[asterisk-users] Asterisk 16.23.0 strange issue where Answer request succeeds and able to perform actions but Asterisk never sent 200 OK to answer call
Some background... We use AMI and AsyncAGI to be able to receive events about calls (and other Asterisk details) and control it from our application. Works great and have about 100 sites (some newer, some older) without issues. I was notified this morning about a customer who had something strange happen and I can't explain it. Asterisk 16.23.0 and PJSIP. Call comes into Asterisk. Asterisk sends the Trying. Via AMI, notified of the call and dial plan has it go to AsyncAGI for our application to be able to control the call. Via AMI, we tell Asterisk to Answer. Asterisk processes it and indicates it was answered. The Asterisk AMI/AGI indicates call was answered successfully, call state is Up, etc. Everything appears to be normal. We perform various actions on the call, example play a file, music, tones, etc. However, Asterisk never sent the 200 OK to answer the call. Seems as though Asterisk is in a bizarre state where it thinks it is handling the call, but it really isn't. Reports are this happened to several calls. Eventually, they restarted the entire VM and everything started working well. We think this may be caused by something their switch is doing. Through the grapevine, heard they had some network issue but don't know the details of their switches and architecture for calls coming into Asterisk. We noticed we are seeing two INVITEs happen with the same Call-ID, but 2 additional Record-Route header/value pairs and 3 additional Via header/value pairs. At least in first glance, the rest seems to be the same. I see Asterisk created two different PJSIP calls for each despite same Call-ID, but I am guessing that's because of the additional Via or Record-Route pairs. Is it possible multiple of these double INVITEs could cause Asterisk or PJSIP on this older software to get into a bad state to cause the issues with AMI and Asterisk state? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on the RTP packet header
I am working on a project that uses Asterisk ARI ExternalMedia request to stream the RTP audio from Asterisk to an UDP/RTP receiver project. Using slin16 format. 1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is this correct? 2) Is there some place I can find a description of the 12-byte packet header fields? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some links on new docs asterisk org not working
Not sure where to mention this. Very minor/trivial issue. Just wanted to let someone know. If you go to docs.asterisk.org and the Asterisk REST Interface (at least in both 18 and 20 versions). Go to the Channels. There is a list of Method and path links. Most work, but a few do not. Not sure if the problem is the link should use all lower case or if the section should be a blend up upper/lower case. Originate with ID attempts to jump to originateWithId (should be originatewithid) Continue Dialplan attempts to jump to continueInDialplan (the link that works is continueindialplan) Play with playback id attempts to jump to playWithId (should be playwithid) The snoop attempts to jump to the snoopChannel (should be snoopchannel) Snoop with channel id attempts to jump to snoopChannelWithId (should be snoopchannelwithid) ExternalMedia attempts to go to externalMedia (link that works is externalmedia) Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the best way to disable rtp and jitter information from debugging
I don't recall seeing the rtp and jitter entries being logged regularly at other customer sites, so I am probably missing something obvious. This site is running asterisk 18.17.1. I enabled debugging to try to help track down an issue. The problem is the debug_log file is filling rapidly with entries like the following. [08/10 09:36:53.141] DEBUG[36475][C-009d] res_rtp_asterisk.c: 1691676998.426: pkt: 60711 Arrival sec: 1213.926 Arrival ts: 58268468 RX ts: 705913732 Transit samp: -647645264 Last transit samp: -647645218 d: 46 Curr jitter: \ 2( 0.000) Prev Jitter: 21( 0.000) New Jitter: 22( 0.000) [08/10 09:36:53.141] DEBUG[36475][C-009d] translate.c: Sample size different 960 vs 160 [08/10 09:36:53.146] DEBUG[35070][C-008e] res_rtp_asterisk.c: 1691674100.354: pkt: 205617 Arrival sec: 4112.079 Arrival ts: 197379770 RX ts: 1180840826 Transit samp: -983461056 Last transit samp: -983461053 d:3 Curr jitter: \ -2(89478.485) Prev Jitter: 30( 0.001) New Jitter: 29( 0.001) [08/10 09:36:53.146] DEBUG[35070][C-008e] translate.c: Sample size different 960 vs 160 [08/10 09:36:53.147] DEBUG[35341][C-0093] res_rtp_asterisk.c: 1691674549.397: pkt: 183110 Arrival sec: 3661.920 Arrival ts: 175772138 RX ts: 3440116713 Transit samp: 1030622721 Last transit samp: 1030623191 d: 470 Curr jitter: \ 6( 0.000) Prev Jitter: 374( 0.008) New Jitter: 380( 0.008) [08/10 09:36:53.148] DEBUG[35341][C-0093] translate.c: Sample size different 960 vs 160 [08/10 09:36:53.149] DEBUG[34847][C-0086] res_rtp_asterisk.c: 1691673680.332: pkt: 226561 Arrival sec: 4530.928 Arrival ts: 217484526 RX ts: 2864049486 Transit samp: 1648402336 Last transit samp: 1648402394 d: 58 Curr jitter: \ 0( 0.000) Prev Jitter: 53( 0.001) New Jitter: 53( 0.001) [08/10 09:36:53.149] DEBUG[34847][C-0086] translate.c: Sample size different 960 vs 160 I tried the following to disable these entries, but none of them stopped the entries in the log core set debug category off rtp core set debug category off rtcp rtcp set debug off rtp set debug off Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Encountered a crash, what is best way to tell if it has been fixed or now
