Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive
Looks like version 11.3 did not fix my issue. http://pastebin.com/gd291Bqz On Thu, Apr 4, 2013 at 1:23 PM, Duane Larson duane.lar...@gmail.com wrote: Thanks Jim. Searched through the change log for deadlock but nothing really stuck out. I'll upgrade to 11.3 and see if that makes a difference. On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas li...@cmsws.com wrote: On 04/03/2013 08:15 PM, Duane Larson wrote: So it just happened again on both machines at the same time and I was running debug on both servers. I am running OpenSIPS and load balancing between both servers so I am guessing when the invite was sent to the first server it was frozen for some reason and then OpenSIPS sent the invite to the second server and that server was also frozen/deadlocked because of the SIP message. I noticed on both servers the last log that was posted with Asterisk deadlocked was the following Asterisk version 11.0.1 [Apr 3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11805 instead Asterisk version 11.2.1 [Apr 3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12423 instead In my last email I posted the debug from the Asterisk server with 11.0.1 version of code. Here is a post of the debug for the Asterisk server with version 11.2.1 http://pastebin.com/mbjSSAWM This has to be a bug right? I am thinking of opening an issue on the Asterisk JIRA system A number of deadlocks were fixed in the current release of 11.3. Please read the change log to see if any fit your issue. http://downloads.asterisk.org/**pub/telephony/asterisk/** ChangeLog-11-currenthttp://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11-current On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote: It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson duane.lar...@gmail.com wrote: I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_**sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_**id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- __**__** _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 SIP deadlocks - update_provisional_keepalive
Thanks Jim. Searched through the change log for deadlock but nothing really stuck out. I'll upgrade to 11.3 and see if that makes a difference. On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas li...@cmsws.com wrote: On 04/03/2013 08:15 PM, Duane Larson wrote: So it just happened again on both machines at the same time and I was running debug on both servers. I am running OpenSIPS and load balancing between both servers so I am guessing when the invite was sent to the first server it was frozen for some reason and then OpenSIPS sent the invite to the second server and that server was also frozen/deadlocked because of the SIP message. I noticed on both servers the last log that was posted with Asterisk deadlocked was the following Asterisk version 11.0.1 [Apr 3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11805 instead Asterisk version 11.2.1 [Apr 3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12423 instead In my last email I posted the debug from the Asterisk server with 11.0.1 version of code. Here is a post of the debug for the Asterisk server with version 11.2.1 http://pastebin.com/mbjSSAWM This has to be a bug right? I am thinking of opening an issue on the Asterisk JIRA system A number of deadlocks were fixed in the current release of 11.3. Please read the change log to see if any fit your issue. http://downloads.asterisk.org/**pub/telephony/asterisk/** ChangeLog-11-currenthttp://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11-current On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote: It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson duane.lar...@gmail.com wrote: I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_**sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_**id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 SIP deadlocks - update_provisional_keepalive
It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson duane.lar...@gmail.com wrote: I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 SIP deadlocks - update_provisional_keepalive
So it just happened again on both machines at the same time and I was running debug on both servers. I am running OpenSIPS and load balancing between both servers so I am guessing when the invite was sent to the first server it was frozen for some reason and then OpenSIPS sent the invite to the second server and that server was also frozen/deadlocked because of the SIP message. I noticed on both servers the last log that was posted with Asterisk deadlocked was the following Asterisk version 11.0.1 [Apr 3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 11805 instead Asterisk version 11.2.1 [Apr 3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 12423 instead In my last email I posted the debug from the Asterisk server with 11.0.1 version of code. Here is a post of the debug for the Asterisk server with version 11.2.1 http://pastebin.com/mbjSSAWM This has to be a bug right? I am thinking of opening an issue on the Asterisk JIRA system On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson duane.lar...@gmail.com wrote: It just happened again on the 11.0.1 box and I was able to grab a debug. I am hoping someone can tell me if this is a bug or something wrong with my config. gdb asterisk-bin/sbin/asterisk 29048 Go here for the debug output http://pastebin.com/DGXx0BSk On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson duane.lar...@gmail.comwrote: I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 SIP deadlocks - update_provisional_keepalive
