[asterisk-users] ARA with MySQL or PostgreSQL
Hi everybody! I'm starting to do some test with Asterisk using Realtime Architecture. I would like to know your opinion about using MySQL or PostgreSQL in this schema. Which do you recomend? Are any benefits in any of them? Thanks in advance, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy Load Asterisk Array
Thanks for the reponse to both uf us. I'll be doing this soon, I hope. On Mon, Jul 21, 2008 at 9:25 PM, Jai Rangi [EMAIL PROTECTED] wrote: We also have the similar setup, 2 ser server with heartbeat doing the load balance and 4 asterisk servers handling the media. Of course the data is on MySQL Cluster. Jai Rangi www.bingotelecom.com On Mon, Jul 21, 2008 at 5:13 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote: I have used the OpenSer dispatcher module to load the calls (hash by caller id) to a group of asterisk boxes (In my case, 2 servers). The Asterisk boxes both use ARA and MySQL Master/Master replication. In a case like yours, I think you can use MySQL cluster, and you can still use Dispatcher to balance the load. On Mon, Jul 21, 2008 at 5:22 PM, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Heavy Load Asterisk Array
Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco Unity?
Hi! I've managed to make Asterisk 1.2.26 work with CCM 4.2 using SIP. With H.323 I had some issues. I'm working to integrate Asterisk with Unity (CCM Voicemail) with VPIMv2. To make it I am developing something that make Asterisk VPIM capable. Hope it helps. Greets. On Thu, Feb 28, 2008 at 7:28 PM, Consuelo Vega [EMAIL PROTECTED] wrote: Hello , about this implementacion , i have a issue with ASterisk 1.4.2 and Cisco Unity , the VM doesn't work fine the calls are good but when enter the VM ( cisco Unity ) it didn't work . Somebody has one implementacion ? To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Thu, 28 Feb 2008 20:35:09 + Subject: Re: [asterisk-users] Asterisk and Cisco Unity? Thanks for the info, Dan Peder. It helps me to know the right questions to ask the customer! Cheers Tony In article [EMAIL PROTECTED], Dan Austin [EMAIL PROTECTED] wrote: Tony wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. As Peder mentioned, Unity is only a VM platform. I actually started using Asterisk to replace a Cisco Conferencing package that never worked right. We have had it running internally for three+ years now, and have been very happy with the results. I am currently using chan_ooh323, but SIP is possible if you have CCM 4.2 or higher. You'll also want to run a later release of Asterisk 1.4 which has a work-around for an odd CCM hold implementation. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sigue al minuto las principales noticias de tu ciudad MSN Deportes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX bat phone.
Grandstream HT386 also has that feature. Into the configuration you can find a field called 'Audial Off-hook', there you can set any extension so the ATA will dial as soon as you pick up the handset. On 8/6/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 8/3/07, Michael Munger [EMAIL PROTECTED] wrote: Is there a way to setup an IAX bat phone (immediate=yes) or is this a privilege only reserved for ZAP channels? As I understand it, this would have to be supported by your specific hard/soft phone. It's the same with SIP - taking a handset off-hook doesn't cause any traffic to go to Asterisk. The first packet from the user agent is sent when the phone tries to dial something. Depending on the user agent, this could be as soon as someone presses a single key (so-called early dial with SIP 484 responses), or more typically when an entire number has been dialed and a timeout has occurred or send button has been pressed. Zap FXS ports can tell when a handset has gone off-hook and take some action based on that due to the change in electrical impedance. Some soft-phones support bat-phone operation, though you have to hunt through the docs to get it to work. My Linksys SPA942 desk phone has a dial plan syntax that allows this: (:S0) Which means prefix whatever I type with and match an empty string, dialing as soon as you have a match, which causes the phone to calll as soon as I take it off hook. But it's obviously device-specific, and has nothing to do with SIP or IAX or Asterisk for that matter. When the call arrives at my server, it doesn't look any different than a call to from a phone with a more traditional dialplan. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox 2 and MFC/R2
On 4/23/07, Carlos Chavez [EMAIL PROTECTED] wrote: Can anyone recommend which versions of spandsp, libsupertone, libunicall and libmfcr2 to use to install Unicall on a Trixbox 2.0 machine? [...] Carlos, Use the latest snapshots for Asterisk 1.2 . They are working pretty well. Let us know if you have any issue. Regards. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about bluetooth channel
Iban, For me, it seems to be the codec. Which one are you using? On 3/1/07, Steve Totaro [EMAIL PROTECTED] wrote: Dave Cotton wrote: On Wed, 2007-02-28 at 12:14 -0500, Steve Totaro wrote: Iban Lopetegi Zinkunegi wrote: 28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the next lines: I thought chan_bluetooth only worked with 1.4 head? You thought wrong, he is talking about chan_bluetooth you are talking about chan_cellphone. Yeah, I realized that after I posted. I apologize if I confused anyone more than myself. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Argentine Asterisk Wiki
Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me maint this. I can manage to get a free (perhaps for a limited time) reliable hosting with the benefits of being able to install everything we want (like mediawiki, drupal, tiki-wiki or whatever) with complete access to mysql databases. Please, anyone who is interested in this send me a private e-mail. Best regards! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, VoIP and Linux Blog.