I was able to put the crash through the gdb on the original VM that encountered the problem. (Not sure why the VM I initially tried to analyze the crash dump on didn’t do this correctly, but not concerned about it now). It’s providing additional details. Would this be considered a better example of the crash? I will go through the version comparisons and see if there are any updates since 18.17.1 to see if I can spot any fixes or recent commits. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x55e7c091ed95 in __ao2_ref (user_data=user_data@entry=0x1, delta=delta@entry=1, tag=tag@entry=0x0, file=file@entry=0x7f773800e012 "res_pjsip_session.c", line=line@entry=3639, func=func@entry=0x7f7738011d20 <__PRETTY_FUNCTION__.38105> "ast_sip_dialog_get_session") at astobj2.c:501 501 astobj2.c: No such file or directory. [Current thread is 1 (Thread 0x7f772c1a1700 (LWP 124120))] (gdb) bt #0 0x55e7c091ed95 in __ao2_ref (user_data=user_data@entry=0x1, delta=delta@entry=1, tag=tag@entry=0x0, file=file@entry=0x7f773800e012 "res_pjsip_session.c", line=line@entry=3639, func=func@entry=0x7f7738011d20 <__PRETTY_FUNCTION__.38105> "ast_sip_dialog_get_session") at astobj2.c:501 #1 0x7f773800a0da in ast_sip_dialog_get_session (dlg=dlg@entry=0x7f777415de48) at res_pjsip_session.c:3639 #2 0x7f773800d3e7 in session_outgoing_nat_hook (tdata=0x7f773c633ac8, transport=0x7f7754082048) at res_pjsip_session.c:5567 #3 0x7f773801965d in nat_invoke_hook (obj=, arg=arg@entry=0x7f772c1a0a50, flags=flags@entry=0) at res_pjsip_nat.c:299 #4 0x55e7c09218c0 in internal_ao2_traverse (self=self@entry=0x7f774014cc18, flags=flags@entry=OBJ_SEARCH_NONE, cb_fn=cb_fn@entry=0x7f7738019640 , arg=arg@entry=0x7f772c1a0a50, tag=tag@entry=0x0, file=file@entry=0x7f773801b009 "res_pjsip_nat.c", line=, func=, type=AO2_CALLBACK_DEFAULT, data=0x0) at astobj2_container.c:328 #5 0x55e7c0921d79 in __ao2_callback (c=c@entry=0x7f774014cc18, flags=flags@entry=OBJ_SEARCH_NONE, cb_fn=cb_fn@entry=0x7f7738019640 , arg=arg@entry=0x7f772c1a0a50, tag=tag@entry=0x0, file=file@entry=0x7f773801b009 "res_pjsip_nat.c", line=470, func=0x7f773801b4b8 <__PRETTY_FUNCTION__.29250> "process_nat") at astobj2_container.c:414 #6 0x7f7738019ddf in process_nat (tdata=0x7f773c633ac8) at res_pjsip_nat.c:470 #7 nat_on_tx_message (tdata=0x7f773c633ac8) at res_pjsip_nat.c:479 #8 0x7f777bc6fc66 in endpt_on_tx_msg (endpt=, tdata=0x7f773c633ac8) at ../src/pjsip/sip_endpoint.c:1115 #9 0x7f777bc77c69 in pjsip_transport_send (tr=0x55e7c1ac7708, tdata=tdata@entry=0x7f773c633ac8, addr=addr@entry=0x7f773c633cb8, addr_len=addr_len@entry=16, token=token@entry=0x7f773c635950, cb=cb@entry=0x7f777bc71610 ) at ../src/pjsip/sip_transport.c:935 #10 0x7f777bc717d9 in stateless_send_transport_cb (token=token@entry=0x7f773c635950, tdata=tdata@entry=0x7f773c633ac8, sent=, sent@entry=-70002) at ../src/pjsip/sip_util.c:1276 #11 0x7f777bc71b3a in stateless_send_resolver_callback (status=, token=0x7f773c635950, addr=) at ../src/pjsip/sip_util.c:1377 #12 0x7f77380c3398 in sip_resolve_invoke_user_callback (data=0x7f773c4563c8) at res_pjsip/pjsip_resolver.c:206 #13 0x55e7c0a6f0e3 in ast_taskprocessor_execute (tps=tps@entry=0x7f775416ebc0) at taskprocessor.c:1302 #14 0x55e7c0a76d28 in execute_tasks (data=0x7f775416ebc0) at threadpool.c:1352 #15 0x55e7c0a6f0e3 in ast_taskprocessor_execute (tps=0x55e7c2697720) at taskprocessor.c:1302 #16 0x55e7c0a7760c in threadpool_execute (pool=0x55e7c269be10) at threadpool.c:367 #17 worker_active (worker=0x7f7770001740) at threadpool.c:1137 #18 worker_start (arg=arg@entry=0x7f7770001740) at threadpool.c:1056 #19 0x55e7c0a7f868 in dummy_start (data=) at utils.c:1574 #20 0x7f777b4ba609 in start_thread (arg=) at pthread_create.c:477 #21 0x7f777b23c133 in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:95 Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Wednesday, August 9, 2023 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Encountered a crash, what is best way to tell if it has been fixed or now On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp mailto:dcr...@amtelco.com>> wrote: I have a customer who just encountered a crash while running Asterisk 18.17.1 version. I’m trying to adapt to the changes so not sure where best to look or how to possibly report this. I started by going through https://github.com/asterisk/asterisk/compare/18.17.1...18.19.0 to see if any of the changes seemed to apply to code reported by the backtrace. Entirely possible I missed something, but I didn’t notice anything that applies. I do see a commit was done today to the res_pjsip_nat.c file, but not sure if that would apply to the issue. Any suggestions for where I shoul
Re: [asterisk-users] [External] Encountered a crash, what is best way to tell if it has been fixed or now
Thank you Joshua. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Wednesday, August 9, 2023 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Encountered a crash, what is best way to tell if it has been fixed or now On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp mailto:dcr...@amtelco.com>> wrote: I have a customer who just encountered a crash while running Asterisk 18.17.1 version. I’m trying to adapt to the changes so not sure where best to look or how to possibly report this. I started by going through https://github.com/asterisk/asterisk/compare/18.17.1...18.19.0 to see if any of the changes seemed to apply to code reported by the backtrace. Entirely possible I missed something, but I didn’t notice anything that applies. I do see a commit was done today to the res_pjsip_nat.c file, but not sure if that would apply to the issue. Any suggestions for where I should look or ask? That is how you generally look, by seeing the commits between the two versions, analyzing, and seeing if anything is relevant. Issues themselves are reported on Github. I can say already though that the backtrace is incomplete and doesn't show the full story of what happened, it may be optimized or something. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Encountered a crash, what is best way to tell if it has been fixed or now
I have a customer who just encountered a crash while running Asterisk 18.17.1 version. I'm trying to adapt to the changes so not sure where best to look or how to possibly report this. I started by going through https://github.com/asterisk/asterisk/compare/18.17.1...18.19.0 to see if any of the changes seemed to apply to code reported by the backtrace. Entirely possible I missed something, but I didn't notice anything that applies. I do see a commit was done today to the res_pjsip_nat.c file, but not sure if that would apply to the issue. Any suggestions for where I should look or ask? (gdb) bt #0 0x55e7c091ed95 in __ao2_ref (user_data=user_data@entry=0x1, delta=delta@entry=1, tag=tag@entry=0x0, file=file@entry=0x7f773800e012 "res_pjsip_session.c", line=line@entry=3639, func=func@entry=0x7f7738011d20 <__PRETTY_FUNCTION__.38105> "ast_sip_dialog_get_session") at astobj2.c:501 #1 0x7f773800a0da in ast_sip_dialog_get_session (dlg=dlg@entry=0x7f777415de48) at res_pjsip_session.c:3639 #2 0x7f773800d3e7 in session_outgoing_nat_hook (tdata=0x7f773c633ac8, transport=0x7f7754082048) at res_pjsip_session.c:5567 #3 0x7f773801965d in nat_invoke_hook (obj=, arg=arg@entry=0x7f772c1a0a50, flags=flags@entry=0) at res_pjsip_nat.c:299 #4 0x55e7c09218c0 in internal_ao2_traverse (self=self@entry=0x7f774014cc18, flags=flags@entry=OBJ_SEARCH_NONE, cb_fn=cb_fn@entry=0x7f7738019640 , arg=arg@entry=0x7f772c1a0a50, tag=tag@entry=0x0, file=file@entry=0x7f773801b009 "res_pjsip_nat.c", line=, func=, type=AO2_CALLBACK_DEFAULT, data=0x0) at astobj2_container.c:328 #5 0x55e7c0921d79 in __ao2_callback (c=c@entry=0x7f774014cc18, flags=flags@entry=OBJ_SEARCH_NONE, cb_fn=cb_fn@entry=0x7f7738019640 , arg=arg@entry=0x7f772c1a0a50, tag=tag@entry=0x0, file=file@entry=0x7f773801b009 "res_pjsip_nat.c", line=470, func=0x7f773801b4b8 <__PRETTY_FUNCTION__.29250> "process_nat") at astobj2_container.c:414 #6 0x7f7738019ddf in process_nat (tdata=0x7f773c633ac8) at res_pjsip_nat.c:470 #7 nat_on_tx_message (tdata=0x7f773c633ac8) at res_pjsip_nat.c:479 #8 0x7f777bc6fc66 in ?? () #9 0x7f777bc71610 in ?? () #10 0x7f773c633ac8 in ?? () #11 0x55e7c1ac7708 in ?? () #12 0x7f777bc77c69 in ?? () #13 0x in ?? () Have a good day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Thank you Joshua The networking guys did this change on a test box and their jaws dropped to the floor when it did exactly as you explained. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Tuesday, February 7, 2023 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage On Tue, Feb 7, 2023 at 11:18 AM Dan Cropp mailto:d...@amtelco.com>> wrote: Thank you Joshua. Going back to your idea of the ice_host_candidates. (Again, apologize for my ignorance on networking). Do I understand correctly? We could use this formula for systems that have no one accessing the (where 192.168.1.10 is the internal IP) and 1.2.3.4 is the NAT’s public IP for Asterisk? 192.168.1.10 => 1.2.3.4,include_local_address Using this, would we no longer need the stunaddr configured? You don't need the include_local_address option but otherwise yes. This will cause the ICE host candidates to be 1.2.3.4 instead of the local IP address 192.168.1.10 removing the need to use STUN to discover the public IP address. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Thank you Joshua. I’m not very good at networking. We have a couple people who manage that and they were the ones who configured the boxes to use an external STUN server for all 4 systems that experienced problems accessing this STUN server at the same time. When all the boxes encountered the problem, each one was stuck in the stun timeout. My understanding is 3 customers reported the problem and we rebooted the Asterisk VM. (They service engineer did not understand he could have forced an rtp.conf reload and it should have fixed the STUN issue). After each of these VMs were reset, everything worked. The fourth customer didn’t report an issue for 8 hours. (Agents who would have been busy apparently notified no one to the issue). When this VM was reset, it also started working again. If I understand correctly, this is a good indication the DNS sent a TTL longer than 8 hours. I was told they ran a test and the DNS server was indicating the TTL was 120 seconds (but this was 2 days after the issue). I wonder if the AWS DNS had a hiccup and sent a TTL of 24 hours, but then either self-corrected or was manually corrected for the TTL we require. Going back to your idea of the ice_host_candidates. (Again, apologize for my ignorance on networking). Do I understand correctly? We could use this formula for systems that have no one accessing the (where 192.168.1.10 is the internal IP) and 1.2.3.4 is the NAT’s public IP for Asterisk? 192.168.1.10 => 1.2.3.4,include_local_address Using this, would we no longer need the stunaddr configured? Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Monday, February 6, 2023 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage On Mon, Feb 6, 2023 at 6:05 PM Dan Cropp mailto:d...@amtelco.com>> wrote: A quick follow-up. Looking at other customers running 18.12.1 who reported problems at the exact same time with AWS issue described below. We are seeing similar behavior. For these systems, the third STUN failure occurs. We were able to answer the call because the SIP provider didn’t CANCEL the call. However, upstream from the service provider the calls were terminated. Resulting in a call from the SIP provider to Asterisk that’s live, but there is no caller so it appears to be dead air. Does the res_rtp_asterisk stunaddr DNS TTL expiration mentioned in change ID I7955a046293f913ba121bbd82153b04439e3465f require the dnsmgr.conf to be enabled? It doesn't use dnsmgr so it's not required to be enabled. If the TTL is long, or it's cached locally then it could stick around longer. Fundamentally though is there a reason you're using STUN in the first place? Can you not just configure the public IP address and not rely on an external STUN server? rtp.conf has ice_host_candidates specifically for situations like AWS. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
A quick follow-up. Looking at other customers running 18.12.1 who reported problems at the exact same time with AWS issue described below. We are seeing similar behavior. For these systems, the third STUN failure occurs. We were able to answer the call because the SIP provider didn't CANCEL the call. However, upstream from the service provider the calls were terminated. Resulting in a call from the SIP provider to Asterisk that's live, but there is no caller so it appears to be dead air. Does the res_rtp_asterisk stunaddr DNS TTL expiration mentioned in change ID I7955a046293f913ba121bbd82153b04439e3465f require the dnsmgr.conf to be enabled? Dan From: Dan Cropp Sent: Monday, February 6, 2023 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Asterisk rtp.conf stunaddr setting - what happens if there is an outage Over the weekend, we had several customers running at AWS. AWS had an outage during this time. This customer is running Asterisk 16.23.0 (which has the STUN timeout crash fix). >From what I have been told, other customers are running newer Asterisk 18.12.1 >but encountered similar issues. (I haven't had a chance to verify this) All these customers should be running PJSIP, but I haven't had a chance to verify. The logs show Asterisk was reporting problems communicating with the STUN address