Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Hi, I have started asterisk using strace, and the log is listed in pastebin http://pastebin.com/ry2Q1e6x Moreover, for some peoples Asterisk is properly installed in OpenSuse 12.1 (i586), can you please correct me with the installation steps what I did, my steps as follows, 1. OpenSuse fresh installation 2. Login to root in terminal (sudo -i) 3. Download libpri, dahdi and Asterisk 4. Install libpri and dahdi (even though I am not using any dahdi hardware) - make and make install 5. Installation of Asterisk (./configure, make menuconfig, make, make install and make samples) 6. Start Asterisk (asterisk -c) - here hangs while loading modules. any other packages has to be installed or the installation is fine! please advice! Regards. On Thu, Aug 30, 2012 at 7:03 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote: On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk Asterisk will fork into the background and the process you trace will exit. strace -f asterisk #? strace asterisk -f #? Just in case you wonder, 'asterisk -f strace' will not work as you might have expected from the above examples. Nither will '-f strace asterisk'. '-U asterisk ' may also come in handy. -- 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 hangs while starting in OpenSuse 12.2
Gopalakrishnan I download and compile from libpri, dahdi, and asterisk. You have to insure that you have all of the dependencies when you compile there are many. Also compile and install spandsp as well. I did a 12.1 system over the weekend without issue. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Monday, September 03, 2012 5:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi, I have started asterisk using strace, and the log is listed in pastebin http://pastebin.com/ry2Q1e6x Moreover, for some peoples Asterisk is properly installed in OpenSuse 12.1 (i586), can you please correct me with the installation steps what I did, my steps as follows, OpenSuse fresh installation Login to root in terminal (sudo -i) Download libpri, dahdi and Asterisk Install libpri and dahdi (even though I am not using any dahdi hardware) - make and make install Installation of Asterisk (./configure, make menuconfig, make, make install and make samples) Start Asterisk (asterisk -c) - here hangs while loading modules. any other packages has to be installed or the installation is fine! please advice! Regards. On Thu, Aug 30, 2012 at 7:03 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote: On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk Asterisk will fork into the background and the process you trace will exit. strace -f asterisk #? strace asterisk -f #? Just in case you wonder, 'asterisk -f strace' will not work as you might have expected from the above examples. Nither will '-f strace asterisk'. '-U asterisk ' may also come in handy. -- 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 hangs while starting in OpenSuse 12.2
Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Regards. On Tue, Aug 28, 2012 at 10:47 PM, Bryant Zimmerman brya...@zktech.comwrote: I would install both dahdi and libpri. I brought up a 12.2 RC-2 VM on hyper-v Windows 8 and followed our standard asterisk build and have no issues yet but we have not run full testing to confirm. Also a point of not 12.2 is RC for the next 8 days or so. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Tuesday, August 28, 2012 1:13 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 If I don't need to install dahdi hardware, is it really I need to have libpri installed? Regards. On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote: Check Jason Parker’s post from today and see if you skipped any of the preliminary build steps. It is possible that something like libpri is biting you. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Tuesday, August 28, 2012 11:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 I tried that too, what happens is asterisk is loading but after that if I try to start any one module for example chan_sip.so, asterisk hangs. Regards. On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote: Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 11:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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
Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk That should give you some low level info what's going on. More info about strace and available options can be found in: $ man strace Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote: On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk Asterisk will fork into the background and the process you trace will exit. strace -f asterisk #? strace asterisk -f #? Just in case you wonder, 'asterisk -f strace' will not work as you might have expected from the above examples. Nither will '-f strace asterisk'. '-U asterisk ' may also come in handy. -- 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 hangs while starting in OpenSuse 12.2
Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
I tried that too, what happens is asterisk is loading but after that if I try to start any one module for example chan_sip.so, asterisk hangs. Regards. On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote: Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 11:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 ** ** Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
Check Jason Parker's post from today and see if you skipped any of the preliminary build steps. It is possible that something like libpri is biting you. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Tuesday, August 28, 2012 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 I tried that too, what happens is asterisk is loading but after that if I try to start any one module for example chan_sip.so, asterisk hangs. Regards. On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote: Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
If I don't need to install dahdi hardware, is it really I need to have libpri installed? Regards. On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote: Check Jason Parker’s post from today and see if you skipped any of the preliminary build steps. It is possible that something like libpri is biting you. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Tuesday, August 28, 2012 11:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 ** ** I tried that too, what happens is asterisk is loading but after that if I try to start any one module for example chan_sip.so, asterisk hangs. Regards. On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote: Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 11:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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 -- _ -- 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 hangs while starting in OpenSuse 12.2
I would install both dahdi and libpri. I brought up a 12.2 RC-2 VM on hyper-v Windows 8 and followed our standard asterisk build and have no issues yet but we have not run full testing to confirm. Also a point of not 12.2 is RC for the next 8 days or so. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Tuesday, August 28, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 If I don't need to install dahdi hardware, is it really I need to have libpri installed? Regards. On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote: Check Jason Parker's post from today and see if you skipped any of the preliminary build steps. It is possible that something like libpri is biting you. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Tuesday, August 28, 2012 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 I tried that too, what happens is asterisk is loading but after that if I try to start any one module for example chan_sip.so, asterisk hangs. Regards. On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote: Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 11:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this.Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this.Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