I don't know what's happened, but now is fixed. Sorry. On 1/30/07, Lenz [EMAIL PROTECTED] wrote: That's what I get: The requested URL / was not found on this server :) l. On Mon, 29 Jan 2007 23:16:47 +0100, Facundo Ameal [EMAIL PROTECTED] wrote: Hello everyone! In my humble try of creating a Blog, I've made this: http://fameal.blogdns.org. By now, it's hosted in my own server but shortly it'll be moved to a serious hosting. All post are written in spanish, so it's only for spanish talking people, I will try to make it grow and have english articles. If someone is interested in writing in english (obiously I would help) I can create categories for english talking people. To write a post, the only thing you have to do is register yourself, every article has to be aproved by a moderator, if it's well written, there will be no problem. I hope you like it. Regards. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, VoIP and Linux Blog.
Hello everyone! In my humble try of creating a Blog, I've made this: http://fameal.blogdns.org. By now, it's hosted in my own server but shortly it'll be moved to a serious hosting. All post are written in spanish, so it's only for spanish talking people, I will try to make it grow and have english articles. If someone is interested in writing in english (obiously I would help) I can create categories for english talking people. To write a post, the only thing you have to do is register yourself, every article has to be aproved by a moderator, if it's well written, there will be no problem. I hope you like it. Regards. -- Facundo Ameal. famealatgmaildotcom http://fameal.blogdns.org Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Thanks for the response, I 've already matched codecs. I have no problems with that. Do rxgain and txgain have something to do with R2 protocol errors? Regards. On 1/28/07, Angel Heart [EMAIL PROTECTED] wrote: Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal [EMAIL PROTECTED] wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
I'll try it during weekend then. Thanks for the help. I appreciate it. On 1/29/07, Angel Heart [EMAIL PROTECTED] wrote: Hi, I'm not sure, but I experienced it before with our Nortel Meridian I MFC/R2. When set to both zero(0), calls drop once answered. I tried to vary its values until I finally got it stabled. I'd been in the Datacoms/Telecoms for 16 years now, only with Asterisk I experienced beyond technical theory (out of the book). But bottom line is, it works. Magic ! Angel. *Facundo Ameal [EMAIL PROTECTED]* wrote: Thanks for the response, I 've already matched codecs. I have no problems with that. Do rxgain and txgain have something to do with R2 protocol errors? Regards. On 1/28/07, Angel Heart wrote: Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal [EMAIL PROTECTED] wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva [EMAIL PROTECTED] wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su
[asterisk-users] International Carriers
Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re:sip giving problems, please help.
SIP/15552830438-990b doesn't seem to be a valid channel name, try doing an fsck. On 9/4/06, Ma Zhiyong [EMAIL PROTECTED] wrote: Yes, I also get these problems occasionally Sep 4 17:44:49 WARNING[1365]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:44:49 WARNING[1364]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 60 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 120 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva [EMAIL PROTECTED] wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfer problem
I don't think that the first priority (exten = _44XX,1,Answer) is ok,
have you tried without it?
Try not answering and post what happens.
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't work. I have a traditional PBX connected with a zap channel
to Asterisk in the following way:
IAX/SIP client -- Asterisk (FXO) -- (FXS) traditional PBX --- OFFICE
Phones
Asterisk is connected to the PBX with an internal number configured inside it.
In other words i keep an internal line an i connect it to an fxo port of
asterisk while asterisk is connected to internet and from here comes iax calls
to talk with other numbers in the office connected to the traditional PBX.
Well, calls to a SIP clients defined in asterisk works fine, but calls to Zap
clients doesn't work. In the most basic form I do:
exten = _44XX,1,Answer
exten = _44XX,n,Dial(Zap/g1/${EXTEN:2}|20|tTr)
exten = _44XX,n,Hangup
and the console logs for this are:
Executing Dial(SIP/sipuser-081d13f0, Zap/g1/38|20|tTr) in new stack
-- Called g1/38
-- Zap/1-1 answered SIP/ggonzalez-081d13f0
Here Dial cmd do one ring and nothing more, Zap channel has answered but the
number dialed never RING, what is wrong? what i have to do get this working
fine?. Thanks for any help
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--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
Share your knowledge, use free software.
