[asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
Hi, I'm trying to connect two asterisk instances using the method described here.. http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html under the section Connecting two Asterisk systems together with SIP I have an user named venu in serverA and vijay in serverB the serverA ip is 192.168.0.35 serverB is 192.168.0.36 Here are the details of the config files (extension sip): http://paste.kde.org/737888 When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the *Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)* but I want to make a call to vijay.. can anyone please let me know where I am going wrong? I have the same error when I try to make a call from sip client to a analog phone in a single server asterisk setup... :-\ I'm running Asterisk 11.3 on Ubuntu 12.04 on a KVM virtualized instance.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
@Alec, Thanks.. That was the error.. got it working now.. :) On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sandeep Raju Sent: Sunday, 5 May 2013 8:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP' snip When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but I want to make a call to vijay.. can anyone please let me know where I am going wrong? The clue is 21.-- Executing [999@incoming:2] Dial(SIP/serverA-0004, SIP/vijay@serverB) in new stack 24. getaddrinfo(serverB, (null), ...): Name or service not known 25. No such host: serverB I believe extension 999 in server B is wrong. It should be; # extensions.conf in serverB [incoming] exten = 999,1,Answer() exten = 999,n,Dial(SIP/vijay) exten = 999,n,HangUp() Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP'
@Alec, Now I can dial user vijay but the call gets cut after a few seconds and i get this error in the serverA's console.. http://paste.kde.org/737924 PS: recolgo is the hostname of the system from which I am initialting the call (using a sip client) Thanks On Sun, May 5, 2013 at 2:41 PM, Sandeep Raju sandeepr...@practo.com wrote: @Alec, Thanks.. That was the error.. got it working now.. :) On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nzwrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sandeep Raju Sent: Sunday, 5 May 2013 8:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connecting Multiple Asterisk instances getting Unable to create channel of type 'SIP' snip When i make a call to extension 998 in using user as venu, here is the output i get.. http://paste.kde.org/737894 The problem is that, I'm getting the Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but I want to make a call to vijay.. can anyone please let me know where I am going wrong? The clue is 21.-- Executing [999@incoming:2] Dial(SIP/serverA-0004, SIP/vijay@serverB) in new stack 24. getaddrinfo(serverB, (null), ...): Name or service not known 25. No such host: serverB I believe extension 999 in server B is wrong. It should be; # extensions.conf in serverB [incoming] exten = 999,1,Answer() exten = 999,n,Dial(SIP/vijay) exten = 999,n,HangUp() Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Hi, I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed the official user manual and the blog post here http://www.skelleton.net/2012/08/02/linksys-spa-3102/ When I call an extension say 225 from the analog phone, I can get the IVR I have setup in my dialplan. But when I Call the analog phone extension using a sip phone I get the following error message: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) If any more information is required, i'd be glad to post it here. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on Virtual Machine
Hi Hans, If we use the pre-built packages on say ubuntu (my server os), can i enable options like when i do when i compile and do a menuselect? I mean can i enable the cdr odbc, del odbc etc modules that I need? On Tue, Apr 23, 2013 at 1:13 PM, Hans Witvliet aster...@a-domani.nl wrote: Could it be distro-related? I have various versions of asterisk (from 1.4 upto 11.3) running paravirtualized or HW-virtualized with XEN. Normally i use the pre-build packages from suse, only when i want to try a release-candidates i need them myself. hw -Original Message- From: Sandeep Raju sandeepr...@practo.