in the rtp.conf [02/04 00:15:03.812] NOTICE[5943] stun.c: Attempt 1 to send STUN request to 'x.x.x.x' timed out. [02/04 00:15:06.812] NOTICE[5943] stun.c: Attempt 2 to send STUN request to ''x.x.x.x ' timed out. [02/04 00:15:09.813] WARNING[5943] stun.c: Attempt 3 to send STUN request to 'x.x.x.x' timed out. Check that the server address is correct and reachable. Until Asterisk was reset, the same pattern kept happening. Asterisk received INVITEs Immediately sends the 100 Trying 7 seconds later, Asterisk receives a CANCEL from the SIP provider. Another half second later, Asterisk receives a second CANCEL A second later, Asterisk receives a third CANCEL After the third failed to send STUN request, Asterisk sends a 200 OK response for the CSeq CANCEL Followed by a 487 Request Terminated Then a second 200 OK response for the CANCEL CSeq Then a third 200 OK response for the CANCEL CSeq We have an AMI connection. At this point, we are seeing the Newchannel event for this channel. It immediately sends various events for the Channel, including the Event: Hangup indicating the channel is ended. 63 ms later, it receives an ACK which completes the Call-ID processing. This went on for over 8 hours. When they restarted the Asterisk box, everything was fine. I have been told, they had to restart each Asterisk we had running at AWS to resolve the failed to send to STUN error. No calls/channels would work until that was resolved. I wonder if the STUN address lookup happens only one time and AWS DNS may have modified something during this outage/recovery? Is there a recommendation on how to prevent this from happening? Any thoughts? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Over the weekend, we had several customers running at AWS. AWS had an outage during this time. This customer is running Asterisk 16.23.0 (which has the STUN timeout crash fix). >From what I have been told, other customers are running newer Asterisk 18.12.1 >but encountered similar issues. (I haven't had a chance to verify this) All these customers should be running PJSIP, but I haven't had a chance to verify. The logs show Asterisk was reporting problems communicating with the STUN address in the rtp.conf [02/04 00:15:03.812] NOTICE[5943] stun.c: Attempt 1 to send STUN request to 'x.x.x.x' timed out. [02/04 00:15:06.812] NOTICE[5943] stun.c: Attempt 2 to send STUN request to ''x.x.x.x ' timed out. [02/04 00:15:09.813] WARNING[5943] stun.c: Attempt 3 to send STUN request to 'x.x.x.x' timed out. Check that the server address is correct and reachable. Until Asterisk was reset, the same pattern kept happening. Asterisk received INVITEs Immediately sends the 100 Trying 7 seconds later, Asterisk receives a CANCEL from the SIP provider. Another half second later, Asterisk receives a second CANCEL A second later, Asterisk receives a third CANCEL After the third failed to send STUN request, Asterisk sends a 200 OK response for the CSeq CANCEL Followed by a 487 Request Terminated Then a second 200 OK response for the CANCEL CSeq Then a third 200 OK response for the CANCEL CSeq We have an AMI connection. At this point, we are seeing the Newchannel event for this channel. It immediately sends various events for the Channel, including the Event: Hangup indicating the channel is ended. 63 ms later, it receives an ACK which completes the Call-ID processing. This went on for over 8 hours. When they restarted the Asterisk box, everything was fine. I have been told, they had to restart each Asterisk we had running at AWS to resolve the failed to send to STUN error. No calls/channels would work until that was resolved. I wonder if the STUN address lookup happens only one time and AWS DNS may have modified something during this outage/recovery? Is there a recommendation on how to prevent this from happening? Any thoughts? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application
Thank you Joshua!
Retrieving the Application details is a great solution for this. I completely
forgot about that support.
On a reconnect…
Code and start monitoring for StasisStart events for new channel can control.
Also can perform a GET with it’s own Application for the details.
Then, it can use that information for the bridge ids and channel ids that it’s
responsible for.
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, January 30, 2023 5:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Is there a list of Channel ARI
requests that are allowed when the call is not handed off to the Stasis
application
On Mon, Jan 30, 2023 at 7:30 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We have used AMI for many years and I’m in the process of migrating to ARI.
My understanding is the call should be handed off to Stasis for the ARI
application to control it.
I was playing around with things and discovered the ARI hangup (DELETE
/channels/{channelId}) allowed me to hangup calls even when no StasisStart is
received.
I tried some other requests and they did not seem to work. This is what I
expected to happen for the hangup.
Are there other commands that are allowed on channels when the call is not in
the Stasis app? (Obviously creating a channel and externalMedia will work
because they create new channels).
There's not really a list, some just work due to the internal way they work in
Asterisk.
Also, to be fault tolerant, I noticed a call handed off to Stasis app will
remain in the Stasis app, even if the ARI/WebSocket connection drops (power
outage, etc). When establishing the ARI/WebSocket connection, the first thing
I am planning to do is GET a list of the channels. This returns all of the
channels in the system and not just the channels that are in this Stasis apps
control. I plan to go through the list and identify the channels dialplan
data. Look for app_name of Stasis and the app_data (comma-delimited).
If app_name = “Stasis” and app_data’s first section of the comma-delimited
parse portion matches the Stasis app name this instance is used, I take control
of this channel.
I am planning this additional check because I noticed the Stasis power outage
scenario resulted in channels stuck in the Stasis app. If I don’t take control
of these channels, it’s possible to eventually have hundreds/thousands of
channels. For SIP calls, the other end eventually hangs up. However, this
isn’t the case with Local channels. Particularly when both ends are locally
controlled by Stasis.
