Hi guys, I'm trying to write hangup causes from asterisk into the CDR record.
Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326 M:+6799344012 New Zealand www.ocis.com.au T: +649 980 7022 -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Thursday, 21 May 2009 8:28 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 58, Issue 56 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-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@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: DAHDI fun and games (Danny Nicholas) 2. Re: Step-by-Step Asterisk and MeetMe Help (Tzafrir Cohen) 3. Re: Channels configuration with DAHDI (Dave Fullerton) 4. Re: ...is circuit busy message (Jeff LaCoursiere) 5. Re: Dialplan Priorities and Sort Order... (Alex Samad) 6. Re: Step-by-Step Asterisk and MeetMe Help (Jimmy Ezell) 7. Re: Open source SIP client (marek cervenka) 8. Re: Step-by-Step Asterisk and MeetMe Help (Jonathan Thurman) 9. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business) 10. Re: Channels configuration with DAHDI (Daniel Bareiro) 11. 1.4.24.1 -> 1.6.0.9: segfault (sean darcy) 12. Voicemail playback NEWEST first vs. OLDEST first (Karl Fife) 13. Re: Step-by-Step Asterisk and MeetMe Help (Jeff LaCoursiere) 14. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business) 15. Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor() (Barry L. Kline) 16. PSTN Connection (Manoj Panicker - FOES) 17. Re: Open source SIP client (Alex Samad) 18. Re: PSTN Connection (Paul Hales) 19. interruption in queue (Rilawich Ango) 20. Re: PSTN Connection (--[ UxBoD ]--) 21. Polycom Productivity Suite (Matt Darnell) 22. Fwd: Asterisk CCM, CME Integration (Arun Kumar) ---------------------------------------------------------------------- Message: 1 Date: Wed, 20 May 2009 16:07:48 -0500 From: "Danny Nicholas" <da...@debsinc.com> Subject: Re: [asterisk-users] DAHDI fun and games To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <2897b95e2e394a7d9fad95bff31bf...@db0002> Content-Type: text/plain; charset="us-ascii" Using "r/m" because DAHDI takes 10-15 seconds to get TELCO rings. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Wednesday, May 20, 2009 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI fun and games Danny Nicholas wrote: > Hi Listers, > > I'm running 1.4.25-rc1 on opensuse 11.0 with > dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. > Incoming calls work fine. Outgoing calls made directly (exten => > s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to > let Asterisk make the call (exten => s,1,Dial(DAHDI/G1/5551212,,r). If I > use "m" (moh) the music plays 5-8 seconds after the other end picks up. > When using "r", I get 2-3 rings after other end picks up. I've went through > every flavor of dahdi-linux from 2.0.0 to 2.1.0-rc4 (which crashed me) with > no joy. Any suggestions? Hardware is Dell Poweredge 1650/1550 and > TDM410P/TDM400P. Any reason you're using the r/m option at all? Since this is an analog card I would leave the r/m off and just let asterisk use the in-band progress from the telco. -Dave _______________________________________________ -- 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 ------------------------------ Message: 2 Date: Thu, 21 May 2009 00:11:24 +0300 From: Tzafrir Cohen <tzafrir.co...@xorcom.com> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help To: asterisk-users@lists.digium.com Message-ID: <20090520211124.gm3...@xorcom.com> Content-Type: text/plain; charset=us-ascii On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: > multi-processor machine ( I had to remember to specify smp for the kernel) I repeat: why bother with such an old system? Really? Recall the comment from the book. That book had nothing really specific to Centos 4. Why do you shoot yourself in the foot by installing Centos4 now? (not to mention Zaptel) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ------------------------------ Message: 3 Date: Wed, 20 May 2009 17:12:04 -0400 From: Dave Fullerton <dfullertaster...@shorelinecontainer.com> Subject: Re: [asterisk-users] Channels configuration with DAHDI To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <4a147224.6060...@shorelinecontainer.com> Content-Type: text/plain; charset=UTF-8; format=flowed Daniel Bareiro wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > Hi Tzafrir. > > El mi?rcoles 20 de mayo del 2009 a las 10:00:46 -0300, > Tzafrir Cohen escribi?: > >> On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote: > >> Hint: you don't need to set 'signalling' for analog channels. Or just >> set it explicitly to "auto". This is for Asterisk >= 1.6.0 . Simply >> reduces the complication a bit... > > Thanks for the tip. I will remember it for when I use Asterisk 1.6 :-) > >>> I load the modules wctdm and dahdi. But when I execute in Asterisk >>> CLI "dahdi show channels", I get the following error message: >>> >>> >>> No such command 'dahdi show channels' (type 'help dahdi show' for >>> other possible commands) > >> Try running: >> >> asterisk -r >> >> and in that prompt: >> >> module unload chan_dadhi.so >> module load chan_dadhi.so >> >> and tell us the output you got. > > > # asterisk -r > Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <marks...@digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' > for details. > This is free software, with components licensed under the GNU General > Public > License version 2 and other licenses; you are welcome to redistribute it > under > certain conditions. Type 'core show license' for details. > ======================================================================== = > Connected to Asterisk 1.4.24.1 currently running on alderamin (pid = > 19777) > Verbosity is at least 7 > alderamin*CLI> > alderamin*CLI> module unload chan_dadhi.so > alderamin*CLI> module load chan_dadhi.so > [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module: > Error loading module 'chan_dadhi.so': > /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: > No such file or directory > [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module > 'chan_dadhi.so' could not be loaded. > alderamin*CLI> > > > Mmmm... it would seem to be a bug: > > /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: > No such file or directory > Sounds like DAHDI was installed/compiled *after* Asterisk was compiled. Recompile Asterisk again and make sure /usr/lib/asterisk/modules/chan_dahdi.so is created when you make install. -Dave ------------------------------ Message: 4 Date: Wed, 20 May 2009 21:16:03 +0000 (UTC) From: Jeff LaCoursiere <j...@jeff.net> Subject: Re: [asterisk-users] ...is circuit busy message To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <alpine.bsf.2.00.0905202112310.46...@phoenix.jeff.net> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Wed, 20 May 2009, John Regal wrote: > Thanks for the reply and apologize for the double post. My original post > landed in another thread and thought it may have been missed... > > I questioned my voip provider before posting and they told me they have > other asterisk customers that are making hundreds of simultaneous calls > without problems with the same account type that I have. They indicated that > they do not limit my simultaneous connections. I am now going to have them > trace my connection but hoped to learn of a possible configuration setting I > could check first. > Thanks again for the help. The message you are hearing is coming from your asterisk server, and it is because there was either no response to your call attempt or your call attempt weas refused by your VoIP provider. In the "no response" scenario it may be because you are strapped for bandwidth, but if you were THAT strapped you would have many additional problems, and the calls that were going through would have serious audio problems. So I would focus on the idea that your VoIP provider is refusing the calls that are failing. You could prove this with a packet trace. Use wireshark/tcpdump to capture your attempts and see if you can find the session that is refused. You will either see the refusal come back or the request timed out... Good luck, j ------------------------------ Message: 5 Date: Thu, 21 May 2009 07:35:26 +1000 From: Alex Samad <a...@samad.com.au> Subject: Re: [asterisk-users] Dialplan Priorities and Sort Order... To: asterisk-users@lists.digium.com Message-ID: <20090520213526.ge11...@samad.com.au> Content-Type: text/plain; charset="us-ascii" On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote: > > > Alex Samad wrote: > > On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote: > > [snip] > > > I left the busy after dial because this is what the original poster > had. In this case, if the channel does not get hungup then the next > execution will be a busy, letting the caller know the call was not > completed. In the dialplan macro I normally use it checks for the call > status and acts accordingly as seen below. If you are new to Asterisk > syntax this is probably confusing. If you are following it, I don't > exit on a BUSY because I frequently get a BUSY when there is actually a > congestion or channel problem. Anyhow, how often is a line actually > busy these days? > > exten => s,n,Set(DIALS1="IAX2/xxxxx...@carrier1-out/${ARG1},90,T") > exten => s,n,Set(DIALS2="IAX2/xxxxx...@carrier2-out/${ARG1},90,T") > exten => s,n,Set(DIALS3="SIP/${ar...@carrier3-out,90,T") > > exten => s,n,Set(DialNum=3) > exten => s,n,Set(DialCount=0) > > exten => s,n(dial),Set(DialCount=$[1 + ${DialCount}]) > exten => s,n,GotoIf($[${DialCount} > ${DialNum}]?h,1) > exten => s,n,Dial(${DIALS${DialCount}}) > exten => s,n,Goto(dial) > > exten => s-CONGESTION,1,Congestion(5) > exten => s-CONGESTION,n,Macro(rhangup) > > exten => s-BUSY,1,Playtones(busy) > exten => s-BUSY,n,Busy(5) > exten => s-BUSY,n,Macro(rhangup) > > exten => h,1,GotoIf($[${DIALSTATUS} = BUSY]?s-BUSY,1) > exten => h,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?s-CONGESTION,1) > exten => h,n,GotoIf($[${DIALSTATUS} = CONGESTION]?s-CONGESTION,1) > exten => h,n,Macro(rhangup) > > exten => t,1,Macro(rhangup) > Wow thats a neet way to dial multiple providers, can you make it into a macro and passin an array of numbers ? and maybe another param to specify how many elements in the array ? [snip] > -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/ec fe0455/attachment-0001.pgp ------------------------------ Message: 6 Date: Wed, 20 May 2009 14:36:38 -0700 From: "Jimmy Ezell" <jez...