[asterisk-users] Asterisk+h324m gateway issue
Hi , i worked with h324m gateway for 3g video calling .It configured successfully . my code in extensions.conf is [from-zaptel] exten = _X.,1,h324m_gw(0@mainmenu) exten=_X.,n,Hangup [mainmenu] exten = 0,1,h324m_gw_answer() exten = 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video call (either sip or through pri) , asterisk cli shows the following error -- Executing [123@from-zaptel:1] h324m_gw(SIP/100-b7602680, 0@mainmenu) in new stack localhost*CLI Disconnected from Asterisk server Executing last minute cleanups when i routed the call directly to [mainmenu] call stack at h324m_gw_answer() please help me ... Thanks Regards, Pankaj Pandey -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! [Jan 14 11:28:54] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:28:54] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28044] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:28:54] DEBUG[28075] channel.c: Avoiding deadlock for channel '0x114af2c0' Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Kind regards, Jonas. -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Sometimes I can see this messages too - but with no impact. It is a debug-message and should not indicate any problems. What does it mean when you say "Asterisk was no longer responding"? -Thorsten- -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Sometimes I can see this messages too - but with no impact. It is a debug-message and should not indicate any problems. What does it mean when you say Asterisk was no longer responding? -Thorsten- Hello, the debug-file is flooded with this message during 2 à 3 seconds and counts about 300 à 400 lines... So I don't think it's just a debug-message. Asterisk was not responding as in core show channels had no output, sip show peers had no output, core restart now did nothing... The Asterisk proces was still running though... Also: all registrations of SIP peers were lost. I could see that the IP-phones lost their registration to the Asterisk server. And they did not re-register untill the server was finally rebooted. Kind regards, Jonas. -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Gllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Sometimes I can see this messages too - but with no impact. It is a debug-message and should not indicate any problems. What does it mean when you say "Asterisk was no longer responding"? -Thorsten- Hello, the debug-file is flooded with this message during 2 3 seconds and counts about 300 400 lines... So I don't think it's just a debug-message. Asterisk was not responding as in "core show channels" had no output, "sip show peers" had no output, "core restart now" did nothing... The Asterisk proces was still running though... Also: all registrations of SIP peers were lost. I could see that the IP-phones lost their registration to the Asterisk server. And they did not re-register untill the server was finally rebooted. This message is repeated over 100 times. (You can take a look at the source code.) Which Asterisk-Version do you use? Did it happen before or again? -Thorsten- -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
On 01/14/2011 02:22 PM, Thorsten Göllner wrote: Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Sometimes I can see this messages too - but with no impact. It is a debug-message and should not indicate any problems. What does it mean when you say Asterisk was no longer responding? -Thorsten- Hello, the debug-file is flooded with this message during 2 à 3 seconds and counts about 300 à 400 lines... So I don't think it's just a debug-message. Asterisk was not responding as in core show channels had no output, sip show peers had no output, core restart now did nothing... The Asterisk proces was still running though... Also: all registrations of SIP peers were lost. I could see that the IP-phones lost their registration to the Asterisk server. And they did not re-register untill the server was finally rebooted. This message is repeated over 100 times. (You can take a look at the source code.) Which Asterisk-Version do you use? Did it happen before or again? -Thorsten- Hello, I use asterisk 1.6.2.10 As I said, this is the first time I experience this. I used 1.4 before, never had this. I'm using 1.6.2.10 now for about 5 à 6 months and this is the first time. Kind regards, Jonas. -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
Hi, We had the same problems. These problems accours when we try to send (from different servers) a lot of IAX calls to one server. (see couple of 100 calls at the same time) When we upgraded asterisk to version 1.8 we didn't get these problems. Regards, Arjan Kroon Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jonas Kellens Verzonden: 14-01-2011 14:31 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' On 01/14/2011 02:22 PM, Thorsten Göllner wrote: Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Sometimes I can see this messages too - but with no impact. It is a debug-message and should not indicate any problems. What does it mean when you say Asterisk was no longer responding? -Thorsten- Hello, the debug-file is flooded with this message during 2 à 3 seconds and counts about 300 à 400 lines... So I don't think it's just a debug-message. Asterisk was not responding as in core show channels had no output, sip show peers had no output, core restart now did nothing... The Asterisk proces was still running though... Also: all registrations of SIP peers were lost. I could see that the IP-phones lost their registration to the Asterisk server. And they did not re-register untill the server was finally rebooted. This message is repeated over 100 times. (You can take a look at the source code.) Which Asterisk-Version do you use? Did it happen before or again? -Thorsten- Hello, I use asterisk 1.6.2.10 As I said, this is the first time I experience this. I used 1.4 before, never had this. I'm using 1.6.2.10 now for about 5 à 6 months and this is the first time. Kind regards, Jonas. -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Kind regards, Jonas. There are many updates from 1.6.2.10 to stable. Try updating to stable 1.6.2.15 (16 any minute now) to fix this and other issues... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
On 01/14/2011 02:40 PM, Andrew Latham wrote: On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellensjonas.kell...@telenet.be wrote: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Kind regards, Jonas. There are many updates from 1.6.2.10 to stable. Try updating to stable 1.6.2.15 (16 any minute now) to fix this and other issues... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ Hello, so this can be fixed with a simple upgrade ?? Are there many changes when upgrading from 1.6.2.10 to 1.6.2.15 ? Because I don't like messing up things by upgrading to a version with features I don't know about. Kind regards, Jonas. -- _ -- 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] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
On Fri, Jan 14, 2011 at 10:46 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 01/14/2011 02:40 PM, Andrew Latham wrote: On Fri, Jan 14, 2011 at 7:55 AM, Jonas Kellensjonas.kell...@telenet.be wrote: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Kind regards, Jonas. There are many updates from 1.6.2.10 to stable. Try updating to stable 1.6.2.15 (16 any minute now) to fix this and other issues... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ Hello, so this can be fixed with a simple upgrade ?? Are there many changes when upgrading from 1.6.2.10 to 1.6.2.15 ? Because I don't like messing up things by upgrading to a version with features I don't know about. Kind regards, Jonas. For sub versions there should be no features and you can always read the CHANGES http://svn.asterisk.org/svn/asterisk/branches/1.6.2/CHANGES ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ftarz Sent: Thursday, January 13, 2011 9:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has answered the call. I've had people hang-up since they don't hear anything when they answer my calls. If I try the exact same call using an IAX route, the call is connected at my end just as soon as the PSTN number answers. I don't have any connection delays for incoming FXO calls. They are connected as soon as I answer the calls. Can anyone give me some pointers on where to start looking? Frank Two possible answers - #1 you have somehow got tonedur set to a value in the 100's (should be around 80) #2 you're calling a cell phone - this takes an extra 3-4 seconds. -- _ -- 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] CallerID and URL pop up for windows...