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Installing Asterisk on Virtual Machine Date: Tue, 23 Apr 2013 10:18:00 +0530 Hi Tzafrir, I have installed Asterisk 11.2 on ubuntu 12.04 64 bit server instance running on my private openstack cloud. My bare machine is Intel® Core™ i7-2600K CPU @ 3.40GHz × 8 with 8GB ram and running at 64 bit ubuntu 12.04 desktop edition with Kernel Linux 3.2.0-23-generic. output of uname -a on my ubuntu cloud instance where i'm trying to setup asterisk.. Linux asterisk 3.2.0-23-virtual #36-Ubuntu SMP Tue Apr 10 22:29:03 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux Here is my backtrace.. http://paste.kde.org/730316/ Sorry for the late reply... On Mon, Apr 22, 2013 at 5:13 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote: Hi, I'm trying to install Asterisk 11.2 on a virtual machine in my private opestack cloud.. When I compile Asterisk 11.2 from source (./configure, make, make install) as specified in the Asterisk book and run it, it gives me the error: Illegal instruction (core dumped). Any ideas how I can solve this? What operating system do you have installed there? What CPU? What is the output of: uname -a Illegal instruction means that you tried running an instruction that the CPU cann't run. Maybe an incorrect choice of optimization flags? Maybe this is due to libraries not matching your architecture? Next thing to do: get a trace from the core file that was dumped using gdb. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on Virtual Machine
@Hans, Now I feel its distro related as I am getting the same error when I try to compile and run asterisk 1.8.. what distro are you using? I think I need to change the distro I'm running on.. On Tue, Apr 23, 2013 at 1:44 PM, Sandeep Raju sandeepr...@practo.comwrote: Hi Hans, If we use the pre-built packages on say ubuntu (my server os), can i enable options like when i do when i compile and do a menuselect? I mean can i enable the cdr odbc, del odbc etc modules that I need? On Tue, Apr 23, 2013 at 1:13 PM, Hans Witvliet aster...@a-domani.nlwrote: Could it be distro-related? I have various versions of asterisk (from 1.4 upto 11.3) running paravirtualized or HW-virtualized with XEN. Normally i use the pre-build packages from suse, only when i want to try a release-candidates i need them myself. hw -Original Message- From: Sandeep Raju sandeepr...@practo.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Installing Asterisk on Virtual Machine Date: Tue, 23 Apr 2013 10:18:00 +0530 Hi Tzafrir, I have installed Asterisk 11.2 on ubuntu 12.04 64 bit server instance running on my private openstack cloud. My bare machine is Intel® Core™ i7-2600K CPU @ 3.40GHz × 8 with 8GB ram and running at 64 bit ubuntu 12.04 desktop edition with Kernel Linux 3.2.0-23-generic. output of uname -a on my ubuntu cloud instance where i'm trying to setup asterisk.. Linux asterisk 3.2.0-23-virtual #36-Ubuntu SMP Tue Apr 10 22:29:03 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux Here is my backtrace.. http://paste.kde.org/730316/ Sorry for the late reply... On Mon, Apr 22, 2013 at 5:13 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote: Hi, I'm trying to install Asterisk 11.2 on a virtual machine in my private opestack cloud.. When I compile Asterisk 11.2 from source (./configure, make, make install) as specified in the Asterisk book and run it, it gives me the error: Illegal instruction (core dumped). Any ideas how I can solve this? What operating system do you have installed there? What CPU? What is the output of: uname -a Illegal instruction means that you tried running an instruction that the CPU cann't run. Maybe an incorrect choice of optimization flags? Maybe this is due to libraries not matching your architecture? Next thing to do: get a trace from the core file that was dumped using gdb. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on Virtual Machine
@Hans, I just tried installing from pre-built packages (which has asterisk 1.8). Its working fine! :) only the compiled installed versions were giving me the error!.. PS: sorry for spamming with multiple mails.. On Tue, Apr 23, 2013 at 2:10 PM, Sandeep Raju sandeepr...@practo.comwrote: @Hans, Now I feel its distro related as I am getting the same error when I try to compile and run asterisk 1.8.. what distro are you using? I think I need to change the distro I'm running on.. On Tue, Apr 23, 2013 at 1:44 PM, Sandeep Raju sandeepr...@practo.comwrote: Hi Hans, If we use the pre-built packages on say ubuntu (my server os), can i enable options like when i do when i compile and do a menuselect? I mean can i enable the cdr odbc, del odbc etc modules that I need? On Tue, Apr 23, 2013 at 1:13 PM, Hans Witvliet aster...@a-domani.nlwrote: Could it be distro-related? I have various versions of asterisk (from 1.4 upto 11.3) running paravirtualized or HW-virtualized with XEN. Normally i use the pre-build packages from suse, only when i want to try a release-candidates i need them myself. hw -Original Message- From: Sandeep Raju sandeepr...@practo.