Re: [asterisk-users] Possible bug - queue doesn't play hold music
On Wed, 2012-12-19 at 11:16 -0600, Richard Mudgett wrote: On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... The incoming call leg does not appear to be answered yet so I would not expect the caller to be able to hear MOH. Richard The queue command in the dialplan does not have the r option, therefore it should play the moh as specified in the queue's specific config rather than a ringing tone. This has been the behaviour in previous 1.8 and in 1.4. Also, why does the log indicate moh starting? -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Possible bug - queue doesn't play hold music
On Thu, 2012-12-20 at 09:23 +, Ishfaq Malik wrote: On Wed, 2012-12-19 at 11:16 -0600, Richard Mudgett wrote: On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... The incoming call leg does not appear to be answered yet so I would not expect the caller to be able to hear MOH. Richard The queue command in the dialplan does not have the r option, therefore it should play the moh as specified in the queue's specific config rather than a ringing tone. This has been the behaviour in previous 1.8 and in 1.4. Also, why does the log indicate moh starting? My apologies, you are correct. Every other time I've used MoH in a queue it has been afetr a Playback or Background, both of which answer the channel... -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Congestion() forcing PRI channels to be not available
On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote: You probably already know this, but 1.4x is very old (released in 2006) and is officially end-of-life. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You might get more help or better behavior by updating to a newer more current version of Asterisk, such as 1.8 which will be receiving bug fixes into October 2014. On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel. Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a 'dahdi restart' is issued. Thanks. -- James I believe that what you are describing is a very old bug, which is fixed somewhere in the 1.8 timeline when the interface between DAHDI and Asterisk is changed slightly. I encountered the same issue some time ago. I do not recall the exact conditions under which the issue happens, but I believe it is the attempt to cancel an unanswered inbound call with a specific subset of cause codes. If you are using an older Asterisk version, the only workaround is to use Playtones + Hangup() instead of sending the Congestion() or Busy() cause codes. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk conferencing |MEETME or app_conference
Hii, we implemented the same senario with modification of meet-me application and using php and mysql. When user want to talk, he press some predefined digit on the phone. Admin can view who raise hands (press digit) in php page and press predeifned digit and that user get unmuted. As soon as user complete his talk, admin again press some digit and that user get muted. Regards, Bharat Lalcheta On Wed, Dec 19, 2012 at 7:41 PM, pankaj pandey pankaj.n...@yahoo.comwrote: conference, when QA session begins, is there a way for participants to raise hands, if they have any questions so Leader can unmute them? -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Recommended T.38 settings for receiving faxes from Cisco AS5350XM
Hi, What are the recommended T.38 settings for sending/receiving faxes from Cisco AS5350XM gateways? The chan_sip.conf file has a remark about what Cisco is doing wrong and says that the values received from the gateway should be overridden, but doesn't say what settings to use for maximum success. Can anyone give me some suggestions? I don't know much about T.38 and I've been told I have to solve this... Thanks! -- Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 11 and no RTP
I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network. Tried looking at the config.log for rtp anything and it looks ok. Anyone know what might be the issue? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Congestion() forcing PRI channels to be not available
On Thu, Dec 20, 2012 at 2:22 AM, Steve Davies davies...@gmail.com wrote: On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote: You probably already know this, but 1.4x is very old (released in 2006) and is officially end-of-life. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You might get more help or better behavior by updating to a newer more current version of Asterisk, such as 1.8 which will be receiving bug fixes into October 2014. On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna jlama...@gmail.comwrote: Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel. Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a 'dahdi restart' is issued. Thanks. -- James I believe that what you are describing is a very old bug, which is fixed somewhere in the 1.8 timeline when the interface between DAHDI and Asterisk is changed slightly. I encountered the same issue some time ago. I do not recall the exact conditions under which the issue happens, but I believe it is the attempt to cancel an unanswered inbound call with a specific subset of cause codes. If you are using an older Asterisk version, the only workaround is to use Playtones + Hangup() instead of sending the Congestion() or Busy() cause codes. Regards, Steve Thanks Steve. It must have been introduced between DAHDI 2.4.0 and 2.6.1 or between Asterisk 1.4.35 and 1.4.44. I had a box running Asterisk 1.4.35 + DAHDI 2.4.0 and I never had any issues. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and no RTP