I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command netstat -nap |grep 5060 and see that Asterisk has a lot under the Recv-Q column. It usually takes about 10 minutes before Asterisk becomes responsive again or else before 10 minutes is up I could restart Asterisk and everything will be back to normal. I see in the message logs the following errors On the 11.0.1 Asterisk server WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID 11473. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4406). On the 11.2.1 Asterisk server WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID 30810. This is probably a bug (chan_sip.c: update_provisional_keepalive, line 4683). When I look in chan_sip.c on both servers I see that they are the same line of code AST_SCHED_DEL_UNREF(sched, pvt-provisional_keepalive_sched_id, dialog_unref(pvt, when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr)); What could be causing this because it seems to happen at least once a day. -- _ -- 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] Queue not sending call to Agent
After a Good Call from a PSTN phone if I do a sip prune realtime peer 9013XX9XX8 (9013XX9XX8 being the phone number of the Agent/Member) then I can call the number again and not get the issue. So this has something to do with the stuff that is put in my peer table after a call. Any ideas? On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson duane.lar...@gmail.comwrote: One more piece to add. I had mentioned before that I could get a call from a PSTN user to work the first time. So here is all the output of a Good call from a PSTN user after I have performed a RELOAD on asterisks CLI http://pastebin.com/9RSvQsmN And when the caller or agent hangs this call up all calls from the PSTN afterward get put in the queue automatically and the agent never gets called. On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson duane.lar...@gmail.comwrote: Ok. Something isn't right. With a user that is local to my SIP user database calls the queue phone number everything works without issue. It is when a remote user (like someone from the PSTN) calls the queue phone number that the caller gets put into the queue and the agent/member doesn't receive the call. I have captured debugs from OpenSIPS and Asterisk and I can't really see any difference. I also executed the commands you told me where I could. Here are the debugs Good call from local SIP user to Queue LocalUser - OpenSIPSProxy - Asterisk (then asterisk calls the agent/member) - OpenSIPSProxy - Agent http://pastebin.com/Fa9y3CXQ Bad call from PSTN Caller to Queue PSTN Gatway - OpenSIPSB2BUA - OpenSIPSProxy - Asterisk (then asterisk doesn't call Agent/Member for some reason) http://pastebin.com/VBA9nGAs Thanks for looking at this. Currently this happens every time. Any call from a local user gets put in queue and agent is called right away, but any call from PSTN user gets put in queue and agent isn't called but the agent shows as Asterisk18*CLI queue show irock.com has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime, 131s talktime), W:0, C:12, A:15, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 1991 secs ago) Callers: 1. SIP/9013XX9XX8-002d (wait: 0:02, prio: 0) When it is a good call and I do queue show I see this Asterisk18*CLI queue show irock.com has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime, 131s talktime), W:0, C:12, A:16, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 2079 secs ago) No Callers *How come with the Bad Call the Agent/Member shows up in a queue show as being a Member and a Caller???* On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot satish4aster...@gmail.com wrote: I am not sure but seems like Agent channel not being released from Asterisk. Next time when this happens, try 'core show channels' to check whether Agent channel is released or not. [SATISH] On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson duane.lar...@gmail.comwrote: Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a column of 'qualify' in my database for the sip users. For my config I am using OpenSIPS as the register and proxy. Asterisk is only used for voicemail and ACD/Hunt groups. On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot satish4aster...@gmail.com wrote: Provide the entry for Agent SIP/9013XX9XX8 along with parameters 'callcounter' and 'qualify' from sip.conf. Also provide CLI outputs of 'core show channels',sip show peers' and 'queue show' when... (1)First caller enters the Queue (2)First caller gets connected with Agent (3)First caller gets disconnected from Agent (4)Second caller enters the Queue You may have sequences changed for step no 3 and 4 in your scenario. [SATISH] -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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
Re: [asterisk-users] Queue not sending call to Agent
Ok. Something isn't right. With a user that is local to my SIP user database calls the queue phone number everything works without issue. It is when a remote user (like someone from the PSTN) calls the queue phone number that the caller gets put into the queue and the agent/member doesn't receive the call. I have captured debugs from OpenSIPS and Asterisk and I can't really see any difference. I also executed the commands you told me where I could. Here are the debugs Good call from local SIP user to Queue LocalUser - OpenSIPSProxy - Asterisk (then asterisk calls the agent/member) - OpenSIPSProxy - Agent http://pastebin.com/Fa9y3CXQ Bad call from PSTN Caller to Queue PSTN Gatway - OpenSIPSB2BUA - OpenSIPSProxy - Asterisk (then asterisk doesn't call Agent/Member for some reason) http://pastebin.com/VBA9nGAs Thanks for looking at this. Currently this happens every time. Any call from a local user gets put in queue and agent is called right away, but any call from PSTN user gets put in queue and agent isn't called but the agent shows as Asterisk18*CLI queue show irock.com has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime, 131s talktime), W:0, C:12, A:15, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 1991 secs ago) Callers: 1. SIP/9013XX9XX8-002d (wait: 0:02, prio: 0) When it is a good call and I do queue show I see this Asterisk18*CLI queue show irock.com has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime, 131s talktime), W:0, C:12, A:16, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 2079 secs ago) No Callers *How come with the Bad Call the Agent/Member shows up in a queue show as being a Member and a Caller???