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Installing Asterisk on Virtual Machine Date: Tue, 23 Apr 2013 10:18:00 +0530 Hi Tzafrir, I have installed Asterisk 11.2 on ubuntu 12.04 64 bit server instance running on my private openstack cloud. My bare machine is Intel® Core™ i7-2600K CPU @ 3.40GHz × 8 with 8GB ram and running at 64 bit ubuntu 12.04 desktop edition with Kernel Linux 3.2.0-23-generic. output of uname -a on my ubuntu cloud instance where i'm trying to setup asterisk.. Linux asterisk 3.2.0-23-virtual #36-Ubuntu SMP Tue Apr 10 22:29:03 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux Here is my backtrace.. http://paste.kde.org/730316/ Sorry for the late reply... On Mon, Apr 22, 2013 at 5:13 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote: Hi, I'm trying to install Asterisk 11.2 on a virtual machine in my private opestack cloud.. When I compile Asterisk 11.2 from source (./configure, make, make install) as specified in the Asterisk book and run it, it gives me the error: Illegal instruction (core dumped). Any ideas how I can solve this? What operating system do you have installed there? What CPU? What is the output of: uname -a Illegal instruction means that you tried running an instruction that the CPU cann't run. Maybe an incorrect choice of optimization flags? Maybe this is due to libraries not matching your architecture? Next thing to do: get a trace from the core file that was dumped using gdb. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] Installing Asterisk on Virtual Machine
@Tzafrir, I uninstalled the version 11.2 and compiled the version 1.8.12.2 as mentioned in that page... its working fine now.. as my virtual machine was running on KVM.. i think i faced the same issue mentioned in that issue report.. I even went further and uninstalled 1.8.12.2 and install 1.8.22 and again the problem was back.. so, i think the problem is same as the one in the issue... On Tue, Apr 23, 2013 at 6:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Apr 23, 2013 at 02:17:47PM +0530, Sandeep Raju wrote: @Hans, I just tried installing from pre-built packages (which has asterisk 1.8). Its working fine! :) only the compiled installed versions were giving me the error!.. PS: sorry for spamming with multiple mails.. Distro packages naturally disable BUILD_NATIVE. In the Debian package build rules: # Make sure the configure script gets an CFLAGS parameter. Otherwise # it will build with -march=native What is the minimal code that will get asterisk crash on your system when built with -march=native? It would b einteresting to make this an autoconf test (see the existing test for NATIVE on configure.ac). The bug report notes that this is a gcc issue, but I don't see any link to a gcc bug report anywhere. Here we have gcc 4:4.6.3-1ubuntu5 (right? That what I got from packages.ubuntu.com) still buggy. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on Virtual Machine
my gcc version is as follows gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3 On Tue, Apr 23, 2013 at 6:12 PM, Sandeep Raju sandeepr...@practo.comwrote: @Tzafrir, I uninstalled the version 11.2 and compiled the version 1.8.12.2 as mentioned in that page... its working fine now.. as my virtual machine was running on KVM.. i think i faced the same issue mentioned in that issue report.. I even went further and uninstalled 1.8.12.2 and install 1.8.22 and again the problem was back.. so, i think the problem is same as the one in the issue... On Tue, Apr 23, 2013 at 6:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Apr 23, 2013 at 02:17:47PM +0530, Sandeep Raju wrote: @Hans, I just tried installing from pre-built packages (which has asterisk 1.8). Its working fine! :) only the compiled installed versions were giving me the error!.. PS: sorry for spamming with multiple mails.. Distro packages naturally disable BUILD_NATIVE. In the Debian package build rules: # Make sure the configure script gets an CFLAGS parameter. Otherwise # it will build with -march=native What is the minimal code that will get asterisk crash on your system when built with -march=native? It would b einteresting to make this an autoconf test (see the existing test for NATIVE on configure.ac). The bug report notes that this is a gcc issue, but I don't see any link to a gcc bug report anywhere. Here we have gcc 4:4.6.3-1ubuntu5 (right? That what I got from packages.ubuntu.com) still buggy. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing Asterisk on Virtual Machine
Hi, I'm trying to install Asterisk 11.2 on a virtual machine in my private opestack cloud.. When I compile Asterisk 11.2 from source (./configure, make, make install) as specified in the Asterisk book and run it, it gives me the error: Illegal instruction (core dumped). Any ideas how I can solve this? Thanks srp_ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on Virtual Machine