@hmhca.com> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <e77ab304d41c084090d498682873252fc1b...@nts-10.ca.hmhengineers.com> Content-Type: text/plain; charset="iso-8859-1" >On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: > >> multi-processor machine ( I had to remember to specify smp >for the kernel) > >I repeat: why bother with such an old system? Really? > >Recall the comment from the book. That book had nothing really specific >to Centos 4. Why do you shoot yourself in the foot by >installing Centos4 >now? > >(not to mention Zaptel) > >-- > Tzafrir Cohen Tzafrir thanks for the comments. I am not done playing with this and in the end I may well use newer software as you suggest. According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago) is that really consider that old? I am looking to setup a phone system that I would hope would not require any major software upgrades for many years. Jimmy > ------------------------------ Message: 7 Date: Thu, 21 May 2009 01:17:24 +0200 (CEST) From: marek cervenka <cerv...@fpf.slu.cz> Subject: Re: [asterisk-users] Open source SIP client To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <alpine.lrh.2.00.0905210116570.17...@axpsu.fpf.slu.cz> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > can anybody help me to give Opensource SIP client information which can be modified as per our requirment http://www.qutecom.org --------------------------------------- Marek Cervenka ======================================= ------------------------------ Message: 8 Date: Wed, 20 May 2009 16:33:15 -0700 From: Jonathan Thurman <jthurma...@gmail.com> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help To: asterisk-users@lists.digium.com Message-ID: <f7cbcc6e0905201633n75f67015m7143d0652c691...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" >From the front page ( http://wiki.centos.org/FrontPage ): "*What is CentOS?* CentOS is an Enterprise Linux distribution based on the freely available sources from Red Hat Enterprise Linux<ftp://ftp.redhat.com/pub/redhat/linux/enterprise/>. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer hardware. This results in a secure, low-maintenance, reliable, predictable and reproducible Linux environment." CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ): "We intend to support CentOS-4 updates until Feb 29, 2012" CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ): "We intend to support CentOS 5 until Mar 31st, 2014" So if you don't want major upgrades for a while you might want to go with the latest version. To put it into Microsoft terms... the minor version is like a service pack. So CentOS 4.7 is really a base lined version 4, service pack 7. You get the new features in major releases (like there are no more "smp" kernels in 5 to deal with) -Jonathan On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jez...@hmhca.com> wrote: > > >On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: > > > >> multi-processor machine ( I had to remember to specify smp > >for the kernel) > > > >I repeat: why bother with such an old system? Really? > > > >Recall the comment from the book. That book had nothing really specific > >to Centos 4. Why do you shoot yourself in the foot by > >installing Centos4 > >now? > > > >(not to mention Zaptel) > > > >-- > > Tzafrir Cohen > > Tzafrir thanks for the comments. I am not done playing with this and in > the end I may well use newer software as you suggest. > > According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago) is > that really consider that old? I am looking to setup a phone system that I > would hope would not require any major software upgrades for many years. > > > Jimmy > > > > _______________________________________________ > -- 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/12 dc7cf2/attachment.htm ------------------------------ Message: 9 Date: Wed, 20 May 2009 20:00:06 -0400 From: "ContactTel Business" <li...@contacttel.com> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <004901c9d9a7$19d16950$4d743b...@com> Content-Type: text/plain; charset="us-ascii" Many years in telecom and computer world is around 100 year in real life.. 10 years ago i was a millionaire in the dot com boom and 24 years old with a P2 300 computer.., 20 years ago i was military engineer and running on 3.76 MHz 386's amber screens.. last year it was dual cores, today its quad/opt cores, and tomorrow morning it's going to be quantum physics/organic computers and VOIP will be of the past, since Voice over Something else will arrive. You can't put a system and let it go for 3-4 years unless you don't have any growth, ( new drives = new technology , IDE/SATA/ISCSI) new RAM/ NEW CPU/ etc all these need software upgrades eventually.. As far as my personal experience i reformat my desktops /fully, semi annually, and all servers get a facelift every other month ( new glib for new freeswitch updates, new ZAP hardware ? then you need new zaptel.. wait zaptel aka dhadi needs X, X needs Y.. and so on.. Mike ContacTel.COM From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: May-20-09 7:33 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help >From the front page ( http://wiki.centos.org/FrontPage ): "What is CentOS? CentOS is an Enterprise Linux distribution based on the freely available <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat Enterprise Linux. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer hardware. This results in a secure, low-maintenance, reliable, predictable and reproducible Linux environment." CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ): "We intend to support CentOS-4 updates until Feb 29, 2012" CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ): "We intend to support CentOS 5 until Mar 31st, 2014" So if you don't want major upgrades for a while you might want to go with the latest version. To put it into Microsoft terms... the minor version is like a service pack. So CentOS 4.7 is really a base lined version 4, service pack 7. You get the new features in major releases (like there are no more "smp" kernels in 5 to deal with) -Jonathan On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jez...@hmhca.com> wrote: >On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: > >> multi-processor machine ( I had to remember to specify smp >for the kernel) > >I repeat: why bother with such an old system? Really? > >Recall the comment from the book. That book had nothing really specific >to Centos 4. Why do you shoot yourself in the foot by >installing Centos4 >now? > >(not to mention Zaptel) > >-- > Tzafrir Cohen Tzafrir thanks for the comments. I am not done playing with this and in the end I may well use newer software as you suggest. According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago) is that really consider that old? I am looking to setup a phone system that I would hope would not require any major software upgrades for many years. Jimmy > _______________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/05 1c70dd/attachment.htm ------------------------------ Message: 10 Date: Wed, 20 May 2009 21:19:18 -0300 From: Daniel Bareiro <daniel-lis...@gmx.net> Subject: Re: [asterisk-users] Channels configuration with DAHDI To: asterisk-users@lists.digium.com Message-ID: <slrnh197i3.3j6.daniel-lis...@marian.freesoftware.org> Content-Type: text/plain; charset=UTF-8 -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Dave. El mi?rcoles 20 de mayo del 2009 a las 18:12:04 -0300, Dave Fullerton escribi?: >>>> I load the modules wctdm and dahdi. But when I execute in Asterisk >>>> CLI "dahdi show channels", I get the following error message: >>>> >>>> >>>> No such command 'dahdi show channels' (type 'help dahdi show' for >>>> other possible commands) >>> Try running: >>> >>> asterisk -r >>> >>> and in that prompt: >>> >>> module unload chan_dadhi.so >>> module load chan_dadhi.so >>> >>> and tell us the output you got. >> # asterisk -r >> Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. >> Created by Mark Spencer <marks...@digium.com> >> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' >> for details. >> This is free software, with components licensed under the GNU General >> Public >> License version 2 and other licenses; you are welcome to redistribute it >> under >> certain conditions. Type 'core show license' for details. >> ======================================================================== = >> Connected to Asterisk 1.4.24.1 currently running on alderamin (pid = >> 19777) >> Verbosity is at least 7 >> alderamin*CLI> >> alderamin*CLI> module unload chan_dadhi.so >> alderamin*CLI> module load chan_dadhi.so >> [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module: >> Error loading module 'chan_dadhi.so': >> /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: >> No such file or directory >> [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module >> 'chan_dadhi.so' could not be loaded. >> alderamin*CLI> >> >> >> Mmmm... it would seem to be a bug: >> >> /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: >> No such file or directory > Sounds like DAHDI was installed/compiled *after* Asterisk was > compiled. Recompile Asterisk again and make sure > /usr/lib/asterisk/modules/chan_dahdi.so is created when you make > install. Mmmm... but I believe that it had done already in that order. In fact, I reviewed the existence of the module and it was in the directory. For that reasonI said that perhaps it was bug by the following thing: [May 20 20:49:07] WARNING[23599]: loader.c:359 load_dynamic_module: Error loading module 'chan_dadhi.so': /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file: ^^^^^^^^^^^^^ No such file or directory [May 20 20:49:07] WARNING[23599]: loader.c:653 load_resource: Module 'chan_dadhi.so' could not be loaded. Apparently Asterisk is looking for the module using an incorrect name. Whatever happens, I compile Asterisk again but I got the same error message. Thanks for your reply. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoUnkMACgkQZpa/GxTmHTfKeACffr1Q2vgJnyDVrA+hCN3DtHd5 e4UAoJUSl6JjjMCX+SiUA2/cyHJhwtfN =wCSV -----END PGP SIGNATURE----- ------------------------------ Message: 11 Date: Wed, 20 May 2009 20:22:16 -0400 From: sean darcy <seandar...@gmail.com> Subject: [asterisk-users] 1.4.24.1 -> 1.6.0.9: segfault To: asterisk-users@lists.digium.com Message-ID: <gv26rp$38...@ger.gmane.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed I'm testing an upgrade of an i686 production machine running 1.4.24.1 to 1.6.0.9. I've installed dahdi-linux-2.1.0.4. But: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <marks...@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ======================================================================== = == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/logger.conf': == Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found == Parsing '/etc/asterisk/dnsmgr.conf': == Found == Parsing '/etc/asterisk/http.conf': == Found ................ == Parsing '/etc/asterisk/manager.conf': == Found [May 20 18:43:54] NOTICE[750]: manager.c:3903 __init_manager: Invalid keyword <displaysystemname> = <yes> in manager.conf [general ........ == Parsing '/etc/asterisk/smdi.conf': == Found [May 20 18:43:54] NOTICE[750]: res_smdi.c:1272 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. ........... == Parsing '/etc/asterisk/musiconhold.conf': == Found [May 20 18:43:54] WARNING[750]: res_musiconhold.c:1496 load_moh_classes: A directory must be specified for class 'default'! [May 20 18:43:54] WARNING[750]: res_musiconhold.c:1657 load_module: No music on hold classes configured, disabling music on hold. == Registered application 'MusicOnHold' ............... == Registered application 'DateTime' app_sayunixtime.so => (Say time) == Registered application 'SetCallerPres' app_setcallerid.so => (Set CallerID Presentation Application) == Registered file format gsm, extension(s) gsm format_gsm.so => (Raw GSM data) == Registered application 'BackgroundDetect' app_talkdetect.so => (Playback with Talk Detection) Segmentation fault strace was little help: strace asterisk -c ....... .open("/usr/lib/asterisk/modules/format_gsm.so", O_RDONLY) = 12 read(12, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0 \10\0\000"..., 512) = 512 fstat64(12, {st_mode=S_IFREG|0755, st_size=150128, ...}) = 0 mmap2(NULL, 16240, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 12, 0) = 0xb7364000 mmap2(0xb7367000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 12, 0x2) = 0xb7367000 close(12) = 0 gettimeofday({1242859458, 600093}, NULL) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) = 0 gettid() = 921 futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1 futex(0x8195988, FUTEX_WAKE, 1) = 1 futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1 futex(0x8195988, FUTEX_WAKE, 1) = 1 .open("/usr/lib/asterisk/modules/app_talkdetect.so", O_RDONLY) = 12 read(12, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\v\0"..., 512) = 512 fstat64(12, {st_mode=S_IFREG|0755, st_size=155069, ...}) = 0 mmap2(NULL, 12176, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 12, 0) = 0xb7361000 mmap2(0xb7363000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 12, 0x1) = 0xb7363000 close(12) = 0 gettimeofday({1242859458, 601194}, NULL) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) = 0 stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) = 0 gettid() = 921 futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1 futex(0x8195988, FUTEX_WAKE, 1) = 1 futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1 futex(0x8195988, FUTEX_WAKE, 1) = 1 .open("/usr/lib/asterisk/modules/app_random.so", O_RDONLY) = 12 read(12, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\260\10"..., 512) = 512 fstat64(12, {st_mode=S_IFREG|0755, st_size=135210, ...}) = 0 mmap2(NULL, 9356, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 12, 0) = 0xb735e000 mmap2(0xb7360000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 12, 0x1) = 0xb7360000 close(12) = 0 --- SIGSEGV (Segmentation fault) @ 0 (0) --- +++ killed by SIGSEGV +++ Process 921 detached Anyone else seen this? sean ------------------------------ Message: 12 Date: Wed, 20 May 2009 19:28:19 -0500 From: "Karl Fife" <karlf...@gmail.com> Subject: [asterisk-users] Voicemail playback NEWEST first vs. OLDEST first To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <bc16df1437384d7c88be5da5ecbf8...@kfife2> Content-Type: text/plain; charset="iso-8859-1" Is there a way to make the asterisk voicemail app play back messages in NEWEST FIRST order, instead of OLDEST FIRST? I see the situation repeatedly where someone needs to dip into their voicemail archive to get something from a recently saved voicemail message, and they have to slog through lots of irrelevant stuff to get there. I have seen this question come up previously on this list without an answer. I'm hoping that someone can shed light on how to do it, or confirm that it is NOT currently supported. I've looked at the new 1.6 voicemail.conf and it doesn't seem have any parameters that speak to that feature, nor an voicemailmain parameter in 1.4 or 1.6. Can anyone confirm that this is not supported, or enlighten us on how-to? Many thanks. -Karl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/00 8201e2/attachment-0001.htm ------------------------------ Message: 13 Date: Thu, 21 May 2009 00:42:46 +0000 (UTC) From: Jeff LaCoursiere <j...