* On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot satish4aster...@gmail.comwrote: I am not sure but seems like Agent channel not being released from Asterisk. Next time when this happens, try 'core show channels' to check whether Agent channel is released or not. [SATISH] On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson duane.lar...@gmail.comwrote: Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a column of 'qualify' in my database for the sip users. For my config I am using OpenSIPS as the register and proxy. Asterisk is only used for voicemail and ACD/Hunt groups. On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot satish4aster...@gmail.com wrote: Provide the entry for Agent SIP/9013XX9XX8 along with parameters 'callcounter' and 'qualify' from sip.conf. Also provide CLI outputs of 'core show channels',sip show peers' and 'queue show' when... (1)First caller enters the Queue (2)First caller gets connected with Agent (3)First caller gets disconnected from Agent (4)Second caller enters the Queue You may have sequences changed for step no 3 and 4 in your scenario. [SATISH] -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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] Queue not sending call to Agent
One more piece to add. I had mentioned before that I could get a call from a PSTN user to work the first time. So here is all the output of a Good call from a PSTN user after I have performed a RELOAD on asterisks CLI http://pastebin.com/9RSvQsmN And when the caller or agent hangs this call up all calls from the PSTN afterward get put in the queue automatically and the agent never gets called. On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson duane.lar...@gmail.comwrote: Ok. Something isn't right. With a user that is local to my SIP user database calls the queue phone number everything works without issue. It is when a remote user (like someone from the PSTN) calls the queue phone number that the caller gets put into the queue and the agent/member doesn't receive the call. I have captured debugs from OpenSIPS and Asterisk and I can't really see any difference. I also executed the commands you told me where I could. Here are the debugs Good call from local SIP user to Queue LocalUser - OpenSIPSProxy - Asterisk (then asterisk calls the agent/member) - OpenSIPSProxy - Agent http://pastebin.com/Fa9y3CXQ Bad call from PSTN Caller to Queue PSTN Gatway - OpenSIPSB2BUA - OpenSIPSProxy - Asterisk (then asterisk doesn't call Agent/Member for some reason) http://pastebin.com/VBA9nGAs Thanks for looking at this. Currently this happens every time. Any call from a local user gets put in queue and agent is called right away, but any call from PSTN user gets put in queue and agent isn't called but the agent shows as Asterisk18*CLI queue show irock.com has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime, 131s talktime), W:0, C:12, A:15, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 1991 secs ago) Callers: 1. SIP/9013XX9XX8-002d (wait: 0:02, prio: 0) When it is a good call and I do queue show I see this Asterisk18*CLI queue show irock.com has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime, 131s talktime), W:0, C:12, A:16, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 2079 secs ago) No Callers *How come with the Bad Call the Agent/Member shows up in a queue show as being a Member and a Caller???* On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot satish4aster...@gmail.com wrote: I am not sure but seems like Agent channel not being released from Asterisk. Next time when this happens, try 'core show channels' to check whether Agent channel is released or not. [SATISH] On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson duane.lar...@gmail.comwrote: Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a column of 'qualify' in my database for the sip users. For my config I am using OpenSIPS as the register and proxy. Asterisk is only used for voicemail and ACD/Hunt groups. On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot satish4aster...@gmail.com wrote: Provide the entry for Agent SIP/9013XX9XX8 along with parameters 'callcounter' and 'qualify' from sip.conf. Also provide CLI outputs of 'core show channels',sip show peers' and 'queue show' when... (1)First caller enters the Queue (2)First caller gets connected with Agent (3)First caller gets disconnected from Agent (4)Second caller enters the Queue You may have sequences changed for step no 3 and 4 in your scenario. [SATISH] -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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] Queue not sending call to Agent
Yesterday I rebooted the server and it seems to be working again. Not sure
what the reboot might have changed. Hopefully it doesn't happen again but I
can't be sure. To answer your question I have the sip.conf in my mysql
database and in MySQL I have callcounter set to yes. I don't have a column
of 'qualify' in my database for the sip users. For my config I am using
OpenSIPS as the register and proxy. Asterisk is only used for voicemail and
ACD/Hunt groups.