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Re: [asterisk-users] MFC/R2 problems
These are the different meanings for the diferrent error codes: T1 TIMEOUT = 32769 T2 TIMEOUT = 32770 T3 TIMEOUT = 32771 UNEXPECTED MF SIGNAL= 32772 UNEXPECTED CAS = 32773 INVALID STATE = 32774 SET_CAS FAILURE = 32775 SEIZE ACK TIMEOUT = 32776 DEVICE IO ERROR = 32777 T1B TIMEOUT = 32778 I hope it helps. Greets On 1/8/07, yusuf [EMAIL PROTECTED] wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does - Unicall/1 protocol error. Cause 32769 mean, and can anyone help me. -- -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with IAX2 and Real Time configuration
Hello everybody, I'm having a problem trying to dial with an IAX2 extensions. I connect trough iaxComm and try to dial an extensions, then in asterisk CLI appears this: Aug 3 01:14:29 NOTICE[20915]: chan_iax2.c:7357 socket_read: Rejected connect attempt from 192.168.1.128, requested/capability 0x2/0x2 incompatible with our capability 0xf90c. I googled it but nothing appears. I have asterisk, zaptel and libpri from SVN brach 1.2. Thanks in advance. Greets. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Issue with IAX2 and Real Time configuration
It's solved. The problem was that the softphone has only one codec allowed and asterisk was configured to no allow that codec. On 8/2/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hello everybody, I'm having a problem trying to dial with an IAX2 extensions. I connect trough iaxComm and try to dial an extensions, then in asterisk CLI appears this: Aug 3 01:14:29 NOTICE[20915]: chan_iax2.c:7357 socket_read: Rejected connect attempt from 192.168.1.128, requested/capability 0x2/0x2 incompatible with our capability 0xf90c. I googled it but nothing appears. I have asterisk, zaptel and libpri from SVN brach 1.2. Thanks in advance. Greets. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk GNUDialer issue
Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer 0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone. The agents from GNUDialer log ok, and everything is fine until the GNUDialer makes a call, as soon as it engages (the phone starts to ring) asterisk crashes with these messages: Channel Zap/1-1 was answered. -- Executing Answer(Zap/1-1, ) in new stack Channel Zap/1-1 was answered. -- Executing Answer(Zap/1-1, ) in new stack localhost*CLI Ouch ... error while writing audio data: : Broken pipe Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). [1]+ Segmentation fault asterisk - I 'll really appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with USB
But my cell phone is recognised as a ttyACM device... Is it the same? 2006/2/7, Joseph Tanner [EMAIL PROTECTED]: Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with USB Date: Tue, 7 Feb 2006 11:55:07 -0300 Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with USB
If you can do it with a bluetooth conectionm, why not with the USB? Do you know which is the differece? Does it detect the cellphone as another device? I don't have a phone with bluetooth capability so I cannot test it to see how te OS recognizes it. 2006/2/8, Joseph Tanner [EMAIL PROTECTED]: I believe it's just being recognized as a modem. Feel free to try it out, but I haven't seen anything describing how to accomplish what you want with just a data cable (and I have searched). Please prove me wrong, I'd love to ditch the bluetooth dongle (already have too many 2.4GHz devices as it is, I think they're starting to cause interference). This is just going on gut instinct here, but if you're really persistent, maybe you can use the data cable to send the dial commands, and have some kind of adapter cable going from the 2.5 plug you have, to a 3.5, put that into the line-in of a sound card, and then configure asterisk to send the dial commands (to dial numbers, hangup, anything that needs a key pressed on the phone) through usb (should be able to access the tty device and issue commands there), and use the soundcard for audio. If I'm not mistaken, that's basically what the dock-n-talk and cellsocket devices do. You may run into a few problems, but I think it'd work. Joseph Tanner On 2/8/06, Facundo Ameal [EMAIL PROTECTED] wrote: But my cell phone is recognised as a ttyACM device... Is it the same? 2006/2/7, Joseph Tanner [EMAIL PROTECTED]: Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with USB Date: Tue, 7 Feb 2006 11:55:07 -0300 Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source
Re: [Asterisk-Users] Asterisk with USB
Yes, you are right, the point is that cells pass voice over bluetooth because there are bluetooth hands free, I think thats the point. Now I understand, thnk you all. 2006/2/8, Morgan Gilroy [EMAIL PROTECTED]: I assume the bluetooth connects as a hands free device and not a data cable? Iv not seen any mobile that will pass voice down the data cable. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Facundo Ameal Sent: 08 February 2006 13:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with USB If you can do it with a bluetooth conectionm, why not with the USB? Do you know which is the differece? Does it detect the cellphone as another device? I don't have a phone with bluetooth capability so I cannot test it to see how te OS recognizes it. 2006/2/8, Joseph Tanner [EMAIL PROTECTED]: I believe it's just being recognized as a modem. Feel free to try it out, but I haven't seen anything describing how to accomplish what you want with just a data cable (and I have searched). Please prove me wrong, I'd love to ditch the bluetooth dongle (already have too many 2.4GHz devices as it is, I think they're starting to cause interference). This is just going on gut instinct here, but if you're really persistent, maybe you can use the data cable to send the dial commands, and have some kind of adapter cable going from the 2.5 plug you have, to a 3.5, put that into the line-in of a sound card, and then configure asterisk to send the dial commands (to dial numbers, hangup, anything that needs a key pressed on the phone) through usb (should be able to access the tty device and issue commands there), and use the soundcard for audio. If I'm not mistaken, that's basically what the dock-n-talk and cellsocket devices do. You may run into a few problems, but I think it'd work. Joseph Tanner On 2/8/06, Facundo Ameal [EMAIL PROTECTED] wrote: But my cell phone is recognised as a ttyACM device... Is it the same? 2006/2/7, Joseph Tanner [EMAIL PROTECTED]: Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with USB Date: Tue, 7 Feb 2006 11:55:07 -0300 Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com
[Asterisk-Users] Asterisk with USB
Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
First, about the Jabber library: I'm using Asterisk Perl and the Jabber module for Perl. About dinmically loading the jabberid list, welll that's the problem I had and now I'm developing that. I thought about (and it's what I'm doing) generate a little database in XML in which you would put jabberid and extension so if you know the extension, you know the jabberid... what do you think about that? 2006/2/3, Andrew Kohlsmith [EMAIL PROTECTED]: On Friday 03 February 2006 10:21, Facundo Ameal wrote: I 'm developing something similar. It a perl script which tells you who is calling but it do it by sendind a jabber message. it's my first perl script so it's not finished yet. i'll share it so you can contribute if you want... http://www.mixdown.ca/~andrew/astbot -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
If you wnt to do it quick, I've seen this in another post of this
list, and I think is good:
exten = s,1,System(/bin/echo -n -e '${CALLERIDNAME}
${CALLERIDNUM}'| nc -w 1 192.168.1.16 10629)
then you have tyo be monitoring that port and capture the information,
you can do that in VB.