@jeff.net> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <alpine.bsf.2.00.0905210041350.46...@phoenix.jeff.net> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed So you were fourteen and a military engineer? j On Wed, 20 May 2009, ContactTel Business wrote: > Many years in telecom and computer world is around 100 year in real life.. > 10 years ago i was a millionaire in the dot com boom and 24 years old with a > P2 300 computer.., 20 years ago i was military engineer and running on 3.76 > MHz 386's amber screens.. last year it was dual cores, today its quad/opt > cores, and tomorrow morning it's going to be quantum physics/organic > computers and VOIP will be of the past, since Voice over Something else will > arrive. > > > > You can't put a system and let it go for 3-4 years unless you don't have any > growth, ( new drives = new technology , IDE/SATA/ISCSI) new RAM/ NEW CPU/ > etc all these need software upgrades eventually.. > > > > As far as my personal experience i reformat my desktops /fully, semi > annually, and all servers get a facelift every other month ( new glib for > new freeswitch updates, new ZAP hardware ? then you need new zaptel.. wait > zaptel aka dhadi needs X, X needs Y.. and so on.. > > > > Mike > > ContacTel.COM > > > > > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan > Thurman > Sent: May-20-09 7:33 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help > > > > From the front page ( http://wiki.centos.org/FrontPage ): > > "What is CentOS? > CentOS is an Enterprise Linux distribution based on the freely available > <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat > Enterprise Linux. Each CentOS version is supported for 7 years (by means of > security updates). A new CentOS version is released every 2 years and each > CentOS version is regularly updated (every 6 months) to support newer > hardware. This results in a secure, low-maintenance, reliable, predictable > and reproducible Linux environment." > > CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ): > "We intend to support CentOS-4 updates until Feb 29, 2012" > > CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ): > "We intend to support CentOS 5 until Mar 31st, 2014" > > > So if you don't want major upgrades for a while you might want to go with > the latest version. To put it into Microsoft terms... the minor version is > like a service pack. So CentOS 4.7 is really a base lined version 4, > service pack 7. You get the new features in major releases (like there are > no more "smp" kernels in 5 to deal with) > > -Jonathan > > > > On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jez...@hmhca.com> wrote: > > >> On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: >> >>> multi-processor machine ( I had to remember to specify smp >> for the kernel) >> >> I repeat: why bother with such an old system? Really? >> >> Recall the comment from the book. That book had nothing really specific >> to Centos 4. Why do you shoot yourself in the foot by >> installing Centos4 >> now? >> >> (not to mention Zaptel) >> >> -- >> Tzafrir Cohen > > Tzafrir thanks for the comments. I am not done playing with this and in the > end I may well use newer software as you suggest. > > According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago) is > that really consider that old? I am looking to setup a phone system that I > would hope would not require any major software upgrades for many years. > > > Jimmy > >> > > _______________________________________________ > -- 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 > > > > ------------------------------ Message: 14 Date: Wed, 20 May 2009 21:02:21 -0400 From: "ContactTel Business" <li...@contacttel.com> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <007101c9d9af$cc4f19d0$64ed4d...@com> Content-Type: text/plain; charset="us-ascii" Hehe i meant 15 but i knew one would spot that.. I was 17 in fact, left at 22, yeah demolition, construction, sniper, road demolish, anti tank craters, and all the bells and whistles, >>-----Original Message----- >>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- >>boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere >>Sent: May-20-09 8:43 PM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help >> >> >>So you were fourteen and a military engineer? >> >>j >> >>On Wed, 20 May 2009, ContactTel Business wrote: >> >>> Many years in telecom and computer world is around 100 year in real >>life.. >>> 10 years ago i was a millionaire in the dot com boom and 24 years old >>with a >>> P2 300 computer.., 20 years ago i was military engineer and running >>on 3.76 >>> MHz 386's amber screens.. last year it was dual cores, today its >>quad/opt >>> cores, and tomorrow morning it's going to be quantum physics/organic >>> computers and VOIP will be of the