Does this sound like I am on the right track for migrating from AMI to Stasis,
ARI/Websocket support?
You may be able to get the application details[1][2] which would tell you what
the application is subscribed to, which would include the channels.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Applications+REST+API#Asterisk20ApplicationsRESTAPI-get
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Application
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com<http://www.sangoma.com> and
www.asterisk.org<http://www.asterisk.org>
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[asterisk-users] Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application
We have used AMI for many years and I'm in the process of migrating to ARI.
My understanding is the call should be handed off to Stasis for the ARI
application to control it.
I was playing around with things and discovered the ARI hangup (DELETE
/channels/{channelId}) allowed me to hangup calls even when no StasisStart is
received.
I tried some other requests and they did not seem to work. This is what I
expected to happen for the hangup.
Are there other commands that are allowed on channels when the call is not in
the Stasis app? (Obviously creating a channel and externalMedia will work
because they create new channels).
Also, to be fault tolerant, I noticed a call handed off to Stasis app will
remain in the Stasis app, even if the ARI/WebSocket connection drops (power
outage, etc). When establishing the ARI/WebSocket connection, the first thing
I am planning to do is GET a list of the channels. This returns all of the
channels in the system and not just the channels that are in this Stasis apps
control. I plan to go through the list and identify the channels dialplan
data. Look for app_name of Stasis and the app_data (comma-delimited).
If app_name = "Stasis" and app_data's first section of the comma-delimited
parse portion matches the Stasis app name this instance is used, I take control
of this channel.
I am planning this additional check because I noticed the Stasis power outage
scenario resulted in channels stuck in the Stasis app. If I don't take control
of these channels, it's possible to eventually have hundreds/thousands of
channels. For SIP calls, the other end eventually hangs up. However, this
isn't the case with Local channels. Particularly when both ends are locally
controlled by Stasis.
Does this sound like I am on the right track for migrating from AMI to Stasis,
ARI/Websocket support?
Dan
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Re: [asterisk-users] Question on ARI externalMedia
Please disregard, I figured out what I was doing wrong.
Dan
From: Dan Cropp
Sent: Friday, January 20, 2023 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Question on ARI externalMedia
A couple years ago, I know I had ARI externalMedia working. Trying to figure
out what I'm doing wrong today.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
My ari.conf
[general]
enabled = yes
pretty = no
allowed_origins = *
[MyApp]
type = user
read_only = no
password_format = plain
password = Password
I send this curl -v -u MyApp:Password -X POST
"http://localhost:8088/ari/channels/externalMedia?channelId=1234abcd5678=MyApp_host=192.168.33.32%3A1053=slin16;
I can make other ARI commands work, so it must be something specific to my
externalMedia command and the parameters.
The output is the following...
{"id":"1234abcd5678","name":"UnicastRTP/192.168.33.32:1053-0x7fcffc020300","state":"Down","protocol_id":"","caller":{"name":"","number":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"default","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-01-20T10:59:24.569-0600","language":"en","channelvars":{"UNICASTRTP_LOCAL_PORT":"19194","UNICASTRTP_LOCAL_ADDRESS":"192.168.33.31"}}
Dan
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[asterisk-users] Question on ARI externalMedia
A couple years ago, I know I had ARI externalMedia working. Trying to figure
out what I'm doing wrong today.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
My ari.conf
[general]
enabled = yes
pretty = no
allowed_origins = *
[MyApp]
type = user
read_only = no
password_format = plain
password = Password
I send this curl -v -u MyApp:Password -X POST
"http://localhost:8088/ari/channels/externalMedia?channelId=1234abcd5678=MyApp_host=192.168.33.32%3A1053=slin16;
I can make other ARI commands work, so it must be something specific to my
externalMedia command and the parameters.
The output is the following...
{"id":"1234abcd5678","name":"UnicastRTP/192.168.33.32:1053-0x7fcffc020300","state":"Down","protocol_id":"","caller":{"name":"","number":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"default","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-01-20T10:59:24.569-0600","language":"en","channelvars":{"UNICASTRTP_LOCAL_PORT":"19194","UNICASTRTP_LOCAL_ADDRESS":"192.168.33.31"}}
Dan
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Re: [asterisk-users] [External] monitor files gsm format split
We use the sox (SoX - Sound eXchange) package to perform many audio manipulation routines our customers require. Dan -Original Message- From: asterisk-users On Behalf Of astuserl...@mytelpbx.com Sent: Friday, December 9, 2022 5:53 AM To: asterisk-users@lists.digium.com Subject: [External] [asterisk-users] monitor files gsm format split Hi, I would like to split some monitor files in gsm format to smaller audio files and look for an program running on debian. Did somebody knows a program on cpnsole, easy to use. Bye Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of
Thank you Joshua. I am going to dig some more. I can’t see anything in the logs to indicate any reason for why it’s suddenly able to send the OPTIONS response. There are zero SIP packets from Kamailio between the one that fails to send and the one that succeeds. Very strange. If I can’t get it working, I will create an issue report. Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Wednesday, December 7, 2022 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time) On Wed, Dec 7, 2022 at 11:34 AM Joshua C. Colp mailto:jc...@sangoma.com>> wrote: On Wed, Dec 7, 2022 at 11:26 AM Dan Cropp mailto:d...@amtelco.com>> wrote: On two VMs, we encounter a strange behavior when we upgrade from 18.12.1 to 18.15.0 (also tried 18.15.1 last night). When we roll the VMs back to 18.12.1, we don’t see the behavior repeat. We have a Kamailio VM front ending the asterisk. It sends OPTIONS messages periodically. After startup (and also after reloading configuration settings), we see periods where the response can’t be sent. After a period of time, it suddenly starts working. I haven't seen this before, and haven't seen any other reports of it so far. The OPTIONS code itself hasn't changed between the two. There was a fix for a crash in send_stateful_response so adding log messages to the error cases is likely needed to see in particular which one is failing. Ha, those changes haven't even landed yet. It's pretty much a thin wrapper over PJSIP stuff in 18.15.1. The PJSIP versions are also fairly close too. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
ATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Accept-Encoding: identity Accept-Language: en Server: Asterisk PBX 18.15.1 Content-Length: 0 Portion of the pjsip.conf settings... [Kamailio] type = aor authenticate_qualify = yes contact = sip:192.168.10.235 ;outbound_proxy=sip:192.168.12.10 [identify158] type = identify endpoint = Kamailio match = 192.168.12.10 [Kamailio] type = endpoint context = IS transport = transport1 aors = Kamailio accountcode = 29 dtmf_mode = inband device_state_busy_at = 48 force_rport = no moh_passthrough = no identify_by = username,ip,header disallow = all allow = ulaw asymmetric_rtp_codec = yes acl = acl6 outbound_proxy=sip:192.168.12.10 Was there some configuration change introduced after 18.12.1 that I missed? Any thoughts? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What conditions require the AMI_VERSION number to be bumped?