On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot satish4aster...@gmail.comwrote:
Provide the entry for Agent SIP/9013XX9XX8 along with parameters
'callcounter' and 'qualify' from sip.conf.
Also provide CLI outputs of 'core show channels',sip show peers' and 'queue
show' when...
(1)First caller enters the Queue
(2)First caller gets connected with Agent
(3)First caller gets disconnected from Agent
(4)Second caller enters the Queue
You may have sequences changed for step no 3 and 4 in your scenario.
[SATISH]
On Sat, Jun 11, 2011 at 2:56 AM, duane.lar...@gmail.com wrote:
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the agent dials a number to log
into the queue. When someone calls the queue the first time the call is sent
to the agent without issue. The issue is that any calls after the first are
placed in the queue and never sent to the agent who is logged in and
available. Before I call the queue I do a show queue and it shows the
agent as
Asterisk18*CLI queue show
irock.com has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/9013XX9XX8 (dynamic) (Not in use) has taken no calls yet
No Callers
Then the call comes into the queue and the callee just sits in the queue.
When I do a show queue again when the callee is in the queue it shows the
agent as busy
Asterisk18*CLI queue show
irock.com has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/9013XX9XX8 (dynamic) (Busy) has taken no calls yet
Callers:
1. SIP/9013XX9XX8-0001 (wait: 0:12, prio: 0)
So I am not sure what happened because the agent was free before the call.
If I do a reload at the Asterisk CLI and then call again the agent gets the
call and then the second call is once again placed in the queue. I will
attach a SIP Debug that shows what is going on. I don't see any SIP invites
leaving Asterisk to invite the agent to the call.
One other thing Currently in my config I have the agent show up as
just the username which is the phone number. If I set it so that the agent
shows up as phonenumber@blah then I can call the agent constantly without
any issue. The only problem here is that when I do a queue show the agent
shows up as unknown status. So when the agent is on a call and someone
else calls the agent will be interrupted.
This is what I have in queues.conf
[irock.com]
strategy=ringall
ringinuse=no
joinempty=yes
leavewhenempty=no
announce-frequency=30
min-announce-frequency=15
periodic-announce-frequency=60
announce-holdtime=yes
announce-position=yes
; (You are now first in line.)
queue-youarenext = queue-youarenext
; (There are)
queue-thereare = queue-thereare
; (calls waiting.)
queue-callswaiting = queue-callswaiting
; (The current est. holdtime is)
queue-holdtime = queue-holdtime
; (minutes.)
queue-minutes = queue-minutes
; (seconds.)
queue-seconds = queue-seconds
; (Thank you for your patience.)
queue-thankyou = queue-thankyou
; (Hold time)
queue-reporthold = queue-reporthold
; (All reps busy / wait for next)
periodic-announce = queue-periodic-announce
This is what I have in extensions.conf
exten = 9012XX1XX1,1,Answer()
exten = 9012XX1XX1,n,Set(QUEUE_MAX_PENALTY=0);
exten = 9012XX1XX1,n,Queue(irock.com,t)
exten = 9012XX1XX1,n,Hangup()
exten = *50,1,Answer
exten = *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4})
exten = *50,n,Hangup
exten = *51,1,Answer
exten = *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4})
exten = *51,n,Hangup
[macro-queue-login]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone});
exten = s,n,AddQueueMember(${queue});
exten = s,n,Playback(agent-loginok)
[macro-queue-logout]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone});
exten = s,n,RemoveQueueMember(${queue});
exten = s,n,Playback(agent-loggedoff)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
[asterisk-users] Queue not sending call to Agent
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the agent dials a number to log
into the queue. When someone calls the queue the first time the call is
sent to the agent without issue. The issue is that any calls after the
first are placed in the queue and never sent to the agent who is logged in
and available. Before I call the queue I do a show queue and it shows the
agent as
Asterisk18*CLI queue show
irock.com has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/9013XX9XX8 (dynamic) (Not in use) has taken no calls yet
No Callers
Then the call comes into the queue and the callee just sits in the queue.
