Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system
On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote: hello people, I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some reason I have noticed that only after a few test calls, the asterisk process is running between 95% - 99.9% CPU when there's absolutely nothing on the system. This is a clean Asterisk system in an internal network with nothing else on it with no calls on it but it's still sitting with 96% CPU. I'm not a developer so not that ept with using debug tools etc to figure out why it's doing that. Could anyone please tell me how I can figure out why it's doing this and/or help debug this. Makes no sense for it to be using CPU with nothing happening on the system The first thing I'd do is run 'top', press shift H, and see what is/are the offending thread(s). Is it a single thread? Two? More? Is it all user time? Much of it is system time? If you strace the PID of the top thread (strace -p PID), what do you see? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cant find asterisk src dir for FreePBX full distro
On Tue, Jul 05, 2011 at 04:07:28PM +0200, Tobias Steen wrote: It seems that the full distro package from FreePBX with Asterisk 1.8.1.4 someway hides (deletes?) the source directory for asterisk after installation. Is it installed from rpm packages? If so: are those available on-line? If so: look for .src..rpm packages near by. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] single keypress short-circuits to invalid extension handler
Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0throughdigits/0digit s/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0digits/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] single keypress short-circuits to invalid extension handler
You can't use WaitExten to receive two digits. Use Read() command. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, July 06, 2011 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] single keypress short-circuits to invalid extension handler Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this. Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0throughdigits/0digit s/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output. -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0digits/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Agents outbound calls to be recorded
Hi All; I know that incoming calls for the agent can be recorded, but how I can let the outbound calls for the agents to be recorded? I can determine the directory to store the outbound calls of the agents to be other than the directory to store the incoming calls of the agents? 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] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system
On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote: hello people, I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some reason I have noticed that only after a few test calls, the asterisk process is running between 95% - 99.9% CPU when there's absolutely nothing on the system. This is a clean Asterisk system in an internal network with nothing else on it with no calls on it but it's still sitting with 96% CPU. I'm not a developer so not that ept with using debug tools etc to figure out why it's doing that. Could anyone please tell me how I can figure out why it's doing this and/or help debug this. Makes no sense for it to be using CPU with nothing happening on the system The first thing I'd do is run 'top', press shift H, and see what is/are the offending thread(s). Is it a single thread? Two? More? Is it all user time? Much of it is system time? If you strace the PID of the top thread (strace -p PID), what do you see? Hi Tzafrir, thanks for the comments and suggestions. So I'd done all of that and what I'd found was - After I'd done Shift-h, There was only one / single thread that was taking all of the CPU - 33% was Sser and 66% was System times - when I'd run an strace on the PID of the offending thread it just rolled some message past my screen which I couldn't capture and can't remember what it said :( Anyway I've killed that process, updated the packages the system, upgraded to 1.8.4.4 and will give it another shot and see what happens. Would've helped if I'd kept the system as it was so people could help me figure out what was going on, but the fact that it stopped responding to commands which were trying to kill the hung channels, reloading configs, or even trying to stop the system wouldn't work is bizarre. I hope the developers pay attention to that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Recording SIP history
Hi, can anyone help with this? Thanks Lee From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: 05 July 2011 16:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Recording SIP history Hi all, can someone explain what siphistory is supposed to do as it appears to record nothing to my log files. When I sip show history callid it's fine but it's not logging anything. My logger.conf has debug = debug and the debug file grows. Is my understanding correct in that at the end of the call the entire sip show history callid should be dumped to the debug file? I am using 1.6.2.19. ;--- SIP DEBUGGING --- sipdebug=yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) dumphistory=yes; Dump SIP history at end of SIP dialogue ; SIP history is output to the DEBUG logging channel Thanks Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Agents outbound calls to be recorded
