choppy connection :) I need the quality of a POTs line. Not a satellite phone in the middle of the pacific ocean :) From cfowler at outpostsentinel.com Sat Feb 3 15:26:58 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] reply to group In-Reply-To: <45C4BD95.1010308@vecsector.com> References: <45C4BB29.4000308@itmonger.com> <45C4BD95.1010308@vecsector.com> Message-ID: <1170534418.19758.459.camel@shuttle.linxdev.com> On Sat, 2007-02-03 at 11:51 -0500, Dustin Wildes wrote: > I vote 'yes' - we can always delete, but the mailing list is designed to > share info, IMO. > Exactly. The ALE list I believe was changed a few years back to this. When i hit reply now it goes to the sender and I have to paste in the Aaug address. I may be directing my reply to the sender but usually the group needs to hear it. We are all discussing the same thing. From cfowler at outpostsentinel.com Sat Feb 3 15:32:59 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <45C4BD5D.6020902@vecsector.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> <45C4B28E.4090500@cosgrove.cc> <45C4BD5D.6020902@vecsector.com> Message-ID: <1170534780.19758.466.camel@shuttle.linxdev.com> Since I've got the FXO board and I do have the phone line I think that I am going to install that as a trunk too. I believe I can make Asterisk fail over to that line if my main VOIP trunk (now is voipjet) goes down. I'm going to install another one at a friend's house (Louie) tonight. I told him that all his extension should start at 300 and mine at 200. I think I can create a link between the two PBX systems so that if I dial 300 then his phone rings. He dials 200 then mine rings. I've got a group setup so that ext 2 is all my phones. 3 will be all his. No I plan to set him up a voipjet trunk to play with as well. What would be really cool is if my voipjet trunk failed but his was up then all my outbounds would go via the Internet to hisp lace about 5m away and then out his voip trunk. We're are just playing around to try and learn it. I ordered a Sipura SPA-2000 ATA off eBay. If iy works well when I get this PBX installed in our data center then I may be ordering many more and distribute them among our sales reps. I just need to get inbound DID taken care of and IVR since we have no receptionist. We do use a company that handles all our phone needs including a receptionist and we could do a little bit better if I install some of this technology. They use Meridian Mail and I _hate_ Meridian Mail. On Sat, 2007-02-03 at 11:50 -0500, Dustin Wildes wrote: > I've used TelIAX for about 2.5 years now as my main only telco for my > business. > They've always been responsive and reliable - and very good about > alerting everyone about maintenance and outages. > They also give you 'beta' proxies to test their upcoming servers, so you > can address and potential problems. > > For playing around with outbound calling - go for the cheapest one, but > for business-quality uptime, go for reliability - don't get caught up > with a $.01/min versus $.02/min provider. Even at 1000 minutes - that's > only $10 and having your phones down sure does mean more than $10 to me > in the end. ;-) > > Anyway - my $.02 cents... > > > Dustin Wildes > VecSector, LLC > > > Anthony Cosgrove wrote: > > There is also voicepulse connect $11/mo for one DID that comes with 4 > > channels in and 4 channels out. http://connect.voicepulse.com - they > > are BYOD and LD rates can be .01 once you verify your account. > > > > Anthony > > Itaki Networks > > > > Brian Whigham wrote: > > > >>> Are there any providers that are good that will allow > >>> > >>>> me to use either an ATA or Asterisk? Then that line would be my real- > >>>> world line for the PBX. > >>> > >> > >> Chris, > >> > >> That's the idea behind protocols. Asterisk speaks several, including > >> MGCP, H.323, SIP, IAX, & IAX2. If the provider speaks that protocol, > >> you can connect Asterisk directly to it. There are ATAs that speak IAX2 > >> or SIP (probably requiring a firmware change). > >> > >> And, as you found, you may have a different provider for outbound calls > >> vs. inbound calls. Look for DID to find a direct-inward-dial provider. > >> You'll find a local phone number for as little as $2.92/mo. The lowest > >> price I've found is SIPphone.com (aka Gizmo Project). Conveniently, > >> I've also found them to be one of the most reliable and cheapest > >> termination providers. They're $0.01/min. outbound. Inbound minutes > >> are free. Since I use my cell phone most of the time, I'd only get my > >> money's worth out of Packet8 if I used 1,709 minutes/mo. I've been > >> using my pre-paid $10 to Gizmo for months now. I use other providers in > >> case of failure. I've found Voxee and voipjet to be cheap. But, voipjet > >> recently did something with requiring volume. And Voxee has been > >> unreliable for me. There are also some free DID providers, though voice > >> quality hasn't impressed me with them. Look for < 50ms latency to the > >> provider to avoid excessive delay. > >> > >> Down the road, you might also be interested in having several incoming > >> numbers and termination providers. There's a functionality called > >> least-cost-routing (LCR) which looks up the destination and chooses the > >> cheapest provider dynamically. > >> > >> http://voip-info.org has a page with the cheapest ATAs, DIDs, and > >> termination providers. > >> > >> see how fun VOIP is? > >> > >> Brian > >> _______________________________________________ > >> Aaug mailing list > >> Aaug@atlaug.com > >> http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From dustin at vecsector.com Sat Feb 3 15:42:23 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Failover Dialplan In-Reply-To: <1170534780.19758.466.camel@shuttle.linxdev.com> References: <1170476060.19758.433.camel@shuttle.linxdev.com> <45C4B1F5.5030307@itmonger.com> <45C4B28E.4090500@cosgrove.cc> <45C4BD5D.6020902@vecsector.com> <1170534780.19758.466.camel@shuttle.linxdev.com> Message-ID: <45C4F3AF.2070006@vecsector.com> Hey Christopher - you can definitely do this. I use macros for everything, so here's an example of a failover: [outgoing] exten => _X.,1,Macro(callVoipJet2PSTN) |[macro-callVoipJet2PSTN] exten => s,1,Dial(IAX2/@voipjet/{$MACRO_EXTEN},15,tT) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Goto(s-FAILOVER,1) exten => s-BUSY,1,||Goto(s-FAILOVER,1) ||exten => s-CONGESTION,1,Goto(s-FAILOVER,1) ||exten => s-CHANUNAVAIL,1,Goto(s-FAILOVER,1)| | exten => s-FAILOVER,1,||Dial(Zap/g1/$MACRO_EXTEN},15,tT) Hope it helps! | Christopher Fowler wrote: > Since I've got the FXO board and I do have the phone line I think that I > am going to install that as a trunk too. I believe I can make Asterisk > fail over to that line if my main VOIP trunk (now is voipjet) goes down. > > I'm going to install another one at a friend's house (Louie) tonight. I > told him that all his extension should start at 300 and mine at 200. I > think I can create a link between the two PBX systems so that if I dial > 300 then his phone rings. He dials 200 then mine rings. I've got a > group setup so that ext 2 is all my phones. 3 will be all his. > > No I plan to set him up a voipjet trunk to play with as well. What > would be really cool is if my voipjet trunk failed but his was up then > all my outbounds would go via the Internet to hisp lace about 5m away > and then out his voip trunk. We're are just playing around to try and > learn it. > > I ordered a Sipura SPA-2000 ATA off eBay. If iy works well when I get > this PBX installed in our data center then I may be ordering many more > and distribute them among our sales reps. I just need to get inbound > DID taken care of and IVR since we have no receptionist. We do use a > company that handles all our phone needs including a receptionist and we > could do a little bit better if I install some of this technology. They > use Meridian Mail and I _hate_ Meridian Mail. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.atlaug.com/pipermail/aaug/attachments/20070203/841428c4/attachment.html From steve.odom at verso.com Sat Feb 3 15:52:14 2007 From: steve.odom at verso.com (Odom, Steve) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] reply to group Message-ID: <9103FBDAC462C140A84247F8AF65D5F706196F2A@ATL01-MXCCL01.verso.com> I have a full blown class 5 soft switch in our lab we use for testing purposes. I may be willing to give up some free DID's with multiple inbound call capabilities in return for some free fun Asterisk development work and support. Is anyone interested? Steve -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Saturday, February 03, 2007 3:27 PM To: Dustin Wildes Cc: aaug@atlaug.com Subject: Re: [Aaug] reply to group On Sat, 2007-02-03 at 11:51 -0500, Dustin Wildes wrote: > I vote 'yes' - we can always delete, but the mailing list is designed to > share info, IMO. > Exactly. The ALE list I believe was changed a few years back to this. When i hit reply now it goes to the sender and I have to paste in the Aaug address. I may be directing my reply to the sender but usually the group needs to hear it. We are all discussing the same thing. _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From dustin at vecsector.com Sat Feb 3 15:59:57 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] reply to group In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F706196F2A@ATL01-MXCCL01.verso.com> References: <9103FBDAC462C140A84247F8AF65D5F706196F2A@ATL01-MXCCL01.verso.com> Message-ID: <45C4F7CD.7070309@vecsector.com> Hey Steve! I'm not needing any DIDs, but I'd be willing to lend a hand. Odom, Steve wrote: > I have a full blown class 5 soft switch in our lab we use for testing > purposes. I may be willing to give up some free DID's with multiple > inbound call capabilities in return for some free fun Asterisk > development work and support. Is anyone interested? > Steve > > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf > Of Christopher Fowler > Sent: Saturday, February 03, 2007 3:27 PM > To: Dustin Wildes > Cc: aaug@atlaug.com > Subject: Re: [Aaug] reply to group > > On Sat, 2007-02-03 at 11:51 -0500, Dustin Wildes wrote: > >> I vote 'yes' - we can always delete, but the mailing list is designed >> > to > >> share info, IMO. >> >> > > Exactly. The ALE list I believe was changed a few years back to this. > When i hit reply now it goes to the sender and I have to paste in the > Aaug address. I may be directing my reply to the sender but usually the > group needs to hear it. We are all discussing the same thing. > > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.atlaug.com/pipermail/aaug/attachments/20070203/81dcd6bf/attachment.html From cfowler at outpostsentinel.com Sat Feb 3 16:07:32 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent Message-ID: <1170536852.5909.5.camel@shuttle.linxdev.com> Are there any howtos on setting up an automated attendent? I want to setup a "receptionist" on ext 0 (operator) and have it be the AA. "Dial 200 to call Chris" .... and so forth I see that festival is loaded so maybe I can use that as the speech. I don't feel like using my voice. From dustin at vecsector.com Sat Feb 3 16:13:21 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <1170536852.5909.5.camel@shuttle.linxdev.