When I do a show queue again when the callee is in the queue it shows the
agent as busy
Asterisk18*CLI queue show
irock.com has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime,
0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/9013XX9XX8 (dynamic) (Busy) has taken no calls yet
Callers:
1. SIP/9013XX9XX8-0001 (wait: 0:12, prio: 0)
So I am not sure what happened because the agent was free before the call.
If I do a reload at the Asterisk CLI and then call again the agent gets the
call and then the second call is once again placed in the queue. I will
attach a SIP Debug that shows what is going on. I don't see any SIP invites
leaving Asterisk to invite the agent to the call.
One other thing Currently in my config I have the agent show up as just
the username which is the phone number. If I set it so that the agent shows
up as phonenumber@blah then I can call the agent constantly without any
issue. The only problem here is that when I do a queue show the agent
shows up as unknown status. So when the agent is on a call and someone
else calls the agent will be interrupted.
This is what I have in queues.conf
[irock.com]
strategy=ringall
ringinuse=no
joinempty=yes
leavewhenempty=no
announce-frequency=30
min-announce-frequency=15
periodic-announce-frequency=60
announce-holdtime=yes
announce-position=yes
; (You are now first in line.)
queue-youarenext = queue-youarenext
; (There are)
queue-thereare = queue-thereare
; (calls waiting.)
queue-callswaiting = queue-callswaiting
; (The current est. holdtime is)
queue-holdtime = queue-holdtime
; (minutes.)
queue-minutes = queue-minutes
; (seconds.)
queue-seconds = queue-seconds
; (Thank you for your patience.)
queue-thankyou = queue-thankyou
; (Hold time)
queue-reporthold = queue-reporthold
; (All reps busy / wait for next)
periodic-announce = queue-periodic-announce
This is what I have in extensions.conf
exten = 9012XX1XX1,1,Answer()
exten = 9012XX1XX1,n,Set(QUEUE_MAX_PENALTY=0);
exten = 9012XX1XX1,n,Queue(irock.com,t)
exten = 9012XX1XX1,n,Hangup()
exten = *50,1,Answer
exten = *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4})
exten = *50,n,Hangup
exten = *51,1,Answer
exten = *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4})
exten = *51,n,Hangup
[macro-queue-login]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone});
exten = s,n,AddQueueMember(${queue});
exten = s,n,Playback(agent-loginok)
[macro-queue-logout]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone});
exten = s,n,RemoveQueueMember(${queue});
exten = s,n,Playback(agent-loggedoff)
--
_
-- 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] Forward voicemail not working
Thanks Chad. I will try the patch. On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Sun, 02 Jan 2011 17:44:19 + duane.lar...@gmail.com wrote: I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I had the same issue. It was a regression caused by a fix for ODBC storage, and it seems to have affected every recent release of Asterisk. There's a patch here: https://issues.asterisk.org/view.php?id=18358 Looks like the fix will be incorporated into 1.8.3. You'll have to use the patch until then. -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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] Forward voicemail not working
Patch worked like a charm. Thanks Chad. Thought I had done something wrong when installing. Really appreciate it. On Mon, Jan 10, 2011 at 2:27 PM, Duane Larson duane.lar...@gmail.comwrote: Thanks Chad. I will try the patch. On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace cwall...@lodgingcompany.com wrote: On Sun, 02 Jan 2011 17:44:19 + duane.lar...@gmail.com wrote: I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I had the same issue. It was a regression caused by a fix for ODBC storage, and it seems to have affected every recent release of Asterisk. There's a patch here: https://issues.asterisk.org/view.php?id=18358 Looks like the fix will be incorporated into 1.8.3. You'll have to use the patch until then. -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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] Forward voicemail not working
