FRANCISCO PEREZ-LANDAETA
2006-Sep-09 12:48 UTC
[asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54
hi i need helpl configuring a quintum tenor analog gateway using sip with asterisk. anyone, help is appreciated the model of the gteway is asm200 i need the settings to configure it with asterisk. for some reason it registers with asterisk but when try to call the extension from the quintum it is not recognized. help help help thanks>From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: asterisk-users Digest, Vol 26, Issue 54 >Date: Sat, 9 Sep 2006 12:00:25 -0700 (MST) >MIME-Version: 1.0 >Received: from lists.digium.com ([69.16.138.164]) by >bay0-mc2-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, 9 >Sep 2006 12:03:59 -0700 >Received: from digium-69-16-138-164.phx1.puregig.net (localhost >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 3C2CA41D5;Sat, 9 >Sep 2006 12:00:25 -0700 (MST) >X-Message-Info: LsUYwwHHNt1Qrly5/IdcOLxnJ5Hdz4bhYGyQtYHi6jU>X-BeenThere: asterisk-users@lists.digium.com >X-Mailman-Version: 2.1.5 >Precedence: list >List-Id: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users.lists.digium.com> >List-Unsubscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=unsubscribe> >List-Archive: <http://lists.digium.com/pipermail/asterisk-users> >List-Post: <mailto:asterisk-users@lists.digium.com> >List-Help: <mailto:asterisk-users-request@lists.digium.com?subject=help> >List-Subscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=subscribe> >Errors-To: asterisk-users-bounces@lists.digium.com >Return-Path: asterisk-users-bounces@lists.digium.com >X-OriginalArrivalTime: 09 Sep 2006 19:04:00.0431 (UTC) >FILETIME=[B57B13F0:01C6D442] > >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 > asterisk-users-request@lists.digium.com > >You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of asterisk-users digest..." > > >Today's Topics: > > 1. Re: Call Forwarding in SIP.conf (broadbandvoice@comcast.net) > 2. RE: Call Processing Slow 11 seconds (G.Jacobsen) > 3. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) > 4. RE: Call Processing Slow 11 seconds (broadbandvoice@comcast.net) > 5. Re: Call Processing Slow 11 seconds (Alberto Sagredo) > 6. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) > 7. Re: What don't I get about SIP? (John Marvin) > 8. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) > 9. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) > 10. RE: What don't I get about SIP? (Mike) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Sat, 09 Sep 2006 17:12:54 +0000 >From: broadbandvoice@comcast.net >Subject: Re: [asterisk-users] Call Forwarding in SIP.conf >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: > <090920061712.20356.4502F61600032D6F00004F84220588644208010B020E9B02@comcast.net> > >Content-Type: text/plain; charset="us-ascii" > >Skipped content of type multipart/alternative-------------- next part >-------------- >An embedded message was scrubbed... >From: "Tim St. Pierre" <tim@communicatefreely.net> >Subject: Re: [asterisk-users] Call Forwarding in SIP.conf >Date: Sat, 9 Sep 2006 16:52:40 +0000 >Size: 2109 >Url: >http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/828bebdd/attachment-0001.eml > >------------------------------ > >Message: 2 >Date: Sat, 9 Sep 2006 19:17:23 +0300 >From: "G.Jacobsen" <g_jacobsen@yahoo.co.uk> >Subject: RE: [asterisk-users] Call Processing Slow 11 seconds >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> >Message-ID: <CPEBJFBCDCKKIHJAODHCCEPGCLAA.g_jacobsen@yahoo.co.uk> >Content-Type: text/plain; charset="us-ascii" > >In case you use an adapter or voip phone: Did you try to press hash # after >the number ? - then the adapter/voip phone dials immediately and doesnt >wait >for the next digit timeout. > >Cheers > >Gerry > > -----Original Message---- > From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of >broadbandvoice@comcast.net > Sent: Samstag, 9. September 2006 15:15 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Call Processing Slow 11 seconds > > > I'm having some slowness issue with Asterisk. When a number is dialed it >takes 11 seconds before it rings out. I been considering using openser for >the call processing and leaving asterisk for voicemail and conference >bridge. I get a dialtone rightaway when the receiver is picked up but after >dialing the number but within asterisk extensions and pstn numbers takes 11 >seconds before ringing out. Anyone else experiencing this. I use Asterisk >1.2.3 >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/632afcb4/attachment-0001.htm > >------------------------------ > >Message: 3 >Date: Sat, 09 Sep 2006 18:23:37 +0100 >From: Daniel Pocock <daniel@readytechnology.co.uk> >Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453 >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <4502F899.4010602@readytechnology.co.uk> >Content-Type: text/plain; charset=us-ascii; format=flowed > > > >Jason Lee wrote: > > > Hi, > > > > I was testing the intel based G729 codec on SVN-trunk-r42453 following > > the > > new instructions for compiling with SVN trunk and it in preliminary > > tests it > > works ok for some calls but I found when one end of the call is an IVR >or > > Music On Hold the sound gets all distorted and asterisk segfaults. You > > can > > view the backtrace at http://pastebin.ca/165220 > > > > Any assistance on this would be appreciated. > > >Have you compiled with debugging symbols instead of CPU optimization? > >Can you type `bt' after the segfault, to give us some more detail? > >How long into the call does this happen? > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >--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 > > > > > > >------------------------------ > >Message: 4 >Date: Sat, 09 Sep 2006 17:27:15 +0000 >From: broadbandvoice@comcast.net >Subject: RE: [asterisk-users] Call Processing Slow 11 seconds >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: > <090920061727.5745.4502F9730006E06300001671220699973508010B020E9B02@comcast.net> > >Content-Type: text/plain; charset="us-ascii" > >Skipped content of type multipart/alternative-------------- next part >-------------- >An embedded message was scrubbed... >From: "G.Jacobsen" <g_jacobsen@yahoo.co.uk> >Subject: RE: [asterisk-users] Call Processing Slow 11 seconds >Date: Sat, 9 Sep 2006 17:20:05 +0000 >Size: 818 >Url: >http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/a8051465/attachment-0001.eml > >------------------------------ > >Message: 5 >Date: Sat, 09 Sep 2006 19:47:23 +0200 >From: Alberto Sagredo <asagredo@peoplecall.com> >Subject: Re: [asterisk-users] Call Processing Slow 11 seconds >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <4502FE2B.1020200@peoplecall.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Yes you could script a dialplan putting xxxx... and S0 (zero) at the end. > >An example : > >(xxxxxxS0) It will dial 6 digits directly when you enter the 6th. > >You could learn how to adapt your Linksys dialplan looking this wiki. > >http://voip.wikispaces.com/ > >broadbandvoice@comcast.net escribió: > > Yes that works. I'm using Linksys adapter, is there a code I can put > > in the dial plan to prevent users from putting # after the number? I > > have a lot of people on the server and cannot ask them all to be > > pushing # after every call. Thanks for the tip and any help will be > > appreciated. > > > > > > -------------- Original message -------------- > > From: "G.Jacobsen" <g_jacobsen@yahoo.co.uk> > > In case you use an adapter or voip phone: Did you try to press > > hash # after the number ? - then the adapter/voip phone dials > > immediately and doesnt wait for the next digit timeout. > > > > Cheers > > > > Gerry > > > > > > -----Original Message---- > > *From:* asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com]*On Behalf Of > > *broadbandvoice@comcast.net > > *Sent:* Samstag, 9. September 2006 15:15 > > *To:* asterisk-users@lists.digium.com > > *Subject:* [asterisk-users] Call Processing Slow 11 seconds > > > > I'm having some slowness issue with Asterisk. When a number is > > dialed it takes 11 seconds before it rings out. I been > > considering using openser for the call processing and leaving > > asterisk for voicemail and conference bridge. I get a dialtone > > rightaway when the receiver is picked up but after dialing the > > number but within asterisk extensions and pstn numbers takes > > 11 seconds before ringing out. Anyone else experiencing this. > > I use Asterisk 1.2.3 > > > > > > ------------------------------------------------------------------------ > > > > Asunto: > > RE: [asterisk-users] Call Processing Slow 11 seconds > > De: > > "G.Jacobsen" <g_jacobsen@yahoo.co.uk> > > Fecha: > > Sat, 9 Sep 2006 17:20:05 +0000 > > Para: > > "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users@lists.digium.com> > > > > Para: > > "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users@lists.digium.