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> Message-ID: <45C4FAF1.9090603@vecsector.com> Auto Attendants aren't that bad. Here's a sample: [attendant] exten => s,1,Answer() exten => s,2,Wait(2) exten => s,3,Set(count=1) exten => s,4,While($[${count} < 3]) exten => s,5,Set(count=$[${count} + 1]) exten => s,6,Background(autoattendant-greeting) exten => s,7,Set(TIMEOUT(digit)=2) exten => s,8,Set(TIMEOUT(response)=5) exten => s,9,EndWhile exten => s,10,Hangup() exten => i,1,Playback(invalid) exten => i,2,Goto(s,3) exten => t,1,Goto(s,3) exten => 0,1,Goto(extensions,100,1) exten => 2,1,Goto(extensions,102,1) exten => 3,1,Goto(extensions,103,1) exten => 4,1,Goto(extensions,104,1) Christopher Fowler wrote: > Are there any howtos on setting up an automated attendent? I want to > setup a "receptionist" on ext 0 (operator) and have it be the AA. "Dial > 200 to call Chris" .... and so forth > > I see that festival is loaded so maybe I can use that as the speech. I > don't feel like using my voice. > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > From cfowler at outpostsentinel.com Sat Feb 3 16:31:44 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <45C4FAF1.9090603@vecsector.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> <45C4FAF1.9090603@vecsector.com> Message-ID: <1170538305.5909.7.camel@shuttle.linxdev.com> What is starting to confuse me is that it seems I can't do all this in FreePBX? I may need to ditch that GUI and use VI On Sat, 2007-02-03 at 16:13 -0500, Dustin Wildes wrote: > Auto Attendants aren't that bad. > Here's a sample: > > [attendant] > exten => s,1,Answer() > exten => s,2,Wait(2) > exten => s,3,Set(count=1) > exten => s,4,While($[${count} < 3]) > exten => s,5,Set(count=$[${count} + 1]) > exten => s,6,Background(autoattendant-greeting) > exten => s,7,Set(TIMEOUT(digit)=2) > exten => s,8,Set(TIMEOUT(response)=5) > exten => s,9,EndWhile > exten => s,10,Hangup() > > exten => i,1,Playback(invalid) > exten => i,2,Goto(s,3) > exten => t,1,Goto(s,3) > > exten => 0,1,Goto(extensions,100,1) > exten => 2,1,Goto(extensions,102,1) > exten => 3,1,Goto(extensions,103,1) > exten => 4,1,Goto(extensions,104,1) > > > Christopher Fowler wrote: > > Are there any howtos on setting up an automated attendent? I want to > > setup a "receptionist" on ext 0 (operator) and have it be the AA. "Dial > > 200 to call Chris" .... and so forth > > > > I see that festival is loaded so maybe I can use that as the speech. I > > don't feel like using my voice. > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From splante at insightsys.com Sat Feb 3 17:21:23 2007 From: splante at insightsys.com (Scott Plante) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <1170538305.5909.7.camel@shuttle.linxdev.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> <45C4FAF1.9090603@vecsector.com> <1170538305.5909.7.camel@shuttle.linxdev.com> Message-ID: <45C50AE3.9020104@insightsys.com> For FreePBX, you set them up under Setup->IVR. -- Scott Plante, CTO Insight Systems, Inc. (+1) 404 873 0058 x104 splante@insightsys.com http://zyross.com Christopher Fowler wrote: > What is starting to confuse me is that it seems I can't do all this in > FreePBX? > > I may need to ditch that GUI and use VI > > On Sat, 2007-02-03 at 16:13 -0500, Dustin Wildes wrote: > >> Auto Attendants aren't that bad. >> Here's a sample: >> >> [attendant] >> exten => s,1,Answer() >> exten => s,2,Wait(2) >> exten => s,3,Set(count=1) >> exten => s,4,While($[${count} < 3]) >> exten => s,5,Set(count=$[${count} + 1]) >> exten => s,6,Background(autoattendant-greeting) >> exten => s,7,Set(TIMEOUT(digit)=2) >> exten => s,8,Set(TIMEOUT(response)=5) >> exten => s,9,EndWhile >> exten => s,10,Hangup() >> >> exten => i,1,Playback(invalid) >> exten => i,2,Goto(s,3) >> exten => t,1,Goto(s,3) >> >> exten => 0,1,Goto(extensions,100,1) >> exten => 2,1,Goto(extensions,102,1) >> exten => 3,1,Goto(extensions,103,1) >> exten => 4,1,Goto(extensions,104,1) >> >> >> Christopher Fowler wrote: >> >>> Are there any howtos on setting up an automated attendent? I want to >>> setup a "receptionist" on ext 0 (operator) and have it be the AA. "Dial >>> 200 to call Chris" .... and so forth >>> >>> I see that festival is loaded so maybe I can use that as the speech. I >>> don't feel like using my voice. >>> >>> >>> _______________________________________________ >>> Aaug mailing list >>> Aaug@atlaug.com >>> http://lists.atlaug.com/mailman/listinfo/aaug >>> >>> >> _______________________________________________ >> Aaug mailing list >> Aaug@atlaug.com >> http://lists.atlaug.com/mailman/listinfo/aaug >> > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > From cfowler at outpostsentinel.com Sat Feb 3 17:28:28 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <45C50AE3.9020104@insightsys.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> <45C4FAF1.9090603@vecsector.com> <1170538305.5909.7.camel@shuttle.linxdev.com> <45C50AE3.9020104@insightsys.com> Message-ID: <1170541708.5909.14.camel@shuttle.linxdev.com> I've done that but I'm not sure how to test to see if it is working. I've goto one named 'AA' but I can't fine 'AA' in any log file in /etc/asterisk. On Sat, 2007-02-03 at 17:21 -0500, Scott Plante wrote: > For FreePBX, you set them up under Setup->IVR. > From splante at insightsys.com Sat Feb 3 18:50:50 2007 From: splante at insightsys.com (Scott Plante) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <1170541708.5909.14.camel@shuttle.linxdev.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> <45C4FAF1.9090603@vecsector.com> <1170538305.5909.7.camel@shuttle.linxdev.com> <45C50AE3.9020104@insightsys.com> <1170541708.5909.14.camel@shuttle.linxdev.com> Message-ID: <45C51FDA.8030005@insightsys.com> Once you hit the red bar, it should get written out to extensions_additional.conf. It doesn't use your name, though. It'll be named ivr-1, ivr-2, etc. You typically use one as the destination for an inbound route or a timecondition. You can create more that are destinations of the main one. Christopher Fowler wrote: > I've done that but I'm not sure how to test to see if it is working. > I've goto one named 'AA' but I can't fine 'AA' in any log file > in /etc/asterisk. > > On Sat, 2007-02-03 at 17:21 -0500, Scott Plante wrote: > >> For FreePBX, you set them up under Setup->IVR. >> From cfowler at outpostsentinel.com Sat Feb 3 22:19:14 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] SIP/IAX2 Provider In-Reply-To: <009901c74807$44b19c50$2002a8c0@qwerty10> References: <009901c74807$44b19c50$2002a8c0@qwerty10> Message-ID: <1170559154.5909.25.camel@shuttle.linxdev.com> On Sat, 2007-02-03 at 21:50 -0500, Jon M. Crate wrote: > Perhaps I'm on a list with your group that I shouldn't be. > I'm getting a dozen a day talking about stuff I know nothing about. > I've been deleting then thinking they were the latest form of scam. > Would you please remove me? >> Would you be kind enough to stop sending me all this stinkin spam??? Just because something is _strange_ to you does not automatically make it Spam. Got to http://www.atlaug.com to un-subscribe to this list. From cfowler at outpostsentinel.com Sat Feb 3 22:23:51 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <45C51FDA.8030005@insightsys.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> <45C4FAF1.9090603@vecsector.com> <1170538305.5909.7.camel@shuttle.linxdev.com> <45C50AE3.9020104@insightsys.com> <1170541708.5909.14.camel@shuttle.linxdev.com> <45C51FDA.8030005@insightsys.com> Message-ID: <1170559431.5909.31.camel@shuttle.linxdev.com> I got it working. I did not know that I would have difficulties dialing the IVR from inside. Since I've not configured Zap then I had no way of testing. When I configured a PC with an FXO tonight it worked beautifully. We seem to get what we wanted accomplished. We were left with 2 technical issues 1. We uploaded a MP3 for music-on-old. We could barely hear it and is was choppy. Perhaps lowering the bitrate and quality would help since we do not want to push 128k over the connection anyway 2. A "reboot" of the box forced a kernel panic when unloading the Zap driver. This would cause it to hang and require a hard reset. Maybe I need to update kernel and drivers? It is working very nice. I noticed that Louie had no LD on his BellSouth line so we went to voipjet and signed up. This was nice because we was able to do follow-me with his cell phone. I'm going to install my hardware next week and run it to our DC on the fast connection. I will then be able to test it very well... On Sat, 2007-02-03 at 18:50 -0500, Scott Plante wrote: > Once you hit the red bar, it should get written out to > extensions_additional.conf. > It doesn't use your name, though. It'll be named ivr-1, ivr-2, etc. > You typically use one as the destination for an inbound route or a > timecondition. > You can create more that are destinations of the main one. > > Christopher Fowler wrote: > > I've done that but I'm not sure how to test to see if it is working. > > I've goto one named 'AA' but I can't fine 'AA' in any log file > > in /etc/asterisk. > > > > On Sat, 2007-02-03 at 17:21 -0500, Scott Plante wrote: > > > >> For FreePBX, you set them up under Setup->IVR. > >> From steve.odom at verso.com Mon Feb 5 00:30:03 2007 From: steve.odom at verso.com (Odom, Steve) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent Message-ID: <9103FBDAC462C140A84247F8AF65D5F706196F9E@ATL01-MXCCL01.verso.com> Did you run genzaptelconf at the command prompt? Steve -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Saturday, February 03, 2007 10:24 PM To: Scott Plante Cc: aaug@atlaug.com Subject: Re: [Aaug] Automated Attendent I got it working. I did not know that I would have difficulties dialing the IVR from inside. Since I've not configured Zap then I had no way of testing. When I configured a PC with an FXO tonight it worked beautifully. We seem to get what we wanted accomplished. We were left with 2 technical issues 1. We uploaded a MP3 for music-on-old. We could barely hear it and is was choppy. Perhaps lowering the bitrate and quality would help since we do not want to push 128k over the connection anyway 2. A "reboot" of the box forced a kernel panic when unloading the Zap driver. This would cause it to hang and require a hard reset. Maybe I need to update kernel and drivers? It is working very nice. I noticed that Louie had no LD on his BellSouth line so we went to voipjet and signed up. This was nice because we was able to do follow-me with his cell phone. I'm going to install my hardware next week and run it to our DC on the fast connection. I will then be able to test it very well... On Sat, 2007-02-03 at 18:50 -0500, Scott Plante wrote: > Once you hit the red bar, it should get written out to > extensions_additional.conf. > It doesn't use your name, though. It'll be named ivr-1, ivr-2, etc. > You typically use one as the destination for an inbound route or a > timecondition. > You can create more that are destinations of the main one. > > Christopher Fowler wrote: > > I've done that but I'm not sure how to test to see if it is working. > > I've goto one named 'AA' but I can't fine 'AA' in any log file > > in /etc/asterisk. > > > > On Sat, 2007-02-03 at 17:21 -0500, Scott Plante wrote: > > > >> For FreePBX, you set them up under Setup->IVR. > >> _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From splante at insightsys.com Mon Feb 5 09:55:22 2007 From: splante at insightsys.com (Scott Plante) Date: Thu May 17 00:41:42 2007 Subject: [Aaug] Automated Attendent In-Reply-To: <1170559431.5909.31.camel@shuttle.linxdev.com> References: <1170536852.5909.5.camel@shuttle.linxdev.com> <45C4FAF1.9090603@vecsector.com> <1170538305.5909.7.camel@shuttle.linxdev.com> <45C50AE3.9020104@insightsys.com> <1170541708.5909.14.camel@shuttle.linxdev.com> <45C51FDA.8030005@insightsys.com> <1170559431.5909.31.camel@shuttle.linxdev.com> Message-ID: <45C7455A.6030207@insightsys.com> An HTML attachment was scrubbed... URL: http://lists.atlaug.com/pipermail/aaug/attachments/20070205/e16ac52a/attachment.html From cfowler at outpostsentinel.com Mon Feb 5 13:45:55 2007 From: cfowler at outpostsentinel.com (cfowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Trunk Failure Notification Message-ID: <1170701156.5150.7.camel@cfowler-laptop> The system is almost ready to go to be installed. I've got 2 trunks configured. #1. FXO port on X100P (Inbound and Outbound) #2. Voipjet VOIP Trunk (Outbound only) I have unlimited LD on #1 but have to watch #2. Also #1 will be my main line for incoming. If that fails and the system routes to #2 can I be notified via email? For example if someone unplugs the phone line from the FXO port can Asterisk tell me this? Can it tell me if it loses connection to #2? From ACSOLOMO at southernco.com Mon Feb 5 14:00:54 2007 From: ACSOLOMO at southernco.com (Solomon, Arnold C.) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Trunk Failure Notification In-Reply-To: <1170701156.5150.7.camel@cfowler-laptop> References: <1170701156.5150.7.camel@cfowler-laptop> Message-ID: <0A6D0540C39FAC42BBD6EF9685B8728838A3C5@GAXGPEX35.southernco.com> Chris: The command "zap show channel " will show a channel in alarm. In the example below the channel has lost loop current "is in alarm". InAlarm: 1 (0 = false, 1 = true) Hope this helps Arnold ----- >zap show channel 1 Channel: 1x*CLI> File Descriptor: 18 Span: 1 Extension: Dialing: no Context: incoming Caller ID string: Destroy: 0 InAlarm: 1 Signalling Type: FXS Kewlstart Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No Actual Hookstate: Onhook -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of cfowler Sent: Monday, February 05, 2007 1:46 PM To: Aaug@atlaug.com Subject: [Aaug] Trunk Failure Notification The system is almost ready to go to be installed. I've got 2 trunks configured. #1. FXO port on X100P (Inbound and Outbound) #2. Voipjet VOIP Trunk (Outbound only) I have unlimited LD on #1 but have to watch #2. Also #1 will be my main line for incoming. If that fails and the system routes to #2 can I be notified via email? For example if someone unplugs the phone line from the FXO port can Asterisk tell me this? Can it tell me if it loses connection to #2? _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From kris at itaki.net Mon Feb 5 14:48:36 2007 From: kris at itaki.net (Kris) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] reply to group updated on AAUG list Message-ID: <45C78A14.1020100@itaki.net> Where are replies to list messages directed? Poster is /strongly/ recommended for most mailing lists. (Details for *reply_goes_to_list*) Poster This list Explicit address Should reply to list now... I would recommend all persons on the list to create an AAUG folder and rule.. If anyone needs to change their respective AAUG list options please look for the email sent every 1st of the month "atlaug.com mailing list memberships reminder" Thanks Kris Sheets kris @ itaki.net 404.422.5753 From cfowler at outpostsentinel.com Wed Feb 7 22:26:02 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Dialing Rules Message-ID: <1170905163.23826.51.camel@shuttle.linxdev.com> I've moved my Trixbox to our Data Center for testing. Here is my outbound setup 9_outside Dialing Rule: 9|. Trunk 0: Zap/g0 Dialing Rule: 678NXXXXXX, 770NXXXXXX, 404NXXXXXX Trunk 1: Voip Jet Dialing Rule: None The analog line has no LD so I want to confirm that when someone dials a local number like 9,678xxxxxxx that it uses that card When someone dials a LD number like 9,1xxx-xxx-xxxx that is uses the voip trunk with LD. Have I configured my dialing rules correctly. I tried to dial 9,1770- XXX-XXXX, my house phone, and I got a message that it is not necessary to dial a 1 before that number. I don't think that call got routed to the second trunk. I got the Sipura-200 in today and it works great! I decided to go up to the DC tonight instead of waiting till in the morning to install the PC. Luckily I have 24x7 access. From cfowler at outpostsentinel.com Thu Feb 8 14:31:23 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Linux Journal Message-ID: <1170963083.23826.90.camel@shuttle.linxdev.com> I just got my issue of 03/2007 LJ. The main focus is Asterisk. From cfowler at outpostsentinel.com Thu Feb 8 15:30:46 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring Message-ID: <1170966646.23826.101.camel@shuttle.linxdev.com> I've plugged my ATA into a Linksys 802.11B router. The router is plugged into my home network. So to make a call the ATA needs to go out of 2 firewalls. The Linksys and then my Linux routers. When I call the DID line and dial 301 to get my extension I immediately go into voice mail. I get a message that "The person at extension 301 is on the phone". My ATA does not ring. About 60(s) later after I've hung up it rings. When I answer it I only hear a dial tone. Trixbox panel is showing the phone on hook. Maybe dual firewalls is causing an issue? From cfowler at outpostsentinel.com Thu Feb 8 15:47:42 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <1170966646.23826.101.camel@shuttle.linxdev.com> References: <1170966646.23826.101.camel@shuttle.linxdev.com> Message-ID: <1170967662.23826.105.camel@shuttle.linxdev.com> I see this in the PBX log: Feb 8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now UNREACHABLE! Last qualify: 69 On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote: > I've plugged my ATA into a Linksys 802.11B router. The router is > plugged into my home network. So to make a call the ATA needs to go out > of 2 firewalls. The Linksys and then my Linux routers. > > When I call the DID line and dial 301 to get my extension I immediately > go into voice mail. I get a message that "The person at extension 301 > is on the phone". My ATA does not ring. About 60(s) later after I've > hung up it rings. When I answer it I only hear a dial tone. Trixbox > panel is showing the phone on hook. Maybe dual firewalls is causing an > issue? > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From blearning at speakeasy.net Thu Feb 8 16:56:05 2007 From: blearning at speakeasy.net (Bill Learning) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <1170967662.23826.105.camel@shuttle.linxdev.com> Message-ID: <00c801c74bcb$eebac3d0$720aa8c0@billdev06> If you haven't got it yet Set the gateway on the ata to the gateway on the DSL not the Router. It will pass through correctly on the routers route table. Nat has to be on both the * server and the ATA, but don't put a NAT address in. The issue is that it is using the router route table rather then where it should be on the Modem, but really in sip speak your are double natted. BTW I wouldn't implement this on a Large scale, it would be nasty Bill Learning Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like excess." -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Thursday, February 08, 2007 3:48 PM To: ATLANTA * USERS GROUP Subject: Re: [Aaug] Delayed ring I see this in the PBX log: Feb 8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now UNREACHABLE! Last qualify: 69 On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote: > I've plugged my ATA into a Linksys 802.11B router. The router is > plugged into my home network. So to make a call the ATA needs to go out > of 2 firewalls. The Linksys and then my Linux routers. > > When I call the DID line and dial 301 to get my extension I immediately > go into voice mail. I get a message that "The person at extension 301 > is on the phone". My ATA does not ring. About 60(s) later after I've > hung up it rings. When I answer it I only hear a dial tone. Trixbox > panel is showing the phone on hook. Maybe dual firewalls is causing an > issue? > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From blearning at speakeasy.net Thu Feb 8 17:00:43 2007 From: blearning at speakeasy.net (Bill Learning) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] NobleSYS resumes Message-ID: <00c901c74bcc$9766d960$720aa8c0@billdev06> Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 11449 bytes Desc: not available Url : http://lists.atlaug.com/pipermail/aaug/attachments/20070208/17209e9f/attachment.jpe From cfowler at outpostsentinel.com Thu Feb 8 17:11:22 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <00c801c74bcb$eebac3d0$720aa8c0@billdev06> References: <00c801c74bcb$eebac3d0$720aa8c0@billdev06> Message-ID: <1170972682.23826.113.camel@shuttle.linxdev.com> On Thu, 2007-02-08 at 16:56 -0500, Bill Learning wrote: > If you haven't got it yet > > Set the gateway on the ata to the gateway on the DSL not the Router. It > will pass through correctly on the routers route table. Nat has to be > on both the * server and the ATA, but don't put a NAT address in. The > issue is that it is using the router route table rather then where it > should be on the Modem, but really in sip speak your are double natted. > what if I do not know the gateway. Read the paragraph below and you'll see what I'm trying to test. The server has a public address so I've not even considered mucking with sip_nat.conf on the server. I assumed since the server is public and is not behind a firewall then I would not have too. > > BTW I wouldn't implement this on a Large scale, it would be nasty I don't plan on it. I'm testing what I can do compared to services like Vonage. When I travel it may be that I will be double natted. I will not control what I plug into. I get what I can get. When I normally travel with my Packet8 adapter I plug it into eth1 on my laptop. I then run MASQ on my laptop with dhcp, named, etc. Just like If my laptop was a router. In those cases the ATA will be at least double natted. One at the hotel and the second my laptop. > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf > Of Christopher Fowler > Sent: Thursday, February 08, 2007 3:48 PM > To: ATLANTA * USERS GROUP > Subject: Re: [Aaug] Delayed ring > > I see this in the PBX log: > > Feb 8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now UNREACHABLE! > Last qualify: 69 > > > On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote: > > I've plugged my ATA into a Linksys 802.11B router. The router is > > plugged into my home network. So to make a call the ATA needs to go > out > > of 2 firewalls. The Linksys and then my Linux routers. > > > > When I call the DID line and dial 301 to get my extension I > immediately > > go into voice mail. I get a message that "The person at extension 301 > > is on the phone". My ATA does not ring. About 60(s) later after I've > > hung up it rings. When I answer it I only hear a dial tone. Trixbox > > panel is showing the phone on hook. Maybe dual firewalls is causing > an > > issue? > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From blearning at speakeasy.net Thu Feb 8 17:15:20 2007 From: blearning at speakeasy.net (Bill Learning) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <1170972682.23826.113.camel@shuttle.linxdev.com> Message-ID: <00d401c74bce$a2547a60$720aa8c0@billdev06> You have to know it or it wont work *SMILE* if you do know it, it will work. That is what the 301 message is. I think it is page 15 in the manual *LOL* Bill Learning Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like excess." -----Original Message----- From: Christopher Fowler [mailto:cfowler@outpostsentinel.com] Sent: Thursday, February 08, 2007 5:11 PM To: blearning@speakeasy.net; ATLANTA * USERS GROUP Subject: RE: [Aaug] Delayed ring On Thu, 2007-02-08 at 16:56 -0500, Bill Learning wrote: > If you haven't got it yet > > Set the gateway on the ata to the gateway on the DSL not the Router. It > will pass through correctly on the routers route table. Nat has to be > on both the * server and the ATA, but don't put a NAT address in. The > issue is that it is using the router route table rather then where it > should be on the Modem, but really in sip speak your are double natted. > what if I do not know the gateway. Read the paragraph below and you'll see what I'm trying to test. The server has a public address so I've not even considered mucking with sip_nat.conf on the server. I assumed since the server is public and is not behind a firewall then I would not have too. > > BTW I wouldn't implement this on a Large scale, it would be nasty I don't plan on it. I'm testing what I can do compared to services like Vonage. When I travel it may be that I will be double natted. I will not control what I plug into. I get what I can get. When I normally travel with my Packet8 adapter I plug it into eth1 on my laptop. I then run MASQ on my laptop with dhcp, named, etc. Just like If my laptop was a router. In those cases the ATA will be at least double natted. One at the hotel and the second my laptop. > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf > Of Christopher Fowler > Sent: Thursday, February 08, 2007 3:48 PM > To: ATLANTA * USERS GROUP > Subject: Re: [Aaug] Delayed ring > > I see this in the PBX log: > > Feb 8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now UNREACHABLE! > Last qualify: 69 > > > On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote: > > I've plugged my ATA into a Linksys 802.11B router. The router is > > plugged into my home network. So to make a call the ATA needs to go > out > > of 2 firewalls. The Linksys and then my Linux routers. > > > > When I call the DID line and dial 301 to get my extension I > immediately > > go into voice mail. I get a message that "The person at extension 301 > > is on the phone". My ATA does not ring. About 60(s) later after I've > > hung up it rings. When I answer it I only hear a dial tone. Trixbox > > panel is showing the phone on hook. Maybe dual firewalls is causing > an > > issue? > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Thu Feb 8 17:38:40 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <00d401c74bce$a2547a60$720aa8c0@billdev06> References: <00d401c74bce$a2547a60$720aa8c0@billdev06> Message-ID: <1170974320.23826.121.camel@shuttle.linxdev.com> On Thu, 2007-02-08 at 17:15 -0500, Bill Learning wrote: > You have to know it or it wont work *SMILE* if you do know it, it will > work. How so? Is it a requirement of SIP? Here is the device I'm currently using: http://www.voip-info.org/wiki-Packet8+DTA310+and+Asterisk I'm using that device with the original P8 firmware and I pay them for service. I can take that device _anywhere_ and get a line out. I can be double natted and it still works no problem. I do not know how it does this but based on your comments I will assume it is not using SIP? My experience with the device is that the only time it failed me was when the site's firewall was blocking it. I would then fire up my laptop convert it to a router and VPN back into our office. In effect NAT #1 was my laptop and NAT #2 was our firewall at the office. I've never even connected to the DTA310 via a web browser to configure any setting. Possibly this is what someone was talking about earlier about the difference between IAX2 and SIP. Something about IAS2 does not have the NAT issues as SIP. Is this what I'm experiencing? If so would purchasing the Digium IAX2 adapter allow me to use Asterisk in virtually any environment even if I'm 2x natted or even 3x natted? That is what I'm testing. I'm testing the ability to attach the Sipura-2000 to _any_ network regardless of design and connect. In most cases I'll have no clue what that network looks like and if a firewall blocks me I can tunnel back to our office. In that case I'm screwed because I'll be 2x natted and this device will not work unless I specifically configure it for that configuration. Each time I move it I may have to configure it again? > That is what the 301 message is. 301 is the extension I've configured for the ATA > > I think it is page 15 in the manual *LOL* > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > > -----Original Message----- > From: Christopher Fowler [mailto:cfowler@outpostsentinel.com] > Sent: Thursday, February 08, 2007 5:11 PM > To: blearning@speakeasy.net; ATLANTA * USERS GROUP > Subject: RE: [Aaug] Delayed ring > > On Thu, 2007-02-08 at 16:56 -0500, Bill Learning wrote: > > If you haven't got it yet > > > > Set the gateway on the ata to the gateway on the DSL not the Router. > It > > will pass through correctly on the routers route table. Nat has to be > > on both the * server and the ATA, but don't put a NAT address in. The > > issue is that it is using the router route table rather then where it > > should be on the Modem, but really in sip speak your are double > natted. > > > > what if I do not know the gateway. Read the paragraph below and you'll > see what I'm trying to test. > > The server has a public address so I've not even considered mucking with > sip_nat.conf on the server. I assumed since the server is public and is > not behind a firewall then I would not have too. > > > > > BTW I wouldn't implement this on a Large scale, it would be nasty > > > I don't plan on it. I'm testing what I can do compared to services like > Vonage. > When I travel it may be that I will be double natted. I will not > control what I plug into. > I get what I can get. > > When I normally travel with my Packet8 adapter I plug it into eth1 on my > laptop. I then run MASQ on my laptop with dhcp, named, etc. Just like > If my laptop was a router. In those cases the ATA will be at least > double natted. One at the hotel and the second my laptop. > > > > > > > Bill Learning > > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > > excess." > > > > -----Original Message----- > > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On > Behalf > > Of Christopher Fowler > > Sent: Thursday, February 08, 2007 3:48 PM > > To: ATLANTA * USERS GROUP > > Subject: Re: [Aaug] Delayed ring > > > > I see this in the PBX log: > > > > Feb 8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now > UNREACHABLE! > > Last qualify: 69 > > > > > > On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote: > > > I've plugged my ATA into a Linksys 802.11B router. The router is > > > plugged into my home network. So to make a call the ATA needs to go > > out > > > of 2 firewalls. The Linksys and then my Linux routers. > > > > > > When I call the DID line and dial 301 to get my extension I > > immediately > > > go into voice mail. I get a message that "The person at extension > 301 > > > is on the phone". My ATA does not ring. About 60(s) later after > I've > > > hung up it rings. When I answer it I only hear a dial tone. > Trixbox > > > panel is showing the phone on hook. Maybe dual firewalls is causing > > an > > > issue? > > > > > > _______________________________________________ > > > Aaug mailing list > > > Aaug@atlaug.com > > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > From blearning at speakeasy.net Thu Feb 8 18:26:40 2007 From: blearning at speakeasy.net (Bill Learning) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <1170974320.23826.121.camel@shuttle.linxdev.com> Message-ID: <00e101c74bd8$99617a70$720aa8c0@billdev06> Well the pros and cons of one implementation over another is for an Asterisk Beer Meeting *SMILE*. A lot of the devices have auto sensing features in them, the sipura apparently doesn't, ok it really doesn't I was just being nice. Bill Learning Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like excess." -----Original Message----- From: Christopher Fowler [mailto:cfowler@outpostsentinel.com] Sent: Thursday, February 08, 2007 5:39 PM To: blearning@speakeasy.net Cc: 'ATLANTA * USERS GROUP' Subject: RE: [Aaug] Delayed ring On Thu, 2007-02-08 at 17:15 -0500, Bill Learning wrote: > You have to know it or it wont work *SMILE* if you do know it, it will > work. How so? Is it a requirement of SIP? Here is the device I'm currently using: http://www.voip-info.org/wiki-Packet8+DTA310+and+Asterisk I'm using that device with the original P8 firmware and I pay them for service. I can take that device _anywhere_ and get a line out. I can be double natted and it still works no problem. I do not know how it does this but based on your comments I will assume it is not using SIP? My experience with the device is that the only time it failed me was when the site's firewall was blocking it. I would then fire up my laptop convert it to a router and VPN back into our office. In effect NAT #1 was my laptop and NAT #2 was our firewall at the office. I've never even connected to the DTA310 via a web browser to configure any setting. Possibly this is what someone was talking about earlier about the difference between IAX2 and SIP. Something about IAS2 does not have the NAT issues as SIP. Is this what I'm experiencing? If so would purchasing the Digium IAX2 adapter allow me to use Asterisk in virtually any environment even if I'm 2x natted or even 3x natted? That is what I'm testing. I'm testing the ability to attach the Sipura-2000 to _any_ network regardless of design and connect. In most cases I'll have no clue what that network looks like and if a firewall blocks me I can tunnel back to our office. In that case I'm screwed because I'll be 2x natted and this device will not work unless I specifically configure it for that configuration. Each time I move it I may have to configure it again? > That is what the 301 message is. 301 is the extension I've configured for the ATA > > I think it is page 15 in the manual *LOL* > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > > -----Original Message----- > From: Christopher Fowler [mailto:cfowler@outpostsentinel.com] > Sent: Thursday, February 08, 2007 5:11 PM > To: blearning@speakeasy.net; ATLANTA * USERS GROUP > Subject: RE: [Aaug] Delayed ring > > On Thu, 2007-02-08 at 16:56 -0500, Bill Learning wrote: > > If you haven't got it yet > > > > Set the gateway on the ata to the gateway on the DSL not the Router. > It > > will pass through correctly on the routers route table. Nat has to be > > on both the * server and the ATA, but don't put a NAT address in. The > > issue is that it is using the router route table rather then where it > > should be on the Modem, but really in sip speak your are double > natted. > > > > what if I do not know the gateway. Read the paragraph below and you'll > see what I'm trying to test. > > The server has a public address so I've not even considered mucking with > sip_nat.conf on the server. I assumed since the server is public and is > not behind a firewall then