Asking because I see there is a new DeadlockStart event added to 18.15.0 but the AMI_VERSION value is still 7.0.2 Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Question on Originate with EarlyMedia
Thank you. May be a nice feature in the future to send an indication event when the call is answered. The reason is if call A needs to be transferred to destination. When calling the destination B (originate the call). Some companies/departments play inband recordings instead of ringing. We have a customer who plays “Your call will be recorded for legal purposes”. Because call A needs to hear that message, we currently place both calls in a ConfBridge. Call A now hears the recording. If/when the call is really answered, we want to remove the calls from the ConfBridge and transfer A to B. (Successful transfer removes call from Asterisk and patches them internally in their switch NEC, Cisco, Avaya, etc) Currently, we require an agent to listen into the ConfBridge and decide when they believe it’s answered. Agent kicks everyone from the ConfBridge and the REFER (transfer) is sent. On some occasions the agent is incorrect or makes a mistake in doing this too soon, so the transfer fails. At that point, add everyone back into the ConfBridge, waiting for the agent to decide when to retry. Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Thursday, September 1, 2022 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Question on Originate with EarlyMedia On Thu, Sep 1, 2022 at 1:32 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Using AMI, we send an Originate with EarlyMedia: true setting If the other end sends a 183, Asterisk When the 183 is received, Asterisk indicates the ChannelState: 6 and ChannelStateDesc: Up values. All is fine up to this point. It may take the caller several seconds before the called party answers. When the called party answers (200 OK received), in the debugging I see Asterisk processing this and debugging show TSX State: Terminated Inv State: EARLY At this point, the call is truly connected. Is there a configuration setting to indicate whether Asterisk should send an event indicating when the early media ends and the call is really Up? There is no option. The Originate code makes the channel appear as answered when the 183 arrives, everything reflects that afterwards. Even if a second answer occurs it gets ignored. The log message you refer to is internal state information to do with the SIP side. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on Originate with EarlyMedia
Using AMI, we send an Originate with EarlyMedia: true setting If the other end sends a 183, Asterisk When the 183 is received, Asterisk indicates the ChannelState: 6 and ChannelStateDesc: Up values. All is fine up to this point. It may take the caller several seconds before the called party answers. When the called party answers (200 OK received), in the debugging I see Asterisk processing this and debugging show TSX State: Terminated Inv State: EARLY At this point, the call is truly connected. Is there a configuration setting to indicate whether Asterisk should send an event indicating when the early media ends and the call is really Up? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] I think there may be a bug in 18.14.0 ${GEOLOC_PROFILE(profile_precedence)}, seems to always return prefer_incoming
Thank you for the explanation George.
This makes it very easy to know if the Geo Location is the configured or an
incoming value.
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Thursday, August 25, 2022 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] I think there may be a bug in 18.14.0
${GEOLOC_PROFILE(profile_precedence)}, seems to always return prefer_incoming
On Wed, Aug 24, 2022 at 7:51 AM George Joseph
mailto:gjos...@sangoma.com>> wrote:
Yeah, that's weird. I opened an issue for this...
https://issues.asterisk.org/jira/browse/ASTERISK-30190
OK, It's actually not weird :)
Let's say the configured profile is set to discard_config or prefer_incoming
and there actually is an incoming profile. In this situation, by the time you
reach the dialplan, we've already discarded the configured profile in favor of
the incoming one so profile_precedence is going to be what's on the incoming
one which will always be prefer_incoming. Is that going to be an issue?
BTW, there still is a bug where effective_location will be blank in this same
situation and there are patches up on Gerrit that fix that and a few other bugs.
On Tue, Aug 23, 2022 at 2:47 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Running into a problem when retrieving the profile_precedence in the
extensions.conf
Creating a very basic geolocation.conf to allow passing through geolocation
values for outbound.
[discard_config]
type = profile
profile_precedence = discard_config
[discard_incoming]
type = profile
profile_precedence = discard_incoming
[prefer_config]
type = profile
profile_precedence = prefer_config
[prefer_incoming]
type = profile
profile_precedence = prefer_incoming
I have tried setting the pjsip.conf geoloc_incoming_call_profile to all four of
these profiles for inbound call testing. The discard_incoming correctly blocks
the geo location information. Other 3 pass the geo location values through
[192.168.33.31]
type = endpoint
context = IS
transport = transport1
aors = 192.168.33.31
accountcode = 20
dtmf_mode = inband
device_state_busy_at = 1600
moh_passthrough = no
identify_by = username,ip,header
disallow = all
allow = ulaw
acl = acl1
geoloc_incoming_call_profile = prefer_config
geoloc_outgoing_call_profile = prefer_config
When I have the following line in the extensions.conf, it’s retrieving the
GEOLOC_PROFILE(profile_precedence) to the variable, but it’s being set to
prefer_incoming even when it should be discard_config or prefer_config.
same => n,Set(MY__GEO_PROFILE_PRECEDENCE=${GEOLOC_PROFILE(profile_precedence)})
Dan
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[asterisk-users] I think there may be a bug in 18.14.0 ${GEOLOC_PROFILE(profile_precedence)}, seems to always return prefer_incoming
Running into a problem when retrieving the profile_precedence in the
extensions.conf
Creating a very basic geolocation.conf to allow passing through geolocation
values for outbound.