Hi Bryant, As you said, I dont have Hyper-V, I avoided virtualbox and tested in normal host directly, even then it hangs while loading modules. *Asterisk Dynamic Loader Starting:* * == Parsing '/etc/asterisk/modules.conf': == Found* *[Aug 27 11:52:21] NOTICE[22886]: loader.c:1133 load_modules: 186 modules will be loaded.* This is really tuff working with OpenSuse. I am clueless how to sort our this. Regards. On Fri, Aug 24, 2012 at 3:55 AM, Hans Witvliet aster...@a-domani.nl wrote: On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote: Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. Have you tried the versions from the OBS? Or perhaps a virtualbox issue? Its notorious for vapourizing cpu-cycles... hw -- _ -- 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 hangs while starting in OpenSuse 12.2
On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- __**__**_ -- 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-**usershttp://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 hangs while starting in OpenSuse 12.2
I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Monday, August 27, 2012 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 ** ** Hi Patrick, ** ** With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. ** ** Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. ** ** Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. While starting Asterisk, it hangs here, *Asterisk Dynamic Loader Starting:* * == Parsing '/etc/asterisk/modules.conf': == Found* *[Aug 23 14:56:14] NOTICE[19340]: loader.c:1133 load_modules: 186 modules will be loaded.* any my linux machine uname -a output is below, *Linux linux-w6le.site 3.1.0-1.2-default #1 SMP Thu Nov 3 14:45:45 UTC 2011 (187dde0) i686 i686 i386 GNU/Linux* * * Any suggestion would be much appreciated. Regards, Gopal. On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Ok Thanks Bryant, let me try with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote: I have the current version of 8.x and 10.x on systems. I am using OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of time. Asterisk on all of our boxes are complied from source. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Monday, August 20, 2012 10:11 AM *To*: Bryant Zimmerman brya...@zktech.com *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 It's really glad that asterisk is installed at your machine in open suse. Can you let me know which version you are using and the architecture. Regards. On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote: I compile from source.. Sent from my Verizon Wireless Phone - Reply message - From: Gopalakrishnan N gopalakrishnan...@gmail.com Date: Mon, Aug 20, 2012 8:15 am Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.comshould resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
Have you tried vmware or hyper-v as your host. I have had issues with OpenSuse 12.x with Virtual Box. Asterisk not starting was one of them. Also in a virtual env I found that I had to alwyas build asterisk from source to make things work don't know why but that was the mix that worked for me. I moved to Hyper-V. OpenSuse 12.x as a VM is kind of a black art with asterisk for some reason. Once you get it working it works great. You have to watch how your virtual nic's are setup that can really mess with you as well. But virtual box was a no go for me never spent the time to figure out why. I took the path of least resistance. Thanks Bryant From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Thursday, August 23, 2012 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. While starting Asterisk, it hangs here, Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found [Aug 23 14:56:14] NOTICE[19340]: loader.c:1133 load_modules: 186 modules will be loaded. any my linux machine uname -a output is below, Linux linux-w6le.site 3.1.0-1.2-default #1 SMP Thu Nov 3 14:45:45 UTC 2011 (187dde0) i686 i686 i386 GNU/Linux Any suggestion would be much appreciated. Regards, Gopal. On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Ok Thanks Bryant, let me try with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.com wrote: I have the current version of 8.x and 10.x on systems. I am using OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of time. Asterisk on all of our boxes are complied from source. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Monday, August 20, 2012 10:11 AM To: Bryant Zimmerman brya...@zktech.com Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 It's really glad that asterisk is installed at your machine in open suse. Can you let me know which version you are using and the architecture. Regards. On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote: I compile from source.. Sent from my Verizon Wireless Phone - Reply message - From: Gopalakrishnan N gopalakrishnan...@gmail.com Date: Mon, Aug 20, 2012 8:15 am Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote: Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. Have you tried the versions from the OBS? Or perhaps a virtualbox issue? Its notorious for vapourizing cpu-cycles... hw -- _ -- 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 hangs while starting in OpenSuse 12.2
Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- __**__**_ -- 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-**usershttp://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 hangs while starting in OpenSuse 12.2
From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.comshould resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- __**__** _ -- 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-**usershttp://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 hangs while starting in OpenSuse 12.2
Ok Thanks Bryant, let me try with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote: I have the current version of 8.x and 10.x on systems. I am using OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of time. Asterisk on all of our boxes are complied from source. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Monday, August 20, 2012 10:11 AM *To*: Bryant Zimmerman brya...@zktech.com *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 It's really glad that asterisk is installed at your machine in open suse. Can you let me know which version you are using and the architecture. Regards. On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote: I compile from source.. Sent from my Verizon Wireless Phone - Reply message - From: Gopalakrishnan N gopalakrishnan...@gmail.com Date: Mon, Aug 20, 2012 8:15 am Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.comshould resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- __**__**_ -- 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-**usershttp://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 hangs while starting in OpenSuse 12.2
On 14-08-12 08:29, Gopalakrishnan N wrote: If I change autoload=no then asterisk is starting, but post to that loading modules even chan_sip.so asterisk hangs. Its strange, only in OpenSuse I am facing this. In CentOS, Ubuntu its working fine, same Asterisk version with same hardware. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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 hangs while starting in OpenSuse 12.2
Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with OpenSuse 12.2, its really wired working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well which results to same failure. Regards, Gopal. -- _ -- 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 hangs while starting in OpenSuse 12.2
I am running OpenSuse 12.1 with no issues. I have not tried 12.2 beta yet. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Monday, August 13, 2012 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with OpenSuse 12.2, its really wired working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well which results to same failure. Regards, Gopal. -- _ -- 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 hangs while starting in OpenSuse 12.2