On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B bruceb...@gmail.com wrote: http://bestof.nerdvittles.com/applications/screenpop/But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay for that if it is added to URANG II Has someone tried IdentaPop? www.identafone.com/cidpop.html -- _ -- 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] Blind Transfer not working - 1.4.38
This is a heads up to everyone
Apparently this is a known but in the latest version on asterisk 1.4,
1.6 and 1.8
http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1-6-2-15-and-1-8-0-1
https://issues.asterisk.org/view.php?id=18185
On Thu, 2011-01-06 at 13:10 +, Ishfaq Malik wrote:
On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17
So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off.
I ran a SIP trace of this happening and it appears to be attempting to
do the transfer:
-
--- (12 headers 0 lines) ---
Call 7c5d5a603b2803fd7e451de826e4@x.x.x.x got a SIP call transfer from
caller: (REFER)!
SIP transfer to extension 504@pack-local by pack...@domain.co.uk
--- Transmitting (NAT) to x.x.x.x:52753 ---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
From: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
To: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
Call-ID: 7c5d5a603b2803fd7e451de826e4@x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:incoming mobile number@x.x.x.x
Content-Length: 0
set_destination: Parsing sip:PACK501@192.168.1.105:3072;line=guuuyf05 for
address/port to send to
set_destination: set destination to 192.168.1.105, port 3072
Reliably Transmitting (NAT) to x.x.x.x:52753:
NOTIFY sip:PACK501@192.168.1.105:3072;line=guuuyf05 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
From: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
To: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
Contact: sip:incoming mobile number@x.x.x.x
Call-ID: 7c5d5a603b2803fd7e451de826e4@87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: incoming mobile number sip:incoming mobile
number@x.x.x.x;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
___
But as stated above, extension 504 doesn't ring and the call dies.
Now 504 is a valid extensions in the context pack-local
select * from extensions where exten='_5XX';
+---++---+--+---+---+
| id| context| exten | priority | app | appdata
|
+---++---+--+---+---+
| 65127 | pack-local | _5XX |1 | Macro |
stdexten|${EXTEN}|pack-local|PACK |
+---++---+--+---+---+
Also, attended transfers work without a problem.
Both SIP phones used were Snom phones.
Has anyone encountered an issue like this before?
I spotted something new here, when I try to do the blind transfer I get
the following output on the console
== Spawn extension (pack-local, 504, 0) exited non-zero on
So why would it be looking at priority 0 rather than priority 1?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Blind Transfer not working - 1.4.38
Hi,
1.6.2.16rc1 does not have this problem (that`s why I am running a release
candidate right now). Can`t say about 1.4 versions, but it`s safe to say
whatever they fixed will be out in the next version.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, January 14, 2011 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind Transfer not working - 1.4.38
This is a heads up to everyone
Apparently this is a known but in the latest version on asterisk 1.4,
1.6 and 1.8
http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1
-6-2-15-and-1-8-0-1
https://issues.asterisk.org/view.php?id=18185
On Thu, 2011-01-06 at 13:10 +, Ishfaq Malik wrote:
On Wed, 2011-01-05 at 15:47 +, Ishfaq Malik wrote:
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments
and blind transferring seems to have suddenly stopped working. It
was working fine under 1.4.17
So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off.
I ran a SIP trace of this happening and it appears to be attempting
to do the transfer:
-
--- (12 headers 0 lines) ---
Call 7c5d5a603b2803fd7e451de826e4@x.x.x.x got a SIP call transfer
from caller: (REFER)!
SIP transfer to extension 504@pack-local by pack...@domain.co.uk
--- Transmitting (NAT) to x.x.x.x:52753 ---
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rpor
t=52753
From: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
To: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
Call-ID: 7c5d5a603b2803fd7e451de826e4@x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces
Contact: sip:incoming mobile number@x.x.x.x
Content-Length: 0
set_destination: Parsing
sip:PACK501@192.168.1.105:3072;line=guuuyf05 for address/port to
send to
set_destination: set destination to 192.168.1.105, port 3072
Reliably Transmitting (NAT) to x.x.x.x:52753:
NOTIFY sip:PACK501@192.168.1.105:3072;line=guuuyf05 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
From: incoming mobile number sip:incoming mobile
number@x.x.x.x;tag=as4d0dbc04
To: sip:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp
Contact: sip:incoming mobile number@x.x.x.x
Call-ID: 7c5d5a603b2803fd7e451de826e4@87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: incoming mobile number sip:incoming mobile
number@x.x.x.x;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
___
But as stated above, extension 504 doesn't ring and the call dies.
Now 504 is a valid extensions in the context pack-local select *
from extensions where exten='_5XX';
+---++---+--+---+---
+
| id| context| exten | priority | app | appdata
|
+---++---+--+---+---
+
| 65127 | pack-local | _5XX |1 | Macro |
stdexten|${EXTEN}|pack-local|PACK |
+---++---+--+---+---
+
Also, attended transfers work without a problem.
Both SIP phones used were Snom phones.
Has anyone encountered an issue like this before?
I spotted something new here, when I try to do the blind transfer I
get the following output on the console
== Spawn extension (pack-local, 504, 0) exited non-zero on
So why would it be looking at priority 0 rather than priority 1?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
_
-- 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
--
Ishfaq Malik
Software Developer
Re: [asterisk-users] 5-7 second delay in connecting outgoing FXO calls
On Fri, 14 Jan 2011, Danny Nicholas wrote: I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has answered the call. I've had people hang-up since they don't hear anything when they answer my calls. If I try the exact same call using an IAX route, the call is connected at my end just as soon as the PSTN number answers. I don't have any connection delays for incoming FXO calls. They are connected as soon as I answer the calls. Can anyone give me some pointers on where to start looking? Frank Two possible answers - #1 you have somehow got tonedur set to a value in the 100's (should be around 80) #2 you're calling a cell phone - this takes an extra 3-4 seconds. Another possible is DNS - or the lack of it. Asterisk doing a reverse DNS lookup for the SIP phone at connection time? Gordon -- _ -- 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] 5-7 second delay in connecting outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has answered the call. I've had people hang-up since they don't hear anything when they answer my calls. If I try the exact same call using an IAX route, the call is connected at my end just as soon as the PSTN number answers. I don't have any connection delays for incoming FXO calls. They are connected as soon as I answer the calls. Can anyone give me some pointers on where to start looking? Frank Two possible answers - #1 you have somehow got tonedur set to a value in the 100's (should be around 80) #2 you're calling a cell phone - this takes an extra 3-4 seconds. I have toneduration set as toneduration=80 in chan_dahdi.conf. Is there a tonedur parameter too? And I have this problem with both calls to cell phone and fixed landlines. I only have the delay when calling out over a FXO trunk; no delay issue is seen using the exact same hardware but calling out over an IAX trunk. Any other ideas? Frank -- _ -- 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] CallerID and URL pop up for windows...