2006/2/6, Facundo Ameal [EMAIL PROTECTED]:
First, about the Jabber library: I'm using Asterisk Perl and the
Jabber module for Perl.
About dinmically loading the jabberid list, welll that's the problem I
had and now I'm developing that. I thought about (and it's what I'm
doing) generate a little database in XML in which you would put
jabberid and extension so if you know the extension, you know the
jabberid... what do you think about that?
2006/2/3, Andrew Kohlsmith [EMAIL PROTECTED]:
On Friday 03 February 2006 10:21, Facundo Ameal wrote:
I 'm developing something similar. It a perl script which tells you
who is calling but it do it by sendind a jabber message.
it's my first perl script so it's not finished yet.
i'll share it so you can contribute if you want...
http://www.mixdown.ca/~andrew/astbot
-A.
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--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793
Open your mind, use open source.
--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793
Open your mind, use open source.
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Re: [Asterisk-Users] change languages from an IVR
You have your three choices (1, 2, and 3), so you have to identify them. You set a variable to contain the laguage like this: exten= 1,1,NoOp(LANGUAGE=en) exten= 2,1,NoOp(LANGUAGE=sp) exten= 3,1,NoOp(LANGUAGE=fr) then you can control the dialplan's flow with GotoIf. It's the first thing that came to my mind, I hope it helps. Regards! 2006/2/6, Mark Phillips [EMAIL PROTECTED]: A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID popup
I 'm developing something similar. It a perl script which tells you who is calling but it do it by sendind a jabber message. it's my first perl script so it's not finished yet. i'll share it so you can contribute if you want... 2006/2/3, C F [EMAIL PROTECTED]: You should write a proxy and not connect directly, the reasons are as follows: 1. You don't want asterisk to crash because of problems with the manager app over the network, which Asterisk is known not to handle very well (as per the wiki). 2. Security, if you have every computer connecting to asterisk manager over the network, then you are giving the users a way to login to the system to do much more than they need, with a proxy however, you can always validate (and you should make sure to do that) everything before its submitted to asterisk. On 2/3/06, Mimmus [EMAIL PROTECTED] wrote: It works. Thanks a lot. With 15/20 users, is it better to use a manager proxy or to connect directly to the Asterisk server? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni Miano Sent: Friday, February 03, 2006 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID popup Link event ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
are you sure your sip phone is registering ok?
2006/2/1, abc def [EMAIL PROTECTED]:
Thanks Facundo for instruction but it didn't work. there is nothing new in
your suggestion compare to my conf files nevertheless I tried it but it
didn't work. I can make call from my sip phone but can't receive any phone
call. I am sure some one had had the same problem and solved it.
as always I appreciate for your suggestion, advice and/or correction to my
config files.
if you know how to solve this problem please give me some hint.
thank you
Facundo Ameal [EMAIL PROTECTED] wrote:
i've tested it with this config files and i worked:
extensions.conf
exten = 55,1,Dial(SIP/2271,20)
sip.conf
[2271]
type=friend
host=dynamic
secret=sip
allow=all
qualify=200
nat=no
Instead of 2271 you can put whatever you want.
good luck.
2006/1/31, Facundo Ameal :
Are you using a SIP Softphone or an ATA?
2006/1/31, Facundo Ameal :
does it registers well?
although i think you have to add context=default to the stargate1
section.
try that and see what happens.
2006/1/31, abc def :
Hi all, I am resending this message, so far no one has helped me with
this
incoming call issue. there is no problem with outbound call but there
is no
inbound call to my sip phone. the only message I get when I call from
pstn
is unable to create local channel for call forward to
'Local/[EMAIL PROTECTED]' (case =0). my configuration files are
attached
below. any help would be greatly appreciated. many thanks in advance.
ABC
abc def wrote:
there is no error message coming up on the pbx for in-bound calls
(there is
only debugging messages for outbound calls).
thanks in advance for any hint or suggestion.