Hi All; I know that incoming calls for the agent can be recorded, but how I can let the outbound calls for the agents to be recorded? I can determine the directory to store the outbound calls of the agents to be other than the directory to store the incoming calls of the agents? Regards Bilal Hi Bilal, use MixMonitor() in the outgoing extension for your agents. Regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 on Debian / Sparc taking up 95%+ CPU with No calls on the system
On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote: On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote: hello people, I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some reason I have noticed that only after a few test calls, the asterisk process is running between 95% - 99.9% CPU when there's absolutely nothing on the system. This is a clean Asterisk system in an internal network with nothing else on it with no calls on it but it's still sitting with 96% CPU. I'm not a developer so not that ept with using debug tools etc to figure out why it's doing that. Could anyone please tell me how I can figure out why it's doing this and/or help debug this. Makes no sense for it to be using CPU with nothing happening on the system The first thing I'd do is run 'top', press shift H, and see what is/are the offending thread(s). Is it a single thread? Two? More? Is it all user time? Much of it is system time? If you strace the PID of the top thread (strace -p PID), what do you see? Hi Tzafrir, thanks for the comments and suggestions. So I'd done all of that and what I'd found was - After I'd done Shift-h, There was only one / single thread that was taking all of the CPU - 33% was Sser and 66% was System times - when I'd run an strace on the PID of the offending thread it just rolled some message past my screen which I couldn't capture and can't remember what it said :( Just press ctrl-c . Anyway I've killed that process, updated the packages the system, upgraded to 1.8.4.4 and will give it another shot and see what happens. Would've helped if I'd kept the system as it was so people could help me figure out what was going on, but the fact that it stopped responding to commands which were trying to kill the hung channels, reloading configs, or even trying to stop the system wouldn't work is bizarre. I hope the developers pay attention to that. Developers need some data to work with :-( -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system
On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote: On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote: hello people, I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some reason I have noticed that only after a few test calls, the asterisk process is running between 95% - 99.9% CPU when there's absolutely nothing on the system. This is a clean Asterisk system in an internal network with nothing else on it with no calls on it but it's still sitting with 96% CPU. I'm not a developer so not that ept with using debug tools etc to figure out why it's doing that. Could anyone please tell me how I can figure out why it's doing this and/or help debug this. Makes no sense for it to be using CPU with nothing happening on the system The first thing I'd do is run 'top', press shift H, and see what is/are the offending thread(s). Is it a single thread? Two? More? Is it all user time? Much of it is system time? If you strace the PID of the top thread (strace -p PID), what do you see? Hi Tzafrir, thanks for the comments and suggestions. So I'd done all of that and what I'd found was - After I'd done Shift-h, There was only one / single thread that was taking all of the CPU - 33% was Sser and 66% was System times - when I'd run an strace on the PID of the offending thread it just rolled some message past my screen which I couldn't capture and can't remember what it said :( Just press ctrl-c . haha I did that but since that I did a 100 other things in my ssh window which is only buffered for 5000 lines and those messages have gone past. Anyway I've killed that process, updated the packages the system, upgraded to 1.8.4.4 and will give it another shot and see what happens. Would've helped if I'd kept the system as it was so people could help me figure out what was going on, but the fact that it stopped responding to commands which were trying to kill the hung channels, reloading configs, or even trying to stop the system wouldn't work is bizarre. I hope the developers pay attention to that. Developers need some data to work with :-( Haha of course. Although I have a feeling it'll happen again as this is the 2nd time this has happened. Will keep the system in that state till we can try and resolve this and capture enough info. if I had better memory, I'd have actually remembered what the message was, but anyway, what I was trying to say was that it's much more than just taking up all the CPU tells me that some thread has just gone loco. But the fact the CLI and AMI commands become unresponsive when trying to kill these zombie channels or trying to do a core reload or core stop now etc. tells me that this is a bigger issue than just some thread gone nuts and the channels being hung -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 on Debian / Sparc taking up 95%+ CPU with No calls on the system
On Wed, Jul 06, 2011 at 07:11:26AM -0400, A E [Gmail] wrote: On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote: - when I'd run an strace on the PID of the offending thread it just rolled some message past my screen which I couldn't capture and can't remember what it said :( Just press ctrl-c . haha I did that but since that I did a 100 other things in my ssh window which is only buffered for 5000 lines and those messages have gone past. If the process / thread is in a loop, the messages tend to repeat themselves. Also: anything interesting in /var/log/asterisk/messages ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring connection to VoIP provider?