past, since Voice over Something >>else will >>> arrive. >>> >>> >>> >>> You can't put a system and let it go for 3-4 years unless you don't >>have any >>> growth, ( new drives = new technology , IDE/SATA/ISCSI) new RAM/ NEW >>CPU/ >>> etc all these need software upgrades eventually.. >>> >>> >>> >>> As far as my personal experience i reformat my desktops /fully, semi >>> annually, and all servers get a facelift every other month ( new glib >>for >>> new freeswitch updates, new ZAP hardware ? then you need new zaptel.. >>wait >>> zaptel aka dhadi needs X, X needs Y.. and so on.. >>> >>> >>> >>> Mike >>> >>> ContacTel.COM >>> >>> >>> >>> >>> >>> >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of >>Jonathan >>> Thurman >>> Sent: May-20-09 7:33 PM >>> To: asterisk-users@lists.digium.com >>> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help >>> >>> >>> >>> From the front page ( http://wiki.centos.org/FrontPage ): >>> >>> "What is CentOS? >>> CentOS is an Enterprise Linux distribution based on the freely >>available >>> <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red >>Hat >>> Enterprise Linux. Each CentOS version is supported for 7 years (by >>means of >>> security updates). A new CentOS version is released every 2 years and >>each >>> CentOS version is regularly updated (every 6 months) to support newer >>> hardware. This results in a secure, low-maintenance, reliable, >>predictable >>> and reproducible Linux environment." >>> >>> CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ): >>> "We intend to support CentOS-4 updates until Feb 29, 2012" >>> >>> CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ): >>> "We intend to support CentOS 5 until Mar 31st, 2014" >>> >>> >>> So if you don't want major upgrades for a while you might want to go >>with >>> the latest version. To put it into Microsoft terms... the minor >>version is >>> like a service pack. So CentOS 4.7 is really a base lined version 4, >>> service pack 7. You get the new features in major releases (like >>there are >>> no more "smp" kernels in 5 to deal with) >>> >>> -Jonathan >>> >>> >>> >>> On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jez...@hmhca.com> >>wrote: >>> >>> >>>> On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote: >>>> >>>>> multi-processor machine ( I had to remember to specify smp >>>> for the kernel) >>>> >>>> I repeat: why bother with such an old system? Really? >>>> >>>> Recall the comment from the book. That book had nothing really >>specific >>>> to Centos 4. Why do you shoot yourself in the foot by >>>> installing Centos4 >>>> now? >>>> >>>> (not to mention Zaptel) >>>> >>>> -- >>>> Tzafrir Cohen >>> >>> Tzafrir thanks for the comments. I am not done playing with this and >>in the >>> end I may well use newer software as you suggest. >>> >>> According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months >>ago) is >>> that really consider that old? I am looking to setup a phone system >>that I >>> would hope would not require any major software upgrades for many >>years. >>> >>> >>> Jimmy >>> >>>> >>> >>> _______________________________________________ >>> -- 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 >>> >>> >>> >>> >> >>_______________________________________________ >>-- 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 ------------------------------ Message: 15 Date: Wed, 20 May 2009 21:52:18 -0400 From: "Barry L. Kline" <blkl...@attglobal.net> Subject: [asterisk-users] Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor() To: asterisk-users@lists.digium.com Message-ID: <4a14b3d2.5050...@attglobal.net> Content-Type: text/plain; charset=ISO-8859-1 I wrote a note earlier about this problem but have done quite a bit more debugging. Now I'm stuck at what to do next. I have inbound calls being answered by our Asterisk box, which then dials our answering service and bridges those calls. The inbound and outbound are both PRIs. The answering service takes our calls on a PRI. If I don't use the Monitor() application, things work find and have been for a few thousand calls. If I add the Monitor() application, no audio ever gets passed from the caller to the answering service. I have noted the following things while testing with Monitor(): 1) If I have it call my cell phone instead of the service, it works fine. 2) If I have it call my home phone instead of the services, it works fine. 3) I tried calling another number (in another state) that I know terminates into a PRI and it worked fine. 4) If I call the service without Monitor(), it works fine. Throw in Monitor() and it's virtually guaranteed not to work. My dial plan and debug output for both the working and failing call is at http://www.pastebin.ca/1429504 . Things start to diverge around lines 28-31 and 68-72. Can anyone tell me what I can do to further trace this problem? Thanks in advance for anything you may be able to offer. Barry ------------------------------ Message: 16 Date: Thu, 21 May 2009 07:06:02 +0400 From: "Manoj Panicker - FOES" <manoj.panic...@emirates.com> Subject: [asterisk-users] PSTN Connection To: <asterisk-users@lists.digium.com> Message-ID: <ac5f42f85475254aab74b5a2b0653e4401f76...