@Doug, Yes.. I can compile other applications. I discussed this issue on the #asterisk irc and they pointed me to this, https://issues.asterisk.org/jira/browse/ASTERISK-20128 I think the issue is with my asterisk version (which is 11.2)... not sure though! Any help would be grateful :) On Mon, Apr 22, 2013 at 4:42 PM, Doug Lytle supp...@drdos.info wrote: it gives me the error: Illegal instruction (core dumped). Doesn't sound like you have a stable environment. Can you compile other applications without a core dump? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on Virtual Machine
Hi Tzafrir, I have installed Asterisk 11.2 on ubuntu 12.04 64 bit server instance running on my private openstack cloud. My bare machine is Intel® Core™ i7-2600K CPU @ 3.40GHz × 8 with 8GB ram and running at 64 bit ubuntu 12.04 desktop edition with Kernel Linux 3.2.0-23-generic. output of uname -a on my ubuntu cloud instance where i'm trying to setup asterisk.. Linux asterisk 3.2.0-23-virtual #36-Ubuntu SMP Tue Apr 10 22:29:03 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux Here is my backtrace.. http://paste.kde.org/730316/ Sorry for the late reply... On Mon, Apr 22, 2013 at 5:13 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote: Hi, I'm trying to install Asterisk 11.2 on a virtual machine in my private opestack cloud.. When I compile Asterisk 11.2 from source (./configure, make, make install) as specified in the Asterisk book and run it, it gives me the error: Illegal instruction (core dumped). Any ideas how I can solve this? What operating system do you have installed there? What CPU? What is the output of: uname -a Illegal instruction means that you tried running an instruction that the CPU cann't run. Maybe an incorrect choice of optimization flags? Maybe this is due to libraries not matching your architecture? Next thing to do: get a trace from the core file that was dumped using gdb. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forwarding-in india
Hi All, Can any body tell how to enable call forward facility in INDAI for an asterisk IPPBX. Regards, Sandeep.S___ -- 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
[asterisk-users] call forward facility in INDIA
Hi All, Can any body tell how to enable call forward facility in INDIA for an asterisk IPPBX. Regards, Sandeep.S___ -- 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
[asterisk-users] (no subject)
hi, how to write a advanced dial plan for example: dial to a extension(123).if the user didnot pick the call, caller should get a ivr script(Enter 1 to to dial operator and 2 to go to voicemail) If caller press 1 it should dial to the operator,else if he dials 2 it should go to the voicemail of calle's extension. thanks sandeep.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 43, Issue 30
hi all,
how to establish a call between two asterisk servers for the sip users
registered for the servers.
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 10, 2008 11:30 PM
Subject: asterisk-users Digest, Vol 43, Issue 30
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
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or, via email, send a message with subject or body 'help' to
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You can reach the person managing the list at
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When replying, please edit your Subject line so it is more specific
than Re: Contents of asterisk-users digest...
Today's Topics:
1. Re: Domainname for outgoing uri-dialing (B. Haje)
2. Re: oneway audio with asterisk behind cisco pix 506 (Adam KOSA)
3. Re: Asterisk Scalability (Bryan M. Johns)
--
Message: 1
Date: Sun, 10 Feb 2008 18:11:01 +0100
From: B. Haje [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Domainname for outgoing uri-dialing
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
[EMAIL PROTECTED] wrote:
8 feb 2008 kl. 13.24 skrev Bjoern Haje:
Hi,
I use outgoing URI-dialing for my sip-phones as suggested in
http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
The relevant extensions look like this:
[dial-uri]
exten = _[a-z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _[A-Z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,1,Macro(uridial,[EMAIL PROTECTED])
[macro-uridial]
exten = s,1,Set(dialuri=${CUT(ARG1,\;,1)})
exten = s,n,Set(CALLERID(number)=${CALLERID(number)[EMAIL PROTECTED])
exten = s,n,Dial(SIP/${dialuri},120,tr)
exten = s,n,Congestion()
I end up with an outgoing SIP-Invite with contact and from-headers
like [EMAIL PROTECTED]@IP-address
That obviously is not what I want. I can set the fromdomain value in
the general-part of my sip.conf and leave away the setting of the
callerid which fixes the problem. But as I want to use different
domains for the outgoing calls depending on the user, that is not a
solution for me. Can I influence the generation of the outgoing
domainname somehow?