Hi, Thanks for your comments. Even I tried with 12.1 also, its the same issue, I am not sure whether it may be hardware related. Logs below, OS details - uname -a Linux laptop-prasad 3.3.0-2-desktop #1 SMP PREEMPT Sat Mar 24 00:11:53 UTC 2012 (7e9dd21) x86_64 x86_64 x86_64 GNU/Linux while executing asterisk -c from the root prompt, its stuck as below and the CPU usage is fully utilized, == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree == Parsing '/etc/asterisk/enum.conf': == Found == Registered application 'CallCompletionRequest' == Registered application 'CallCompletionCancel' == Parsing '/etc/asterisk/ccss.conf': == Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found [Aug 14 10:48:36] NOTICE[3805]: loader.c:1133 load_modules: 184 modules will be loaded. Any advice would be much appreciated. Regards, Gopal. On Tue, Aug 14, 2012 at 3:37 AM, Bryant Zimmerman brya...@zktech.comwrote: I am running OpenSuse 12.1 with no issues. I have not tried 12.2 beta yet. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Monday, August 13, 2012 8:19 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with OpenSuse 12.2, its really wired working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well which results to same failure. Regards, Gopal. -- _ -- 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 hangs while starting in OpenSuse 12.2
On Tue, 14 Aug 2012, Gopalakrishnan N wrote: while executing asterisk -c from the root prompt, its stuck as below and the CPU usage is fully utilized, [snip] == Parsing '/etc/asterisk/modules.conf': == Found [Aug 14 10:48:36] NOTICE[3805]: loader.c:1133 load_modules: 184 modules will be loaded. I'm just a 1.2 Luddite, but I'll take a stab... I'm guessing you're autoloading everything. (My personal preference is to turn autoloading off and explicitly load just what I need.) Mung a directory listing of your modules so each module name is prefixed with 'noload' Paste this into you modules.conf. Comment out the first half of the 'noloads.' If Asterisk still hangs, the problem is somewhere in the second half. If not, un-comment the ones you just commented and comment out the second half. Continue this process (bi-section search) until you identify the errant module. You'll have to fiddle a bit as you discover module inter-dependencies. You could probably make some educated guesses and start with modules that touch hardware like dahdi or any timing cruft, but the above process will work -- even after a couple of beers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 hangs up call after 20s
On Wed, 22 Dec 2010 13:22:47 -0500, Bruce B bruceb...@gmail.com wrote: This is a NAT issue like noted before. Try: localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0 instead of: localnet=192.168.0.0/24 http://192.168.0.0/24Also, make sure you have all your VPN connections as localnet and other side subnet as localnet as well if you are using VPN. Otherwise, open the neccessary ports needed for SIP and RTP. If you note your router type someone might be able to help more specifically. Thanks Bruce for the tip, but Asterisk still hangs up after 20s when the call originates from the remote user on the Net. The router is built by my ISP, so it has no brand/model. I believe it's based on OpenBSD. I have no VPN: The remote SIP user connects out using STUN. While going through the debug messages, I can see that at some point in the call, Asterisk tries to send SIP messages to the remote user... using Asterisk's public IP address instead of the remote user's IP address :-/ But then, I'm not clear at how to set things up so that remote users can register with Asterisk and be part of the dialplan just like they were on the LAN, with both Asterisk/local and remote users behind their respective NAT firewall, so I would have been very lucky to get this working without more investigation :-) If someone has a working configuration where... 1) Asterisk and some users are on a private LAN behind a NAT firewall 2) some roadwarriors, behind their own NAT firewall, are allowed to register with Asterisk, and make/receive calls just like they were in the office 3) the NAT firewall protecting the Asterisk server has SIP and RTP/RTCP ports mapped, while the NAT firewall protecting the remote user has its ports open dynamically using STUN ... I'm interested in how you set things up. Thank you. -- _ -- 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 hangs up call after 20s
Hi Gilles, If someone has a working configuration where... 1) Asterisk and some users are on a private LAN behind a NAT firewall 2) some roadwarriors, behind their own NAT firewall, are allowed to register with Asterisk, and make/receive calls just like they were in the office 3) the NAT firewall protecting the Asterisk server has SIP and RTP/RTCP ports mapped, while the NAT firewall protecting the remote user has its ports open dynamically using STUN I have (almost) that and it is working fine. One user needed to use a different port than 5060 because his modem really loves to interfere with packets on 5060. Probably because their ISP can also provide a SIP/phone line and the ISP modem is assuming that all packets on 5060 are for that phone line even though it is not enabled. I don't use STUN anywhere. Anyway, in all cases it has worked as soon as the phone was able to register on my server (that was usually the hard part!). In sip.conf I have added for all the remote users the setting canreinvite=no. The downside to that setting is that Asterisk is always in the audio path. For my situation that does not really matter. Best regards, Jeroen Eeuwes -- _ -- 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 hangs up call after 20s
On Thu, 23 Dec 2010 15:54:59 +0100, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: In sip.conf I have added for all the remote users the setting canreinvite=no. The downside to that setting is that Asterisk is always in the audio path. For my situation that does not really matter. Thanks Jeroen. After more reading, I found what it was: To cut down on the hacking attempts I saw (since posting in this mailing list...), I decided to reconfigure my NAT router to use another port than UDP5060 for SIP while leaving it as-is on the inside so internal SIP clients would still connect to the usual port. But www.smartvox.co.uk/astfaq_configbehindnat.htm says: When configuring your NAT/firewall/router device, you will probably need to find the settings for port forwarding or one-to-one NAT. Make sure your NAT device does not use port address translation. i.e. if your Asterisk server expects to receive SIP messages on port 5060, make sure you also use port 5060 on the WAN port of your NAT device to forward these messages. Similarly, make sure the same range of port numbers are forwarded on the WAN port for RTP as will receive the RTP on the Asterisk server. Reconfiguring the NAT router back to UDP5060 solved the problem: Calls originating from the remote SIP client are not longer cut off at 20s by Asterisk. Thanks everyone for the help. -- _ -- 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 hangs up call after 20s
Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: == WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' == I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying Call established: www.pastebin.com/x6MgnrpG There's also this oddity on line 50: Scheduling destruction of SIP dialog. FWIW, in sip.conf, for the remote XLite user, I tried nat=no and nat=yes, with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using t1min=500 in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated. -- _ -- 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 hangs up call after 20s
On 22 December 2010 12:44, Gilles codecompl...@free.fr wrote: Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: == WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' == I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying Call established: www.pastebin.com/x6MgnrpG There's also this oddity on line 50: Scheduling destruction of SIP dialog. FWIW, in sip.conf, for the remote XLite user, I tried nat=no and nat=yes, with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using t1min=500 in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated. Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Cheers, Steve -- _ -- 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 hangs up call after 20s
