Re: [asterisk-users] Installing Asterisk on Virtual Machine
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
Re: [asterisk-users] h323-sip: one way connection
i know what is the exactly problem. i enable debug for h323 and it says: could not find user by name 200 or address 192.168.0.146 when i change peer-146 to 200 every thing is ok and i can call from two side. but it is not good for me because 200 is the name of extension and when i config asterisk systems, i don't know the name of extensions, therefore i should use addresses not name of extensions. do you know how i should define address of the other end in h323.conf file? i define the address by host=192.168.0.146 but asterisk can not find it? why? On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote: please post cli output for both calls. On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote: hello everybody i want to have sip connection between two asterisk systems (145 and 146). connection from 145 to 146 is ok but i can not call from 146 to 145. this is h323.conf file in 145: [peer146] host=192.168.0.146 type=friend context=from-trunk [to-146] type=peer host=192.168.0.146 faststart=yes tunneling=no progress_audio=yes disallow=all allow=alaw allow=ulaw this is mu extensions.conf file in 145: [from-trunk] exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1}) [line-231] exten=_2.,1,Dial(H323/to-146/2${EXTEN:1}) i have this error: dropping call because extensions '100', 's' and 'i' doesn't exists in context default. if i change peer146 to general, every thing is ok and i can call from two side. my question is: in h323 connection, is it a MUST to have general context in h323.conf? if not, why i have this error and how i can solve it? thanks in advance sam -- _ -- 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
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:
[asterisk-users] Jitter Buffer in asterisk 1.8.11.0
I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in server. I am having some issue in audio quality. I want to enable jitter buffer on asterisk but don't know, how to do. Any one can help me. -- _ -- 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] Dialplan reload not reloading everything
Good morning, We recently fell back to the most recent build of asterisk 1.8 down from 11.3 and I believe we've crossed some sort of limit for 1.8. Our dialplan is 515723 entries long with 6263 distinct contexts. Both are loaded realtime via odbc (mysql). Previously at the end of a dialplan reload we would get a summary of how long it took to reload everything. Now it just shows the last line that it loaded as seen below: -- Registered extension context '4959_5095_0'; registrar: pbx_config -- Including switch 'realtime/@' in context '4959_5095_0' -- Registered extension context '4960_5096_0'; registrar: pbx_config -- Including switch 'realtime/@' in context '4960_5096_0' asterisk*CLI I've tried turning on debug and there's no extra information. I tried running it from the command line with -rx and it says dialplan reloaded. Yet a bunch of the newest contexts aren't recognized (asterisk reports that they don't exist then call dies). I've confirmed they're in the database. The context it ends with is not always the same. Sometimes it's in the 3000 range and sometimes it's in the 4000s. Has anyone else seen this? Is there a maximum string length for the contexts or something that could be causing this? -- _ -- 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] asterisk music on hold recommendations
Hi all, I'm wondering what the recommendations are for using music on hold on asterisk. As far as I understood from various pages on the web and a response from the IRC channel, I am to avoid using mp3 files because of licensing and transcoding issues. correct? I am currently using asterisk 1.8 with the mpg123 processes (mode=mp3 or mode=quietmp3 in the conf file). This means that there is one single shared stream of moh for all channels that are using the same class of moh. If I were to start using wav files (mode=files), is there a way to have the same kind of shared stream of moh to reduce the load on the machine in the case where a lot calls are on hold? Is it even worth it to try reducing the load (maybe asterisk handles playing wav files very efficiently and the extra load generated by it is negligible)? I am looking to upgrade to asterisk 11 in the future. Is any of this different for that version? Thanks for any responses! Frederic -- _ -- 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
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
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
Re: [asterisk-users] asterisk music on hold recommendations
On Tue, Apr 23, 2013 at 02:30:24PM +0200, Frederic Van Espen wrote: Hi all, I'm wondering what the recommendations are for using music on hold on asterisk. As far as I understood from various pages on the web and a response from the IRC channel, I am to avoid using mp3 files because of licensing and transcoding issues. correct? Short version: Not really. But just use the built in The earliest moh support Asterisk had was playing of MP3 files (or piping the output of an external command). Only later on native MoH was developed - playing any file Asterisk could play. At the time Digium licensed a set of mp3 files from FreePlay Music that could be freely used as MoH files with Asterisk. Later on a certain more subtle licensing issue came up and Digium chose to stop distributing those MoH files with Asterisk. They were replaced with a set of five files which are: * Longer * Better licensed (CC-BY-SA 3.0) * Available in all the required formats So the licensing issues in question are: * MP3 is patent-encumbered and some Linux distribution keep out even MP3 playing code (other only remove MP3 encoding code). * If you don't intend to play it to a MP3 channel, why waste CPU resources on transcoding it? The newer files are available in more convinient formats. IIRC the license of the FPM ones prevented Digium from distributing modified copies. I am currently using asterisk 1.8 with the mpg123 processes (mode=mp3 or mode=quietmp3 in the conf file). If you use that mode, you're probably doing something wrong following an ancient guide. This means that there is one single shared stream of moh for all channels that are using the same class of moh. If I were to start using wav files (mode=files), is there a way to have the same kind of shared stream of moh to reduce the load on the machine in the case where a lot calls are on hold? Is it even worth it to try reducing the load (maybe asterisk handles playing wav files very efficiently and the extra load generated by it is negligible)? I am looking to upgrade to asterisk 11 in the future. Is any of this different for that version? -- 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