Ama
I just post my configuration file here for sip phone:
extensions.conf
-
[globals]
[default]
include = incoming
include = outgoing
include = iax
inculde = sip
include = sccp
[sip]
exten = 2171,1,Dial(SIP/stargate1,20)
;exten = 2171,1,Dial(SIP/2171,20)
exten = 2171,2,Hangup
gt; exten = 2172,1,Dial(SIP/stargate2,20)
;exten = 2172,1,Dial(SIP/2172,20)
exten = 2172,2,Hangup
exten = 2173,1,Dial(SIP/stargate3,20)
;exten = 2173,1,Dial(SIP/2173,20)
exten = 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten = _214943[5-9]6,2,Hangup
[outgoing]
exten = _,1,Dial(Zap/g1/${EXTEN})
exten = _,2,Hangup
-
sip.conf
-
[general]
context=default ; Default context for incoming calls
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
register = stargate1:[EMAIL PROTECTED]/2171
register = stargate2:[EMAIL PROTECTED]/2172
register = stargate3:[EMAIL PROTECTED]/2173
;-- NAT
SUPPORT
nat=no ; Global NAT settings (Affects all peers and
users)
[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
gt ; type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid=21495071 2171
allow=all
qualify=200
nat=no
defaultip=10.47.200.140
[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141 ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid=21495072 2172
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid=starg ate3 2173
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
PaulH
- Original Message -
From: abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive
calls
Hi all,
I read many posts on asterisk mail site and been trying many
different
things but still I can't get my sip phones to work with asterisk.
I have a full blown-up voip netwok with two asterisk servers connected
to pstn
Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, more on new and used cars. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
Are you using a SIP Softphone or an ATA?
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message, so far no one has helped me with this
incoming call issue. there is no problem with outbound call but there is no
inbound call to my sip phone. the only message I get when I call from pstn
is unable to create local channel for call forward to
'Local/[EMAIL PROTECTED]' (case =0). my configuration files are attached
below. any help would be greatly appreciated. many thanks in advance.
ABC
abc def [EMAIL PROTECTED] wrote:
there is no error message coming up on the pbx for in-bound calls (there is
only debugging messages for outbound calls).
thanks in advance for any hint or suggestion.
Ama
I just post my configuration file here for sip phone:
extensions.conf
-
[globals]
[default]
include = incoming
include = outgoing
include = iax
inculde = sip
include = sccp
[sip]
exten = 2171,1,Dial(SIP/stargate1,20)
;exten = 2171,1,Dial(SIP/2171,20)
exten = 2171,2,Hangup
exten = 2172,1,Dial(SIP/stargate2,20)
;exten = 2172,1,Dial(SIP/2172,20)
exten = 2172,2,Hangup
exten = 2173,1,Dial(SIP/stargate3,20)
;exten = 2173,1,Dial(SIP/2173,20)
exten = 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten = _214943[5-9]6,2,Hangup
[outgoing]
exten = _,1,Dial(Zap/g1/${EXTEN})
exten = _,2,Hangup
-
sip.conf
-
[general]
context=default ; Default context for incoming calls
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
register = stargate1:[EMAIL PROTECTED]/2171
register = stargate2:[EMAIL PROTECTED]/2172
register = stargate3:[EMAIL PROTECTED]/2173
;-- NAT SUPPORT
nat=no ; Global NAT settings (Affects all peers and
users)
[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid=21495071 2171
allow=all
qualify=200
nat=no
defaultip=10.47.200.140
[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141 ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid=21495072 2172
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid=starg ate3 2173
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
PaulH
- Original Message -
From: abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive calls
Hi all,
I read many posts on asterisk mail site and been trying many different
things but still I can't get my sip phones to work with asterisk.
I have a full blown-up voip netwok with two asterisk servers connected
to pstn network with iax phones and cisco sccp phones which all work fine.
however, I have been struggeling to configure my sip phones (polycom 601,
Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip
phones to anywhere else but not receive phone calls. I can see the phones on
sip show registry and sip show peers but no track phone calls for sip.
can you please shed some light on me how to go about solving this
problem?
thank you and best regards,
Ama
HR SIZE=1 Do you Yahoo!?
With a free 1 GB, there's more in store with Yahoo! Mail.
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Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
i've tested it with this config files and i worked:
extensions.conf
exten = 55,1,Dial(SIP/2271,20)
sip.conf
[2271]
type=friend
host=dynamic
secret=sip
allow=all
qualify=200
nat=no
Instead of 2271 you can put whatever you want.
good luck.
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
Are you using a SIP Softphone or an ATA?
2006/1/31, Facundo Ameal [EMAIL PROTECTED]:
does it registers well?
although i think you have to add context=default to the stargate1 section.
try that and see what happens.
2006/1/31, abc def [EMAIL PROTECTED]:
Hi all, I am resending this message, so far no one has helped me with this
incoming call issue. there is no problem with outbound call but there is
no
inbound call to my sip phone. the only message I get when I call from pstn
is unable to create local channel for call forward to
'Local/[EMAIL PROTECTED]' (case =0). my configuration files are attached
below. any help would be greatly appreciated. many thanks in advance.
ABC
abc def [EMAIL PROTECTED] wrote:
there is no error message coming up on the pbx for in-bound calls (there
is
only debugging messages for outbound calls).
thanks in advance for any hint or suggestion.