I tried a lot of these softwares in the past few days and lots of them are just a pile of .. lots of compatibility issues with various versions of Outlook and Windows or simply don't do either of inbound or outbound. However, I have been testing Ingeniussoftware and their product so far works with Inbound and pulls up Outlook contact. Haven't tried outbound. On Fri, Jan 14, 2011 at 9:19 AM, Gilles codecompl...@free.fr wrote: On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B bruceb...@gmail.com wrote: http://bestof.nerdvittles.com/applications/screenpop/But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay for that if it is added to URANG II Has someone tried IdentaPop? www.identafone.com/cidpop.html -- _ -- 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] Selecing the E1 cards for the call center
Hi All; We would like to build a call center having 2 E1, but we would like to know which card to select: Sangoma or Digium? And card type to be PCI express or PCI 5.0V or PCI 3.3V ? Any advise or special recommendations for the call center? Regards Bilal -- _ -- 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] Ghost ringing
We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Everything is connected to an on-net Asterisk box. We have sniffed SIP traffic and haven't found anything out of the ordinary, in fact, the RTP data doesn't appear to contain the ringing even though it is audible on the phone. Any thoughts? Anyone ever experience any similar issues? Thanks much in advance, Joe -- _ -- 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] Ghost ringing
I had this reported, but it has nothing to do with Asterisk (as far as I could tell). The Polycom firmware (3.3.x) was the problem. I don`t remember if it was 3.3.1 or 3.3.0, but if you`re running one of those try the other. It helped me, I haven`t had the complaint since. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Everything is connected to an on-net Asterisk box. We have sniffed SIP traffic and haven't found anything out of the ordinary, in fact, the RTP data doesn't appear to contain the ringing even though it is audible on the phone. Any thoughts? Anyone ever experience any similar issues? Thanks much in advance, Joe -- _ -- 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] Ghost ringing
Hey Mike, I originally thought the same thing; however, I have swapped their phone with another one here in the office. The one on-site is still experiencing issues; the office isn't. If it were firmware, I'd assume the issue would 'travel' with the phone. Though I can give re-flashing it a shot. Thanks, Joe I had this reported, but it has nothing to do with Asterisk (as far as I could tell). The Polycom firmware (3.3.x) was the problem. I don`t remember if it was 3.3.1 or 3.3.0, but if you`re running one of those try the other. It helped me, I haven`t had the complaint since. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Everything is connected to an on-net Asterisk box. We have sniffed SIP traffic and haven't found anything out of the ordinary, in fact, the RTP data doesn't appear to contain the ringing even though it is audible on the phone. Any thoughts? Anyone ever experience any similar issues? Thanks much in advance, Joe -- _ -- 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] Ghost ringing
It`s possible the firmware problem is caused by higher (or lower?) latencies. I can only report on my own experience, which is peace and quiet since I switch (I think the good version to have with respect to that issue was 3.3.0) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, I originally thought the same thing; however, I have swapped their phone with another one here in the office. The one on-site is still experiencing issues; the office isn't. If it were firmware, I'd assume the issue would 'travel' with the phone. Though I can give re-flashing it a shot. Thanks, Joe I had this reported, but it has nothing to do with Asterisk (as far as I could tell). The Polycom firmware (3.3.x) was the problem. I don`t remember if it was 3.3.1 or 3.3.0, but if you`re running one of those try the other. It helped me, I haven`t had the complaint since. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Everything is connected to an on-net Asterisk box. We have sniffed SIP traffic and haven't found anything out of the ordinary, in fact, the RTP data doesn't appear to contain the ringing even though it is audible on the phone. Any thoughts? Anyone ever experience any similar issues? Thanks much in advance, Joe -- _ -- 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
[asterisk-users] Why are 4 ports used for a single call?
Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are 4 ports used for a single call?
I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecing the E1 cards for the call center
While we're at it, can someone please tell me whether I should be using vi or emacs? ;-) Many thanks, Tom PS: Bilal: You have asked a nearly unanswerable question. Some prefer one, some prefer the other. Both cards are quality items. I can say that I only have experience with Sangoma T1/E1 cards, but our Digium FXO/FXS card works well, too. On 01/14/2011 12:42 PM, bilal ghayyad wrote: Hi All; We would like to build a call center having 2 E1, but we would like to know which card to select: Sangoma or Digium? And card type to be PCI express or PCI 5.0V or PCI 3.3V ? Any advise or special recommendations for the call center? Regards Bilal -- _ -- 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] Why are 4 ports used for a single call?
Each media stream will use two, one for RTP and one for RTCP. In your case 10200/10201 are an RTP/RTCP pair, and 10504/10505 are another pair. RTP is always and even numbered port, and RTCP is always RTP port + 1. Yes, it's in the RFC for RTP. The fact that you have two pairs means that two media streams are being negotiated, perhaps one for audio and one for video? Your phone config or wireshark captures will tell you for certain. Of course, I'm assuming that those ports are for one endpoint (phone). If one pair is for caller and one pair is for callee, then this is a normal simplest scenario, one pair for each side. You didn't specify whose ports they were. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Friday, January 14, 2011 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are 4 ports used for a single call? I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are 4 ports used for a single call?
RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Selecing the E1 cards for the call center
Tom Rymes wrote: While we're at it, can someone please tell me whether I should be using vi or emacs? ;-) Many thanks, Tom That is certainly a religious argument that will NEVER have one right answer. vi is probably found on most systems, but that is certainly no reason to use it! John Novack PS: Bilal: You have asked a nearly unanswerable question. Some prefer one, some prefer the other. Both cards are quality items. I can say that I only have experience with Sangoma T1/E1 cards, but our Digium FXO/FXS card works well, too. On 01/14/2011 12:42 PM, bilal ghayyad wrote: Hi All; We would like to build a call center having 2 E1, but we would like to know which card to select: Sangoma or Digium? And card type to be PCI express or PCI 5.0V or PCI 3.3V ? Any advise or special recommendations for the call center? Regards Bilal -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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 1.4.39 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix bugs in saying numbers using the Swedish language syntax (Closes issue #18355. Reported, patched by oej) * Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). Patched by jpeeler. * Improve handling of REGISTER requests with multiple contact headers. Patched by jpeeler. * app_followme: Don't create a Local channel if the target extension does not exist. (Closes issue #18126. Reported, patched by junky) * Revert code that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, tested by twilson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39 Thank you for your continued support of 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
[asterisk-users] Asterisk 1.6.2.16 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.16. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.16 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (Closes issue #18384. Reported, patched, tested by bjm, tilghman) * app_followme: Don't create a Local channel if the target extension does not exist. (Closes issue #18126. Reported, patched by junky) * Revert code that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, tested by twilson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16 Thank you for your continued support of 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
[asterisk-users] Asterisk 1.8.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * 'sip notify clear-mwi' needs terminating CRLF. (Closes issue #18275. Reported, patched by klaus3000) * Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (Closes issue #18031. Reported by rain. Patched by bbryant) * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. (Closes issue #18342. Reported, patched by nivek.) * Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (Closes issue #18272. Reported by klaus3000. Patched by Marquis42) * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer-cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2 Thank you for your continued support of 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] Why are 4 ports used for a single call?
Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Why are 4 ports used for a single call?
Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports - I think that the 4 port requirement is probably for niceties like conferencing and transfers. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Friday, January 14, 2011 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are 4 ports used for a single call? Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Spectralink 8002
Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- 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 1.8.2 Now Available (not 1.8.3)
Sorry, subject is incorrect. The released version is Asterisk 1.8.2. On 11-01-14 03:12 PM, Asterisk Development Team wrote: The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * 'sip notify clear-mwi' needs terminating CRLF. (Closes issue #18275. Reported, patched by klaus3000) * Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (Closes issue #18031. Reported by rain. Patched by bbryant) * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. (Closes issue #18342. Reported, patched by nivek.) * Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (Closes issue #18272. Reported by klaus3000. Patched by Marquis42) * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer-cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2 Thank you for your continued support of 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] Why are 4 ports used for a single call?
Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that as well? maybe video transmission stuff? Thanks again, On Fri, Jan 14, 2011 at 4:12 PM, Bruce B bruceb...@gmail.com wrote: Got it. Thanks. Makes sense to keep an extra two in mind for conference etc Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas da...@debsinc.com wrote: Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports – I think that the 4 port requirement is probably for “niceties” like conferencing and transfers. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Friday, January 14, 2011 2:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Why are 4 ports used for a single call?
Got it. Thanks. Makes sense to keep an extra two in mind for conference etc Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas da...@debsinc.com wrote: Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports – I think that the 4 port requirement is probably for “niceties” like conferencing and transfers. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Friday, January 14, 2011 2:15 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen solsta...@gmail.com wrote: RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Spectralink 8002
If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- 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] Spectralink 8002
I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- 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] Spectralink 8002
The only time I've seen no net found on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time frame? On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Friday 14 January 2011 15:12:29 Bruce B wrote: Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiquette, the Delete key becomes the first line of defense. -- Tilghman -- _ -- 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] Spectralink 8002
It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller is basically a rebranded Aruba MC-800 (don't know about the APs). I've also tried on my WRT-54G at home, and it does the same thing. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:45:55 PM Subject: Re: [asterisk-users] Spectralink 8002 The only time I've seen no net found on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time frame? On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- 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:
Re: [asterisk-users] Spectralink 8002
Do you see any attempt on the wireless controller from the phone to connect to anything on the network after the TFTP exchange? Any traffic at all on the network from the phone? Have you tried to capture packets with Wireshark or something similar? On Fri, Jan 14, 2011 at 4:15 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller is basically a rebranded Aruba MC-800 (don't know about the APs). I've also tried on my WRT-54G at home, and it does the same thing. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:45:55 PM Subject: Re: [asterisk-users] Spectralink 8002 The only time I've seen no net found on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time frame? On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On 01/14/2011 4:19 PM, Bruce B wrote: Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that as well? maybe video transmission stuff? More likely, it's because only one client behind NAT can use port 5060, so other clients need to use other ports. Could be another reason, though. Tom -- _ -- 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] Why are 4 ports used for a single call?
So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? On Fri, Jan 14, 2011 at 5:13 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 14 January 2011 15:12:29 Bruce B wrote: Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiquette, the Delete key becomes the first line of defense. -- Tilghman -- _ -- 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] Spectralink 8002
The traffic I originally posted was all (except the DHCP request/response) that the phone did since power on. That was sniffed at the output of the wireless controller (all APs tunnel back to the controller). The wireless controller shows the phone as connected, but I haven't gone much further with troubleshooting there... A call to Polycom may be in order, but I don't know what kind of support I get as an end user. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 4:29:17 PM Subject: Re: [asterisk-users] Spectralink 8002 Do you see any attempt on the wireless controller from the phone to connect to anything on the network after the TFTP exchange? Any traffic at all on the network from the phone? Have you tried to capture packets with Wireshark or something similar? On Fri, Jan 14, 2011 at 4:15 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller is basically a rebranded Aruba MC-800 (don't know about the APs). I've also tried on my WRT-54G at home, and it does the same thing. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:45:55 PM Subject: Re: [asterisk-users] Spectralink 8002 The only time I've seen no net found on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time frame? On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download
Re: [asterisk-users] Why are 4 ports used for a single call?
On Fri, 14 Jan 2011 17:29:26 -0500, Tom Rymes try...@rymes.com wrote: On 01/14/2011 4:19 PM, Bruce B wrote: Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that as well? maybe video transmission stuff? More likely, it's because only one client behind NAT can use port 5060, so other clients need to use other ports. Could be another reason, though. FWIW, by default, GrandStream Budget phones use UDP 5004 for RTP. -- _ -- 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] logging
We have queuemetrics, qloaderd and mysql running on our asterisk server in order to streamline call reporting. Now in order to get internal call logs, I need to get thirdlane Master.csv file. I see there is an option in thirdlane to import this into mysql which would make it easier to work with across multiple locations. Is anyone already doing this... is this recommended? Also, any general suggestions on handling all call data both external and internal would be appreciated. Thanks, Nick -- _ -- 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] Ghost ringing
Hey Mike, Is there any chance that you still have this firmware around or can get it? It's unavailable through the Polycom site and through their support. Thanks much in advance, Joe It`s possible the firmware problem is caused by higher (or lower?) latencies. I can only report on my own experience, which is peace and quiet since I switch (I think the good version to have with respect to that issue was 3.3.0) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, I originally thought the same thing; however, I have swapped their phone with another one here in the office. The one on-site is still experiencing issues; the office isn't. If it were firmware, I'd assume the issue would 'travel' with the phone. Though I can give re-flashing it a shot. Thanks, Joe I had this reported, but it has nothing to do with Asterisk (as far as I could tell). The Polycom firmware (3.3.x) was the problem. I don`t remember if it was 3.3.1 or 3.3.0, but if you`re running one of those try the other. It helped me, I haven`t had the complaint since. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as well, although their hardware is newer). It can be fixed if the person using the phone puts the other person on hold and then takes them off hold. They have just been doing this by double pressing hold so the other party doesn't even realize it. Everything is connected to an on-net Asterisk box. We have sniffed SIP traffic and haven't found anything out of the ordinary, in fact, the RTP data doesn't appear to contain the ringing even though it is audible on the phone. Any thoughts? Anyone ever experience any similar issues? Thanks much in advance, Joe -- _ -- 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] Selecing the E1 cards for the call center
On Fri, 14 Jan 2011, Tom Rymes wrote: While we're at it, can someone please tell me whether I should be using vi or emacs? ;-) What 'rymes' with flame bait? -- 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] Why are 4 ports used for a single call?
On Jan 14, 2011, at 5:24 PM, Bruce B wrote: So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? It may be a sin here, but it is certainly impolite many places, and illogical everywhere. This is because we normally read top to bottom, but top-posting forces you to read bottom to top. Tom -- _ -- 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] Why are 4 ports used for a single call?
You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Back to my other questions, now that UDP is clear for me, what ports does SIP require? TCP/UDP 5060 ? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that as well? maybe video transmission stuff? Thanks On Fri, Jan 14, 2011 at 6:32 PM, Tom Rymes try...@rymes.com wrote: On Jan 14, 2011, at 5:24 PM, Bruce B wrote: So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? It may be a sin here, but it is certainly impolite many places, and illogical everywhere. This is because we normally read top to bottom, but top-posting forces you to read bottom to top. Tom -- _ -- 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] Tools to Monitor Asterisk Servers and VMs
Hi Everyone, Are there any generally accepted and widely used tools made and tailored to be used for purpose of monitoring Asterisk servers? I am wondering if there is anything that the Asterisk community mostly uses or are there lots of manual scripts written and nothing really exists that every one kind of uses (e.g. Fail2ban for security). Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are 4 ports used for a single call?