Jerry Geis wrote: I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network. Tried looking at the config.log for rtp anything and it looks ok. Anyone know what might be the issue? Thanks, Jerry Start by making sure the rtp engine module was compiled and added to your modules directory. If you are using default directories, you can do this with the following: $ ls /usr/lib/asterisk/modules -l | grep res_rtp_ For me this shows two engine resources which register, those being: -rwxr-xr-x 1 root root 990101 2012-12-19 15:25 res_rtp_asterisk.so -rwxr-xr-x 1 root root 207519 2012-12-19 15:25 res_rtp_multicast.so If those aren't present, and especially if res_rtp_asterisk isn't present, you'll likely need use make menuselect to make sure they are enabled. If they aren't fixing this could be pretty trivial. If they are, make sure you aren't getting a build error when compiling. If they are present (and I'm kinda guessing they are), check Asterisk's log messages for problems with registering the RTP engines. Specifically you are looking for log messages from rtp_engine.c which have a form similar to: RTP Engine 'name of engine' failed sanity check so it was not registered. Alternatively the register rtp engine function can fail if the engine is already in the list, but that seems unlikely. If that IS the case, you'll have an error more similar to: An RTP engine with the name 'name of engine' has already been registered. I imagine you'll much more likely see the first one though. I don't think we've put much effort into supporting Asterisk on ARM platforms, so I don't really know much about the specifics on why this would be failing. Hopefully the above helps though. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt file: * The PRI channels in chan_dahdi can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ispan/number[:subaddress]-sequence-number * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with no-media as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. ok - can I use a different method of doing my check to see if a line is available by using the AMI call ExtensionState or ChanIsAvail? Doing Action: ExtensionState Parameters: DAHDI/1 says Error Message: Extension not specified and Action: Command Command: ChanIsAvail Parameters: DAHDI/1 says Error No such command ChanIsAvail I'm clearly missing something? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.19.0 - [2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL
On 12/18/2012 10:56 AM, Tim Nelson wrote: I'm getting this error message on my Asterisk CLI, and in the logs, roughly every 10-20 seconds: [2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL While it doesn't appear to be actually affecting anything, I'm curious to know what the error represents, where it's coming from, and of course, if there is a fix for it. All info appreciated, thanks! --Tim Yeah, that's not a good thing. Its an error message from Asterisk's reference counted object library, indicating one of two things: 1) The object attempting to be manipulated is not a reference counted object 2) The object attempting to be manipulated was (potentially) already destroyed Either way, you'll usually need some context to figure out what object is being manipulated inappropriately. You can get (very) detailed reference count logs by enabling reference count debugging in Asterisk [1]. Note that this isn't available in menuselect, as it has to be enabled in the specific modules you whose objects you want to track. [1] https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11 and DAHDI/i4
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt file: * The PRI channels in chan_dahdi can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ispan/number[:subaddress]-sequence-number * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with no-media as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. ok - can I use a different method of doing my check to see if a line is available by using the AMI call ExtensionState or ChanIsAvail? Doing Action: ExtensionState Parameters: DAHDI/1 says Error Message: Extension not specified It is Action: ExtensionState Exten: 5551212 Context: fubar This will return the status of the dialplan exten hint. and Action: Command Command: ChanIsAvail Parameters: DAHDI/1 says Error No such command ChanIsAvail ChanIsAvail is a dialplan application not a CLI command. It also will not work for what you want in this case. I'm clearly missing something? Quite possibly. :) Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
It is Action: ExtensionState Exten: 5551212 Context: fubar This will return the status of the dialplan exten hint. / and // Action: Command // Command: ChanIsAvail // Parameters: DAHDI/1 // // says Error // No such command ChanIsAvail / ChanIsAvail is a dialplan application not a CLI command. It also will not work for what you want in this case. / I'm clearly missing something? / Quite possibly. :) Richard OK - so what I am trying to do is through the AMI interface ask if channel DAHDI/1 is busy, on hook or available. How do I tell that In the past I simply did a core show channels and see if DAHDI/1 was present. It it was I new it was in use... How do check now in asterisk 11 if the channels are reported as DAHDI/i4 etc... Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