Hello, you have a typicall nat issue. Asterisk receives messages from the phone but cannot send any messages back (thats why it tries to resend the 200 ok message 6 times). try setting qualify=yes to your sip peers config to keep the nat port open. best regards stefan Am 22.12.10 13:44, schrieb Gilles: Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine; However, when the _remote_ XLite calls the local XLite, things work OK until precisely 20s, where Asterisk decides to hang up, and displays the following error message in the console: == WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno 2 (Critical Response) WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no reply to our critical packet. == Spawn extension (my-phones, local-xlite-extension, 1) exited non-zero on 'SIP/unused-008008e4' == I'm no SIP expert, but based on the debug session, before deciding to hang up, Asterisk tries 6 times to send an OK message to the remote XLite, and doesn't seem to get an answer. FWIW, after Asterisk has hung up, the remote XLite remains off-hook, oblivious to this error and keeps displaying Call established: www.pastebin.com/x6MgnrpG There's also this oddity on line 50: Scheduling destruction of SIP dialog. FWIW, in sip.conf, for the remote XLite user, I tried nat=no and nat=yes, with no difference. I'm actually not sure how to configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using t1min=500 in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated. -- _ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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 hangs up call after 20s
On Wed, 22 Dec 2010 14:31:32 +0100, Stefan Schmidt s...@sil.at wrote: you have a typicall nat issue. Asterisk receives messages from the phone but cannot send any messages back (thats why it tries to resend the 200 ok message 6 times). try setting qualify=yes to your sip peers config to keep the nat port open. Thanks for the idea, but all users are defined with qualify=yes: = /etc/asterisk cat sip.conf [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = yes ;allowexternalinvites=yes externip=public IP localnet=192.168.0.0/24 ;Other IPs can still REGISTER :-/ deny=0.0.0.0/0 permit=VOSP IP/255.255.255.255 permit = 192.168.0.0/255.255.255.0 alwaysauthreject=yes ;for safety context = dummmy ;all RTP packets go through Asterisk canreinvite=no ;makes no difference: still hangs up ;t1min=500 disallow=all allow=ulaw allow=alaw allow=gsm register = me:p...@vosp.com [vosp_outgoing] type=peer host=vosp.com username=me secret=mysecret fromuser=me fromdomain=vosp.com nat=yes canreinvite=no qualify=yes [vosp_incoming] type=peer host=vosp.com context=from_vosp nat=yes canreinvite=no insecure=port,invite qualify=yes ;(!) means it's a template [sets](!) type=friend context=my-phones host=dynamic qualify=yes nat=no [local-xlite](sets) secret=mysecret [remote-xlite](sets) secret=mysecret ;remote extension behind own NAT: nat=yes or nat=no? ;makes no difference : still hangs up ;nat=yes nat=no = What's weird, is that the remote XLite can successfully call the local XLite and I get sound both ways, and it's only 20s into the call that Asterisk decides to give up and hang up (while the remote side still thinks everything's OK). I tried SJphone instead of XLite, same result. Could it be some wrong configuration in Asterisk? Thank you. -- _ -- 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 hangs up call after 20s
On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com wrote: Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Thanks for the tip, but I get the same problem with SJPhone and PhonerLite, so it looks like a problem in Asterisk. -- _ -- 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 hangs up call after 20s
This is a NAT issue like noted before. Try: localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0 instead of: localnet=192.168.0.0/24 http://192.168.0.0/24Also, make sure you have all your VPN connections as localnet and other side subnet as localnet as well if you are using VPN. Otherwise, open the neccessary ports needed for SIP and RTP. If you note your router type someone might be able to help more specifically. -Bruce On Wed, Dec 22, 2010 at 12:27 PM, Gilles codecompl...@free.fr wrote: On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com wrote: Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Thanks for the tip, but I get the same problem with SJPhone and PhonerLite, so it looks like a problem in Asterisk. -- _ -- 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] Asterisk hangs up for some calls
Dear list;
I'm trying for forward some calls to an others asterisk using IAX2 protocol.
But My asterisk can forward some calls and for others it hangs up automaticaly.
Before my asterisk was working perfectly, i do not know what is happening!!
When i try directly zoiper with my provider's asterisk it works perfectly.
Here is the output from the cli when i made a call that asterisk hangs up:
Verbosity is at least 3
-- Accepting AUTHENTICATED call from 192.168.1.5:
requested format = unknown,
requested prefs = (ulaw|slin|alaw),
actual format = ulaw,
host prefs = (gsm|ulaw|alaw),
priority = mine
-- Executing [00212675410...@pstn:1] Set(IAX2/#000105-12477,
calleeNumber=011212675410113) in new stack
-- Executing [00212675410...@pstn:2] AGI(IAX2/#000105-12477,
agi://localhost/ManageCalls.agi?when=beforecalleeNumber=011212675410113) in
new stack
-- AGI Script
agi://localhost/ManageCalls.agi?when=beforecalleeNumber=011212675410113
completed, returning 0
-- Executing [00212675410...@pstn:3] Dial(IAX2/#000105-12477,
IAX2/mylo...@pstn/011212675410113||S(348)) in new stack
-- Setting call duration limit to 348 seconds.
-- Called mylo...@pstn/011212675410113
-- Call accepted by 8.17.37.23 (format ulaw)
-- Format for call is ulaw
-- Hungup 'IAX2/pstn-533'
-- No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'IAX2/#000105-12477' status is 'NOANSWER'
-- Executing [...@pstn:1] DeadAGI(IAX2/#000105-12477,
agi://localhost/ManageCalls.agi?when=after) in new stack
-- AGI Script agi://localhost/ManageCalls.agi?when=after completed,
returning 0
-- Hungup 'IAX2/#000105-12477'
here is my config:
[pstn]
exten=_00X.,1,Set(calleeNumber=011${EXTEN:2})
exten=_00X.,n,AGI(agi://localhost/ManageCalls.agi?when=beforecalleeNumber=${calleeNumber})
exten =_00X.,n,Dial(IAX2/mylo...@pstn/${calleeNumber},,S(${SECONDS-REMAINING}))
exten = h,1,DeadAGI(agi://localhost/ManageCalls.agi?when=after)
Thanks in advance for your help
--
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Re: [asterisk-users] Asterisk hangs up for some calls
On Tue, 15 Jun 2010, Adil Zaaraoui wrote: I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. 1) What is different about the numbers? Are some international or to countries restricted by your provider? 2) What does your provider say when you tell them a particular destination failed? 3) If you enable IAX debugging, you may get a clue or get output that may be helpful to others. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 hangs on STRPTIME
On Sunday 07 October 2007 20:56, Baji Panchumarti wrote: could this be the reason for my problem ? ( I am using a 64 bit AMD processor ) 2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED] * main/stdtime/private.h, main/stdtime/tzfile.h, include/asterisk/localtime.h, main/stdtime/localtime.c: Working on issue #10531 exposed a rather nasty 64-bit issue on ast_mktime, so we updated the localtime.c file from source. Next we'll have to write ast_strptime to match. In a word, yes. -- Tilghman ___ --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 hangs on STRPTIME
On 10/8/07, Tilghman Lesher wrote: On Sunday 07 October 2007 20:56, Baji Panchumarti wrote: could this be the reason for my problem ? ( I am using a 64 bit AMD processor ) 2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED] * main/stdtime/private.h, main/stdtime/tzfile.h, include/asterisk/localtime.h, main/stdtime/localtime.c: Working on issue #10531 exposed a rather nasty 64-bit issue on ast_mktime, so we updated the localtime.c file from source. Next we'll have to write ast_strptime to match. In a word, yes. thank you for replying and for documenting the issue in the first place. I am migrating my * stuff to a 32bit machine, I would be happy to test and report back this and any other 64-bit bug/feature on my AMD box when needed. -baji. -- ___ --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 hangs on STRPTIME
Successful Post mortem :
Output for below as seen on the 32-bit machine for STRPTIME
-- Executing [EMAIL PROTECTED]:6] Set(IAX2/4883-2, v_ts=) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(IAX2/4883-2, v_ts=2007-10-08) in
new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(IAX2/4883-2, 2007-10-08) in new
stack
-- Executing [EMAIL PROTECTED]:9] Set(IAX2/4883-2, v_ts=) in new stack
-- Executing [EMAIL PROTECTED]:10] Set(IAX2/4883-2, v_ts=1191665043) in
new stack
-- Executing [EMAIL PROTECTED]:11] NoOp(IAX2/4883-2, 1191665043) in new
stack
Tilghman = Gracee
-baji.