On Jan 14, 2011, at 6:45 PM, Bruce B wrote: You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Suffice it to say, Bruce, this subject has been hashed over thousands, nay, hundreds of thousands of times, and I doubt anything new can be had from doing it again. FYI, It is also considered good etiquette to remove any non-relevant information from the quoted text to keep it short and easy to parse, especially removing the automatically generated footers from the list. As for your question about ports (see, I can stay on topic occasionally!), someone already mentioned something about some equipment using 5004 for RTP, IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP clients behind NAT. There may be other reasons, too. Tom -- _ -- 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] Why are 4 ports used for a single call?
On Fri, Jan 14, 2011 at 6:53 PM, Tom Rymes try...@rymes.com wrote: On Jan 14, 2011, at 6:45 PM, Bruce B wrote: You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Suffice it to say, Bruce, this subject has been hashed over thousands, nay, hundreds of thousands of times, and I doubt anything new can be had from doing it again. FYI, It is also considered good etiquette to remove any non-relevant information from the quoted text to keep it short and easy to parse, especially removing the automatically generated footers from the list. As for your question about ports (see, I can stay on topic occasionally!), someone already mentioned something about some equipment using 5004 for RTP, IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP clients behind NAT. There may be other reasons, too. Tom Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it as well? I am talking strictly in case of Asterisk. -Bruce -- _ -- 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] Top Posting
Bruce et al. I'm posting a new thread with the Top Posting subject so I won't draw complaints about hijacking the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm actively following a thread, the most-recent information appears at the top of the message I receive. If I've missed part of the thread, I need to look only at the most recent message and scroll down a bit to see what's been happening. Bottom Posting requires me to scroll through all of the history before I see the newest addition. While scrolling down, I may see something new and realize that the sender has interleaved responses, addressing multiple points with individual responses. It's been a while, but when I researched Top Posting I found this Wikipedia description: Top-posting is a natural consequence of the behavior of the reply function in many current e-mail readers, such as Microsoft Outlook http://en.wikipedia.org/wiki/Microsoft_Outlook , Gmail http://en.wikipedia.org/wiki/Gmail , and others. By default, these programs insert into the reply message a copy of the original message (without headers and often without any extra indentation or quotation markers), and position the editing cursor http://en.wikipedia.org/wiki/Cursor_%28computers%29 above it. Moreover, a bug present on most flavours of Microsoft Outlook caused the quotation markers to be lost when replying in plain text to a message that was originally sent in HTML/RTF. In addition, users of mobile devices http://en.wikipedia.org/wiki/Handheld_device , like BlackBerries http://en.wikipedia.org/wiki/BlackBerry , are encouraged to use top-posting, because the devices only download the beginning of a message for viewing. The rest of the message is only retrieved when needed, which takes additional download time. Putting the relevant content at the beginning of the message requires less bandwidth, less time, and less scrolling for the Blackberry user.[4] http://en.wikipedia.org/wiki/Posting_style#cite_note-3 [5] http://en.wikipedia.org/wiki/Posting_style#cite_note-4 [6] http://en.wikipedia.org/wiki/Posting_style#cite_note-5 For these and possibly other reasons, many users seem to accept top-posting as the standard reply style. .and an explanation of why people complain about it: Objections to top-posting on newsgroups, as a rule, seem to come from persons who first went online in the earlier days of Usenet http://en.wikipedia.org/wiki/Usenet , and in communities that date to Usenet's early days. Until the mid-90s, top-posting was unknown and interleaved posting an obvious standard that all net.newcomers had to learn. Among the most vehement communities are those in the Usenet http://en.wikipedia.org/wiki/Comp.*_hierarchy comp.lang hierarchy, especially comp.lang.c and comp.lang.c++. Top-posting is more tolerated on the http://en.wikipedia.org/wiki/Alt.*_hierarchy alt hierarchy. Newer online participants, especially those with limited experience of Usenet, tend to be less sensitive to arguments about posting style. When I post (which is rarely, as I have little to offer the list), I top post and explain that it's my preference and I don't know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- _ -- 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] Top Posting
Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al Im posting a new thread with the Top Posting subject so I wont draw complaints about hijacking the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If Im actively following a thread, the most-recent information appears at the top of the message I receive. If Ive missed part of the thread, I need to look only at the most recent message and scroll down a bit to see whats been happening. Bottom Posting requires me to scroll through all of the history before I see the newest addition. While scrolling down, I may see something new and realize that the sender has interleaved responses, addressing multiple points with individual responses. Its been a while, but when I researched Top Posting I found this Wikipedia description: Top-posting is a natural consequence of the behavior of the "reply" function in many current e-mail readers, such as Microsoft Outlook, Gmail, and others. By default, these programs insert into the reply message a copy of the original message (without headers and often without any extra indentation or quotation markers), and position the editing cursor above it. Moreover, a bug present on most flavours of Microsoft Outlook caused the quotation markers to be lost when replying in plain text to a message that was originally sent in HTML/RTF. In addition, users of mobile devices, like BlackBerries, are encouraged to use top-posting, because the devices only download the beginning of a message for viewing. The rest of the message is only retrieved when needed, which takes additional download time. Putting the relevant content at the beginning of the message requires less bandwidth, less time, and less scrolling for the Blackberry user.[4][5][6] For these and possibly other reasons, many users seem to accept top-posting as the "standard" reply style. and an explanation of why people complain about it: Objections to top-posting on newsgroups, as a rule, seem to come from persons who first went online in the earlier days of Usenet, and in communities that date to Usenet's early days. Until the mid-90s, top-posting was unknown and interleaved posting an obvious standard that all net.newcomers had to learn. Among the most vehement communities are those in the Usenet comp.lang hierarchy, especially comp.lang.c and comp.lang.c++. Top-posting is more tolerated on the alt hierarchy. Newer online participants, especially those with limited experience of Usenet, tend to be less sensitive to arguments about posting style. When I post (which is rarely, as I have little to offer the list), I top post and explain that its my preference and I dont know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- _ -- 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
Re: [asterisk-users] AGI-Macro w/Agruments
So, I take it nobody has a solution to this?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, January 07, 2011 12:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] AGI-Macro w/Agruments
OK, I need to dial a macro from AGI and needs to pass an argument.
Ok, I found an bug report, but it was stated un fixable? really after 5
years?
https://issues.asterisk.org/view.php?id=2470
I found this email in the archive, but no solution other then the dodgy work
around?
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg85048.html
I have tried this, but it doesn't work.
$AGI-set_variable('DAILNO', $BranchPhone);
$AGI-exec(Macro,agidial);
And my macro:
[macro-agidial]
exten = s,1,AGI(getcid.pl,${CALLERID(NUM)},1)
exten = s,2,NoOp(DIALNO=${DIALNO})
exten = s,3,Dial(SIP/${DIALNO}@SIPPROVIDER,60)
exten = s,4,GotoIf($[${DIALSTATUS} = CONGESTION]?10)
exten = s,6,Hangup()
exten = s,10,Dial(IAX2/SERVER2/${DIALNO})
exten = s,12,Hangup()
but when the macro is called, Dialno = nothing.