The Asterisk 11 part is irrelevant. You need to use an AGI or local call to use the ChanIsAvail function. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 20, 2012 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 11 and DAHDI/i4 It is Action: ExtensionState Exten: 5551212 Context: fubar This will return the status of the dialplan exten hint. and Action: Command Command: ChanIsAvail Parameters: DAHDI/1 says Error No such command ChanIsAvail ChanIsAvail is a dialplan application not a CLI command. It also will not work for what you want in this case. I'm clearly missing something? Quite possibly. :) Richard OK - so what I am trying to do is through the AMI interface ask if channel DAHDI/1 is busy, on hook or available. How do I tell that In the past I simply did a core show channels and see if DAHDI/1 was present. It it was I new it was in use... How do check now in asterisk 11 if the channels are reported as DAHDI/i4 etc... Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
It is Action: ExtensionState Exten: 5551212 Context: fubar This will return the status of the dialplan exten hint. and Action: Command Command: ChanIsAvail Parameters: DAHDI/1 says Error No such command ChanIsAvail ChanIsAvail is a dialplan application not a CLI command. It also will not work for what you want in this case. I'm clearly missing something? Quite possibly. :) Richard OK - so what I am trying to do is through the AMI interface ask if channel DAHDI/1 is busy, on hook or available. How do I tell that In the past I simply did a core show channels and see if DAHDI/1 was present. It it was I new it was in use... How do check now in asterisk 11 if the channels are reported as DAHDI/i4 etc... You should just cache the AMI DAHDIChannel event information in your program. If you really must you could use the CLI command pri show channels. However, it is not intended to be repeatedly run for performance reasons. It blocks processing of ISDN messages while it is running. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
You should just cache the AMI DAHDIChannel event information in your program. If you really must you could use the CLI command pri show channels. However, it is not intended to be repeatedly run for performance reasons. It blocks processing of ISDN messages while it is running. I am not continually logged in to the AMI to catch those events... Can I make a call to a local channel, run some context+ extension, there call ChanIsAvail for the channel I am interested in - but they how do I get that info back to my C program? Also is that a big overhead? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
IMO the local channel call should be the lowest overhead option available. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, December 20, 2012 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 11 and DAHDI/i4 You should just cache the AMI DAHDIChannel event information in your program. If you really must you could use the CLI command pri show channels. However, it is not intended to be repeatedly run for performance reasons. It blocks processing of ISDN messages while it is running. I am not continually logged in to the AMI to catch those events... Can I make a call to a local channel, run some context+ extension, there call ChanIsAvail for the channel I am interested in - but they how do I get that info back to my C program? Also is that a big overhead? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
You should just cache the AMI DAHDIChannel event information in your program. If you really must you could use the CLI command pri show channels. However, it is not intended to be repeatedly run for performance reasons. It blocks processing of ISDN messages while it is running. I am not continually logged in to the AMI to catch those events... Can I make a call to a local channel, run some context+ extension, there call ChanIsAvail for the channel I am interested in - but they how do I get that info back to my C program? Also is that a big overhead? Whether the overhead is going to affect performance to be a problem depends on how often you execute the command. You can also use the AMI DAHDIShowChannels action. If the channel has a B channel it will be listed with which B channel it currently is attached. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
IMO the local channel call should be the lowest overhead option available. What about: Action: Command Command: dahdi show channels I can just look to see if Extension has anything for the Chan I am interested in? is that a big overhead and block anything? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
On 12/20/2012 01:00 PM, Jerry Geis wrote: IMO the local channel call should be the lowest overhead option available. What about: Action: Command Command: dahdi show channels I can just look to see if Extension has anything for the Chan I am interested in? is that a big overhead and block anything? Jerry Looks like the dahdi show channels does not work for me because if I make a call originating from a polycom phone going out a line (DAHDI/1) it is not reflected there. So I cannot tell the line is in use. It only shows a call as active if its an incoming call. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 11 and DAHDI/i4
I have a little dialplan context now...
[check-chanisavail]
exten = s,1,ChanIsAvail(${agi_channel})
exten = s,n,System(/bin/echo ${AVAILCHAN} /tmp/${agi_file})
exten = s,n,Hangup()
and a call file:
Channel: Local/s@check-chanisavail/n
Context: check-chanisavail
Extension: s
Priority: 1
SetVar: agi_file=jerry
SetVar: agi_channel=DAHDI/1
To tell me if a channel is busy or not.