--
On 10/7/07, Baji Panchumarti [EMAIL PROTECTED] wrote:
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten = s,n,Set(v_ts=)
exten =
s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)})
exten = s,n,NoOp(${v_ts})
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRPTIME(2007-10-06
05:04:03|America/New_York|%Y-%m-%d %H:%M:%S)})
exten = s,n,NoOp(${v_ts})
;
I get the output :
-- Executing [ [EMAIL PROTECTED]:6] Set(IAX2/4883-1, v_ts=) in new
stack
-- Executing [EMAIL PROTECTED]:7] Set(IAX2/4883-1, v_ts=2007-10-07) in
new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(IAX2/4883-1, 2007-10-07) in new
stack
-- Executing [EMAIL PROTECTED]:9] Set(IAX2/4883-1, v_ts=) in new
stack
If this is a reported bug that has been fixed in 1.4.12, I can upgrade to
it,
but I'd like to know.
tia.
-baji.
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[asterisk-users] asterisk hangs on STRPTIME
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)})
exten = s,n,NoOp(${v_ts})
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRPTIME(2007-10-06
05:04:03|America/New_York|%Y-%m-%d %H:%M:%S)})
exten = s,n,NoOp(${v_ts})
;
I get the output :
-- Executing [EMAIL PROTECTED]:6] Set(IAX2/4883-1, v_ts=) in new stack
-- Executing [EMAIL PROTECTED]:7] Set(IAX2/4883-1, v_ts=2007-10-07) in
new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(IAX2/4883-1, 2007-10-07) in new
stack
-- Executing [EMAIL PROTECTED]:9] Set(IAX2/4883-1, v_ts=) in new stack
If this is a reported bug that has been fixed in 1.4.12, I can upgrade to
it,
but I'd like to know.
tia.
-baji.
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Re: [asterisk-users] asterisk hangs on STRPTIME
could this be the reason for my problem ?
( I am using a 64 bit AMD processor )
2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED]
* main/stdtime/private.h, main/stdtime/tzfile.h,
include/asterisk/localtime.h, main/stdtime/localtime.c: Working
on issue #10531 exposed a rather nasty 64-bit issue on
ast_mktime, so we updated the localtime.c file from source.
Next we'll have to write ast_strptime to match.
1.4.12 changelog
http://svn.digium.com/view/asterisk/tags/1.4.12/ChangeLog?view=markup
thnx,
-baji.
--
On 10/7/07, I wrote:
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)})
exten = s,n,NoOp(${v_ts})
;
exten = s,n,Set(v_ts=)
exten = s,n,Set(v_ts=${STRPTIME(2007-10-06
05:04:03|America/New_York|%Y-%m-%d %H:%M:%S)})
exten = s,n,NoOp(${v_ts})
;
I get the output :
-- Executing [ [EMAIL PROTECTED]:6] Set(IAX2/4883-1, v_ts=) in new
stack
-- Executing [EMAIL PROTECTED]:7] Set(IAX2/4883-1, v_ts=2007-10-07) in
new stack
-- Executing [EMAIL PROTECTED]:8] NoOp(IAX2/4883-1, 2007-10-07) in new
stack
-- Executing [EMAIL PROTECTED]:9] Set(IAX2/4883-1, v_ts=) in new
stack
If this is a reported bug that has been fixed in 1.4.12, I can upgrade
to it,
but I'd like to know.
tia.
-baji.
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Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
I found a subtle difference between the two traces you sent (the call that works and the call that gets dropped). This may or may not be what's causing the problem. The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC). There is nothing wrong with the re-transmission as such, but I noticed a potential bug in Asterisk in the way it responds to an INVITE retransmission. Asterisk is bumping up the session version number in the retransmitted 200 OK's SDP. This is as if Asterisk is treating the INVITE retransmission as a RE-INVITE. Asterisk sends 200 OK: o=root 16300 16300 IN IP4 203.89.nnn.nnn Asterisk sends 200 OK (retransmission): o=root 16300 16301 IN IP4 203.89.nnn.nnn Ideally, this bug should have nothing to do with why Asterisk is ignoring the ACK (which is why it keeps reatrasmitting the 200 OK and eventually drops the call). However, if you can confirm that all dropped calls have INVITE retransmission then that might give us a clue? Raj On 4/1/07, kjcsb [EMAIL PROTECTED] wrote: One potential reason could be that the ACK request being sent to Asterisk is malformed. Notice branch=0 in the top Via. This should start with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction. While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction. Clearly, Asterisk is dropping this ACK on the floor. OK. But in the calls that don't get dropped, the branch=0 is present also. See below for an example: -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 03:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 11402 11402 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 39686 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:39686 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received= 147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b7908550, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b7908550, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b7908550, ) in new stack We're at 203.89.nnn.nnn port 15804 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received= 147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED];tag=as7ecf44d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn
Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
One potential reason could be that the ACK request being sent to Asterisk is malformed. Notice branch=0 in the top Via. This should start with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction. While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction. Clearly, Asterisk is dropping this ACK on the floor. OK. But in the calls that don't get dropped, the branch=0 is present also. See below for an example: -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 03:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 11402 11402 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 39686 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:39686 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b7908550, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b7908550, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b7908550, ) in new stack We're at 203.89.nnn.nnn port 15804 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED];tag=as7ecf44d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 15804 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait(SIP/649977-b7908550, 1) in new stack capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK5ba4f251;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED];tag=as7ecf44d1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 --- (12 headers 0 lines) --- -- Executing Set(SIP/649977-b7908550, TIMEOUT(digit)=3) in new stack -- Digit timeout set to 3 -- Executing Set(SIP/649977-b7908550, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing BackGround(SIP/649977-b7908550, custom/11000111000-welcome)
[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
I have the following scenario: PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip = 203.89.nnn.nnn disallow=all allow=ulaw allow=alaw language=nz [DLS] username=649977 type=peer host=domain.co.nz context=from-trunk canreinvite=no Note that Asterisk registers with proxy: 649977:[EMAIL PROTECTED]/649977 sip debug peer DLS -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 13636 13636 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 36274 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:36274 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Goto(SIP/649977-b791bb60, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b791bb60, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b791bb60, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b791bb60, ) in new stack We're at 203.89.nnn.nnn port 11648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 11648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait(SIP/649977-b791bb60, 1) in new stack capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type:
Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