@dxbhqmbex10.corp.emirates.com> Content-Type: text/plain; charset="us-ascii" Hi Which is the best interface card to connect PSTN line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/84 91ae86/attachment-0001.htm ------------------------------ Message: 17 Date: Thu, 21 May 2009 14:13:31 +1000 From: Alex Samad <a...@samad.com.au> Subject: Re: [asterisk-users] Open source SIP client To: asterisk-users@lists.digium.com Message-ID: <20090521041330.ga30...@samad.com.au> Content-Type: text/plain; charset="us-ascii" On Tue, May 19, 2009 at 10:38:24AM +1000, Paul Hales wrote: > > Not true. I am always wrong. > (wait...is that a paradox?) only on the 42nd time > > PaulH > > [snip] > ContactTel Business wrote: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/fc 197395/attachment-0001.pgp ------------------------------ Message: 18 Date: Thu, 21 May 2009 14:20:21 +1000 From: Paul Hales <pdha...@optusnet.com.au> Subject: Re: [asterisk-users] PSTN Connection To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <4a14d685.2040...@optusnet.com.au> Content-Type: text/plain; charset=ISO-8859-1 Digium PSTN cards seem to work. PaulH Manoj Panicker - FOES wrote: > > Hi > Which is the best interface card to connect* PSTN* line with > Asterisk. Can somebody please help. My intention is to route the > incoming PSTN calls to internal IP Phones through Asterisk and Vice > versa. The Asterisk is in LAN and is reachable from all the IP phones > in the LAN. > > Thanks > Manoj > > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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 ------------------------------ Message: 19 Date: Thu, 21 May 2009 14:10:18 +0800 From: Rilawich Ango <maillist...@gmail.com> Subject: [asterisk-users] interruption in queue To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <6fbb529e0905202310u42d75f2boe5bec86b81bb4...@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 HI, I want to allow user to press 0 to the voicemail if the user don't want to wait in the queue. Below is what I set but it doesn't work. Anyone can help? ango file: features.conf [applicationmap] opervm => 0,self/both,Macro,opervm file: extensions.conf ... exten => 5555,n(queue),Set(DYNAMIC_FEATURES=opervm) exten => 5555,n,Queue(5555|tThH|||180) ... [macro-opervm] exten => s,1,NoOp(--openvm--) exten => s,n,VoiceMail(3...@default,u) exten => s,n,Hangup ------------------------------ Message: 20 Date: Thu, 21 May 2009 09:01:39 +0100 (BST) From: "--[ UxBoD ]--" <ux...@splatnix.net> Subject: Re: [asterisk-users] PSTN Connection To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <30533477.221242892899791.javamail.r...@office.splatnix.net> Content-Type: text/plain; charset=utf-8 ----- "Paul Hales" <pdha...@optusnet.com.au> wrote: > Digium PSTN cards seem to work. > > > > PaulH OpenVox works well. Best Regards, -- SplatNIX IT Services :: Innovation through collaboration ------------------------------ Message: 21 Date: Wed, 20 May 2009 22:04:21 -1000 From: Matt Darnell <mattdarn...@gmail.com> Subject: [asterisk-users] Polycom Productivity Suite To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <52b8ace90905210104u4239b2e2t3a539b34d7559...@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Has anyone been able to do the following: 1. Set the phone to automatically record all calls to the USB stick, now you have to press three keys. 2. Put Record on the main screen when a call is active. This would eliminate having to press the 'more' softkey. Thanks, Matt ------------------------------ Message: 22 Date: Thu, 21 May 2009 13:58:10 +0530 From: Arun Kumar <arunv...@gmail.com> Subject: [asterisk-users] Fwd: Asterisk CCM, CME Integration To: "ccie_vo...@onlinestudylist.com" <ccie_vo...@onlinestudylist.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, Commercial and Business-Oriented Asterisk Discussion <asterisk-...@lists.digium.com> Message-ID: <a70a109b0905210128u32b3a41cude2dc79189178...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi All, please provide some help. I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in the above setup I'm able to call from Asterisk to my CME and vice-versa. here is my problem: when I call from 6004 to my cme extension 4615, on 4615 I've configured noans timeout to 15 and then it goes to my unity express (cue) for voicemail so when I call my cme extension it rings for few seconds and then on my asterisk cli I see "500 Internal Server Error" back from my CCM IP and getting standard asterisk message saying "all circuits are busy now" . as per my understanding it should go to my cue. please advise and let me know if you need any other details. Regards Arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/12 12870a/attachment.htm ------------------------------ _______________________________________________ --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 End of asterisk-users Digest, Vol 58, Issue 56 ********************************************** _______________________________________________ -- 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