[discard_config]
type = profile
profile_precedence = discard_config
[discard_incoming]
type = profile
profile_precedence = discard_incoming
[prefer_config]
type = profile
profile_precedence = prefer_config
[prefer_incoming]
type = profile
profile_precedence = prefer_incoming
I have tried setting the pjsip.conf geoloc_incoming_call_profile to all four of
these profiles for inbound call testing. The discard_incoming correctly blocks
the geo location information. Other 3 pass the geo location values through
[192.168.33.31]
type = endpoint
context = IS
transport = transport1
aors = 192.168.33.31
accountcode = 20
dtmf_mode = inband
device_state_busy_at = 1600
moh_passthrough = no
identify_by = username,ip,header
disallow = all
allow = ulaw
acl = acl1
geoloc_incoming_call_profile = prefer_config
geoloc_outgoing_call_profile = prefer_config
When I have the following line in the extensions.conf, it's retrieving the
GEOLOC_PROFILE(profile_precedence) to the variable, but it's being set to
prefer_incoming even when it should be discard_config or prefer_config.
same => n,Set(MY__GEO_PROFILE_PRECEDENCE=${GEOLOC_PROFILE(profile_precedence)})
Dan
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Re: [asterisk-users] [External] [External] Geo location 18.14.0-rc1 question
Thank you George. As you pointed out, my mistake of the double equal sign caused the problem. Using the passthrough profile and in the AMI Originate setting the Variable: GEOLOC_PROFILE(name) is exactly what we need. My software will receive the GEO settings from third party software. If third party passed a field/value that doesn’t match the Asterisk defaults, our software will add the GEOLOC_PROFILE(name) to the Originate Variable field. Then I send the Originate packet to Asterisk via AMI. Thank you for all your work on this!!! Dan From: asterisk-users On Behalf Of George Joseph Sent: Tuesday, August 16, 2022 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] Geo location 18.14.0-rc1 question On Mon, Aug 15, 2022 at 1:59 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Thank you George. Good idea on the passthrough profile. Is there a way to set the GEOLOC_PROFILE values from the AMI Originate command? I tried the following, but it doesn’t like the GEOLOC_PROFILE values in the variable parameter. If there is a way to do this, the passthrough would solve the problem of Geo Location information settings needing to be provided by a third-party application. Action: Originate Channel: PJSIP/1234@192.168.33.31<mailto:1234@192.168.33.31> Exten: createcall Context: mycontext Priority: 1 Timeout: 6 CallerID: John Smith <8005551234> Variable: GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(method)=Manual,GEOLOC_PROFILE(location_info)=="country=US,A1=Florida,A3=Orlando,HNO=100,RD=Main,STS=Street",CALLERID(num-pres)=allowed_passed_screen Async: true You've got 2 equals signs when you set location_info :). I just tried GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(location_info)="country=US,A3=\"New York\"",GEOLOC_PROFILE(pidf_element)=device and it worked. I believe this portion believe indicates Asterisk treats the GEOLOC_PROFILE(xxx) as GEOLOCPROFILESTATUS variable name. GEOLOCPROFILESTATUS is the variable GEOLOC_PROFILE sets to indicate success or failure. The value of "0" indicates success. What was the actual result in the channel? [08/15 13:41:41.609] DEBUG[42424] manager.c: Examining AMI event: Event: VarSet^M Privilege: dialplan,all^M Channel: PJSIP/192.168.33.31-^M ChannelState: 0^M ChannelStateDesc: Down^M CallerIDNum: ^M CallerIDName: ^M ConnectedLineNum: ^M ConnectedLineName: ^M Language: en^M AccountCode: 20^M Context: IS^M Exten: s^M Priority: 1^M Uniqueid: 1660588901.0^M Linkedid: 1660588901.0^M Variable: GEOLOCPROFILESTATUS^M Value: 0^M ^M Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question
George, Is it possible to set the GEOLOC_PROFILE fields similar to the way PJSIP_HEADER(add, …) works? Asking, because we already use the PJSIP_HEADER(add, xxx) from AMI Originate to add PJSIP headers to the outbound originate. Action: Originate ActionID: S62 Channel: PJSIP/1234@192.168.12.34 Exten: createcall Context: mycontext Priority: 1 Timeout: 6 CallerID: John Smith <8005551212> Variable: PJSIP_HEADER(add,abc)=123,CALLERID(num-pres)=allowed_passed_screen Async: true Codecs: ulaw Dan From: Dan Cropp Sent: Monday, August 15, 2022 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [External] [asterisk-users] Geo location 18.14.0-rc1 question Thank you George. Good idea on the passthrough profile. Is there a way to set the GEOLOC_PROFILE values from the AMI Originate command? I tried the following, but it doesn’t like the GEOLOC_PROFILE values in the variable parameter. If there is a way to do this, the passthrough would solve the problem of Geo Location information settings needing to be provided by a third-party application. Action: Originate Channel: PJSIP/1234@192.168.33.31<mailto:PJSIP/1234@192.168.33.31> Exten: createcall Context: mycontext Priority: 1 Timeout: 6 CallerID: John Smith <8005551234> Variable: GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(method)=Manual,GEOLOC_PROFILE(location_info)=="country=US,A1=Florida,A3=Orlando,HNO=100,RD=Main,STS=Street",CALLERID(num-pres)=allowed_passed_screen Async: true I believe this portion believe indicates Asterisk treats the GEOLOC_PROFILE(xxx) as GEOLOCPROFILESTATUS variable name. [08/15 13:41:41.609] DEBUG[42424] manager.c: Examining AMI event: Event: VarSet^M Privilege: dialplan,all^M Channel: PJSIP/192.168.33.31-^M ChannelState: 0^M ChannelStateDesc: Down^M CallerIDNum: ^M CallerIDName: ^M ConnectedLineNum: ^M ConnectedLineName: ^M Language: en^M AccountCode: 20^M Context: IS^M Exten: s^M Priority: 1^M Uniqueid: 1660588901.0^M Linkedid: 1660588901.0^M Variable: GEOLOCPROFILESTATUS^M Value: 0^M ^M Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question
Thank you George. Good idea on the passthrough profile. Is there a way to set the GEOLOC_PROFILE values from the AMI Originate command? I tried the following, but it doesn’t like the GEOLOC_PROFILE values in the variable parameter. If there is a way to do this, the passthrough would solve the problem of Geo Location information settings needing to be provided by a third-party application. Action: Originate Channel: PJSIP/1234@192.168.33.31 Exten: createcall Context: mycontext Priority: 1 Timeout: 6 CallerID: John Smith <8005551234> Variable: GEOLOC_PROFILE(format)=civicAddress,GEOLOC_PROFILE(method)=Manual,GEOLOC_PROFILE(location_info)=="country=US,A1=Florida,A3=Orlando,HNO=100,RD=Main,STS=Street",CALLERID(num-pres)=allowed_passed_screen Async: true I believe this portion believe indicates Asterisk treats the GEOLOC_PROFILE(xxx) as GEOLOCPROFILESTATUS variable name. [08/15 13:41:41.609] DEBUG[42424] manager.c: Examining AMI event: Event: VarSet^M Privilege: dialplan,all^M Channel: PJSIP/192.168.33.31-^M ChannelState: 0^M ChannelStateDesc: Down^M CallerIDNum: ^M CallerIDName: ^M ConnectedLineNum: ^M ConnectedLineName: ^M Language: en^M AccountCode: 20^M Context: IS^M Exten: s^M Priority: 1^M Uniqueid: 1660588901.0^M Linkedid: 1660588901.0^M Variable: GEOLOCPROFILESTATUS^M Value: 0^M ^M Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] [External] [External] [External] Geo location 18.14.0-rc1 question
Thank you Joshua
rc2 resolved the issue I was seeing.