No, but that would be a good addition to Asterisk. I started
experimenting with that in my caller ID utf8 branch at some point,
but never got time or funding to complete that work.
Thanks for your help again. Would be nice really, but I'll try to find a
workaround to avoid that problem (or ignore it).
Bjoern
--
Message: 2
Date: Sun, 10 Feb 2008 18:44:46 +0100
From: Adam KOSA [EMAIL PROTECTED]
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
pix 506
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
permit udp any host 192.168.5.0 range 1 2 and then I didn't
home users typically use /24 netmask. If this is the case, i don't
understand why do you write keyword host following a network address.
either specify a valid host address, or write 192.168.5.0 255.255.255.0
to specify the whole subnet.
if the netmask isn't /24 then, of course the above 5.0 may be a valid
host address.
regards
adam
--
Message: 3
Date: Sun, 10 Feb 2008 12:54:44 -0500
From: Bryan M. Johns [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk Scalability
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; delsp=yes
We have multiple installs that tested-out at nearly concurrent 400 SIP
channels on a Dell 2950 with 2Xquad core at 1.6 Ghz, 16 GB of RAM.
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Support: [EMAIL PROTECTED]
http://www.sheltonjohns.com
On Feb 8, 2008, at 5:09 AM, Femi wrote:
Hi,
Does anyone have data on the switching capacity of Asterisk based on
the
hardware?
I need to know what type of hardware would be required to switch 100
simultaneous calls as opposed to 1000 or 1 calls, no TDM just
SIP to SIP
VoIP calls
Thanks
Femi
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[asterisk-users] Caller id issue and Dial tone for sip phone on zero dialing
Hi all, I am not getting the dial tone when i dial the zero digit. And i am using analog card,for my operator phone caller id is not displaying on the phone.I am in india. In india is it possible to get the caller id for analog cards. Can any body help me. Please reply. ThanksRegards, sandeep.s___ -- 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
[asterisk-users] caller id issue for INDIA
hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 42, Issue 51
hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51 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: app_voicemail for spanish (Andrew Joakimsen) 2. SVN servers down for maintenance (Russell Bryant) 3. Re: Asterisk 1.4.17 crashing more (Steve Totaro) 4. Zaptel 1.2.23 and 1.4.8 released (The Asterisk Development Team) 5. Re: AGISTATUS is SUCCESS even though my PHP script returned -1 (Matt Riddell) 6. Re: Video Call and Asterisk (Matt Riddell) 7. Re: app_voicemail for spanish (Anton Krall) 8. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Steve Totaro) 9. Re: Asterisk RFC2833 to SIP INFO DTMF conversion erros. (Mayur) 10. Re: AGISTATUS is SUCCESS even though my PHP script returned -1 (Steve Edwards) 11. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Tzafrir Cohen) 12. Park() help, extension not heard (Rob) 13. Re: AGISTATUS is SUCCESS even though my PHP script returned -1 (Brian Hutchinson) 14. Re: Asterisk 1.4.17 crashing more (Brian Hutchinson) 15. Re: app_voicemail for spanish (Andrew Joakimsen) 16. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Andrew Joakimsen) 17. Re: Park() help, extension not heard (Rob) 18. pickupchan without bristuffed version? (Stefan Guenther) 19. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Bruce McAlister) 20. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Thomas Kenyon) 21. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Andrew Joakimsen) 22. Re: G.729 pre-compiled binaries and Asterisk 1.2.x. (Thomas Kenyon) -- Message: 1 Date: Mon, 14 Jan 2008 18:57:34 -0500 From: Andrew Joakimsen [EMAIL PROTECTED] Subject: Re: [asterisk-users] app_voicemail for spanish To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=UTF-8 The language support is supposed to be there I know I've played with it and there are at least SOME grammatical changes (don't recall which right now) But if further language support is needed you should file a bugreport. On Jan 14, 2008 5:04 PM, Anton Krall [EMAIL PROTECTED] wrote: Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying trabajo mensjes would say mensajes de trabajo o mensajes trabajo (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton ___ -- 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: Mon, 14 Jan 2008 17:59:51 -0600 From: Russell Bryant [EMAIL PROTECTED] Subject: [asterisk-users] SVN servers down for maintenance To: undisclosed-recipients:; Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed The Digium svn servers are down, and will likely be down for the rest of the evening, as I perform some system maintenance. I apologize for any inconvenience that this may cause. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. -- Message: 3 Date: Mon, 14 Jan 2008 19:03:21 -0500 From: Steve Totaro [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk 1.4.17 crashing more 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 On Jan 14, 2008 6:23 PM, Abdul [EMAIL PROTECTED] wrote: Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time asterisk not responding anything for clients for registration either for invitation. Please advice us
[asterisk-users] Asterisk is not adding Via field
Hi, I am trying to integrate asterisk with openser for a simple call. I am facing some issues with Asterisk. Below is the explanation: I have a UA1 sending invite to UA2 through Openser and Asterisk with the below sequence. Sequence is UA1-OpenSER-Asterisk-Openser-UA2 When Asterisk gets the INVITE, the INVITE contains two Via headers, one of the UA1 and the other Openser's. As Asterisk acts as a B2BUA, it recreats the Dialog. So before forwarding the INVITE to Openser back, it is removing the Via header of UA1 and also it is not adding its own Via header. So when the INVITE reaches UA2 from Openser, the INVITE will have only one Via header (which is of Openser). So when UA2 responds with 180 Ringing, it will reach Openser, but Openser cannot forward to Asterisk because it does not contain the Via header of Asterisk. Please help me out. Cheers, -Sandeep A ___ --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
[asterisk-users] Asterisk-LDAP Integration?
Hi , Has anyone earlier tried integrating asterisk with LDAP. I am interested to integrate LDAP for authentication purpose for any SIP Incoming calls.. Pl. suggest pointers. Thanks and Regards --Sandeep Kalra Ph: +91-120-4342000-X-2966 : +91-120-4342966 (direct) M- 9810683168 visit: http://www.globalLogic.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
RE: [asterisk-users] Cisco 7940 - NAT Option
. Could it hurt something when they are used inside our LAN with NAT enabled? The answer is no! With my test bed, I found that Asterisk can detect Endpoint behind NAT(match via and src_ip). So, once the EP is on LAN (same side of NAT) then they work as if there is no NAT. The option of nat=yes is immaterial. Thanks and Regards --Sandeep Kalra Ph: +91-120-4342000-X-2966 : +91-120-4342966 (direct) www.globallogic.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Monday, December 18, 2006 9:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7940 - NAT Option I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used. Could it hurt something when they are used inside our LAN with NAT enabled? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.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
RE: [asterisk-users] matching the beginning of an EXTEN
Try Exten = _351217588XXX, 1, Dial ( ... ) Thanks and Regards --Sandeep Kalra Ph: +91-120-4342000-X-2966 : +91-120-4342966 (direct) M- 9810683168 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Thursday, December 14, 2006 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] matching the beginning of an EXTEN Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ? I want match the first 9 digits does Asterisk has any function for this? Thanks Regards Joao Pereira ___ --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
[Asterisk-Users] Hangup problem with other EPBX
From the list I read .. Connecting Asterisk with Digium FXO interface card with any EPABX will have hangup issue. Ie: hangup from EPABX extn will not be recognised by Asterisk. I am facing this problem with TDM FXO interface cards. Is there any work aroud for this .Like if the User press # ,it should hangup like that . Other details Country :India ( Working in default country as US) EPABX :Siemens 64 Port. Version:Asterisk CVS HEAD on 2005-09-27 ( on Redhat 9) I am using default zapdata.conf zaptel.conf with signelling fxsks Is there any solution or work around for it . I am ready to give more details or experimentation. Thanks -Sandeep ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