Re: [asterisk-users] asterisk music on hold recommendations
On 04/23/2013 03:12 PM, Tzafrir Cohen wrote: If you use that mode, you're probably doing something wrong following an ancient guide. Well, these modes are the ones documented in the sample conf files that came with asterisk 1.8.13.0: snip ; valid mode options: ; files -- read files from a directory in any Asterisk supported ; media format ; quietmp3 -- default ; mp3 -- loud ; mp3nb -- unbuffered ; quietmp3nb -- quiet unbuffered ; custom -- run a custom application (See examples below) snip Seems I did miss something from the sample file before though: snip ;cachertclasses=yes ; use 1 instance of moh class for all users who are using it, ; decrease consumable cpu cycles and memory ; disabled by default snip That seems to answer my other question. Anyone got any experience using this? 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
On 2013-04-23 08:47, Sandeep Raju wrote: 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.com wrote: @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.com wrote: 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 [1]). 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 [2]) still buggy. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com [3] iax:gu...@local.xorcom.com/tzafrir [4] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com [5] -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello [6] asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users [7] Links: -- [1] http://configure.ac [2] http://packages.ubuntu.com [3] http://www.xorcom.com [4] http://iax:gu...@local.xorcom.com/tzafrir [5] http://www.api-digital.com [6] http://www.asterisk.org/hello [7] 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 I had the same issue trying to install asterisk 11 on amazon ec2. I installed from source. from compile options in menuselect, I had to uncheck build_native. Not necessarily ideal, but I got asterisk started. Paul TurnOpen.com -- _ -- 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] cdr report
Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? Sincerely yours, Aris -- _ -- 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] cdr report
Well, the question is, what your secretary wants to do. Only see the CDRs or more? Realtime? One simple method would be to mail her the CSV-File, so she can open it with Excel or Calc (Open Office). Am 23.04.2013 16:35, schrieb aristidis tsitras: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? Sincerely yours, Aris -- _ -- 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] cdr report
That would be nice. is there a way to have it ready in xls? if yes, then i could send it put a cron to send it every night/week/month. On 04/23/2013 05:42 PM, Thorsten Göllner wrote: Well, the question is, what your secretary wants to do. Only see the CDRs or more? Realtime? One simple method would be to mail her the CSV-File, so she can open it with Excel or Calc (Open Office). Am 23.04.2013 16:35, schrieb aristidis tsitras: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? Sincerely yours, Aris -- _ -- 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-use -- _ -- 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] cdr report
On Tuesday 23 April 2013, aristidis tsitras wrote: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? CSV files can be opened with any spreadsheet software (such as OpenOffice.org calc or Numbers). Alternatively, you can have the CDR using a database. This can be on another server. Note if you are using MySQL, you will have to enable this yourself; this is because not all of Asterisk is covered by the GPL, and the MySQL CDR code ends up unredistributable. (But it works as well as anything). Then write a Web app on the database server to display wanted CDR entries. -- AJS Answers come *after* questions. -- _ -- 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] cdr report
On 23/04/2013 11:09 AM, A J Stiles wrote: On Tuesday 23 April 2013, aristidis tsitras wrote: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? CSV files can be opened with any spreadsheet software (such as OpenOffice.org calc or Numbers). Alternatively, you can have the CDR using a database. This can be on another server. Note if you are using MySQL, you will have to enable this yourself; this is because not all of Asterisk is covered by the GPL, and the MySQL CDR code ends up unredistributable. (But it works as well as anything). Then write a Web app on the database server to display wanted CDR entries. What about a script to convert the CSV to HTML and ftp the html file to a web server where it can be accessed as a browser page? -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] cdr report
On 04/23/2013 06:23 PM, Ron Wheeler wrote: On 23/04/2013 11:09 AM, A J Stiles wrote: On Tuesday 23 April 2013, aristidis tsitras wrote: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? CSV files can be opened with any spreadsheet software (such as OpenOffice.org calc or Numbers). Alternatively, you can have the CDR using a database. This can be on another server. Note if you are using MySQL, you will have to enable this yourself; this is because not all of Asterisk is covered by the GPL, and the MySQL CDR code ends up unredistributable. (But it works as well as anything). Then write a Web app on the database server to display wanted CDR entries. What about a script to convert the CSV to HTML and ftp the html file to a web server where it can be accessed as a browser page? is it possible to have the script to convert to html? i will send it to a folder -- _ -- 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] cdr report