Hello I was wondering if Asterisk can be configured to monitor a connection to a VoIP provider, whether someone is currently using it for a call or the connection is idle? FWIW, my VoIP provider doesn't run an iperf server on their side. I don't know if ping/traceroute is a good enough solution to monitor an SIP connection. I'd like this so I can check how good the line is before calling or receiving a call. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system
On Wed, Jul 6, 2011 at 7:50 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jul 06, 2011 at 07:11:26AM -0400, A E [Gmail] wrote: On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote: - when I'd run an strace on the PID of the offending thread it just rolled some message past my screen which I couldn't capture and can't remember what it said :( Just press ctrl-c . haha I did that but since that I did a 100 other things in my ssh window which is only buffered for 5000 lines and those messages have gone past. If the process / thread is in a loop, the messages tend to repeat themselves. Also: anything interesting in /var/log/asterisk/messages ? Yup, it surely was in some funky loop...and I wouldn't be surprised if it was looping to check if the channels were hungup or not and ended up taking up the entire CPUI should've tried to just kill that thread with its PID and seen if the operation returns to normal. No, unfortunately nothing interesting found in the logs, other than the indication that when I tried to reload using core reload it was actually loading the configs even though it didn't show anything on the CLI. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] single keypress short-circuits to invalid extension handler
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, July 06, 2011 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] single keypress short-circuits to invalid extension handler Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this... Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0through digits/0digits/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output... -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0dig its/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack You don't have a priority 1 exten = 00,1,MeetMe(SouthAfrica0,dMs) exten = 01,1,MeetMe(Swaziland1,dMs) exten = 02,1,MeetMe(Botswana2,dMs) Etc. WaitExten can accept more than one digit. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] single keypress short-circuits to invalid extension handler
On 7/6/11 3:20 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, July 06, 2011 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] single keypress short-circuits to invalid extension handler Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this... Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0through digits/0digits/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output... -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0dig its/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack You don't have a priority 1 exten = 00,1,MeetMe(SouthAfrica0,dMs) exten = 01,1,MeetMe(Swaziland1,dMs) exten = 02,1,MeetMe(Botswana2,dMs) Etc. WaitExten can accept more than one digit. Thanks Eric - this was it. I knew WaitExten() would read more than 1 digit. I guess I'd been staring at it so long I couldn't see the error. I appreciate the extra eyes! Cassius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Polycom Blf / Directed Pickup
Using a Polycom 650 with 3.3.1, I could not have Directed Pickup working. More precisely, I configured the phone using call and attendant entries as described in this thread. Whenever a call comes in, BLF is blinking green. Pressing the associated key generate generates a general Call Pickup (*8), not a directed Call Pickup. Could you confirm this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] libpri 1.4.12 Now Available
The Asterisk Development Team announces the release of libpri version 1.4.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The following are some of the issues resolved in this release: * Add call transfer exchange of subaddresses support and fix PTMP call transfer signaling. * Invalid PTMP redirecting signaling as TE towards NT. * Add Q931_IE_TIME_DATE to CONNECT message when in network mode. (issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett) * Swap of master/slave in pri_enslave() incorrect. (issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie) * Fix I-frame retransmission quirks. * Crash if NFAS swaps D channels on a call with an active timer. * DMS-100 not receiving caller name anymore. (issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett) * B channel lost by incoming call in BRI NT PTMP mode. * Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls. This release contains several new features, among them: 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support 2.) ETSI Advice Of Charge (AOC) support 3.) ETSI Explicit Call Transfer (ECT) support 4.) ETSI Call Waiting support for ISDN phones 5.) ETSI Malicious Call ID support 6.) Add Display IE text handling options. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12 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] Dropping Conference calls
So I made the change you suggested. That still hasn't worked, but I did manage
to grab some logging from a dropped call.
[Jul 6 16:19:37] DEBUG[25950] channel.c: Got a FRAME_CONTROL (8) frame on
channel DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] res_rtp_asterisk.c: Setting the marker bit due
to a source update
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Requested indication 20 on channel
DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] channel.c: Bridge stops bridging channels
SIP/7531-0077 and DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: Inserting a CDR record.