In india no distributer for digium cards If any body is going to us u can ask them to bring it. I got in that way -sandeep Ankit wrote: where did u purchase ur card frm, im not able to find ne distributor of digium cards in india, and if i order it frm their site it will have to pay arnd 2k rs for shipping :( -ankit On 8/9/05, *Gurminder Arora* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I m using it on POTS line and will start with ISDN soon :-). Cheers Gurminder On 8/9/05, Ankit [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi gurminder, are you using it on isdn line or pots line? On 8/9/05, Gurminder Arora [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Meeting VS Call Confrence
Hi I am tried the patch for outboud call from conferance but the following error : [EMAIL PROTECTED] asterisk]# patch -p0 app_meetme.c_outboundcall_rev3.txt patching file apps/app_meetme.c Hunk #1 succeeded at 34 (offset 1 line). Hunk #2 succeeded at 63 with fuzz 2 (offset 1 line). Hunk #3 succeeded at 112 (offset 3 lines). Hunk #4 succeeded at 163 (offset 4 lines). Hunk #5 FAILED at 191. Hunk #6 FAILED at 553. Hunk #7 succeeded at 682 with fuzz 1 (offset 54 lines). Hunk #8 succeeded at 1112 (offset 131 lines). Hunk #9 FAILED at 1464. Hunk #10 succeeded at 1726 (offset 38 lines). 3 out of 10 hunks FAILED -- saving rejects to file apps/app_meetme.c.rej my asterisk version is : #asterisk -V gives Asterisk CVS-HEAD-04/12/05-18:15:04 Pl suggest me what went wrong. Thanks Sandeep Peter Svensson wrote: On Thu, 2 Jun 2005, Mohamed A. Gombolaty wrote: I was trying to make call confrence available but all the asterisk documents use the meeting room concept, where those who wanna meet have to dial an extension corresponding to the meeting room, while call conference actually means that I am on exten 100 I can dial exten 200 and add it to confrence and again dial 333 and add it to the confrence and so on. Is there any way to make call confrencing available and not meeting room concepts? There is a patch to add call out from within a meetme conference. See bug number 3405 on http://bugs.digium.com/. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP or IAX
For bridging VOIP with PSTN Lines Which one is giving better performance SIP or IAX ? I am looking at a result without NAT in picture ? Can some body give details from experiance ? Also with single SIP/IAX channel can I use more than one call at a time ? Thanks Sandeep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Replacing SIP Trunking With IAX Trunking
I have the sip trunking as below :
I tried with IAX Trunking .But no success
Can some one send IAX Trunking config for the below setup replacing
SIP ?
PBX1 (192.168.10.2)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.1.2
extensions.conf
exten = 1113,1, Dial(SIP/abc1,10,t)
exten = 1158,1, Dial(SIP/xyz1,10,t)
exten = _2XXX,1, Dial(SIP/pbx/${EXTEN})
PBX2 (192.168.1.2)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.10.2
extensions.conf
exten = 2113,1, Dial(SIP/abc2,10,t)
exten = 2158,1, Dial(SIP/xyz2,10,t)
exten = _1XXX,1, Dial(SIP/pbx/${EXTEN})
Thanks
Sandeep
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[Asterisk-Users] TDM400P Channels stop answering after some time .
Hi Need help on bridging SIP with TDM400P(4 FXO Modules ) My setup is as follows US OFFICE -TDM400P(FXO) --SIP--- TDM400P(FXOs)INDIA OFFICE (DSL Line) Asterisk Asterisk PBX(Siemens) /DSL Line Server Server Everithing works fine for one or two calls or maximum 4 calls over the setup. Ie after some time zap channels are not ringing.Then I have to reload asterisk.Once restart everithing works fine for 2 or 3 calls over the setup then the same issue .I need to restart asterisk again . Is it the problem with TDM400P ? OR the problem with 2.6 Kernel ? or Problem with SIP and TDM Card ? How I can troubleshoot ? I am using Fedora core3 Kernel 2.6.9-1.667 My zaptel.conf on both systes: loadzone = us defaultzone=us fxsks=1-4 My zapdata.conf on both systems : signalling=fxs_ks rxwink=300 usecallingpres=yes transfer=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=4.9 txgain=6.9 busydetect=yes callprogress=yes progzone=us musiconhold=default jitterbuffers=4 My sip.conf on both systems [pbx] type=friend username=pbx secret=pbx host=192.168.X.Y dtmfmode=info insecure=very qualify=no disallow=all allow=ulaw Do you want any more details ? thanks -Sandeep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OFF TOPIC] Voip phone sellers in India
check with webtel,Mob:32333033 On Sun, 2005-01-09 at 19:33 +0100, Vikram Rangnekar wrote: I am looking for some in India to buy VOIP phones from. Please get in touch with me off the list on [EMAIL PROTECTED] Sorry for the off topic mail I am just having such a hard time finding any voip phones in India that I got desperate and didnt know which list to post this on. -- Sandeep A.S [EMAIL PROTECTED] Netcontinuum Pvt Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