I would not have too. > > > > > BTW I wouldn't implement this on a Large scale, it would be nasty > > > I don't plan on it. I'm testing what I can do compared to services like > Vonage. > When I travel it may be that I will be double natted. I will not > control what I plug into. > I get what I can get. > > When I normally travel with my Packet8 adapter I plug it into eth1 on my > laptop. I then run MASQ on my laptop with dhcp, named, etc. Just like > If my laptop was a router. In those cases the ATA will be at least > double natted. One at the hotel and the second my laptop. > > > > > > > Bill Learning > > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > > excess." > > > > -----Original Message----- > > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On > Behalf > > Of Christopher Fowler > > Sent: Thursday, February 08, 2007 3:48 PM > > To: ATLANTA * USERS GROUP > > Subject: Re: [Aaug] Delayed ring > > > > I see this in the PBX log: > > > > Feb 8 15:44:43 NOTICE[2831] chan_sip.c: Peer '301' is now > UNREACHABLE! > > Last qualify: 69 > > > > > > On Thu, 2007-02-08 at 15:30 -0500, Christopher Fowler wrote: > > > I've plugged my ATA into a Linksys 802.11B router. The router is > > > plugged into my home network. So to make a call the ATA needs to go > > out > > > of 2 firewalls. The Linksys and then my Linux routers. > > > > > > When I call the DID line and dial 301 to get my extension I > > immediately > > > go into voice mail. I get a message that "The person at extension > 301 > > > is on the phone". My ATA does not ring. About 60(s) later after > I've > > > hung up it rings. When I answer it I only hear a dial tone. > Trixbox > > > panel is showing the phone on hook. Maybe dual firewalls is causing > > an > > > issue? > > > > > > _______________________________________________ > > > Aaug mailing list > > > Aaug@atlaug.com > > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > From cfowler at outpostsentinel.com Thu Feb 8 18:36:20 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Delayed ring In-Reply-To: <00e101c74bd8$99617a70$720aa8c0@billdev06> References: <00e101c74bd8$99617a70$720aa8c0@billdev06> Message-ID: <1170977780.23826.130.camel@shuttle.linxdev.com> On Thu, 2007-02-08 at 18:26 -0500, Bill Learning wrote: > Well the pros and cons of one implementation over another is for an > Asterisk Beer Meeting *SMILE*. A lot of the devices have auto sensing > features in them, the sipura apparently doesn't, ok it really doesn't > I > was just being nice. I think that now makes more sense to me. What I've been doing today is test and compare so I'll know what I'm up against. I have 2 ATAs. One works and one does not. The one that does is the DTA310. The one that does not is the Sipura-2000. I only found out about this when I wanted to plug them both into my network. I only have one port available on the wired segment. I gate the DTA310 that port. I gave the Linksys port to the Sipura. That is when I realized and you pointed out that I need to make a config change on the Sipura. what struck me as odd was that I've placed the DTA310 in those situations before and been successful. Thankfully you just pointed out this is not a protocol limitation but a device one. In this case I can _not_ use the Sipura-2000 in production. Basically no one works at the office. They all work from home or where ever. We travel so we plan on taking these things with us. They need to work where ever we plug them into. I can't have my sales rep calling me stating that ATA does not work at the hotel he is at and then I realize he is 2x natted. I then do not want to have him call the front desk asking questions like "What is the gateway address". Typically hotel Internet is out-sourced. What I need is for him to plug in anywhere an make calls. The Sipura-2000 will not do that. Maybe my firmware is too old? I did buy it used on eBay. I should look into upgrading it tomorrow. These little quirks are important to know before I go to any type of larger scale deployment. I do not want to end up trying to recoup investments selling stuff used on eBay. I'm buying it used there now just for testing purposes. The Sipura-2000 for me is best used when I _know_ the layout of the network. When I don't I need to trust another device. Anyone here know of one that can handle being natted a few times? From cfowler at outpostsentinel.com Mon Feb 12 22:50:49 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds Message-ID: <1171338649.4841.9.camel@shuttle.linxdev.com> Is there a way to generate nice voices like those that come with asterisk in /var/lib/asterisk/sounds? I saw flite and festival but when using them they don't sound very good. I want to generate the IVR menu using the same voice as the one in the sample sounds. From blearning at speakeasy.net Mon Feb 12 23:16:40 2007 From: blearning at speakeasy.net (Bill Learning) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds In-Reply-To: <1171338649.4841.9.camel@shuttle.linxdev.com> Message-ID: <003201c74f25$c62dc0e0$720aa8c0@billdev06> Allison the IVRvoice.... She is our favorite .. Bill Learning Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like excess." -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Monday, February 12, 2007 10:51 PM To: Aaug@atlaug.com Subject: [Aaug] Sounds Is there a way to generate nice voices like those that come with asterisk in /var/lib/asterisk/sounds? I saw flite and festival but when using them they don't sound very good. I want to generate the IVR menu using the same voice as the one in the sample sounds. _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Mon Feb 12 23:25:36 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds In-Reply-To: <003201c74f25$c62dc0e0$720aa8c0@billdev06> References: <003201c74f25$c62dc0e0$720aa8c0@billdev06> Message-ID: <1171340737.4841.11.camel@shuttle.linxdev.com> http://www.theivrvoice.com/html/index.htm On Mon, 2007-02-12 at 23:16 -0500, Bill Learning wrote: > Allison the IVRvoice.... > > She is our favorite .. > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf > Of Christopher Fowler > Sent: Monday, February 12, 2007 10:51 PM > To: Aaug@atlaug.com > Subject: [Aaug] Sounds > > Is there a way to generate nice voices like those that come with > asterisk in /var/lib/asterisk/sounds? I saw flite and festival but when > using them they don't sound very good. I want to generate the IVR menu > using the same voice as the one in the sample sounds. > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Tue Feb 13 09:47:52 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Accessing VM Message-ID: <1171378072.4841.21.camel@shuttle.linxdev.com> I've ditched Trixbox for now and have began working with astlinux. The PC that I've been using has giving me kernel panics and general protection faults constantly. I got tired of having to drive to the data center to restart the thing. I've got intelligent power and headless operation there but this was a cheap ATX that will not come back on during power loss. So I still have to go there. After the 3rd time I removed it. I've got a embedded type system running astlinux. It has 64mb flash and 128mb ram and is a VIA Nehmiah 1GHZ system. I want to rebuild the configs from scratch to represent what I got working in Trixbox. It is the only way to learn this system. Here is what I've done. 1. Began reading "Asterisk: The Future of Telephony" from Safari. This book is a good beginning but does not get into advanced topics. 2. Removed X100P card from failing system and installed in embedded PC. 3. Worked through the book in #1 and got 75% of my config complete from what I had in Trixbox. I still need to work on IVR, VM, Music-On-Hold, Ring Groups and features What I'm trying to get working now is the VM system. I've created 2 extensions *97 (System Vmail) and *98 (User Vmail) *97 works fine. *98 I can't seem to get working In syslog I get this message: Feb 13 09:37:10 pbx local0.notice asterisk[1500]: NOTICE[1500]: app_voicemail.c:5167 in vm_execmain: Specified user ' "301" <301>' not found (check voicemail.conf and/or realtime config). Falling back to authentication mode. voicemail.conf: 301 => 0000,Chris Fowler,cfowler@opsup.com,,attach=yes|saycid=no| envelope=no|delete=no extensions.conf: ; ************************************************************************** ; * Features * ; ************************************************************************** ; System General VoiceMail system exten => *97,1,Macro(vmsystem) exten => *98,1,Answer() exten => *98,n,NoOp(${EXTEN}) exten => *98,n,NoOp(${CHANNEL}) exten => *98,n,NoOp(${CALLERID}) exten => *98,n,NoOp(${CALLERIDNUM}) exten => *98,n,Macro(vmsystem, s${CALLERID}@default) ; Send user to their VM system ; ************************************************************************** ; * Macros * ; ************************************************************************** [macro-voicemail] exten => s,1,Dial(${ARG1},10,r) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten => _s-.,1,Goto(s-NOANSWER,1) [macro-vmsystem] exten => s,1,Answer() exten => s,n,VoicemailMain(${ARG1}) exten => s,n,Hangup Working through the book has made a _HUGE_ difference. Unfortunately that book is the only book on Safari related to *. I need a recommendation on a better book so that I can read the configs that Trixbox generates and it not seem like Greek to me. Thanks, CF From cfowler at outpostsentinel.com Tue Feb 13 09:54:51 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds In-Reply-To: <003301c74f28$ac99df80$720aa8c0@billdev06> References: <003301c74f28$ac99df80$720aa8c0@billdev06> Message-ID: <1171378492.4841.23.camel@shuttle.linxdev.com> $12 ? 1 to 15 words $24 ? 16 to 30 words $36 ? 31 to 45 words, etc. Very cheap for a professional sounding voice. Much better than my voice! On Mon, 2007-02-12 at 23:37 -0500, Bill Learning wrote: > That is her and very reliable and quick. She has done all the prompts > in asterisk. > > Go to the digium site to get pricing ... I think it is a few bucks for > several words. You email her the script tell her how you want it to > sound and witin a few days you upload and you are in business. It makes > for a very professional sounding system. > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > > -----Original Message----- > From: Christopher Fowler [mailto:cfowler@outpostsentinel.com] > Sent: Monday, February 12, 2007 11:26 PM > To: blearning@speakeasy.net; ATLANTA * USERS GROUP > Subject: RE: [Aaug] Sounds > > http://www.theivrvoice.com/html/index.htm > On Mon, 2007-02-12 at 23:16 -0500, Bill Learning wrote: > > Allison the IVRvoice.... > > > > She is our favorite .. > > > > Bill Learning > > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > > excess." > > -----Original Message----- > > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On > Behalf > > Of Christopher Fowler > > Sent: Monday, February 12, 2007 10:51 PM > > To: Aaug@atlaug.com > > Subject: [Aaug] Sounds > > > > Is there a way to generate nice voices like those that come with > > asterisk in /var/lib/asterisk/sounds? I saw flite and festival but > when > > using them they don't sound very good. I want to generate the IVR > menu > > using the same voice as the one in the sample sounds. > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > From steve.odom at verso.com Tue Feb 13 10:25:54 2007 From: steve.odom at verso.com (Odom, Steve) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds Message-ID: <9103FBDAC462C140A84247F8AF65D5F7062F60EE@ATL01-MXCCL01.verso.com> Are you looking for text to speech? I use Cepstral David. Not perfect, but much better than Festival. Steve -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Tuesday, February 13, 2007 9:55 AM To: blearning@speakeasy.net; Aaug@atlaug.com Subject: RE: [Aaug] Sounds $12 ? 