[Jul 6 16:19:37] DEBUG[25950] cdr_mysql.c: SQL command as follows: INSERT INTO
cdr
(`calldate`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`accountcode`,`uniqueid`)
VALUES ('2011-07-06
15:58:57','7531','8883203585','from-sip','SIP/7531-0077','DAHDI/i1/18883203585-7e','Dial','DAHDI/g1/18883203585','1240','1238','ANSWERED','3','\Adam
Witwer\','1309982337.338')
[Jul 6 16:19:37] DEBUG[25950] channel.c: Hanging up channel
'DAHDI/i1/18883203585-7e'
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c:
dahdi_hangup(DAHDI/i1/18883203585-7e)
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value:
ON(1) on DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: sig_pri_hangup 1
[Jul 6 16:19:37] DEBUG[25950] sig_pri.c: Not yet hungup... Calling hangup
once with icause, and clearing call
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Disabled echo cancellation on
channel 1
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option TDD MODE, value: OFF(0)
on DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Updated conferencing on 1, with 0
conference users
[Jul 6 16:19:37] DEBUG[25950] chan_dahdi.c: Set option AUDIO MODE, value:
OFF(0) on DAHDI/i1/18883203585-7e
[Jul 6 16:19:37] VERBOSE[25950] chan_dahdi.c: -- Hungup
'DAHDI/i1/18883203585-7e'
On Jul 1, 2011, at 2:38 PM, Jonathan Thomas wrote:
The exited non-zero is typical when a call has ended. What I would recommend
(easiest method) is for you to enter the CLI using: asterisk
–rvvv
The v’s will provide more verbose logging, the 4 d’s will place the core in
debug mode(4). Once in the CLI, pick a phone you will use as a test unit and
issue a
sip set debug peer XX (X=peer device id, such as 10001)
This will seriously increase the size of your logging – but should provide
you with a very thorough trace of the call as its in flight, including the
SIP dialog between the phone/server.
Perhaps you can enable the above, place a call that drops, then snip that
section of the full log and send it to the list for parsing. It’s the best
way to nail down an issue like this.
JT
From: Mark Rosedale [mailto:mrosed...@oreilly.com]
Sent: Friday, July 01, 2011 2:17 PM
To: jonathan.tho...@us.patersons.net
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dropping Conference calls
So I didn't have sip debug set. So I don't have any SIP TIMER's in my log. I
have that set now.
I would be interested in the debut/logs if you have them.
I do have Spawn extension...exited non-zero on 'SIP/'
Here is the specifics
VERBOSE[10928] pbx.c: == Spawn extension (from-sip, 1***, 1) exited
non-zero on 'SIP/7XXX-09d7'
Not sure if that relates or not, but it is the only hit for the connection
between my sip client and the PRI going outbound right before the hangup.
On Jul 1, 2011, at 11:21 AM, Jonathan Thomas wrote:
The key item in my logs, which would preface the call dropping, was:
[2011-06-28 09:43:49] DEBUG[25563] chan_sip.c: ** SIP TIMER: Cancelling
retransmit of packet (reply received) Retransid #858
For instance - a call would be connected. SIP debug/core debug on. At the
14:30 mark I would begin tailing the full log. Once I saw the SIP TIMER
notice, it would be followed by a new INVITE (re-invite) SIP transmission
that would be sent to the phone currently on call. This re-invite was odd
in that it would be on a different port to the phone than was already
established (for example the NAT outgoing SIP OPTIONS would be sent to the
phone on port 27608 - and this re-invite might go out on port 35780). The
behavior following would be: Asterisk would hang up as though the parties
disconnected - however the phone would show the call was still going and
would continue sending SIP responses to asterisk indicating as such. When
the phone was manually hung up it would send a SIP BYE (as normal) to
asterisk - indicating it had no notice that Asterisk dropped the call.
Adding to sip.conf
session-timers=refuse
Resolved the issue by stopping Asterisk from sending these re-invites during
a live call.
Hope that helps! I have more SIP debugs/logs if they're useful to ya.