When no channels are busy I execute the dialplan above and
it correctly gave me DAHDI/1-1 in my file. As expected.
When I call in from the outside into my asterisk box,
then I execute my dialplan above and I query (DAHDI/1) I get
AVAILCHAN = which is what I expect.
However, if I use a polycom phone to dial out, and then I execute
my dialplan above and I query (DAHDI/1) it says its still available.
Should it not say AVAILCHAN = as I am using that line it is not
available.
Thanks,
Jerry
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Re: [asterisk-users] asterisk 11 and DAHDI/i4
Just for grins, do you have a softphone like xlite that you can try the
outgoing call on? I think it's an outgoing issue, not a polycom one.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 20, 2012 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 11 and DAHDI/i4
I have a little dialplan context now...
[check-chanisavail]
exten = s,1,ChanIsAvail(${agi_channel}) exten = s,n,System(/bin/echo
${AVAILCHAN} /tmp/${agi_file}) exten = s,n,Hangup()
and a call file:
Channel: Local/s@check-chanisavail/n
Context: check-chanisavail
Extension: s
Priority: 1
SetVar: agi_file=jerry
SetVar: agi_channel=DAHDI/1
To tell me if a channel is busy or not.
When no channels are busy I execute the dialplan above and it correctly gave
me DAHDI/1-1 in my file. As expected.
When I call in from the outside into my asterisk box, then I execute my
dialplan above and I query (DAHDI/1) I get AVAILCHAN = which is what I
expect.
However, if I use a polycom phone to dial out, and then I execute my
dialplan above and I query (DAHDI/1) it says its still available.
Should it not say AVAILCHAN = as I am using that line it is not
available.
Thanks,
Jerry
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Re: [asterisk-users] asterisk 11 and DAHDI/i4
Just for grins, do you have a softphone like xlite that you can try the outgoing call on? I think it's an outgoing issue, not a polycom one. I do not have a softphone. I have a yealink VP-2009 and same behavior. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] sip call failed in openbts with asterisk
Cause 20 means your SIP device is not registered or you do not have an IP specified for it in your peer. sip show peers will show that. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Huang Sent: Thursday, December 20, 2012 11:16 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip call failed in openbts with asterisk Hi I met a problem in asterisk, please see message in the following, the detail debug log is in the attached file. can someone help to point out where to correctly configure asterisk, thanks a lot ! BR/Scott --- -- Executing [8690@phones:1] Dial(SIP/IMSI466990004244439-0014, SIP/IMSI466974104638690) in new stack Really destroying SIP dialog '3862c8d23be16ce36e564c3251cbc10c@127.0.1.1:5060' Method: INVITE [Dec 21 00:05:39] WARNING[2838]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466990004244439-0014' status is 'CHANUNAVAIL' -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 1.4.14 Now Available
The Asterisk Development Team has announced the release of libpri 1.4.14. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.14 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix compiler warning in pritest.c. (Closes issue PRI-145. Reported by Tzafrir Cohen) * --- Q.SIG: Allow PROGRESS when in the Active state. (Closes issue PRI-147. Reported by Nick Merrett) * --- Handle optional Recommendation octet 3a in Cause IE. (Closes issue PRI-151. Reported by Tzafrir Cohen) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.14 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] asterisk 11 and no RTP
Error loading module 'res_rtp_asterisk.so': /usr/lib64/libavformat.so.52: undefined symbol: av_tree_node_size This is the error I get when trying to start Asterisk 11 on centos 5. Asterisk 11 works fine on my centos 6 box - I also verified that on centos 6 I do not have the above mentioend file. ls /usr/lib64/libav* libavahi-client.so.3 libavahi-common.so.3 libavahi-core.so.6libavahi-glib.so.1 libavahi-ui.so.0 libavc1394.so.0 libavahi-client.so.3.2.5 libavahi-common.so.3.5.1 libavahi-core.so.6.0.1libavahi-glib.so.1.0.1 libavahi-ui.so.0.1.1 libavc1394.so.0.3.0 Is there another library it can use instead of libavformat? something I have installed on Centos 6 and not on centos 5? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call
When doing a 'dahdi show channel X' from the asterisk console, when the line is not part of a call the echo cancellation line ALWAYS says 'currently OFF'. Once a call is in progress, it will change to 'ON'. Is this a bug, or is the behavior by design? My setup: Asterisk 10.10.0 Dahdi 2.6.1 TE820 T1 card In chan_dahdi.cfg: ... echocancel=yes echocancelwhenbridged=yes echotraining=800 ... And in #included chan_dahdi_additional.conf (this is a freepbx install): ... ;;[3884] signalling=fxo_ks pickupgroup= mailbox=3884@default immediate=no echotraining=800 echocancelwhenbridged=yes echocancel=yes context=from-internal callprogress=no callgroup= callerid=John Doe 3884 busydetect=no busycount=7 accountcode= channel=73 ... -Justin Killen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call