--
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Re: [asterisk-users] Top Posting
Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al… I’m posting a new thread with the “Top Posting” subject so I won’t draw complaints about “hijacking” the 4-port thread. snip When I post (which is rarely, as I have little to offer the list), I top post and explain that it’s my preference and I don’t know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax As mentioned in the past, trimming your post is the best first step on mailing lists. Many of the top post vs bottom post comments happen on the 5+ post on a thread when the size of the email becomes an issue. I have blindly replied in the past and was unable to understand my own email. Take a moment and trim out the messy bits. Use a (snip) or snip to note huge missing areas. As you will note in Don's post there is a history to the argument. Also note Don's multiple signatures which I think he will review after he sees it in action. :) Above all, be polite... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bruce B
You've been officially added to my kill file [1]. The lists are here to get suggestions and assistance with various issues [2]. They are *NOT* your one stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You make it abundantly clear that you're making no effort whatsoever to find answers to the questions you post. And, rather than listen to answers given, or even suggestions about your list etiquette, you instead choose to ignore those suggestions and ask more questions [10]. AND, to make matters worse, this isn't the only list you actively abuse [11][12][13]. Also, since you're unable to seek information on your own, I've taken the liberty of keeping references to all of the above points for you. If I were a mod, I'd drop you from the list. But alas, pushing your useless drivel to /dev/null will have to suffice [14]. I'll just sit here listening to a very relevant song [15] while I get back to the regularly scheduled programming. --Tim [1] http://en.wikipedia.org/wiki/Kill_file [2] http://en.wikipedia.org/wiki/Mailing_list [3] http://lists.digium.com/pipermail/asterisk-users/2011-January/257684.html [4] http://lists.digium.com/pipermail/asterisk-users/2011-January/257685.html [5] http://lists.digium.com/pipermail/asterisk-users/2011-January/257762.html [6] http://lists.digium.com/pipermail/asterisk-users/2011-January/257832.html [7] http://lists.digium.com/pipermail/asterisk-users/2011-January/257888.html [8] http://lists.digium.com/pipermail/asterisk-users/2011-January/257962.html [9] http://lists.digium.com/pipermail/asterisk-users/2011-January/257992.html [10] http://lists.digium.com/pipermail/asterisk-users/2011-January/257991.html [11] http://www.mail-archive.com/support@pfsense.com/msg21300.html [12] http://www.mail-archive.com/support@pfsense.com/msg21307.html [13] http://www.mail-archive.com/support@pfsense.com/msg21119.html [14] http://en.wikipedia.org/wiki//dev/null [15] http://en.wikipedia.org/wiki/Don't_Go_Away_Mad_(Just_Go_Away) http://en.wikipedia.org/wiki/Don%27t_Go_Away_Mad_%28Just_Go_Away%29 -- _ -- 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] Top Posting
Awww...that's no fair. Andrew has bottom-posted to this top-post thread. That really confuses me. Andrew's 'header' appears at the top of the 'stuff,' and his comments at the bottom. Then there's a little sniping that I would have considered really important when I only had 16KB of memory to work with, but it leaves me wondering whose comment was Seconded... We've lost the attribution. What did you mean, Andrew, about Don's multiple signatures which I think he will review? --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Friday, January 14, 2011 7:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al Im posting a new thread with the Top Posting subject so I wont draw complaints about hijacking the 4-port thread. snip When I post (which is rarely, as I have little to offer the list), I top post and explain that its my preference and I dont know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax As mentioned in the past, trimming your post is the best first step on mailing lists. Many of the top post vs bottom post comments happen on the 5+ post on a thread when the size of the email becomes an issue. I have blindly replied in the past and was unable to understand my own email. Take a moment and trim out the messy bits. Use a (snip) or snip to note huge missing areas. As you will note in Don's post there is a history to the argument. Also note Don's multiple signatures which I think he will review after he sees it in action. :) Above all, be polite... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
It was only the people who ONLY asked in a response to go to Google to find answers that annoyed me but slowly posting preference adds up as well. As long as the subject header is not changed all e-mail clients (no matter how stupid they are), now-a-days, create a nice tree. Even so does Hotmail, the worst out there. So I totally don't get what the fuss is. And I was being sarcastic about what top-posting is because I believe it's a stupid personal preference rule that someone made it. I don't care if one is pre-dinosaur age or not. In addition, if all one has to say is Search google or oh my god, my eyes hurt, due to your WHATEVER WAY you post, I see nothing but whining. What is the list for then? If you are smart to answer or have your news up on the air you don't have to bother to answer. I personally never respond to a post if I am going to say use Google or unless I am sure I know what I am talking about. So, in order to be courteous to each other, please move on if you don't want to respond. I can't believe there are people who are setting behind their desks all day waiting to moderate the Asterisk user list while there is no moderation on to be done on this list. I suggest play a game of pacman rather than smart alek responses. At the end I also want to give credit for many smart people out here who without any prejudice do respond and do understand what they are talking about. But there are also the occasional whiners.meh who cares... Thanks for bringing this up. On Fri, Jan 14, 2011 at 8:30 PM, Andrew Latham lath...@gmail.com wrote: Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al… I’m posting a new thread with the “Top Posting” subject so I won’t draw complaints about “hijacking” the 4-port thread. snip When I post (which is rarely, as I have little to offer the list), I top post and explain that it’s my preference and I don’t know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax As mentioned in the past, trimming your post is the best first step on mailing lists. Many of the top post vs bottom post comments happen on the 5+ post on a thread when the size of the email becomes an issue. I have blindly replied in the past and was unable to understand my own email. Take a moment and trim out the messy bits. Use a (snip) or snip to note huge missing areas. As you will note in Don's post there is a history to the argument. Also note Don's multiple signatures which I think he will review after he sees it in action. :) Above all, be polite... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Since I don't want anyone bitch at my spelling again: news up = nose up :-) -Bruce On Fri, Jan 14, 2011 at 8:55 PM, Bruce B bruceb...@gmail.com wrote: It was only the people who ONLY asked in a response to go to Google to find answers that annoyed me but slowly posting preference adds up as well. As long as the subject header is not changed all e-mail clients (no matter how stupid they are), now-a-days, create a nice tree. Even so does Hotmail, the worst out there. So I totally don't get what the fuss is. And I was being sarcastic about what top-posting is because I believe it's a stupid personal preference rule that someone made it. I don't care if one is pre-dinosaur age or not. In addition, if all one has to say is Search google or oh my god, my eyes hurt, due to your WHATEVER WAY you post, I see nothing but whining. What is the list for then? If you are smart to answer or have your news up on the air you don't have to bother to answer. I personally never respond to a post if I am going to say use Google or unless I am sure I know what I am talking about. So, in order to be courteous to each other, please move on if you don't want to respond. I can't believe there are people who are setting behind their desks all day waiting to moderate the Asterisk user list while there is no moderation on to be done on this list. I suggest play a game of pacman rather than smart alek responses. At the end I also want to give credit for many smart people out here who without any prejudice do respond and do understand what they are talking about. But there are also the occasional whiners.meh who cares... Thanks for bringing this up. On Fri, Jan 14, 2011 at 8:30 PM, Andrew Latham lath...@gmail.com wrote: Seconded. Although I've succumbed to bottom posting on occasion when following the convention of the ongoing thread. On 01/14/2011 07:42 PM, Don Kelly wrote: Bruce et al… I’m posting a new thread with the “Top Posting” subject so I won’t draw complaints about “hijacking” the 4-port thread. snip When I post (which is rarely, as I have little to offer the list), I top post and explain that it’s my preference and I don’t know how to do it effectively otherwise. This gives everyone fair warning to delete my posts before reading them. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax As mentioned in the past, trimming your post is the best first step on mailing lists. Many of the top post vs bottom post comments happen on the 5+ post on a thread when the size of the email becomes an issue. I have blindly replied in the past and was unable to understand my own email. Take a moment and trim out the messy bits. Use a (snip) or snip to note huge missing areas. As you will note in Don's post there is a history to the argument. Also note Don's multiple signatures which I think he will review after he sees it in action. :) Above all, be polite... ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bruce B