I still can't figure out why this isn't working. I updated to the latest version of Asterisk 1.8.1 with no luck. I am using Realtime for sipusers and vmusers if that makes any difference. I tested this on a new install and saw the following under the folder where I installed Asterisk I had /home/asterisk/asterisk-bin/spool/asterisk/voicemail so no directories had been created yet. I left a voicemail for 9xx2xx2...@irock.com. That created the directory irock.com under the voicemail folder and also created the directories for user 9XX2XX2009 (INBOX and all the other stuff). Then I called into 9XX2XX2009's mailbox and forwarded the voicemail to 9XX2XX2008. So under the voicemail directory there was no folder for 9XX2XX2008, but since I forwarded the voicemail to 9XX2XX2008 asterisk created the folder for that user and also created the subfolder INBOX, but it didn't copy the voicemail to that directory. If I can't get the forward voicemail option to work with the VoicemailMain() function is there any way to disable this option so that users don't try to use it? On Sun, Jan 2, 2011 at 11:44 AM, duane.lar...@gmail.com wrote: I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the message to the other user from a debug perspective I don't see any errors. The only message I see is == Saving '/home/asterisk/asterisk-bin/spool/asterisk/voicemail/ irock.com/9XX2XX2009/Old/msg.txt': [Jan 2 11:24:18] NOTICE[17036]: app_voicemail.c:5154 copy_message: Copying message from 9xx2xx2...@irock.com to 2...@irock.com Yet when I look in 2011 spool directory I don't see any message at all. It is just not being copied. What could be the issue? -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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] Forward voicemail not working
I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the message to the other user from a debug perspective I don't see any errors. The only message I see is == Saving '/home/asterisk/asterisk-bin/spool/asterisk/voicemail/irock.com/9XX2XX2009/Old/msg.txt': [Jan 2 11:24:18] NOTICE[17036]: app_voicemail.c:5154 copy_message: Copying message from 9xx2xx2...@irock.com to 2...@irock.com Yet when I look in 2011 spool directory I don't see any message at all. It is just not being copied. What could be the issue? -- _ -- 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 Realtime Queue not working
I have configured my mysql database by following this link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
The only difference is that I am using ODBC instead of MySQL with Realtime.
Within extensions.conf I have the following for my queue
exten = 9**2**1611,1,Answer
exten = 9**2**1611,2,Queue(irock.com,tT,,,300)
exten = *50,1,Answer
exten = *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4})
exten = *50,n,Hangup
exten = *51,1,Answer
exten = *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4})
exten = *51,n,Hangup
[macro-queue-login]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone});
exten = s,n,AddQueueMember(${queue});
exten = s,n,Playback(agent-loginok)
[macro-queue-logout]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone});
exten = s,n,RemoveQueueMember(${queue});
exten = s,n,Playback(agent-loggedoff)
When I call 9**2**1611 I get the following error when debugging
-- Goto (irock.com,9012211611,1)
-- Executing [9012211...@irock.com:1] Answer(SIP/9012732004-0001, )
in new stack
-- Executing [9012211...@irock.com:2]
Queue(SIP/9012732004-0001, irock.com,tT,,,300) in new stack
[Dec 26 16:39:57] WARNING[5264]: app_queue.c:5732 queue_exec: Unable to
join queue 'irock.com'
-- Executing [9012211...@irock.com:3]
GotoIf(SIP/9012732004-0001, irock.com !=
irock.com?irock.com,9012211611,1) in new stack
If I do a show queue I don't see the irock.com queue
Here is what I have in my MySQL database
++-+--+---+-+--++--++++---+---+++--+++---+---+++--+-+---++---+-++-+++---+-+
| name | musiconhold | announce | context | timeout | monitor_join |
monitor_format | queue_youarenext | queue_thereare | queue_callswaiting |
queue_holdtime | queue_minutes | queue_seconds | queue_lessthan |
queue_thankyou | queue_reporthold | announce_frequency |
announce_round_seconds | announce_holdtime | retry | wrapuptime | maxlen |
servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus |
eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart |
ringinuse | setinterfacevar |
++-+--+---+-+--++--++++---+---+++--+++---+---+++--+-+---++---+-++-+++---+-+
| 9**2**1611 | NULL | NULL | irock.com | NULL | NULL | NULL |
queue-youarenext | queue-thereare | queue-callswaiting | queue-holdtime |
queue-minutes | queue-seconds | NULL | queue-thankyou | queue-reporthold |
30 | NULL | yes | NULL | NULL | NULL | NULL | leastrecent | yes | no | NULL
| NULL | NULL | NULL | NULL | NULL | 0 | NULL |
++-+--+---+-+--++--++++---+---+++--+++---+---+++--+-+---++---+-++-+++---+-+
My other realtime stuff like sipusers and vmusers works just fine.
extconfig.conf
; Primary Database Connection
sipusers = odbc,proxy01,sipusers,1
sippeers = odbc,proxy01,sipusers,1
voicemail = odbc,proxy01,vmusers,1
meetme = odbc,proxy01,meetme,1
queues = odbc,proxy01,queue_table,1
queue_members = odbc,proxy01,queue_member_table,1
extensions = odbc,proxy01,extensions,1
--
_
-- 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] Wireless Desktop VoIP Phone?