On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote: When doing a 'dahdi show channel X' from the asterisk console, when the line is not part of a call the echo cancellation line ALWAYS says 'currently OFF'. Once a call is in progress, it will change to 'ON'. Is this a bug, or is the behavior by design? This is expected behavior. From a DAHDI perspective: The echocan chip's channels are allocated on demand during channel setup and de-allocated on channel teardown. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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] asterisk 11 and no RTP
Jerry Geis wrote: Error loading module 'res_rtp_asterisk.so': /usr/lib64/libavformat.so.52: undefined symbol: av_tree_node_size This is the error I get when trying to start Asterisk 11 on centos 5. Asterisk 11 works fine on my centos 6 box - I also verified that on centos 6 I do not have the above mentioend file. ls /usr/lib64/libav* libavahi-client.so.3 libavahi-common.so.3 libavahi-core.so.6libavahi-glib.so.1 libavahi-ui.so.0 libavc1394.so.0 libavahi-client.so.3.2.5 libavahi-common.so.3.5.1 libavahi-core.so.6.0.1libavahi-glib.so.1.0.1 libavahi-ui.so.0.1.1 libavc1394.so.0.3.0 Is there another library it can use instead of libavformat? something I have installed on Centos 6 and not on centos 5? I'm not aware of what alternatives if any are available and for which versions of CentOS they would be available for... this sort of thing isn't really something I have a great deal of expertise with. If I had to guess, the problem is that libavahi and libavformat is probably a dependency of pjproject/pjnath which is employed by res_rtp_asterisk Asterisk 11, though I'm not certain about that. You might be able to use the old res_rtp_asterisk by grabbing it from Asterisk 10's source and making whatever tweaks are needed... but I wouldn't recommend it. Is there some reason you can't get the required libraries for CentOS5? Without knowing what the differences are between libs you have on the CentOS5 and CentOS6 machines... I really don't have a lot to go on. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call
This is highly confusing. It would be nice if at least the display gave the configured value as well. -Justin Killen -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russ Meyerriecks Sent: Thursday, December 20, 2012 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote: When doing a 'dahdi show channel X' from the asterisk console, when the line is not part of a call the echo cancellation line ALWAYS says 'currently OFF'. Once a call is in progress, it will change to 'ON'. Is this a bug, or is the behavior by design? This is expected behavior. From a DAHDI perspective: The echocan chip's channels are allocated on demand during channel setup and de-allocated on channel teardown. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] dahdi timing source multiple cards
I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72 span=4,4,0,esf,b8zs bchan=73-96 span=5,5,0,esf,b8zs bchan=97-120 span=6,6,0,esf,b8zs bchan=121-144 span=7,7,0,esf,b8zs bchan=145-168 span=8,8,0,esf,b8zs bchan=169-192 span=9,9,0,esf,b8zs bchan=193-216 span=10,10,0,esf,b8zs bchan=217-240 span=11,11,0,esf,b8zs bchan=241-264 span=12,12,0,esf,b8zs bchan=265-288 dahdi_scan : [1] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=1 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=25 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [3] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 3 name=TE4/0/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=49 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [4] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 4 name=TE4/0/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=73 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [5] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 1 name=TE4/1/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=97 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [6] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 2 name=TE4/1/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=121 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [7] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 3 name=TE4/1/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=145 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [8] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 4 name=TE4/1/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=169 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [9] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 1 name=TE4/2/1 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=193 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [10] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 2 name=TE4/2/2 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=217 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [11] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 3 name=TE4/2/3 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=241 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [12] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 4 name=TE4/2/4 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=265 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [root@aislecom28502 dahdi]# Thanks, Dave