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 > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --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 > > > > > >------------------------------ > >Message: 6 >Date: Sat, 9 Sep 2006 13:03:32 -0500 >From: "Jason Lee" <jason.m.lee@gmail.com> >Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453 >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> >Message-ID: > <c3bac2490609091103l489be6bas75c63061e1a7cf4c@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >I recompiled with debuging options... > >both bt and btfull outputs http://pastebin.ca/165250 >Before I recompiled it gave me a second of audio then I got nothing but >distortion for 5 seconds then asterisk would crash. >I retested after compiling it with just a call between two local devices >one >using ulaw and the other using g729 and I'm getting nothing but distortion. >I then tried calling music on hold and it took 3 minutes to crash the whole >time I got nothing but distortion. > > >On 9/9/06, Daniel Pocock <daniel@readytechnology.co.uk> wrote: > > > > > > > > Jason Lee wrote: > > > > > Hi, > > > > > > I was testing the intel based G729 codec on SVN-trunk-r42453 following > > > the > > > new instructions for compiling with SVN trunk and it in preliminary > > > tests it > > > works ok for some calls but I found when one end of the call is an IVR > > or > > > Music On Hold the sound gets all distorted and asterisk segfaults. You > > > can > > > view the backtrace at http://pastebin.ca/165220 > > > > > > Any assistance on this would be appreciated. > > > > > Have you compiled with debugging symbols instead of CPU optimization? > > > > Can you type `bt' after the segfault, to give us some more detail? > > > > How long into the call does this happen? > > > > > > > >------------------------------------------------------------------------ > > > > > >_______________________________________________ > > >--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 > > > > > >-- >Regards, > >Jason Lee >OmegaServ >jlee@omegaserv.com >Direct Line: (204) 480-1238 >Toll Free: (866) 664-7786 Ext 200 >http://www.omegaserv.com >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/d4e38b74/attachment-0001.htm > >------------------------------ > >Message: 7 >Date: Sat, 09 Sep 2006 12:04:33 -0600 >From: John Marvin <jm-asterisk@themarvins.org> >Subject: Re: [asterisk-users] What don't I get about SIP? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <45030231.4060808@themarvins.org> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Mike wrote: > > > Did I misread the Asterisk wiki pages, because I believed that when a > > pattern was present, the pattern takes precedence over any "real" > > extensions? (i.e. if I have both 1234 and _1XXX as extensions in a >context)? > >It's the opposite. Asterisk always uses the most specific match for an >extension, i.e. anything that matches _1XXX will take precedence over >_XXXX, but if it matches _12XX that will take precedence over _1XXX, etc. > >John > > >------------------------------ > >Message: 8 >Date: Sat, 09 Sep 2006 19:15:31 +0100 >From: Daniel Pocock <daniel@readytechnology.co.uk> >Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453 >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <450304C3.2060505@readytechnology.co.uk> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > >Jason Lee wrote: > > > I recompiled with debuging options... > > > > both bt and btfull outputs http://pastebin.ca/165250 > > Before I recompiled it gave me a second of audio then I got nothing but > > distortion for 5 seconds then asterisk would crash. > > I retested after compiling it with just a call between two local > > devices one > > using ulaw and the other using g729 and I'm getting nothing but > > distortion. > > I then tried calling music on hold and it took 3 minutes to crash the > > whole > > time I got nothing but distortion. > > >This suggests that someone/something gave the command `stop now' > >Can you send the backtrace from a segfault? > > > > > On 9/9/06, Daniel Pocock <daniel@readytechnology.co.uk> wrote: > > > >> > >> > >> > >> Jason Lee wrote: > >> > >> > Hi, > >> > > >> > I was testing the intel based G729 codec on SVN-trunk-r42453 >following > >> > the > >> > new instructions for compiling with SVN trunk and it in preliminary > >> > tests it > >> > works ok for some calls but I found when one end of the call is an >IVR > >> or > >> > Music On Hold the sound gets all distorted and asterisk segfaults. >You > >> > can > >> > view the backtrace at http://pastebin.ca/165220 > >> > > >> > Any assistance on this would be appreciated. > >> > > >> Have you compiled with debugging symbols instead of CPU optimization? > >> > >> Can you type `bt' after the segfault, to give us some more detail? > >> > >> How long into the call does this happen? > >> > >> > >> > >------------------------------------------------------------------------ > >> > >> > > >> >_______________________________________________ > >> >--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 > >> > > > > > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >--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 > > > > > > >------------------------------ > >Message: 9 >Date: Sat, 9 Sep 2006 13:28:55 -0500 >From: "Jason Lee" <jason.m.lee@gmail.com> >Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453 >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> >Message-ID: > <c3bac2490609091128y4235e54dqace530af644cf1a3@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >Sorry about that. I thought I had the right core dump. I retried again and >the output from bt and bt full is at http://pastebin.ca/165289 >It took 1min 50seconds of nothing but distortion before asterisk segfaulted > >-- >Regards, > >Jason > >On 9/9/06, Daniel Pocock <daniel@readytechnology.co.uk> wrote: > > > > > > > > Jason Lee wrote: > > > > > I recompiled with debuging options... > > > > > > both bt and btfull outputs http://pastebin.ca/165250 > > > Before I recompiled it gave me a second of audio then I got nothing >but > > > distortion for 5 seconds then asterisk would crash. > > > I retested after compiling it with just a call between two local > > > devices one > > > using ulaw and the other using g729 and I'm getting nothing but > > > distortion. > > > I then tried calling music on hold and it took 3 minutes to crash the > > > whole > > > time I got nothing but distortion. > > > > > This suggests that someone/something gave the command `stop now' > > > > Can you send the backtrace from a segfault? > > > > > > > > On 9/9/06, Daniel Pocock <daniel@readytechnology.co.uk> wrote: > > > > > >> > > >> > > >> > > >> Jason Lee wrote: > > >> > > >> > Hi, > > >> > > > >> > I was testing the intel based G729 codec on SVN-trunk-r42453 > > following > > >> > the > > >> > new instructions for compiling with SVN trunk and it in preliminary > > >> > tests it > > >> > works ok for some calls but I found when one end of the call is an > > IVR > > >> or > > >> > Music On Hold the sound gets all distorted and asterisk segfaults. > > You > > >> > can > > >> > view the backtrace at http://pastebin.ca/165220 > > >> > > > >> > Any assistance on this would be appreciated. > > >> > > > >> Have you compiled with debugging symbols instead of CPU optimization? > > >> > > >> Can you type `bt' after the segfault, to give us some more detail? > > >> > > >> How long into the call does this happen? > > >> > > >> > > >> > > > >------------------------------------------------------------------------ > > >> > > >> > > > >> >_______________________________________________ > > >> >--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 > > >> > > > > > > > > > > > > >------------------------------------------------------------------------ > > > > > >_______________________________________________ > > >--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 > > >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: >http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/a67f3fb5/attachment-0001.htm > >------------------------------ > >Message: 10 >Date: Sat, 9 Sep 2006 14:58:32 -0400 >From: "Mike" <list@virtutel.ca> >Subject: RE: [asterisk-users] What don't I get about SIP? >To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> >Message-ID: <00bb01c6d441$f36800c0$0a01a8c0@MIKE> >Content-Type: text/plain; charset="iso-8859-1" > >It certainly makes sense, and I tried it...it works, you are right. > >So what do you make of this page : >http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf >+sorting > >Mike > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > > John Marvin > > Sent: September 9, 2006 2:05 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] What don't I get about SIP? > > > > Mike wrote: > > > > > Did I misread the Asterisk wiki pages, because I believed > > that when a > > > pattern was present, the pattern takes precedence over any "real" > > > extensions? (i.e. if I have both 1234 and _1XXX as > > extensions in a context)? > > > > It's the opposite. Asterisk always uses the most specific > > match for an extension, i.e. anything that matches _1XXX will > > take precedence over _XXXX, but if it matches _12XX that will > > take precedence over _1XXX, etc. > > > > John > > _______________________________________________ > > --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 > > >End of asterisk-users Digest, Vol 26, Issue 54 >**********************************************_________________________________________________________________ Check the weather nationwide with MSN Search: Try it now! http://search.msn.com/results.aspx?q=weather&FORM=WLMTAG