LOL what a looser. Are you a fat admin behind a desk who is going to loose his job due to recession and is pissed off? Here is your first response to one of my first posts: I was going to respond with some very insightful and helpful information but I'm not a PRI Guru. Sorry, maybe next time. One stupid useless line because you had an issue with the word Guru. P.S. I have no time to go through your collected list but I should say Good Job. I have a position open for you for data entry. You seem to perform data entry and retrieval jobs very well. I am not being sarcastic! -Bruce On Fri, Jan 14, 2011 at 8:31 PM, Tim Nelson tnel...@fudnet.net wrote: You've been officially added to my kill file [1]. The lists are here to get suggestions and assistance with various issues [2]. They are *NOT* your one stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You make it abundantly clear that you're making no effort whatsoever to find answers to the questions you post. And, rather than listen to answers given, or even suggestions about your list etiquette, you instead choose to ignore those suggestions and ask more questions [10]. AND, to make matters worse, this isn't the only list you actively abuse [11][12][13]. Also, since you're unable to seek information on your own, I've taken the liberty of keeping references to all of the above points for you. If I were a mod, I'd drop you from the list. But alas, pushing your useless drivel to /dev/null will have to suffice [14]. I'll just sit here listening to a very relevant song [15] while I get back to the regularly scheduled programming. --Tim [1] http://en.wikipedia.org/wiki/Kill_file [2] http://en.wikipedia.org/wiki/Mailing_list [3] http://lists.digium.com/pipermail/asterisk-users/2011-January/257684.html [4] http://lists.digium.com/pipermail/asterisk-users/2011-January/257685.html [5] http://lists.digium.com/pipermail/asterisk-users/2011-January/257762.html [6] http://lists.digium.com/pipermail/asterisk-users/2011-January/257832.html [7] http://lists.digium.com/pipermail/asterisk-users/2011-January/257888.html [8] http://lists.digium.com/pipermail/asterisk-users/2011-January/257962.html [9] http://lists.digium.com/pipermail/asterisk-users/2011-January/257992.html [10] http://lists.digium.com/pipermail/asterisk-users/2011-January/257991.html [11] http://www.mail-archive.com/support@pfsense.com/msg21300.html [12] http://www.mail-archive.com/support@pfsense.com/msg21307.html [13] http://www.mail-archive.com/support@pfsense.com/msg21119.html [14] http://en.wikipedia.org/wiki//dev/null [15] http://en.wikipedia.org/wiki/Don't_Go_Away_Mad_(Just_Go_Away) http://en.wikipedia.org/wiki/Don%27t_Go_Away_Mad_%28Just_Go_Away%29 -- _ -- 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] Fax stopped working when upgrading to 1.8.2
Magnus, I finally got to testing the patch myself. Apparently it did not work for me. That means that if you are affected by the same issue it is not fixed yet. The current manifestation an incoming OOH_323 call fails in 30 seconds. After reading your initial message more carefully I realized the problem you experience must be different as the ooh_323 issue affects all releases in 1.8 branch and you state that it worked for you when you built from the svn trunk. I would suggest you analyze the "full" log and make sure the fax application does not complain of any problems, then check the "h323_log" and make sure there are no complaints of codecs incompatibility. If you are not utilizing T.38 then only alaw and ulaw will support fax. -Vladimir On 1/14/2011 12:32 AM, magnu...@inputinterior.se wrote: Did apply the patch and did a recompile, no difference, fax still not working. But I did notice one thing, when I was standing at a fax attched to PSTN and trying to send a fax to a fax attached to the Asterisk: The PSTN fax never switched to saying “Sending...” in the display just “Dialing”, but I can “hear” the Asterisk fax i answering. When I went back to Trunk version and did the same, I saw the fax display going from “Dialing” to “Sending” to “Sending OK”. I am sorry to say that I am not smart enough to know what trace I should start looking at, any knows? From: Vladimir Mikhelson Sent: Thursday, January 13, 2011 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: magnu...@inputinterior.se Subject: Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2 Magnus, Can it be the same as I experienced https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by the ticket subject, it reflects the symptoms as they looked originally You can try the patch if applicable and if you know how to compile Addons in 1.8 separately or if you have a capacity to compile the whole thing. -Vladimir On 1/13/2011 6:31 AM, magnu...@inputinterior.se wrote: Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus -- _ -- 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] Bruce B
Now boys, play nice! BTW - the word is LOSER, and ( later on ) LOSE. Words and their spelling do matter John Novack Bruce B wrote: LOL what a looser. Are you a fat admin behind a desk who is going to loose his job due to recession and is pissed off? Here is your first response to one of my first posts: I was going to respond with some very insightful and helpful information but I'm not a PRI Guru. Sorry, maybe next time. One stupid useless line because you had an issue with the word Guru. P.S. I have no time to go through your collected list but I should say Good Job. I have a position open for you for data entry. You seem to perform data entry and retrieval jobs very well. I am not being sarcastic! -Bruce On Fri, Jan 14, 2011 at 8:31 PM, Tim Nelson tnel...@fudnet.net mailto:tnel...@fudnet.net wrote: You've been officially added to my kill file [1]. The lists are here to get suggestions and assistance with various issues [2]. They are *NOT* your one stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You make it abundantly clear that you're making no effort whatsoever to find answers to the questions you post. And, rather than listen to answers given, or even suggestions about your list etiquette, you instead choose to ignore those suggestions and ask more questions [10]. AND, to make matters worse, this isn't the only list you actively abuse [11][12][13]. Also, since you're unable to seek information on your own, I've taken the liberty of keeping references to all of the above points for you. If I were a mod, I'd drop you from the list. But alas, pushing your useless drivel to /dev/null will have to suffice [14]. I'll just sit here listening to a very relevant song [15] while I get back to the regularly scheduled programming. --Tim [1] http://en.wikipedia.org/wiki/Kill_file [2] http://en.wikipedia.org/wiki/Mailing_list [3] http://lists.digium.com/pipermail/asterisk-users/2011-January/257684.html [4] http://lists.digium.com/pipermail/asterisk-users/2011-January/257685.html [5] http://lists.digium.com/pipermail/asterisk-users/2011-January/257762.html [6] http://lists.digium.com/pipermail/asterisk-users/2011-January/257832.html [7] http://lists.digium.com/pipermail/asterisk-users/2011-January/257888.html [8] http://lists.digium.com/pipermail/asterisk-users/2011-January/257962.html [9] http://lists.digium.com/pipermail/asterisk-users/2011-January/257992.html [10] http://lists.digium.com/pipermail/asterisk-users/2011-January/257991.html [11] http://www.mail-archive.com/support@pfsense.com/msg21300.html [12] http://www.mail-archive.com/support@pfsense.com/msg21307.html [13] http://www.mail-archive.com/support@pfsense.com/msg21119.html [14] http://en.wikipedia.org/wiki//dev/null [15] http://en.wikipedia.org/wiki/Don't_Go_Away_Mad_(Just_Go_Away) http://en.wikipedia.org/wiki/Don%27t_Go_Away_Mad_%28Just_Go_Away%29 http://en.wikipedia.org/wiki/Don%27t_Go_Away_Mad_%28Just_Go_Away%29 -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Why are 4 ports used for a single call?