One potential reason could be that the ACK request being sent to Asterisk is malformed. Notice branch=0 in the top Via. This should start with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction. While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction. Clearly, Asterisk is dropping this ACK on the floor. Raj -- SIP read from 147.202.nnn.nnn:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK61752efe;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 --- (12 headers 0 lines) --- Retransmitting #6 (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received= 147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 11648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0c397910;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 --- (12 headers 0 lines) --- == Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/649977-b791bb60' -- Executing Hangup(SIP/649977-b791bb60, ) in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/649977-b791bb60' Destroying call '[EMAIL PROTECTED]' capetown*CLI Any advice in resolving this issue would be greatly appreciated. Regards Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --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] Asterisk hangs up SIP call after 6 200 retransmits
I have the following scenario: PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip = 203.89.nnn.nnn disallow=all allow=ulaw allow=alaw language=nz [DLS] username=649977 type=peer host=domain.co.nz context=from-trunk canreinvite=no Note that Asterisk registers with proxy: 649977:[EMAIL PROTECTED]/649977 sip debug peer DLS -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 13636 13636 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 36274 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:36274 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Goto(SIP/649977-b791bb60, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b791bb60, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b791bb60, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b791bb60, ) in new stack We're at 203.89.nnn.nnn port 11648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 11648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait(SIP/649977-b791bb60, 1) in new stack capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp
Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while
On Oct 19, 2006, at 3:00 PM, [EMAIL PROTECTED] wrote:Date: Thu, 19 Oct 2006 09:30:38 -0500 From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while 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 Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? No. They are not set. i.e. default___ --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] Asterisk hangs up on incoming analog calls after a while
Robert La Ferla wrote: I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ? ___ --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] Asterisk hangs up on incoming analog calls after a while
I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. How can I diagnose this problem? Has anyone experienced anything similar? ___ --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] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute
Check sip.conf parameters: rtptimeout rtpholdtimeout David Gagnon wrote: I would recommend you to call Unlimitel as they have a very good support. Or just send a copy of your post to : [EMAIL PROTECTED] David _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Envoyé : 7 septembre 2006 11:32 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute Hi, I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in my case). My job requires me to attend conference calls regularly, and I am usually there as a silent listener. Therefore, I mute my phone. I`ve noticed that if I mute my phone, after 10-15 of being muted, the line hangs up. I had the same problem with my GXP-2000 before, so I dismissed the phone as being the problem. If I unmute regularly (or the entire time), the line doesnt hang up (until it reaches max timeout of course, which is much more than 15 minutes). So the problem is my phone is muted. I have observed that about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring issue for sure. What I am left with is Asterisk (or my VoIP provider) as the issue. Since I only have control on my own Asterisk server, I thought I should start there. What setting could cause this? I have a fairly fancy dialplan, but I havent changed anything else than the diaplan. All system-wide Asterisk settings are default as far as I know. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute
Hi Mike, I have a IP 500 and do similar mute for calls etc. No probs here using Trixbox. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon Sent: Thursday, 7 September 2006 9:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute I would recommend you to call Unlimitel as they have a very good support. Or just send a copy of your post to : [EMAIL PROTECTED] David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Envoyé: 7 septembre 2006 11:32 À: 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet: [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute Hi, I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in my case). My job requires me to attend conference calls regularly, and I am usually there as a silent listener. Therefore, I mute my phone. I`ve noticed that if I mute my phone, after 10-15 of being muted, the line hangs up. I had the same problem with my GXP-2000 before, so I dismissed the phone as being the problem. If I unmute regularly (or the entire time), the line doesnt hang up (until it reaches max timeout of course, which is much more than 15 minutes). So the problem is my phone is muted.I have observed that about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring issue for sure. What I am left with is Asterisk (or my VoIP provider) as the issue. Since I only have control on my own Asterisk server, I thought I should start there. What setting could cause this? I have a fairly fancy dialplan, but I havent changed anything else than the diaplan. All system-wide Asterisk settings are default as far as I know. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk hangs up after 10-15 minuteswhenSIPPhone is on mute
Thanks for the words of hope. I actually think I found something, I set LineSeize=`30` in the Polycom config file (I still dont know what that means...) but it seemed to work once (out of one try) "Once" doesn`t mean "always", but it is definitely better than "never" BTW, Someone mentionned how Unlimitel (one of my VoIP providers) has great support and I agree. I just know they can`t help me because the issue is between my PBX and my SIP phone, not with them. Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: September 8, 2006 7:46 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Asterisk hangs up after 10-15 minuteswhenSIPPhone is on mute Hi Mike, I have a IP 500 and do similar mute for calls etc. No probs here using Trixbox. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David GagnonSent: Thursday, 7 September 2006 9:28 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute I would recommend you to call Unlimitel as they have a very good support. Or just send a copy of your post to : [EMAIL PROTECTED] David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de MikeEnvoyé: 7 septembre 2006 11:32À: 'Asterisk Users Mailing List - Non-Commercial Discussion'Objet: [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute Hi, I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in my case). My job requires me to attend conference calls regularly, and I am usually there as a silent listener. Therefore, I mute my phone. I`ve noticed that if I mute my phone, after 10-15 of being muted, the line hangs up. I had the same problem with my GXP-2000 before, so I dismissed the phone as being the problem. If I unmute regularly (or the entire time), the line doesnt hang up (until it reaches max timeout of course, which is much more than 15 minutes). So the problem is my phone is muted.I have observed that about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring issue for sure. What I am left with is Asterisk (or my VoIP provider) as the issue. Since I only have control on my own Asterisk server, I thought I should start there. What setting could cause this? I have a fairly fancy dialplan, but I havent changed anything else than the diaplan. All system-wide Asterisk settings are default as far as I know. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk hangs up after 10-15 minutes when SIP Phone is on mute