Snom Sent from Droid On Dec 17, 2010 12:36 PM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with MySQL Cluster
Awesome. Didn't notice that, but that is my fault for not reading the changelog or the updated sample configs. I will try this out. Thanks all for the comments. On Wed, Dec 1, 2010 at 10:09 AM, Tilghman Lesher tles...@digium.com wrote: On Tuesday 30 November 2010 18:34:17 Duane Larson wrote: I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability. I was wondering if it's possible for Asterisk to also use multiple database servers for Realtime? Currently with Realtime I am only able to point to a single IP address for a database. If that database server goes down that Asterisk is pointed to then Asterisk won't be able to do anything. Any options within Asterisk 1.8 to make it more fault tolerant when it comes to Realtime and databases? Yes, if you refer to configs/extconfig.conf.sample, within the Asterisk 1.8 tree, you'll see that realtime supports multiple lines per realtime family, scored by consecutive priorities. 1 is the default, but you can have as many as you'd like. Additionally, for res_config_odbc, there is a setting in res_odbc.conf called negative_connection_cache, which is the length of time that Asterisk remembers that a connection is down before it will once again attempt to connect. The intention, of course, is that once the primary comes back up, you'll want the Asterisk server to revert back to using it. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 with MySQL Cluster
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability. I was wondering if it's possible for Asterisk to also use multiple database servers for Realtime? Currently with Realtime I am only able to point to a single IP address for a database. If that database server goes down that Asterisk is pointed to then Asterisk won't be able to do anything. Any options within Asterisk 1.8 to make it more fault tolerant when it comes to Realtime and databases? -- _ -- 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 with MySQL Cluster
For me OpenSIPS will do most of the work. Asterisk will only handle Hunt Groups/Queues, IVRs, and Voicemail when OpenSIPS forwards that traffic to Asterisk. And since I already have MySQL Cluster working in a redundant fashion I am not sure I want to try out MMM MySQL. I do like the idea of using DNS with hosts file and monitoring the MySQL service on the remote machine and if the service goes down then rewrite the IP in the hosts file. I will have to test that out. If anyone else has any experience I would love to add that to this thread. On Tue, Nov 30, 2010 at 7:04 PM, Singer X.J. Wang w...@pythian.com wrote: Most of the solutions there are too complex and not really suited for Asterisk's low usage. I would seriously consider using MySQL MMM and two MySQL servers in a master-master role. Have the asterisk server also serve as the MMM-Manager and its not that hard. You have automated failovers in MySQL in the 1-2 second range. Singer On Tue, Nov 30, 2010 at 19:51, David Backeberg dbackeb...@gmail.comwrote: On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson duane.lar...@gmail.com wrote: I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability. I was wondering if it's possible for Asterisk to also use multiple database servers for Realtime? Currently with Realtime I am only able to point to a single IP address for a database. If that database server goes down that Asterisk is pointed to then Asterisk won't be able to do anything. Any options within Asterisk 1.8 to make it more fault tolerant when it comes to Realtime and databases? http://dev.mysql.com/doc/refman/5.0/en/ha-overview.html It's a fair amount of work for what in my opinion is a minimal reward. If you've hardened everything else and this is the only single point of failure left in your entire infrastructure, you should be able to sleep well at night. -- _ -- 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 -- The best compliment you could give Pythian for our service is a referral. -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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 with MySQL Cluster
Thats sounds interesting too. I will look into that also. On Wed, Dec 1, 2010 at 1:30 AM, Stefan Schmidt s...@sil.at wrote: Am 01.12.10 05:10, schrieb Duane Larson: For me OpenSIPS will do most of the work. Asterisk will only handle Hunt Groups/Queues, IVRs, and Voicemail when OpenSIPS forwards that traffic to Asterisk. And since I already have MySQL Cluster working in a redundant fashion I am not sure I want to try out MMM MySQL. I do like the idea of using DNS with hosts file and monitoring the MySQL service on the remote machine and if the service goes down then rewrite the IP in the hosts file. I will have to test that out. If anyone else has any experience I would love to add that to this thread. On Tue, Nov 30, 2010 at 7:04 PM, Singer X.J. Wang w...@pythian.com wrote: You could also try using a Mysql Proxy on your local machine. The proxy should handle the connection to the servers and you have just a service on your machine. If this service is unreachable you will loose again, but the chance is even smaller and you have automagic fallbacks. http://dev.mysql.com/downloads/mysql-proxy/ best regards Stefan -- _ -- 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 -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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] ADSL Load Balancing
Your router would have to do per-destination when it came to load balancing between the two dsl circuits. That way a single call could only use one dsl path. On Nov 2, 2010 7:36 PM, Dan Journo d...@keshercommunications.com wrote: Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan -- _ -- 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] Queue Group not forwaring calls to agents
I am trying to set up Hunt Groups and I am having some issues. Here is what
I am trying to do. All my users actually register with OpenSIPS. Asterisk
is using Realtime and I have set up a MySQL View Table so that Asterisk
see's all the SIP users info that OpenSIPS has. This is what I have
configured
queues.conf
--
[irock.com]
strategy=leastrecent
ringinuse=no
joinempty=yes
leavewhenempty=no
announce-frequency=30
min-announce-frequency=15
periodic-announce-frequency=60
announce-holdtime=yes
announce-position=yes
; (You are now first in line.)