Ama
I just post my configuration file here for sip phone:
extensions.conf
-
[globals]
[default]
include = incoming
include = outgoing
include = iax
inculde = sip
include = sccp
[sip]
exten = 2171,1,Dial(SIP/stargate1,20)
;exten = 2171,1,Dial(SIP/2171,20)
exten = 2171,2,Hangup
exten = 2172,1,Dial(SIP/stargate2,20)
;exten = 2172,1,Dial(SIP/2172,20)
exten = 2172,2,Hangup
exten = 2173,1,Dial(SIP/stargate3,20)
;exten = 2173,1,Dial(SIP/2173,20)
exten = 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten = _214943[5-9]6,2,Hangup
[outgoing]
exten = _,1,Dial(Zap/g1/${EXTEN})
exten = _,2,Hangup
-
sip.conf
-
[general]
context=default ; Default context for incoming calls
; Set this to your host name or domain
name
bindport=5060 ; UDP Port to bind to (SIP standard port
is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
register = stargate1:[EMAIL PROTECTED]/2171
register = stargate2:[EMAIL PROTECTED]/2172
register = stargate3:[EMAIL PROTECTED]/2173
;-- NAT SUPPORT
nat=no ; Global NAT settings (Affects all peers
and
users)
[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid=21495071 2171
allow=all
qualify=200
nat=no
defaultip=10.47.200.140
[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141 ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid=21495072 2172
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid=starg ate3 2173
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
PaulH
- Original Message -
From: abc def
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive calls
Hi all,
I read many posts on asterisk mail site and been trying many different
things but still I can't get my sip phones to work with asterisk.
I have a full blown-up voip netwok with two asterisk servers connected
to pstn network with iax phones and cisco sccp phones which all work fine.
however, I have been struggeling to configure my sip phones (polycom 601,
Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip
phones to anywhere else but not receive phone calls. I can see the phones
on
sip show registry and sip show peers but no track phone calls for sip.
can you please shed some light on me how to go about solving this
problem?
thank you and best regards,
Ama
HR SIZE=1 Do you Yahoo
Re: [Asterisk-Users] Best FXO hardware for home use
I'm using an X100P Clone at home and i had not much trouble, remember I'm just testing and learning a bit at home. I think if you hace to implement it at office you'll have to spend a bit more. 2006/1/25, Joseph Tanner [EMAIL PROTECTED]: Personally, I've had great success with an X101P (it's a clone, but it's the exact same chipset and layout of the original). Now, with Asterisk 1.2 beta2 (I believe it was beta2, I could be wrong though) and a P3 933MHz PC I did get annoying echo that I couldn't get rid of, and only on outgoing calls. If someone called me, even though all the same equipment is being used, there was no echo. Anyways, I upgraded to [EMAIL PROTECTED] 2.2 with Asterisk 1.2.1 and at the same time upgraded to a Celeron 2.93GHz PC, and there's virtually no echo. Only if there's complete silence on the other end and you yell very loud, can you barely make any hint of an echo out. No idea if it was the Asterisk upgrade, the new PC, or both that fixed my problem. Also, somewhere around the pre-1.0 days, I had two of these clones (one was the exact same layout as the actual X101P, the other had a different layout but the same chipset) and the one I used with my Packet8 line had no echo, but my landline did. Didn't matter if I switched the lines, the one connected to the Packet8 device had zero echo, the one connected to my landline had a noticeable echo (again, only on outgoing calls, incoming was fine). Played with rxgain/txgain, all the echo settings, etc. But now all is fine. Guess what I'm trying to say, is a lot depends on the line itself, and your exact setup. If you can pick up an X101P clone for cheap, I'd try that first. Most you're out is a few bucks (I say a few bucks, cause even if you pay $20 and decide it won't work for you, you can sell it for about what you paid). If you build or repair PCs a lot for others, then you'll need a good cheap modem someday anyways, the clone cards work fine for that. Works fine for me, only issue I have now is callerid isn't 100% reliable, but works the majority of the time. Until I troubleshoot it further (i.e., connect a regular phone directly to my landline to at least verify it's getting callerid when asterisk isn't), I can't blame the card for that. As long as the card will work with your setup (odds are it will), I think it's the best solution for home or small business use. Joseph Tanner On 1/25/06, Rich Adamson [EMAIL PROTECTED] wrote: echo cancellation is pretty limited on these cheap devices. the spa3000 manual for example states the AEC is limited to 8ms. good AECs will handle 64ms or more. in my experience the spa3000 echo canceller is cranky. it works most but not all of the time. I have been using one for 6 months without any problems. Make sure you have the most current firmware on it and it should work just fine. Kerry, There is an issue with the spa3k (as well as the TDM04b) in terms of handling echo properly on long pstn loops. You are obviously on a relatively short loop if you've not been exposed to the variable echo cancellation issues. In other words, long pstn loops basically fall outside the limits of the echo cancellation software as someone else already noted. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Zaptel issues