Hi, I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in my case). My job requires me to attend conference calls regularly, and I am usually there as a silent listener. Therefore, I mute my phone. I`ve noticed that if I mute my phone, after 10-15 of being muted, the line hangs up. I had the same problem with my GXP-2000 before, so I dismissed the phone as being the problem. If I unmute regularly (or the entire time), the line doesnt hang up (until it reaches max timeout of course, which is much more than 15 minutes). So the problem is my phone is muted.I have observed that about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring issue for sure. What I am left with is Asterisk (or my VoIP provider) as the issue. Since I only have control on my own Asterisk server, I thought I should start there. What setting could cause this? I have a fairly fancy dialplan, but I havent changed anything else than the diaplan. All system-wide Asterisk settings are default as far as I know. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute
I would recommend you to call Unlimitel as they have a very good support. Or just send a copy of your post to : [EMAIL PROTECTED] David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Envoyé: 7 septembre 2006 11:32 À: 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet: [asterisk-users] Asterisk hangs up after 10-15 minutes when SIPPhone is on mute Hi, I have a Polycom 501 connected to Asterisk 1.2.4 (and then connected to a VOIP provider, Unlimitel in my case). My job requires me to attend conference calls regularly, and I am usually there as a silent listener. Therefore, I mute my phone. I`ve noticed that if I mute my phone, after 10-15 of being muted, the line hangs up. I had the same problem with my GXP-2000 before, so I dismissed the phone as being the problem. If I unmute regularly (or the entire time), the line doesnt hang up (until it reaches max timeout of course, which is much more than 15 minutes). So the problem is my phone is muted.I have observed that about 6 times (out of 6 tries) in the last 4 months. It`s a reccuring issue for sure. What I am left with is Asterisk (or my VoIP provider) as the issue. Since I only have control on my own Asterisk server, I thought I should start there. What setting could cause this? I have a fairly fancy dialplan, but I havent changed anything else than the diaplan. All system-wide Asterisk settings are default as far as I know. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hangs the whole system
In our case, it was cpuspeed (a daemon) interfering with the zaptel drivers. Paul Hales Technical Manager AsteriskIT - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 27, 2006 10:14 PM Subject: Re: [Asterisk-Users] Asterisk Hangs the whole system A.R. Nasir Qureshi wrote: Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. Both are possible. If you watched the cvs/svn commits over the last year or so, several asterisk issues have been identified and corrected relating to mem allocation, dereferencing, etc, etc. I don't know that anyone has actually kept track of bugs vs versions to know which versions might be suspect, but it might help if you'd include which distro/kernel you're running, asterisk version, types of cards installed, etc. You might also try running memtest just to rule out memory failures or issues. ___ --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] Asterisk Hangs the whole system
Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. -- Regards, Nasir. ___ --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] Asterisk Hangs the whole system
Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. People in the past had the problem. I dont remember what the cause of the problem was. Try looking at the archives. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk Hangs the whole system
A.R. Nasir Qureshi wrote: Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. Both are possible. If you watched the cvs/svn commits over the last year or so, several asterisk issues have been identified and corrected relating to mem allocation, dereferencing, etc, etc. I don't know that anyone has actually kept track of bugs vs versions to know which versions might be suspect, but it might help if you'd include which distro/kernel you're running, asterisk version, types of cards installed, etc. You might also try running memtest just to rule out memory failures or issues. ___ --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] Asterisk hangs up on incoming PSTN line to analog extension
I have encountered the following problem with the latest Asterisk source (as of 4/23/2006): Someone calls me on my PSTN line, it then dials my analog extension (I have both SIP and analog phones where all analog phones are a shared extension.) After a while, I get a busy signal. How can I further diagnose this? What could be the problem? ___ --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] Asterisk hangs up on incoming PSTN line to analog extension
Robert, I have the same problem, and I discover that when you use de flash or hang up you need a time toasterisk detect that you not do a flash function. A suggest is put de little time umflash. []' Leonardo Silva 2006/4/23, Robert La Ferla [EMAIL PROTECTED]: I have encountered the following problem with the latest Asterisk source(as of 4/23/2006):Someone calls me on my PSTN line, it then dials my analog extension (I have both SIP and analog phones where all analog phones are a sharedextension.)After a while, I get a busy signal.How can I furtherdiagnose this?What could be the problem?___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Leonardo Silvafone: 16 8146-1143 ___ --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] Asterisk hangs up - h323
This is third time today that my Asterisk hangs up. It seams that I have problems with h323. I'm using ooh323 from Asterisk add-ons. I have the following configuration Asterisk 1.2.1 Asterisk-addons 1.2.1 Fedora Core 4 I'm using SIP phones and h323 trunk to my VoIP provider Like I said this is third time today that he hang's up. First time, I came at work and Asterisk was down. Second time I tried to call, and Asterisk was down (not sure at that wary moment or before I tried to call). So, I decide to start logging and this is what I received just before Asterisk died. Anyway, I tried to reload from CLI and that is when he died. What can I do to check why it's happening? I have plenty of disk space, lots of free ram and processor is idle for more than 80%. I think it could be because of alaw codec that I use (my provider requires it) and this is what is in ooh323.conf file (ONLY ulaw, gsm, g729 and g7231 supported as of now). But Like I said, it works for several hours and then it dies... So I don't think that is it. ooh323.conf [general] bindaddr=xxx.xxx.xxx.xxx h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE context=incomingh323 disallow=all allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 full.pbx Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/manager.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Feb 28 14:04:15 NOTICE[5018] cdr.c: CDR simple logging enabled. Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/rtp.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/rtp.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: == RTP Allocating from port range 1 - 2 Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/musiconhold.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/musiconhold.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_indications.so' (Indications Configuration) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_adsi.so' (ADSI Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_features.so' (Call Features Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/features.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/features.