On Jan 14, 2011, at 7:12 PM, Bruce B wrote: Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it as well? I am talking strictly in case of Asterisk. Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC. Tom -- _ -- 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] Top Posting
On Jan 14, 2011, at 8:52 PM, Don Kelly wrote: I have nothing to add to the nascent flame war that I thought we had so narrowly avoided when I sent my last message. However: What did you mean, Andrew, about Don's multiple signatures which I think he will review? --Don [snip] Andrew meant these multiple signatures, and implied that, once you looked at them, you would realize it's a little redundant and not relevant to list users: --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax It's a free country, but given that you prefer a top-posting style where you don't trim previous messages (not judging here, just saying), you might consider omitting your signature for list posts, as it adds an additional eight lines to each message you send, which can really add up. Will it end hunger or bring about world peace? No. Will it be that little bit easier on everyone's eyes? Yes. I think the main lesson from Andrew's post is that top or bottom posting doesn't matter anywhere near as much as trimming posts, so that only that portion of a previous message that you need for context is included, making the entire message compact and nicely legible. Tom -- _ -- 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] Top Posting
I can agree that the entire signature is not relevant to [the] list. but I hope you won't find an example of it adding eight lines to every post. I generally try to include it in only one post in case someone wants to get in touch with me. With regard to the trimming and snipping, I'd prefer to see the entire history in a single message, allowing me to delete all previous posts without losing any information. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Rymes Sent: Friday, January 14, 2011 9:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Jan 14, 2011, at 8:52 PM, Don Kelly wrote: I have nothing to add to the nascent flame war that I thought we had so narrowly avoided when I sent my last message. However: What did you mean, Andrew, about Don's multiple signatures which I think he will review? --Don [snip] Andrew meant these multiple signatures, and implied that, once you looked at them, you would realize it's a little redundant and not relevant to list users: --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax It's a free country, but given that you prefer a top-posting style where you don't trim previous messages (not judging here, just saying), you might consider omitting your signature for list posts, as it adds an additional eight lines to each message you send, which can really add up. Will it end hunger or bring about world peace? No. Will it be that little bit easier on everyone's eyes? Yes. I think the main lesson from Andrew's post is that top or bottom posting doesn't matter anywhere near as much as trimming posts, so that only that portion of a previous message that you need for context is included, making the entire message compact and nicely legible. Tom -- _ -- 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] Top Posting
Whatever your preferred style, the following post is at least worth considering. http://brooksreview.net/2011/01/interleaved-email/ My belief is that it would be nearly impossible for me to follow a high volume list if top posting was the preferred style. For example, the following email from the LKML would need to be more verbose if all the participants were top posting, because they would all have to set the context for their comments. Instead, you can follow the chain of thought for each of the threads contained in the email. http://article.gmane.org/gmane.linux.kernel/1087665 Anyway, just something to consider, Shaun On 1/14/11 9:44 PM, Don Kelly wrote: I can agree that the entire signature is not relevant to [the] list. but I hope you won't find an example of it adding eight lines to every post. I generally try to include it in only one post in case someone wants to get in touch with me. With regard to the trimming and snipping, I'd prefer to see the entire history in a single message, allowing me to delete all previous posts without losing any information. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Rymes Sent: Friday, January 14, 2011 9:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On Jan 14, 2011, at 8:52 PM, Don Kelly wrote: I have nothing to add to the nascent flame war that I thought we had so narrowly avoided when I sent my last message. However: What did you mean, Andrew, about Don's multiple signatures which I think he will review? --Don [snip] Andrew meant these multiple signatures, and implied that, once you looked at them, you would realize it's a little redundant and not relevant to list users: --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax It's a free country, but given that you prefer a top-posting style where you don't trim previous messages (not judging here, just saying), you might consider omitting your signature for list posts, as it adds an additional eight lines to each message you send, which can really add up. Will it end hunger or bring about world peace? No. Will it be that little bit easier on everyone's eyes? Yes. I think the main lesson from Andrew's post is that top or bottom posting doesn't matter anywhere near as much as trimming posts, so that only that portion of a previous message that you need for context is included, making the entire message compact and nicely legible. Tom -- _ -- 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] Top Posting
On 11-01-14 07:42 PM, Don Kelly wrote: Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm actively following a thread, the most-recent information appears at the top of the message I receive. If I've missed part of the thread, I need to look only at the most recent message and scroll down a bit to see what's been happening. It is not a matter of preference, it is actually a rule [1]. Top-posting is also an annoying practice [2] and NOT the general accepted way to reply. [1] http://www.asterisk.org/community/rules [2] http://linux.sgms-centre.com/misc/netiquette.php#toppost -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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 stops responding
I am having a problem with an Asterisk 1.6.2.15 server that runs a small call center with Queuemetrics. In the past month we've had this problem 3 times. The problem is that Asterisk simply stops responding. No calls in or out and you cannot even get to the CLI. The process seems to be running but there is simple no activity. All I see in the log files is: [Jan 14 16:30:46] WARNING[20747] pbx.c: Failed to create new channel thread [Jan 14 16:30:46] WARNING[20747] chan_sip.c: Failed to start PBX :( [Jan 14 16:30:47] WARNING[20745] pbx.c: Failed to create new channel thread [Jan 14 16:30:47] WARNING[20745] chan_dahdi.c: Unable to start PBX on DAHDI/29-1 After restarting Asterisk everything is back to normal. The time between the first failure and the second was almost a month, between the second and third a few days. We use CentOS 5.5 on a Dell 905 server with 8gb of ram. The first time it happened we were using Asterisk 1.6.2.14 so we upgraded to 1.6.2.15 to try and avoid the problem but no luck. Any recommendations? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- 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