1 to 15 words $24 ? 16 to 30 words $36 ? 31 to 45 words, etc. Very cheap for a professional sounding voice. Much better than my voice! On Mon, 2007-02-12 at 23:37 -0500, Bill Learning wrote: > That is her and very reliable and quick. She has done all the prompts > in asterisk. > > Go to the digium site to get pricing ... I think it is a few bucks for > several words. You email her the script tell her how you want it to > sound and witin a few days you upload and you are in business. It makes > for a very professional sounding system. > > Bill Learning > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > excess." > > -----Original Message----- > From: Christopher Fowler [mailto:cfowler@outpostsentinel.com] > Sent: Monday, February 12, 2007 11:26 PM > To: blearning@speakeasy.net; ATLANTA * USERS GROUP > Subject: RE: [Aaug] Sounds > > http://www.theivrvoice.com/html/index.htm > On Mon, 2007-02-12 at 23:16 -0500, Bill Learning wrote: > > Allison the IVRvoice.... > > > > She is our favorite .. > > > > Bill Learning > > Oscar Wilde said: "Moderation is a fatal thing. Nothing succeeds like > > excess." > > -----Original Message----- > > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On > Behalf > > Of Christopher Fowler > > Sent: Monday, February 12, 2007 10:51 PM > > To: Aaug@atlaug.com > > Subject: [Aaug] Sounds > > > > Is there a way to generate nice voices like those that come with > > asterisk in /var/lib/asterisk/sounds? I saw flite and festival but > when > > using them they don't sound very good. I want to generate the IVR > menu > > using the same voice as the one in the sample sounds. > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Tue Feb 13 10:32:33 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F7062F60EE@ATL01-MXCCL01.verso.com> References: <9103FBDAC462C140A84247F8AF65D5F7062F60EE@ATL01-MXCCL01.verso.com> Message-ID: <1171380754.4841.27.camel@shuttle.linxdev.com> On Tue, 2007-02-13 at 10:25 -0500, Odom, Steve wrote: > Are you looking for text to speech? I use Cepstral David. Not perfect, > but much better than Festival. > Steve In the future yes. I will probably have a need for that since all text will be dynamic. From dustin at vecsector.com Tue Feb 13 10:38:16 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Sounds In-Reply-To: <1171380754.4841.27.camel@shuttle.linxdev.com> References: <9103FBDAC462C140A84247F8AF65D5F7062F60EE@ATL01-MXCCL01.verso.com> <1171380754.4841.27.camel@shuttle.linxdev.com> Message-ID: <45D1DB68.7020206@vecsector.com> Also for speech recognition, you can use LumenVox, if interested Christopher Fowler wrote: > On Tue, 2007-02-13 at 10:25 -0500, Odom, Steve wrote: > >> Are you looking for text to speech? I use Cepstral David. Not perfect, >> but much better than Festival. >> Steve >> > > In the future yes. I will probably have a need for that since all text > will be dynamic. > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.atlaug.com/pipermail/aaug/attachments/20070213/49780246/attachment.html From cfowler at outpostsentinel.com Tue Feb 13 11:12:04 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Tweaking on Sipura Message-ID: <1171383124.4841.39.camel@shuttle.linxdev.com> I've got a 2.4GHZ cordless phone plugged into my Sipura-2000. For some strange reason Asterisk can not "hear" me enter 0000 when I try to enter my VM password. From the X-lite phone it works fine >From Sipura: -- Incorrect password '' for user '301' (context = default) >From X-Lite: -- Incorrect password '000008' for user '301' (context = default) I also notice that when I dial '0' and try to dial 9 [internal] include => outbound-local include => outbound-long-distance exten => 0,1,Answer() exten => 0,n,Background(intro) exten => 9,1,Answer() exten => 9,n,Directory(default,internal) exten => i,1,Playback(invalid) exten => i,n,Goto(internal,0,1) exten => t,1,Goto(internal,0,1) It seems the system hears nothing. It just times out and plays intro again. I just tried a new phone and got same effect. From steve.odom at verso.com Tue Feb 13 11:17:32 2007 From: steve.odom at verso.com (Odom, Steve) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Tweaking on Sipura Message-ID: <9103FBDAC462C140A84247F8AF65D5F7062F6151@ATL01-MXCCL01.verso.com> Try to set the dtmf mode to something else. I use info on my wireless. Steve -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Tuesday, February 13, 2007 11:12 AM To: ATLANTA * USERS GROUP Subject: [Aaug] Tweaking on Sipura I've got a 2.4GHZ cordless phone plugged into my Sipura-2000. For some strange reason Asterisk can not "hear" me enter 0000 when I try to enter my VM password. From the X-lite phone it works fine >From Sipura: -- Incorrect password '' for user '301' (context = default) >From X-Lite: -- Incorrect password '000008' for user '301' (context = default) I also notice that when I dial '0' and try to dial 9 [internal] include => outbound-local include => outbound-long-distance exten => 0,1,Answer() exten => 0,n,Background(intro) exten => 9,1,Answer() exten => 9,n,Directory(default,internal) exten => i,1,Playback(invalid) exten => i,n,Goto(internal,0,1) exten => t,1,Goto(internal,0,1) It seems the system hears nothing. It just times out and plays intro again. I just tried a new phone and got same effect. _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Tue Feb 13 11:35:36 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Tweaking on Sipura In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F7062F6151@ATL01-MXCCL01.verso.com> References: <9103FBDAC462C140A84247F8AF65D5F7062F6151@ATL01-MXCCL01.verso.com> Message-ID: <1171384536.4841.41.camel@shuttle.linxdev.com> I just need to go to Fry's and by a 20port switch. My problem is that I'm all out of switches so I placed that * box on another segment. I had it VPN into the promary segment but only gave it a direct route to my desktop. I had to add a direct route to the ATA too since I introduced it into the network. Works great at default settings now. On Tue, 2007-02-13 at 11:17 -0500, Odom, Steve wrote: > Try to set the dtmf mode to something else. I use info on my wireless. > Steve > > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf > Of Christopher Fowler > Sent: Tuesday, February 13, 2007 11:12 AM > To: ATLANTA * USERS GROUP > Subject: [Aaug] Tweaking on Sipura > > I've got a 2.4GHZ cordless phone plugged into my Sipura-2000. For some > strange reason Asterisk can not "hear" me enter 0000 when I try to enter > my VM password. From the X-lite phone it works fine > > >From Sipura: > -- Incorrect password '' for user '301' (context = default) > > >From X-Lite: > -- Incorrect password '000008' for user '301' (context = default) > > I also notice that when I dial '0' and try to dial 9 > > [internal] > include => outbound-local > include => outbound-long-distance > > exten => 0,1,Answer() exten => 0,n,Background(intro) > > exten => 9,1,Answer() > exten => 9,n,Directory(default,internal) > > exten => i,1,Playback(invalid) > exten => i,n,Goto(internal,0,1) > exten => t,1,Goto(internal,0,1) > > It seems the system hears nothing. It just times out and plays intro > again. I just tried a new phone and got same effect. > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Tue Feb 13 16:21:13 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros Message-ID: <1171401674.12571.11.camel@shuttle.linxdev.com> I'm at the point where I'm playing with the config trying to see what I can do. I'm creating an extension (8) that will give the user 3 chances to enter a valid extension. I thought I'd try it with macros but can't seem to override what I've initially configured for i,t extestions [internal] exten => 8,1,Macro(internalsystemaccess) [macro-internalsystemaccess] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Set(count=1) exten => s,4,While($[${count} < 3]) exten => s,5,Set(count=$[${count} + 1]) exten => s,6,Background(enter-ext-of-person) exten => s,7,Set(TIMEOUT(digit)=2) exten => s,8,Set(TIMEOUT(response)=5) exten => s,9,EndWhile exten => s,10,Hangup() exten => i,1,Playback(invalid) exten => i,2,Goto(s,3) exten => t,1,Goto(s,3)[attendant] exten => s,1,Answer() exten => s,2,Wait(2) exten => s,3,Set(count=1) exten => s,4,While($[${count} < 3]) exten => s,5,Set(count=$[${count} + 1]) exten => s,6,Background(intro) exten => s,7,Set(TIMEOUT(digit)=2) exten => s,8,Set(TIMEOUT(response)=5) exten => s,9,EndWhile exten => s,10,Hangup() exten => 0,1,Goto(incoming,0,1) ; Ring Groups exten => 1,1,Goto(incoming,1,1) exten => 2,1,Goto(incoming,2,1) exten => 3,1,Goto(incoming,3,1) ; Internal Access exten => 8,1,Goto(incoming,8,1) ; Company Directory exten => 9,1,Goto(incoming,9,1) exten => i,1,Playback(invalid) exten => i,2,Goto(s,3) exten => t,1,Goto(s,3) What seems to be happening is that the attendant code is getting executed. Instead of looping back around and playing "enter-ext-of- preson" it is playing "intro" The idea is that I can dial into the FXO from anywhere and hit 8. I then enter a password and I've moved from [incoming] to [internal] contexts. Password is so I'm not paying for LD :) I'm getting tired of typing so I'm starting to experiment what I can do with macros. From john at cohutta.com Tue Feb 13 16:44:42 2007 From: john at cohutta.com (John Mullinix) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros In-Reply-To: <1171401674.12571.11.camel@shuttle.linxdev.com> Message-ID: <20070213214437.NEMK14201.outaamta02.mail.tds.net@NAIROBI> Why don't you do this with DISA. Set the incoming route to match your CID. Enter the password and boom, you are from-internal. John Mullinix www.cohutta.com john@cohutta.com 706-632-3343 -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Tuesday, February 13, 2007 4:21 PM To: Aaug@atlaug.com Subject: [Aaug] Over diring t,i extensions in macros I'm at the point where I'm playing with the config trying to see what I can do. I'm creating an extension (8) that will give the user 3 chances to enter a valid extension. I thought I'd try it with macros but can't seem to override what I've initially configured for i,t extestions [internal] exten => 8,1,Macro(internalsystemaccess) [macro-internalsystemaccess] exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Set(count=1) exten => s,4,While($[${count} < 3]) exten => s,5,Set(count=$[${count} + 1]) exten => s,6,Background(enter-ext-of-person) exten => s,7,Set(TIMEOUT(digit)=2) exten => s,8,Set(TIMEOUT(response)=5) exten => s,9,EndWhile exten => s,10,Hangup() exten => i,1,Playback(invalid) exten => i,2,Goto(s,3) exten => t,1,Goto(s,3)[attendant] exten => s,1,Answer() exten => s,2,Wait(2) exten => s,3,Set(count=1) exten => s,4,While($[${count} < 3]) exten => s,5,Set(count=$[${count} + 1]) exten => s,6,Background(intro) exten => s,7,Set(TIMEOUT(digit)=2) exten => s,8,Set(TIMEOUT(response)=5) exten => s,9,EndWhile exten => s,10,Hangup() exten => 0,1,Goto(incoming,0,1) ; Ring Groups exten => 1,1,Goto(incoming,1,1) exten => 2,1,Goto(incoming,2,1) exten => 3,1,Goto(incoming,3,1) ; Internal Access exten => 