queue-youarenext = queue-youarenext
; (There are)
queue-thereare = queue-thereare
; (calls waiting.)
queue-callswaiting = queue-callswaiting
; (The current est. holdtime is)
queue-holdtime = queue-holdtime
; (minutes.)
queue-minutes = queue-minutes
; (seconds.)
queue-seconds = queue-seconds
; (Thank you for your patience.)
queue-thankyou = queue-thankyou
; (Hold time)
queue-reporthold = queue-reporthold
; (All reps busy / wait for next)
periodic-announce = queue-periodic-announce
extension.conf
-
exten = 9012211611,1,Answer
exten = 9012211611,2,Queue(irock.com,tT,,,300)
exten = *50,1,Answer
exten = *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4})
exten = *50,n,Hangup
exten = *51,1,Answer
exten = *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4})
exten = *51,n,Hangup
;exten = *50,1,AgentLogin();
[macro-queue-login]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone});
exten = s,n,AddQueueMember(${queue});
exten = s,n,Playback(agent-loginok)
[macro-queue-logout]
exten = s,1,Set(agent=${EXTEN:4})
exten = s,n,Set(queue=irock.com)
exten = s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone});
exten = s,n,RemoveQueueMember(${queue});
exten = s,n,Playback(agent-loggedoff)
When I do a queue show I see the following
Asterisk18*CLI queue show
irock.com has 0 calls (max unlimited) in 'leastrecent' strategy (0s
holdtime, 0s talktime), W:0, C:0, A:15, SL:0.0% within 0s
Members:
SIP/9012211610 (dynamic) (Unavailable) has taken no calls yet
No Callers
So after I have logged in the agent by dialing *50 it shows up in the queue
as a member but says Unavailable. So when someone calls the queue number
9012211611 I see the following
Executing [9012211...@irock.com:2] Queue(SIP/9012732009-0045,
irock.com,tT,,,300) in new stack
-- Started music on hold, class 'default', on SIP/9012732009-0045
-- Stopped music on hold on SIP/9012732009-0045
-- SIP/9012732009-0045 Playing 'queue-youarenext.slin' (language
'en')
-- Told SIP/9012732009-0045 in irock.com their queue position (which
was 1)
-- SIP/9012732009-0045 Playing 'queue-thankyou.slin' (language
'en')
-- Started music on hold, class 'default', on SIP/9012732009-0045
And I hear all the announcements, but it never calls the agent.
Here is the output when I do a sip show peer for the agent that should be
called.
Asterisk18*CLI sip show peer 9012211610 load
* Name : 9012211610
Realtime peer: Yes, cached
Secret : Not set
MD5Secret: Not set
Remote Secret: Not set
Context : irock.com
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 9012211610
FromDomain : irock.com Port 5060
Callgroup:
Pickupgroup :
MOH Suggest :
Mailbox : 9012211610
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Busy level : 1
Dynamic : No
Callerid :
MaxCallBR: 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID: No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : aethercommunications.com
Addr-IP : 173.203.87.134:5060
Defaddr-IP : 97.74.144.17:5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 9012211610
SIP Options : (none)
Codecs : 0x8008000e (gsm|ulaw|alaw|h263|testlaw)
Codec Order : (none)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent:
Reg. Contact :
Qualify Freq : 6 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use