i don't see any other solution. you have t orecompile either the kernel or zaptel, I recommend recompiling the kernel because then you can continue using the new gcc version. it is not difficult, if you want i can give you intructions so you can do it in a minute. reagrds, 2006/1/24, Mike Hammett [EMAIL PROTECTED]: Yeah, recompiling the kernel is a bit over my head, but I don't want to install an older gcc, so I'll just have to await some hand-holding from the people that put my kernel together (OpenVZ). Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 23, 2006 8:02 PM Subject: Asterisk-Users Digest, Vol 18, Issue 143 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... -- Message: 15 Date: Mon, 23 Jan 2006 22:35:28 -0300 From: Facundo Ameal [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zaptel issues To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 I think you have comiled your kernel with a version of gcc and zaptel with another one, Compile zaptel drivers with gcc-3.3 and you will solve it, otherwise, you cas recompile your kernel with the new version of gcc. i also had that problem. 2006/1/23, Mike Hammett [EMAIL PROTECTED]: [EMAIL PROTECTED] ~]# which modprobe /sbin/modprobe [EMAIL PROTECTED] ~]# modprobe --version module-init-tools version 3.1-pre5 [EMAIL PROTECTED] ~]# dmesg | tail zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' It looks like my gcc versions are different from the one that made the kernel and the one that made the zaptel stuff. So then of the zt lines, do I only need: install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 4:49 AM Subject: Asterisk-Users Digest, Vol 18, Issue 82 -- Message: 12 Date: Fri, 13 Jan 2006 11:52:20 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zaptel issues To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Jan 12, 2006 at 09:39:18AM -0600, Mike Hammett wrote: On a side note: When poking around, I noticed in the zaptel Makefile that there is a section talking about ztdummy automatically being included on 2.6 kernels. Is this correct? On to the main topic: Any ideas for troubleshooting this? [EMAIL PROTECTED] zaptel-1.2.1]# /etc/rc.d/init.d/zaptel start Loading zaptel framework: FATAL: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format [FAILED] Waiting for zap to come online...Error: missing /dev/zap! [EMAIL PROTECTED] libpri-1.2.1]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid
Re: [Asterisk-Users] Home Test!
So: Grandstream is easy and Sipura is more flexible and complete. Am I right? 2006/1/24, The VoIP Connection [EMAIL PROTECTED]: I think they are both great products, and we have many customers using both successfully. You will probably be happy with either. Both have great sound, both work well with Asterisk. The Grandstream is easier to configure, the Sipura has more options. More Grandsreams show up DOA, more Sipuras die in the field. Grandstreams have a few more bugs, but they have much better support. Slight edge to Grandstream on price for similar features. Slight edge to Sipura on build quality. Grandstream is a small and easy to deal with organization. Sipura is Cisco. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 8:30 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Home Test! Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED]: We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video Conferencing.
But I'm in Argentina... 2006/1/24, The VoIP Connection [EMAIL PROTECTED]: Facundo, If everything goes right, we will be demonstrating an Asterisk based Videoconferencing system at the Internet Telephony expo this week. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 8:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Video Conferencing. I'm looking for point to point Video Conferencing , just because, like I said in other post, I'm doing some tests at homeand I want to try *almost* every feature asterisk has. THank you, I 'll read about it. I also would like to develop for asterisk (it's not for the bounty) but I just don't know much about C or ANSI C. 2006/1/23, Dean Collins [EMAIL PROTECTED]: It's possible to do point to point but not point to multipoint. I tried to get development for this some time ago and no one responded, check out my Video Conference Bounty on www.voip-info.org, since then we have developed our own solution that we have decided to market, it will cost $2,000 for up to 10 users that uses the Macromedia communications server. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Facundo Ameal Sent: Monday, 23 January 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video Conferencing. I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Home Test!
Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing List (JUST A TEST)
we hear you loud and clear 2006/1/23, [EMAIL PROTECTED] [EMAIL PROTECTED]: Sorry, I haven't received a message in a few hours, just testing to see if it is alive. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video Conferencing.
I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial out and message playback
look at this: http://www.voip-info.org/wiki-VICIDIAL+Dialer perhaps it's what you are looking for... 2006/1/23, Danish Samad [EMAIL PROTECTED]: Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
Erick Muchas Gracias por la respuesta. I'm not using any of that projects, it's my own Asterisk installation onto slackware 10. well what can you tell about sipura ones? 2006/1/23, Erick Perez [EMAIL PROTECTED]: Hola Facundo, saludos desde Panama. If you're running asterisk at home or some other asterisk project and you're only concerned about the ATA, well, a HT-286 (entry level, cheap) is a good start. Yes, there are reported issues with the GrandStream equipment but all the others have issues too (ok ok I know, don't start on this one). Since your home installation is not *mission critical* a HT-286 will be good. So far I can tell you that a voice provider in my country uses HT-286 and HT-486 commercially deployed at customer premises and it has been working prefectly. My girlfriend who is at this moment in Belgium has an HT-286 that I sent to her and the ATA register back to Panama with no problems. No echo issues. Maybe due to line conditions in Argentina you need to try different echo cancellers. Cheers, On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED]: We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Video Conferencing.