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Blind Transfer (blindxfer) to sequence '#1' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Attended Transfer (atxfer) to sequence '#2' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature One Touch Monitor (automon) to sequence '#3' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Disconnect Call (disconnect) to sequence '#0' Feb 28 14:04:15 DEBUG[5018] res_features.c: Removed old parking extension [EMAIL PROTECTED] Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Added extension '700' priority 1 to parkedcalls Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_config_mysql.so' (MySQL RealTime Configuration Driver) Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Host: Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Port: 0 Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime User: Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Password: Feb 28 14:04:15 ERROR[5018] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) Feb 28 14:04:15 WARNING[5018] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) Feb 28 14:04:15 VERBOSE[5018] logger.c: == MySQL RealTime reloaded. Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2)) Feb 28 14:04:15 ERROR[5018] chan_iax2.c: Unable to load config iax.conf Feb 28 14:04:15 VERBOSE[5018] logger.c: == Loaded firmware 'iaxy.bin' Feb 28 14:04:15 NOTICE[5018] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled. Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny)) Feb 28 14:04:15 NOTICE[5018] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_local.so' (Local Proxy Channel) Feb 28 14:04:15
Re: [Asterisk-Users] Asterisk hangs on 1.2.1
Mark Johnson wrote: Anyone have any idea what's causing this or how to debug it? I'm pretty sure the root cause is with chan_sccp.so, but not sure how to prove it. I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from 12-17-2005. Now, once or twice a week, I get this on the console: Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf1013e0', 10 retries! Once this happens, all of my sccp phones drop offline and attempt to register. I get no sccp messages on the console. There's really nothing on the console to indicate any sort of problem. If I try to do an unload chan_sccp.so and then load it back, all of my SIP phones lose their registrations, none of my Zap channels work and I have to kill Asterisk and restart it. Is this an Asterisk problem or an SCCP problem? Help!! It did it to me again. I enabled full logging and here's what I get. All the 7910's drop off line and try to reregister. All SCCP messages on the CLI stop. Anytime I try a show channels I get the Avoided deadlock message. Here's what the logfile shows. Any ideas? And is there a way to fix the deadlock without restarting Asterisk? Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf002d10', 10 retries! Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:09 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:09 DEBUG[6606] channel.c: Avoiding deadlock for 'SCCP/204-0205' Feb 1 09:17:09 WARNING[6606] channel.c: Avoided deadlock for '0xbf002d10', 10 retries! ___ --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] Asterisk hangs on 1.2.1
Mark Johnson ha scritto: Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf002d10', 10 retries! Yes, the chan_sccp could lock the asterisk channel. To fix it I need a sccp debug 10 log of the call that is locking the channel Sergio ___ --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] Asterisk hangs on 1.2.1
Anyone have any idea what's causing this or how to debug it? I'm pretty sure the root cause is with chan_sccp.so, but not sure how to prove it. I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from 12-17-2005. Now, once or twice a week, I get this on the console: Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf1013e0', 10 retries! Once this happens, all of my sccp phones drop offline and attempt to register. I get no sccp messages on the console. There's really nothing on the console to indicate any sort of problem. If I try to do an unload chan_sccp.so and then load it back, all of my SIP phones lose their registrations, none of my Zap channels work and I have to kill Asterisk and restart it. Is this an Asterisk problem or an SCCP problem? Help!! Mark ___ --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] Asterisk hangs
Since some CVS Updates the asterisk hangs after command: reload or restart now. Then i have to kill -9 th eprocess. Nothing will be outout inside the CLI but i can type commands. Somebody know this problem? And the CallerID bug still seems to be in there too. Regards rene ___ --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] Asterisk hangs
Hi Rene, Yes, I've seen that but our version from CVS is a month or so old os it may well have been rectified now. On our version reloads cause the process to die about 50% of the time, work fine about 45% and cause it to hang in the way your describe probably 5%. Simon On 19/10/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote: Since some CVS Updates the asterisk hangs after command: reload orrestart now.Then i have to kill -9 th eprocess.Nothing will be outout inside the CLI but i can type commands.Somebody know this problem? And the CallerID bug still seems to be in there too.Regards rene___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
[Asterisk-Users] Asterisk hangs the establised calls
Hello all, I have an asterisk 1.0.3 stable instaled on a box. All works fine with this machine, but the only problem i get is that suddenly the machine hangs up all the establised calls and we have to call again. This problem occurs twice a day and i don not know how to debug it. I read carefully the logs placed in /var/log/asterisk, but I can not find the reason for this hangs. I have to say that when this occurs, sometimes asterisk restart, if i make a "show uptime", I can see that asterisk has recently restart, but other times when the same occurs, I can see that asterisk is running seven hours ago, but our calls was hanged up too. I have to say that automaticaly (By own script) asterisk restart every night How could i debug that? Does your Asterisk hang suddenly all the establised calls? Do you know any command that help me finding the problem? Thanks for your time. Ismael. ___ 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] Asterisk hangs up when a call comes in
Hello, I am trying to setup an asterisk box on a simple isdn line with a fritz card. The Capi4Linux drivers are installed and seem to work correct as I can connect to an ISP, have not tried it with ISDN4Linux yet as I read that CAPI has many advantages over i4l ... but I think I will do this next. Next I have compiled zaptel, libpri (are these really needed for a fritz card?) and asterisk and finally did 'make samples' as these are my first expierience with asterisk and wanted to try the demo context. To get it work with CAPI i have then installed the chan_capi.so driver from junghanns.net ... here some output from the cli. [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 12 03:08:07 NOTICE[1024]: chan_capi.c:2338 mkif: ast_capi_pvt(43910906,43910906,demo,0x2,2) (1,2,64) (0)(0.80/0.80) 0 Feb 12 03:08:07 NOTICE[1024]: chan_capi.c:2338 mkif: ast_capi_pvt(43910906,43910906,demo,0x2,2) (1,2,64) (0)(0.80/0.80) 0 Feb 12 03:08:07 NOTICE[1024]: chan_capi.c:2694 load_module: this box has 1 capi controller(s) -- listening on contr1 CIPmask = 0x1fff03ff -- CAPI[contr1] supports DTMF -- CAPI[contr1] supports supplementary services HOLD/RETRIEVE TERMINAL PORTABILITY ECT 3PTY CF CD MCID CCBS MWI CCNR == Registered channel type 'CAPI' (Common ISDN API Driver (0.3.0) aLaw) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI -- started pbx on channel (callgroup=0)! Feb 12 03:08:19 WARNING[4101]: pbx.c:1778 ast_pbx_run: Channel 'CAPI[contr1/43910906]/0' sent into invalid extension 's' in context 'default', but no invalid handler -- CAPI Hangingup Feb 12 03:08:33 WARNING[5125]: pbx.c:1778 ast_pbx_run: Channel 'CAPI[contr1/43910906]/1' sent into invalid extension 's' in context 'default', but no invalid handler -- CAPI Hangingup -- started pbx on channel (callgroup=0)! Feb 12 03:08:47 WARNING[6149]: pbx.c:1778 ast_pbx_run: Channel 'CAPI[contr1/43910906]/2' sent into invalid extension 's' in context 'default', but no invalid handler -- CAPI Hangingup -- started pbx on channel (callgroup=0)! Feb 12 03:08:48 ERROR[3076]: chan_capi.c:1196 pipe_frame: wrote -1 bytes instead of 40 Any solution ?? bye bodo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