Re: [asterisk-users] Blind Transfer Connected
IMHO, blind tranfer definition is to NOT connect A and B back That is correct, and is why it's called a 'blind' transfer; the transferring party does not know or care what happens to the call after effecting the transfer. That's not what users migrating from some legacy PBXs expect, our old Fujitsu essence will call back the transferrer if the call isn't answered. The good old 'hook flash', dial the extension, then hangup. Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ooh323 does not work fine, what about h323 channel
Hi All; The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by selecting the add-on). But really does not work in good performance, for example: if a call came from gnugk to asterisk and the ooh323 handled it, the performance is bad .. some calls are drop and if it is ringing, then it rings for small duration and then stop ringing In other words, if the call went from gnugk to the provider directly (all the path h323), it is better than coming for Asterisk via the ooh323 channel and then to be translated for SIP to be sent for provider. I would like to try the h323 channel (and not the ooh323), but I do not know what I have to do to compile? Any advise? Did anyone tried yate to do the translation from h323 to sip? How it is? 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] Blind Transfer Connected
On 07/06/2011 04:44 PM, Alec Davis wrote: IMHO, blind tranfer definition is to NOT connect A and B back That is correct, and is why it's called a 'blind' transfer; the transferring party does not know or care what happens to the call after effecting the transfer. That's not what users migrating from some legacy PBXs expect, our old Fujitsu essence will call back the transferrer if the call isn't answered. The good old 'hook flash', dial the extension, then hangup. Well, that would have to be handled in the dialplan somehow, because Asterisk alone can't decide when a call is 'not answered'. However, writing such a dialplan would indeed be non-trivial :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Agents outbound calls to be recorded
On 7/6/2011 4:36 AM, bilal ghayyad wrote:
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let the
outbound calls for the agents to be recorded? I can determine the directory to
store the outbound calls of the agents to be other than the directory to store
the incoming calls of the agents?
Regards
Bilal
This is an example of what we do.
MixMonitor(crm/${STRFTIME(${EPOCH},,%B)}/${STRFTIME(${EPOCH},,%d-%m-%Y)}/${STRFTIME(${EPOCH},,%Y%m%d)}-${EXTEN:3}N-${UNIQUEID}-${CALLERID(NUM)}.wav,v(-1)V(2)b,)
What this does is save the recording in:
/var/spool/asterisk/monitor/crm/July/06-07-2011/ (Date in Euro format)
with name:
mmdd-dialednumber-uniqueid-extensionthatdialed.wav
Warning:
I've seen 1.8 create the directory if it does not exist. Asterisk 1.4
will NOT create it. Don't know what 1.6 does with it.
Alex
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Re: [asterisk-users] Blind Transfer Connected
On 07/06/2011 05:52 PM, Kevin P. Fleming wrote: On 07/06/2011 04:44 PM, Alec Davis wrote: IMHO, blind tranfer definition is to NOT connect A and B back That is correct, and is why it's called a 'blind' transfer; the transferring party does not know or care what happens to the call after effecting the transfer. That's not what users migrating from some legacy PBXs expect, our old Fujitsu essence will call back the transferrer if the call isn't answered. The good old 'hook flash', dial the extension, then hangup. Well, that would have to be handled in the dialplan somehow, because Asterisk alone can't decide when a call is 'not answered'. However, writing such a dialplan would indeed be non-trivial :-) Not to mention the expansive myriad of things that can answer the call these days, like sundry voicemail systems, that do not constitute an answer in the sense desired by the transferring party. On the other hand, if you make the ring timeout too short, that breaks functionality such as call forwarding to a cell phone on the recipient side. It seems to me that keeping blind transfer truly blind is the only viable strategy in the contemporary device, service and feature milieu. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] Monitoring connection to VoIP provider?
Community can help you better if you provide some details about you scenario and requirement. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Wednesday, July 06, 2011 5:03 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Monitoring connection to VoIP provider? Hello I was wondering if Asterisk can be configured to monitor a connection to a VoIP provider, whether someone is currently using it for a call or the connection is idle? FWIW, my VoIP provider doesn't run an iperf server on their side. I don't know if ping/traceroute is a good enough solution to monitor an SIP connection. I'd like this so I can check how good the line is before calling or receiving a call. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] ooh323 does not work fine, what about h323 channel
Hi, As per my experience YATE is the best option for H323=SIP Proxy. Regards, Faisal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, July 07, 2011 2:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ooh323 does not work fine, what about h323 channel Hi All; The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by selecting the add-on). But really does not work in good performance, for example: if a call came from gnugk to asterisk and the ooh323 handled it, the performance is bad .. some calls are drop and if it is ringing, then it rings for small duration and then stop ringing In other words, if the call went from gnugk to the provider directly (all the path h323), it is better than coming for Asterisk via the ooh323 channel and then to be translated for SIP to be sent for provider. I would like to try the h323 channel (and not the ooh323), but I do not know what I have to do to compile? Any advise? Did anyone tried yate to do the translation from h323 to sip? How it is? 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