I'm looking for point to point Video Conferencing , just because, like I said in other post, I'm doing some tests at homeand I want to try *almost* every feature asterisk has. THank you, I 'll read about it. I also would like to develop for asterisk (it's not for the bounty) but I just don't know much about C or ANSI C. 2006/1/23, Dean Collins [EMAIL PROTECTED]: It's possible to do point to point but not point to multipoint. I tried to get development for this some time ago and no one responded, check out my Video Conference Bounty on www.voip-info.org, since then we have developed our own solution that we have decided to market, it will cost $2,000 for up to 10 users that uses the Macromedia communications server. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Facundo Ameal Sent: Monday, 23 January 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video Conferencing. I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel issues
I think you have comiled your kernel with a version of gcc and zaptel with another one, Compile zaptel drivers with gcc-3.3 and you will solve it, otherwise, you cas recompile your kernel with the new version of gcc. i also had that problem. 2006/1/23, Mike Hammett [EMAIL PROTECTED]: [EMAIL PROTECTED] ~]# which modprobe /sbin/modprobe [EMAIL PROTECTED] ~]# modprobe --version module-init-tools version 3.1-pre5 [EMAIL PROTECTED] ~]# dmesg | tail zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' It looks like my gcc versions are different from the one that made the kernel and the one that made the zaptel stuff. So then of the zt lines, do I only need: install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 4:49 AM Subject: Asterisk-Users Digest, Vol 18, Issue 82 -- Message: 12 Date: Fri, 13 Jan 2006 11:52:20 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zaptel issues To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Jan 12, 2006 at 09:39:18AM -0600, Mike Hammett wrote: On a side note: When poking around, I noticed in the zaptel Makefile that there is a section talking about ztdummy automatically being included on 2.6 kernels. Is this correct? On to the main topic: Any ideas for troubleshooting this? [EMAIL PROTECTED] zaptel-1.2.1]# /etc/rc.d/init.d/zaptel start Loading zaptel framework: FATAL: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format [FAILED] Waiting for zap to come online...Error: missing /dev/zap! [EMAIL PROTECTED] libpri-1.2.1]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.8-022stab050.1/extra/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy Could you please provide the output of following: which modprobe modprobe --version To make things simpler, do away with the stuff that the zaptel install puts in /etc/modprobe.d/zaptel (or /etc/modprobe.conf ). (ztdummy needs no ztcfg run after it) Also, please provide the latest relevant kernel log messages: dmesg | tail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] make linux26
i compiled it with make linux26 and had no trouble. try it like that. 2006/1/23, Mike Hammett [EMAIL PROTECTED]: Yeah, that's where I saw contradicting what I saw elsewhere. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 23, 2006 3:48 PM Subject: Asterisk-Users Digest, Vol 18, Issue 141 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... -- Message: 23 Date: Mon, 23 Jan 2006 22:37:55 +0100 From: [EMAIL PROTECTED] Subject: RE : [Asterisk-Users] make linux26 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: !~!UENERkVCMDkAAQACABgARuRp1ly/[EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi Mike, You must continue - for zaptel only - to make linux26, as it is described in the companion file README.Linux26 in the Zaptel folder (/usr/src/zaptel). Read the text from this file, as suggested in its title : To build for Linux 2.6, first you must be sure that you have a symlink to your linux-2.6 sources in /usr/src/linux-2.6. The 2.6 kernel no longer needs the full sourcecode to build against it. You can create the symlink to /lib/modules/`uname -r`/build/ and then you can type: # make linux26 # make install Note that you will also need CRC-CCITT functions compiled with your kernel or as a kernel module. These can be selected from the Library Routines submenu during kernel configuration via make menuconfig It is a good habit to read all this README... files before to do something, as it is important to read any user manual for any sofisticated equipment ;-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Hammett Envoyé : lundi 23 janvier 2006 22:10 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] make linux26 I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060123/9b097a35/attachment-0001.htm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
I haven't said it but if someone believes there's a better choice than buying a sipura or a grandstream ht, please tell me, I considered thaat two because, here, they are popular. 2006/1/23, Facundo Ameal [EMAIL PROTECTED]: Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED]: We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto:[EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Home Test!
i am using a win-modem as a X100P clone. It has an especial motorola chiset which is detailed here: http://www.voip-info.org/wiki/view/X100P+clone it was really hard for me to get this modem. sorry, but I can't help you. If you come here you have to go to every store you see and ask, because it's very difficult to get them. i am part of a LUG (Linux User Group) and i am the only one who could manage to get this specific modem. Sorry. 2006/1/23, Maxi Belino [EMAIL PROTECTED]: Hi, Facundo i'm from Uruguay, i'm plannig to visit Argentina and i would like to know where i can get there the X100p Clone Card and some other VoIP stuff. Is there a website you could recommend me? do you have a phone number of this store ? name or address? Thanks spanish on gracias ! saludos !spanish off Maxi 2006/1/24, Facundo Ameal [EMAIL PROTECTED]: I haven't said it but if someone believes there's a better choice than buying a sipura or a grandstream ht, please tell me, I considered thaat two because, here, they are popular. 2006/1/23, Facundo Ameal [EMAIL PROTECTED]: Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: spanish ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles spanich-off I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection [EMAIL PROTECTED] : We have sold thousands of these with no reports of echo problems. Perhaps the reviews were referring to a different Grandstream product? Some of the phones have had some echo issues. BTW, the Sipura 2000 has been replaced by the 2002. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Facundo Ameal [mailto: [EMAIL PROTECTED] Sent: Monday, January 23, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Home Test! Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