8,1,Goto(incoming,8,1) ; Company Directory exten => 9,1,Goto(incoming,9,1) exten => i,1,Playback(invalid) exten => i,2,Goto(s,3) exten => t,1,Goto(s,3) What seems to be happening is that the attendant code is getting executed. Instead of looping back around and playing "enter-ext-of- preson" it is playing "intro" The idea is that I can dial into the FXO from anywhere and hit 8. I then enter a password and I've moved from [incoming] to [internal] contexts. Password is so I'm not paying for LD :) I'm getting tired of typing so I'm starting to experiment what I can do with macros. _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Tue Feb 13 16:49:46 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros In-Reply-To: <20070213214437.NEMK14201.outaamta02.mail.tds.net@NAIROBI> References: <20070213214437.NEMK14201.outaamta02.mail.tds.net@NAIROBI> Message-ID: <1171403386.12571.13.camel@shuttle.linxdev.com> On Tue, 2007-02-13 at 16:44 -0500, John Mullinix wrote: > Why don't you do this with DISA. Set the incoming route to match your > CID. > Enter the password and boom, you are from-internal. I guess I do not know how to do that with DISA From john at cohutta.com Tue Feb 13 16:53:03 2007 From: john at cohutta.com (John Mullinix) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros In-Reply-To: <1171403386.12571.13.camel@shuttle.linxdev.com> Message-ID: <20070213215256.TPNX782.outaamta01.mail.tds.net@NAIROBI> Are you running Trixbox or straight Asterisk? John Mullinix www.cohutta.com john@cohutta.com 706-632-3343 -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Tuesday, February 13, 2007 4:50 PM To: ATLANTA * USERS GROUP Subject: RE: [Aaug] Over diring t,i extensions in macros On Tue, 2007-02-13 at 16:44 -0500, John Mullinix wrote: > Why don't you do this with DISA. Set the incoming route to match your > CID. > Enter the password and boom, you are from-internal. I guess I do not know how to do that with DISA _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Tue Feb 13 16:58:46 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros In-Reply-To: <20070213215256.TPNX782.outaamta01.mail.tds.net@NAIROBI> References: <20070213215256.TPNX782.outaamta01.mail.tds.net@NAIROBI> Message-ID: <1171403926.12571.15.camel@shuttle.linxdev.com> On Tue, 2007-02-13 at 16:53 -0500, John Mullinix wrote: > Are you running Trixbox or straight Asterisk? Asterisk. More specifically astlinux. My configuration program is VI :) From steve.odom at verso.com Tue Feb 13 16:59:02 2007 From: steve.odom at verso.com (Odom, Steve) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros Message-ID: <9103FBDAC462C140A84247F8AF65D5F702F6A965@ATL01-MXCCL01.verso.com> Custom asterisk -------------------------- Sent from my BlackBerry Wireless Handheld ----- Original Message ----- From: aaug-bounces@atlaug.com To: 'ATLANTA * USERS GROUP' Sent: Tue Feb 13 16:53:03 2007 Subject: RE: [Aaug] Over diring t,i extensions in macros Are you running Trixbox or straight Asterisk? John Mullinix www.cohutta.com john@cohutta.com 706-632-3343 -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Tuesday, February 13, 2007 4:50 PM To: ATLANTA * USERS GROUP Subject: RE: [Aaug] Over diring t,i extensions in macros On Tue, 2007-02-13 at 16:44 -0500, John Mullinix wrote: > Why don't you do this with DISA. Set the incoming route to match your > CID. > Enter the password and boom, you are from-internal. I guess I do not know how to do that with DISA _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From john at cohutta.com Tue Feb 13 17:15:06 2007 From: john at cohutta.com (John Mullinix) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Over diring t,i extensions in macros In-Reply-To: <9103FBDAC462C140A84247F8AF65D5F702F6A965@ATL01-MXCCL01.verso.com> Message-ID: <20070213221506.NQSD14201.outaamta02.mail.tds.net@NAIROBI> OK, Here is the printout from voip-info.org on DISA Synopsis DISA (Direct Inward System Access) Description * DISA(passcode[|context]) * DISA(password file) DISA (Direct Inward System Access) Allows someone from outside the telephone switch (PBX) to obtain an "internal" system dialtone and to place calls from it as if they were placing a call from within the switch. A user calls a number that connects to the DISA application and is given dialtone and context. The user enters their passcode, followed by the pound sign (#). If the passcode is correct, the user is then given system dialtone on which a call may be placed. Obviously, this type of access has SERIOUS security implications, and GREAT care must be taken NOT to compromise your security. There is a possibility of accessing DISA without password. Simply exchange your password with no-password. And here is how it is implemented with FreePBX [disa] include => disa-custom exten => 2,1,Set(TIMEOUT(digit)=5) exten => 2,n,Set(TIMEOUT(response)=10) exten => 2,n,Playback(enter-password) exten => 2,n,DISA(/etc/asterisk/disa-2.conf) exten => 2,n(end),Hangup ; end of [disa] The file disa-2.conf has the following line in it: Password|from-internal You could set up a hidden option on your IVR to access it. HTH John Mullinix www.cohutta.com john@cohutta.com 706-632-3343 -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Odom, Steve Sent: Tuesday, February 13, 2007 4:59 PM To: aaug@atlaug.com Subject: Re: [Aaug] Over diring t,i extensions in macros Custom asterisk -------------------------- Sent from my BlackBerry Wireless Handheld ----- Original Message ----- From: aaug-bounces@atlaug.com To: 'ATLANTA * USERS GROUP' Sent: Tue Feb 13 16:53:03 2007 Subject: RE: [Aaug] Over diring t,i extensions in macros Are you running Trixbox or straight Asterisk? John Mullinix www.cohutta.com john@cohutta.com 706-632-3343 -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Tuesday, February 13, 2007 4:50 PM To: ATLANTA * USERS GROUP Subject: RE: [Aaug] Over diring t,i extensions in macros On Tue, 2007-02-13 at 16:44 -0500, John Mullinix wrote: > Why don't you do this with DISA. Set the incoming route to match your > CID. > Enter the password and boom, you are from-internal. I guess I do not know how to do that with DISA _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Wed Feb 14 08:47:03 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Configuing one * to be an ext from another Message-ID: <1171460823.12571.38.camel@shuttle.linxdev.com> I've got my PBX working well. Few tweaks to iron out but I can do that over time. I will install it tomorrow in the rack at the data center. I'm now installing asterisk onto another server. This server has no zaptel hardware. Here is what I want to do 1. On the main PBX create an extension called 1001. When a user dials 1001 it will call the 2nd asterisk box 2. The second asterisk box will answer that call as the [incoming] context and the IVR will be executed. At this point the 2nd asterisk box is a server that I plan on fully automating the IVR via the AGI. It will dial no one. He will only receive SIP calls from the main PBX. Here are my questions: 1. On the main PBX to I simply create and extension for that server just like I did for the soft-phones and ATAs sip.conf: [1001] type=friend secret=1001 qualify=no ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=internal 2. What do I configure in sip.conf on the server? I'm thinking: [1001] type=peer secret=1001 qualify=no ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=incoming Am I on the right track? From cfowler at outpostsentinel.com Wed Feb 14 11:26:32 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] OT: VoiceCon Message-ID: <1171470392.12571.46.camel@shuttle.linxdev.com> Anyone here going to be at VoiceCon Spring in a few weeks? I'll be working our booth so stop by and say hello. From cfowler at outpostsentinel.com Fri Feb 16 16:04:19 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Re: Asterisk book In-Reply-To: <45D619D4.4050507@perlwizard.org> References: <1169138805.26882.28.camel@shuttle.linxdev.com> <45D619D4.4050507@perlwizard.org> Message-ID: <1171659860.11760.91.camel@shuttle.linxdev.com> I'm reading 2 on safari 1. Asterisk: The Future of Telephony 2. VoIP Hacks Both books are ok but I don't really see the point of buying both. If you have safari then check them both out. I started with astlinux using #1 and had a good system after just a couple hours. #2 showed me how to trunk 2 systems together. Right now I'm working on an IVR system that I'm writing in Perl. It interfaces with our trouble ticket system and gives techs options within those tickets over the phone. I was looking for a good book on AGI but I think that it is simply enough that I've nailed it. I'm seeing much repetition in my code so I'm debating about writing a CPAN module called Asterisk::IVR to go with Asterisk::AGI that can read XML and do dynamic IVR with ease. The ability to create a decent IVR system with just an XML file with callbacks for dynamic stuff would be really nice. I'm extremely pleased with what the AGI system has allowed me to do. On Fri, 2007-02-16 at 15:53 -0500, Jason Noble wrote: > Chris, > > Do you have the title of that Oreilly book you were mentioning last > night? You said it was a step by step discussion on getting Asterisk up > and running. > > Thanks, > Jason From cfowler at outpostsentinel.com Fri Feb 16 16:08:46 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out Message-ID: <1171660126.11760.94.camel@shuttle.linxdev.com> I've used AGI to create a dynamic IVR system. Now I want the server to call me. Is that possible? When an alarm occurs I want the server to call me on my cell phone and speak to me. I've got tts working in the IVR system so all I need to figure out is how to make SIP calls. From john at cohutta.com Fri Feb 16 16:54:00 2007 From: john at cohutta.com (John Mullinix) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <1171660126.11760.94.camel@shuttle.linxdev.com> Message-ID: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> If you are wanting to call from a linux script, you might want to look at the command: Asterisk -rx John Mullinix www.cohutta.com john@cohutta.com 706-632-3343 -----Original Message----- From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of Christopher Fowler Sent: Friday, February 16, 2007 4:09 PM To: Aaug@atlaug.com Subject: [Aaug] Calling out I've used AGI to create a dynamic IVR system. Now I want the server to call me. Is that possible? When an alarm occurs I want the server to call me on my cell phone and speak to me. I've got tts working in the IVR system so all I need to figure out is how to make SIP calls. _______________________________________________ Aaug mailing list Aaug@atlaug.com http://lists.atlaug.com/mailman/listinfo/aaug From cfowler at outpostsentinel.com Fri Feb 16 17:25:58 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> Message-ID: <1171664758.11760.97.camel@shuttle.linxdev.com> On Fri, 2007-02-16 at 16:54 -0500, John Mullinix wrote: > If you are wanting to call from a linux script, you might want to look at > the command: > > Asterisk -rx That may actually work. I could use the command 'console dial ' then transfer the caller to a new context where the extension number could be the trouble ticket I want to tell them about. I use AGI to get the exten number and then I'll know exactly what to do. > > John Mullinix > www.cohutta.com > john@cohutta.com > 706-632-3343 > > -----Original Message----- > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of > Christopher Fowler > Sent: Friday, February 16, 2007 4:09 PM > To: Aaug@atlaug.com > Subject: [Aaug] Calling out > > I've used AGI to create a dynamic IVR system. > > Now I want the server to call me. Is that possible? When an alarm > occurs I want the server to call me on my cell phone and speak to me. > I've got tts working in the IVR system so all I need to figure out is > how to make SIP calls. > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From dustin at vecsector.com Fri Feb 16 18:46:45 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <1171664758.11760.97.camel@shuttle.linxdev.com> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> <1171664758.11760.97.camel@shuttle.linxdev.com> Message-ID: <45D64265.6000302@vecsector.com> Easy way is to generate a .call file and put it in the /var/spool/asterisk/outgoing directory. http://svn.digium.com/view/asterisk/trunk/sample.call?rev=13815 Christopher Fowler wrote: > On Fri, 2007-02-16 at 16:54 -0500, John Mullinix wrote: > >> If you are wanting to call from a linux script, you might want to look at >> the command: >> >> Asterisk -rx >> > > That may actually work. > > I could use the command 'console dial ' then > transfer the caller to a new context where the extension number could be > the trouble ticket I want to tell them about. I use AGI to get the > exten number and then I'll know exactly what to do. > > > > >> John Mullinix >> www.cohutta.com >> john@cohutta.com >> 706-632-3343 >> >> -----Original Message----- >> From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of >> Christopher Fowler >> Sent: Friday, February 16, 2007 4:09 PM >> To: Aaug@atlaug.com >> Subject: [Aaug] Calling out >> >> I've used AGI to create a dynamic IVR system. >> >> Now I want the server to call me. Is that possible? When an alarm >> occurs I want the server to call me on my cell phone and speak to me. >> I've got tts working in the IVR system so all I need to figure out is >> how to make SIP calls. >> >> _______________________________________________ >> Aaug mailing list >> Aaug@atlaug.com >> http://lists.atlaug.com/mailman/listinfo/aaug >> >> _______________________________________________ >> Aaug mailing list >> Aaug@atlaug.com >> http://lists.atlaug.com/mailman/listinfo/aaug >> > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.atlaug.com/pipermail/aaug/attachments/20070216/add7ba23/attachment.html From cfowler at outpostsentinel.com Fri Feb 16 19:15:55 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <45D64265.6000302@vecsector.com> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> <1171664758.11760.97.camel@shuttle.linxdev.com> <45D64265.6000302@vecsector.com> Message-ID: <1171671355.11760.101.camel@shuttle.linxdev.com> That is an excellent idea. It should work. I'm seeing some behavior. Here is path of call Asterisk on server #1 calls x303 via IAX2 on the main PBX. The main PBX then dials my cell phone on Zap/1. What I'm seeing is that server 1 seems to execute the goto before I actually answer the cell. It is as if main PBX said yes I answered while it was dialing on Zap/1. The effect is that the IVR application has started asking for a user id and I've not even answered my phone. On Fri, 2007-02-16 at 18:46 -0500, Dustin Wildes wrote: > Easy way is to generate a .call file and put it in > the /var/spool/asterisk/outgoing directory. > http://svn.digium.com/view/asterisk/trunk/sample.call?rev=13815 > > > > Christopher Fowler wrote: > > On Fri, 2007-02-16 at 16:54 -0500, John Mullinix wrote: > > > > > If you are wanting to call from a linux script, you might want to look at > > > the command: > > > > > > Asterisk -rx > > > > > > > That may actually work. > > > > I could use the command 'console dial ' then > > transfer the caller to a new context where the extension number could be > > the trouble ticket I want to tell them about. I use AGI to get the > > exten number and then I'll know exactly what to do. > > > > > > > > > > > John Mullinix > > > www.cohutta.com > > > john@cohutta.com > > > 706-632-3343 > > > > > > -----Original Message----- > > > From: aaug-bounces@atlaug.com [mailto:aaug-bounces@atlaug.com] On Behalf Of > > > Christopher Fowler > > > Sent: Friday, February 16, 2007 4:09 PM > > > To: Aaug@atlaug.com > > > Subject: [Aaug] Calling out > > > > > > I've used AGI to create a dynamic IVR system. > > > > > > Now I want the server to call me. Is that possible? When an alarm > > > occurs I want the server to call me on my cell phone and speak to me. > > > I've got tts working in the IVR system so all I need to figure out is > > > how to make SIP calls. > > > > > > _______________________________________________ > > > Aaug mailing list > > > Aaug@atlaug.com > > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > _______________________________________________ > > > Aaug mailing list > > > Aaug@atlaug.com > > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > > > > _______________________________________________ > > Aaug mailing list > > Aaug@atlaug.com > > http://lists.atlaug.com/mailman/listinfo/aaug > > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug From dustin at vecsector.com Fri Feb 16 19:19:58 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <1171671355.11760.101.camel@shuttle.linxdev.com> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> <1171664758.11760.97.camel@shuttle.linxdev.com> <45D64265.6000302@vecsector.com> <1171671355.11760.101.camel@shuttle.linxdev.com> Message-ID: <45D64A2E.9090503@vecsector.com> You either need to turn on AMD (answer machine detection) or progress indicator for the Zaptel interface. Both of those options detect 'answer' for a call. Christopher Fowler wrote: > That is an excellent idea. It should work. I'm seeing some behavior. > > Here is path of call > > Asterisk on server #1 calls x303 via IAX2 on the main PBX. The main PBX > then dials my cell phone on Zap/1. > > What I'm seeing is that server 1 seems to execute the goto before I > actually answer the cell. It is as if main PBX said yes I answered > while it was dialing on Zap/1. The effect is that the IVR application > has started asking for a user id and I've not even answered my phone. > > > On Fri, 2007-02-16 at 18:46 -0500, Dustin Wildes wrote: > >> Easy way is to generate a .call file and put it in >> the /var/spool/asterisk/outgoing directory. >> http://svn.digium.com/view/asterisk/trunk/sample.call?rev=13815 >> >> >> >> Christopher Fowler wrote: >> >>> On Fri, 2007-02-16 at 16:54 -0500, John Mullinix wrote: >>> >>> >>>> If you are wanting to call from a linux script, you might want to look at >>>> the command: >>>> >>>> Asterisk -rx >>>> >>>> >>> That may actually work. >>> >>> I could use the command 'console dial ' then >>> transfer the caller to a new context where the extension number could be >>> the trouble ticket I want to tell them about. I use AGI to get the >>> exten number and then I'll know exactly what to do. >>> >>> >>> >>> From cfowler at outpostsentinel.com Fri Feb 16 19:28:34 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <45D64A2E.9090503@vecsector.com> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> <1171664758.11760.97.camel@shuttle.linxdev.com> <45D64265.6000302@vecsector.com> <1171671355.11760.101.camel@shuttle.linxdev.com> <45D64A2E.9090503@vecsector.com> Message-ID: <1171672114.11760.103.camel@shuttle.linxdev.com> On Fri, 2007-02-16 at 19:19 -0500, Dustin Wildes wrote: > You either need to turn on AMD (answer machine detection) or progress > indicator for the Zaptel interface. > Both of those options detect 'answer' for a call. Can you elaborate or point me to a link. That must be it because when I go straight out the voipjet trunk It works great! Once I get this solved I can then work on notification via phones. From dustin at vecsector.com Fri Feb 16 20:07:53 2007 From: dustin at vecsector.com (Dustin Wildes) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <1171672114.11760.103.camel@shuttle.linxdev.com> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> <1171664758.11760.97.camel@shuttle.linxdev.com> <45D64265.6000302@vecsector.com> <1171671355.11760.101.camel@shuttle.linxdev.com> <45D64A2E.9090503@vecsector.com> <1171672114.11760.103.camel@shuttle.linxdev.com> Message-ID: <45D65569.4040708@vecsector.com> AMD stuff: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD Zaptel stuff: Just set: /*callprogress=yes */in /etc/asterisk/zapata.conf I'd recommend only using one or the other. In the past, Zaptel 'callprogress' has been less reliable. Christopher Fowler wrote: > On Fri, 2007-02-16 at 19:19 -0500, Dustin Wildes wrote: > >> You either need to turn on AMD (answer machine detection) or progress >> indicator for the Zaptel interface. >> Both of those options detect 'answer' for a call. >> > > Can you elaborate or point me to a link. That must be it because when I > go straight out the voipjet trunk It works great! Once I get this > solved I can then work on notification via phones. > > > _______________________________________________ > Aaug mailing list > Aaug@atlaug.com > http://lists.atlaug.com/mailman/listinfo/aaug > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.atlaug.com/pipermail/aaug/attachments/20070216/5bb4b649/attachment.html From blearning at speakeasy.net Fri Feb 16 20:20:01 2007 From: blearning at speakeasy.net (Bill Learning) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] XO to present theri VOIP products on FEB 27, Message-ID: <017c01c75231$c2915cd0$720aa8c0@billdev06> Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 13826 bytes Desc: not available Url : http://lists.atlaug.com/pipermail/aaug/attachments/20070216/0e455478/attachment.jpe From cfowler at outpostsentinel.com Fri Feb 16 21:12:14 2007 From: cfowler at outpostsentinel.com (Christopher Fowler) Date: Thu May 17 00:41:43 2007 Subject: [Aaug] Calling out In-Reply-To: <45D64265.6000302@vecsector.com> References: <20070216214851.HVXE782.outaamta01.mail.tds.net@NAIROBI> <1171664758.11760.97.camel@shuttle.linxdev.com> <45D64265.6000302@vecsector.com> Message-ID: <1171678334.11760.114.camel@shuttle.linxdev.com> On Fri, 2007-02-16 at 18:46 -0500, Dustin Wildes wrote: > Easy way is to generate a .call file and put it in > the /var/spool/asterisk/outgoing directory. > http://svn.digium.com/view/asterisk/trunk/sample.call?rev=13815 I've got my solution working. Here is how I did it. #1. I created a context specifically for this system in extensions.conf [hacker] exten => _XXXXX,1,Answer() exten => _XXXXX,n,Wait(1) exten => _XXXXX,n,EAGI(/opt/SAM/AGI/notify.agi) exten => _XXXXX,n,Goto(sam,s,1) #2. I have a program that creates a call file. It could look like this: Channel: Zap/1/XXXXXXXX MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: hacker Extension: 00081 Priority: 1 That x number is really the trouble ticket ID. I used that method to tell the notify.agi program what the trouble ticket # is. #3. I wrote notify.agi in PERL. He gets the ticket # from the value of extension and casts to an int to turn 00081 into 81. He tells the user to 'Press 1 for an important message'. He tells user the new ticket #. He terminates. #4 Call is routed to the main IVR application Maybe there is a better solution to passing the ticket # vs what I tried. Now I have the server answering and making calls. From blearning at speakeasy.net Tue Feb 20 22:22:34 2007 From: blearning at speakeas