Hi I need help configuring a quintum box with asterisk. Anyone has it working ? Thanks, Please let me know what I should do. I want to be able to register the asm200 with an extension, and be able to hopoff calls when calling from my asterisk, Thanks,
On 9/9/06 6:47 PM, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> wrote: > 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..." > > > Today's Topics: > > 1. Re: Another (quick) Polycom 501 question (Kevin Smith) > 2. RE: asterisk-users Digest, Vol 26, Issue 54 > (FRANCISCO PEREZ-LANDAETA) > 3. Re: Call Processing Slow 11 seconds ([EMAIL PROTECTED]) > 4. Re: Zaptel-1.2.9 compile error (Samy Antoun) > 5. Problems configuring Polycom 301 (Jim Freeze) > 6. Re: Zaptel-1.2.9 compile error (Nigel Godfrey) > 7. ztdummy installed but choppy audio warning on load (Nigel Godfrey) > 8. Re: ztdummy installed but choppy audio warning on load > (Daniel Pocock) > 9. Re: Zaptel-1.2.9 compile error (Samy Antoun) > 10. Scope of contexts (Rene) > 11. Re: What don't I get about SIP? (John Marvin) > 12. Re: Scope of contexts (Doug Lytle) > 13. Re: Scope of contexts (Moises Silva) > 14. Re: Grandstream GX-2000, doesn't send calls to free lines > (Zeeshan Zakaria) > 15. Re: How to send correct Caller ID on PRI (Zeeshan Zakaria) > 16. Re: How to use Grandstream GX-2000 phones for paging > (Zeeshan Zakaria) > 17. Re: Grandstream, how to use the configuration tool > (Zeeshan Zakaria) > 18. Re: Roundrobin not working on PRI (Zeeshan Zakaria) > 19. Using option 'r' in queue doesn't announce frequeny etc. > (Zeeshan Zakaria) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 09 Sep 2006 15:24:44 -0400 > From: Kevin Smith <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Another (quick) Polycom 501 question > 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; format=flowed > > Hi Mike, > > As far as I know, you need to at least start the dialing (ie New call, > speaker, etc) for the digitmap to even come into play. > > The only settings that I am aware of that you can try to change are > dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf. > > Kevin > > Mike wrote: >> Hi all, >> >> That's my last one for a while (I hope). >> >> How can I (if at all possible) make the 501 turn on the speaker phone >> as soon as a digit is dialed (if the handset is not lifted)? Sort of >> like what a normal speakerphone does. >> >> The reason I want this is I want the 501 digitmap to be taken into >> consideration even if the handset isnt lifted and the speakerphone >> button isn't consciously pressed. For all those users who don't want >> to press send, but like dialing without lifting the handset (and can't >> be bothered to press the speakerphone button). Yes I know it's >> capricious, but we have the users we have... >> >> Yes, I have read the admin manual, but couldn't find the info. I am >> assuming I just don't know what to look for, but that this >> functionality exists. >> >> >> >> Mike >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> --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: 2 > Date: Sat, 09 Sep 2006 19:48:27 +0000 > From: "FRANCISCO PEREZ-LANDAETA" <[EMAIL PROTECTED]> > Subject: [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54 > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; format=flowed > > > 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: [EMAIL PROTECTED] >> 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-us >> [EMAIL PROTECTED]> >> List-Archive: <http://lists.digium.com/pipermail/asterisk-users> >> List-Post: <mailto:asterisk-users@lists.digium.com> >> List-Help: <mailto:[EMAIL PROTECTED]> >> List-Subscribe: >> <http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-us >> [EMAIL PROTECTED]> >> Errors-To: [EMAIL PROTECTED] >> Return-Path: [EMAIL PROTECTED] >> 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 >> [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..." >> >> >> Today's Topics: >> >> 1. Re: Call Forwarding in SIP.conf ([EMAIL PROTECTED]) >> 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 ([EMAIL PROTECTED]) >> 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: [EMAIL PROTECTED] >> Subject: Re: [asterisk-users] Call Forwarding in SIP.conf >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users@lists.digium.com> >> Message-ID: >> <[EMAIL PROTECTED] >> 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" <[EMAIL PROTECTED]> >> 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/20060909/828bebd >> d/attachment-0001.eml >> >> ------------------------------ >> >> Message: 2 >> Date: Sat, 9 Sep 2006 19:17:23 +0300 >> From: "G.Jacobsen" <[EMAIL PROTECTED]> >> Subject: RE: [asterisk-users] Call Processing Slow 11 seconds >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <asterisk-users@lists.digium.com> >> Message-ID: <[EMAIL PROTECTED]> >> 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: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] Behalf Of >> [EMAIL PROTECTED] >> 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/20060909/632afcb >> 4/attachment-0001.htm >> >> ------------------------------ >> >> Message: 3 >> Date: Sat, 09 Sep 2006 18:23:37 +0100 >> From: Daniel Pocock <[EMAIL PROTECTED]> >> 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: <[EMAIL PROTECTED]> >> 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: [EMAIL PROTECTED] >> Subject: RE: [asterisk-users] Call Processing Slow 11 seconds >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users@lists.digium.com> >> Message-ID: >> <[EMAIL PROTECTED] >> et> >> >> Content-Type: text/plain; charset="us-ascii" >> >> Skipped content of type multipart/alternative-------------- next part >> -------------- >> An embedded message was scrubbed... >> From: "G.Jacobsen" <[EMAIL PROTECTED]> >> 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/a805146 >> 5/attachment-0001.eml >> >> ------------------------------ >> >> Message: 5 >> Date: Sat, 09 Sep 2006 19:47:23 +0200 >> From: Alberto Sagredo <[EMAIL PROTECTED]> >> Subject: Re: [asterisk-users] Call Processing Slow 11 seconds >> 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; 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/ >> >> [EMAIL PROTECTED] 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" <[EMAIL PROTECTED]> >>> 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:* [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] Behalf Of >>> [EMAIL PROTECTED] >>> *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" <[EMAIL PROTECTED]> >>> 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" <[EMAIL PROTECTED]> >> 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: >> <[EMAIL PROTECTED]> >> 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 <[EMAIL PROTECTED]> 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 >> [EMAIL PROTECTED] >> 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/d4e38b7 >> 4/attachment-0001.htm >> >> ------------------------------ >> >> Message: 7 >> Date: Sat, 09 Sep 2006 12:04:33 -0600 >> From: John Marvin <[EMAIL PROTECTED]> >> 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: <[EMAIL PROTECTED]> >> 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 <[EMAIL PROTECTED]> >> 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: <[EMAIL PROTECTED]> >> 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 <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> >> 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: >> <[EMAIL PROTECTED]> >> 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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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/a67f3fb >> 5/attachment-0001.htm >> >> ------------------------------ >> >> Message: 10 >> Date: Sat, 9 Sep 2006 14:58:32 -0400 >> From: "Mike" <[EMAIL PROTECTED]> >> 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: <[EMAIL PROTECTED]> >> 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: [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] 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 > > > > ------------------------------ > > Message: 3 > Date: Sat, 09 Sep 2006 20:27:38 +0000 > From: [EMAIL PROTECTED] > Subject: Re: [asterisk-users] Call Processing Slow 11 seconds > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > <[EMAIL PROTECTED] > et> > > Content-Type: text/plain; charset="us-ascii" > > Thanks, I tried that and did not work for me. My users are calling US number > and without the # at the end of the last digit dials it takes 11 seconds > before it starts ringing. > > -------------- Original message -------------- > From: Alberto Sagredo <[EMAIL PROTECTED]> > >> 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/ >> >> [EMAIL PROTECTED] 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" >>> 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:* [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] Behalf Of >>> [EMAIL PROTECTED] >>> *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" >>> Fecha: >>> Sat, 9 Sep 2006 17:20:05 +0000 >>> Para: >>> "Asterisk Users Mailing List - Non-Commercial Discussion" >>> >>> >>> Para: >>> "Asterisk Users Mailing List - Non-Commercial Discussion" >>> >>> >>> >>> _______________________________________________ >>> --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/20060909/99a624f4 > /attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Sat, 9 Sep 2006 13:41:43 -0700 (PDT) > From: Samy Antoun <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Zaptel-1.2.9 compile error > 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 > > --- Bill Maidment <[EMAIL PROTECTED]> wrote: > >> Hi >> I've just tried to compile the zaptel-1.2.9 release and I get the >> following error: > > > Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when > compiling zap: > > make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found > make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found > make[3]: *** No rule to make target > `/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed > by `/usr/src/zaptel/wct4xxp/vpm450m.o'. Stop. > make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2 > make[1]: *** [_module_/usr/src/zaptel] Error 2 > make: *** [linux26] Error 2 > > Hope someone has a workaround for this problem > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > > ------------------------------ > > Message: 5 > Date: Sat, 9 Sep 2006 15:46:07 -0500 > From: "Jim Freeze" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Problems configuring Polycom 301 > 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" > > Hi > > I have successfully been running with several Polycom SoundPoint 501 > phones and recently purchased some Polycom 301 phones. > However, I can't seem to get the phones to register. The phone sees > the asterisk server, but all calls our are busy. > > The only difference for 'sip show peer xxx' for a working 501 phone and > a non working 301 phone is: > asterisk1*CLI> > > Addr->IP : 192.168.80.204 Port 5060 # 501 > > Addr->IP : (Unspecified) Port 0 # 301 > > 'sip show peers' returns: > > asterisk1*CLI> sip show peers > Name/username Host Dyn Nat ACL Port Status > 720/720 (Unspecified) D 0 UNKNOWN > 712/712 192.168.8.205 D 5060 OK (80 ms) > 711/711 192.168.8.203 D 5060 OK (84 ms) > 710/710 192.168.8.204 D 5060 OK (98 ms) > > Any 301 configuration tips would be appreciated. > > Thanks _______________________________________________ --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