On 23/04/2013 11:42 AM, aristidis tsitras wrote: On 04/23/2013 06:23 PM, Ron Wheeler wrote: On 23/04/2013 11:09 AM, A J Stiles wrote: On Tuesday 23 April 2013, aristidis tsitras wrote: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? CSV files can be opened with any spreadsheet software (such as OpenOffice.org calc or Numbers). Alternatively, you can have the CDR using a database. This can be on another server. Note if you are using MySQL, you will have to enable this yourself; this is because not all of Asterisk is covered by the GPL, and the MySQL CDR code ends up unredistributable. (But it works as well as anything). Then write a Web app on the database server to display wanted CDR entries. What about a script to convert the CSV to HTML and ftp the html file to a web server where it can be accessed as a browser page? is it possible to have the script to convert to html? i will send it to a folder echo htmlbodytable input.html; while read INPUT ; do echo trtd${INPUT//,//tdtd}/td/tr input.html ; done input.csv ; echo /table/body/html input.html This Linux shell script will take your log after being copied to input.csv and make it into an ugly web page called input.html. You can add more HTML to add a css style sheet and some column headings to make it look nice. You probably want to fix the file names to reflect the date. Ron -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] h323-sip: one way connection
try type=peer instead of friend. On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote: i know what is the exactly problem. i enable debug for h323 and it says: could not find user by name 200 or address 192.168.0.146 when i change peer-146 to 200 every thing is ok and i can call from two side. but it is not good for me because 200 is the name of extension and when i config asterisk systems, i don't know the name of extensions, therefore i should use addresses not name of extensions. do you know how i should define address of the other end in h323.conf file? i define the address by host=192.168.0.146 but asterisk can not find it? why? On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote: please post cli output for both calls. On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote: hello everybody i want to have sip connection between two asterisk systems (145 and 146). connection from 145 to 146 is ok but i can not call from 146 to 145. this is h323.conf file in 145: [peer146] host=192.168.0.146 type=friend context=from-trunk [to-146] type=peer host=192.168.0.146 faststart=yes tunneling=no progress_audio=yes disallow=all allow=alaw allow=ulaw this is mu extensions.conf file in 145: [from-trunk] exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1}) [line-231] exten=_2.,1,Dial(H323/to-146/2${EXTEN:1}) i have this error: dropping call because extensions '100', 's' and 'i' doesn't exists in context default. if i change peer146 to general, every thing is ok and i can call from two side. my question is: in h323 connection, is it a MUST to have general context in h323.conf? if not, why i have this error and how i can solve it? thanks in advance sam -- _ -- 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] cdr report
On Tue, 2013-04-23 at 17:35 +0300, aristidis tsitras wrote: Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on CPU, it has to be as low on CPU/RAM consumption as possible. any ideas? You can have the cdr_odbc or cdr_mysql module loaded and have the cdr in an external database Once in there you can get any report/format you want with minimum programming. The issue is that you need a machine running the database 24/7 Another option is to use the manager interface and an external client to collect the cdr events The manager interface can be setup to output only the cdr events and the resource requirements on the machine running asterisk are minimal. The downside is that you also need an external client running 24/7 to collect them. Stelios smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Dialplan reload not reloading everything
- Original Message - From: Brandon Mackie bmac...@awktane.com We recently fell back to the most recent build of asterisk 1.8 down from 11.3 and I believe we’ve crossed some sort of limit for 1.8. Our dialplan is 515723 entries long with 6263 distinct contexts. Both are loaded realtime via odbc (mysql). That's a big dialplan! snip I’ve tried turning on debug and there’s no extra information. snip How did you enable debug? Did you follow the directions here? https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information If you can pastebin a log showing the end of the reload with WARNING,ERROR,NOTICE, plus VERBOSE and DEBUG turned up (try 5 or higher) then maybe there will be something interesting.. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com -- _ -- 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] Asterisk 11.4.0-rc1 refuses to use the TURN server
After struggling with one way audio issues as a result of STUN binding errors on both the Asterisk side and the Chrome side, we've decided to just simply go with a TURN relay for RTP packets until the issues are resolved. I configured rtp.conf so that all of the STUN related entries are commented out, and I use the following TURN configuration instead: turnaddr=numb.viagenie.ca:3478 ; ; Username used to authenticate with TURN relay server. turnusername=myusername%40gmail.com ; ; Password used to authenticate with TURN relay server. turnpassword=p@ssw0rd I also use the same configuration on the client side. When running a tcpdump, I see that there is traffic to/from the TURN relay: 10.0.1.18.53875 blues.viagenie.ca.nat-stun-port: UDP, length 20 blues.viagenie.ca.nat-stun-port 10.0.1.18.53875: UDP, length 56 10.0.1.18.51435 blues.viagenie.ca.nat-stun-port: UDP, length 28 blues.viagenie.ca.nat-stun-port 10.0.1.18.51435: UDP, length 100 10.0.1.18.51435 blues.viagenie.ca.nat-stun-port: UDP, length 144 10.0.1.18.51435 blues.viagenie.ca.nat-stun-port: UDP, length 144 blues.viagenie.ca.nat-stun-port 10.0.1.18.51435: UDP, length 100 But it's dead silent when doing a tcpdump on the Asterisk server side. The candidates on both sides don't contain relay candidates. Oddly, the client side still has srflx candidates, suggesting STUN is still at work, but the Asterisk side only contains host candidates. Is TURN fully enabled in Asterisk 11? If so, how does one enable it and make it the priority? Thank you, -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com -- _ -- 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