However, it sounds like it would be best for me to configure the location_info
without the leading underscore for the variable name.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Friday, August 12, 2022 8:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] [External] [External]
[External] Geo location 18.14.0-rc1 question
On Fri, Aug 12, 2022 at 10:01 AM George Joseph
mailto:gjos...@sangoma.com>> wrote:
On Thu, Aug 11, 2022 at 8:43 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thank you George.
I am still running on asterisk 18.14.0-rc1 and have not retrieved the patches
yet.
Did this version have a bug with the variables?
It's quite possible. RC2 was just released so you should try that.
I’m trying the location_info and variables in the AMI Originate you recommended
at the end of the previous e-mail.
In case it’s not coming through correctly via e-mail, the variable names are
preceeded with a single underscore in the AMI and in the location_info values.
[IS_loc_5]
type = location
format = civicAddress
location_info = country=${_MY_GEO_COUNTRY}
location_info = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
location_info = A2=${_MY_GEO_NATSUB}
location_info = A3=${_MY_GEO_CITY}
location_info = HNO=${_MY_GEO_HNO}
location_info = RD=${_MY_GEO_RD}
location_info = STS=${_MY_GEO_STS}
location_info = PC=${_MY_GEO_PC}
Underscore refers to variable inheritance. For usage you don't include the
underscore. I don't think the code strips out the underscore when substituting
or retrieving, so it could very well end up empty during that process.
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com<http://www.sangoma.com> and
www.asterisk.org<http://www.asterisk.org>
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Re: [asterisk-users] [External] [External] [External] Geo location 18.14.0-rc1 question
Thank you George.
Looking forward to working with the changes. I will retrieve them when the
next release candidate comes out.
A quick question on using variables to pass custom Geo Location settings on via
an AMI Originate.
If my AMI originate request looks something like this…
Action: Originate
Channel: PJSIP/1234@192.168.x.x
Exten: createcall
Context: mycontext
Priority: 1
Timeout: 6
CallerID: John Smith <8005551234>
Variable:
_MY_GEO_COUNTRY=US,_MY_GEO_NATSUB=Florida,_MY_GEO_CITY=Orlando,_MY_GEO_HNO=100,_MY_GEO_RD=Main,_MY_GEO_STS=Street
Async: true
Do I need to program the location_variables in the profile like this?
[1]
type = profile
pidf_element = device
profile_action = discard_incoming
usage_rules = retransmission_allowed=yes
location_variables = country=${_MY_GEO_COUNTRY}
location_variables = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
location_variables = A2=${_MY_GEO_NATSUB}
location_variables = A3=${_MY_GEO_CITY}
location_variables = HNO=${_MY_GEO_HNO}
location_variables = RD=${_MY_GEO_RD}
location_variables = STS=${_MY_GEO_STS}
location_variables = PC=${_MY_GEO_PC}
Or would I need to program the location_info_refinements in the profile to use
those variables?
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Wednesday, August 10, 2022 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] [External] Geo location
18.14.0-rc1 question
Sorry for the delay but this turned out to be a bit more complex than I
anticipated.
There are reviews up on Gerrit for the 16 and 18 branches that address the
issues below as well as clean up the implementation, plug some memory leaks,
etc.
16: https://gerrit.asterisk.org/c/asterisk/+/18896
18: https://gerrit.asterisk.org/c/asterisk/+/18897
I anticipate these will make it into the next set of release candidates which
are due to be cut tomorrow.
Give them a try.
On Wed, Aug 3, 2022 at 1:51 PM George Joseph
mailto:gjos...@sangoma.com>> wrote:
Looks like it'll be tomorrow before I can get the patch up. I ran into some
strange issues.
On Tue, Aug 2, 2022 at 1:43 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thank you George
From: asterisk-users
mailto:asterisk-users-boun...@lists.digium.com>>
On Behalf Of George Joseph
Sent: Tuesday, August 2, 2022 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [External] [asterisk-users] [External] Geo location 18.14.0-rc1
question
On Tue, Aug 2, 2022 at 1:35 PM George Joseph
mailto:gjos...@sangoma.com>> wrote:
On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Is the allow_routing setting on the geolocation Wiki Profile also not fully
implemented?
Well, 99% of the code is there. The 1% is parsing the config option. Not sure
how I missed that.
I'll have a patch up first thing in the morning UTC-6.
I'll call it "allow_use_for_routing" in profile.
Actually just "allow_routing_use"
In the code, I see geolocation_routing used instead of allow_routing.
Tried both and Asterisk indicates it cannot find suitable setting so it doesn’t
create the profile object.
Dan
From: Dan Cropp
Sent: Tuesday, August 2, 2022 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Subject: RE: [External] [asterisk-users] Geo location 18.14.0-rc1 question
Thank you George.
From: asterisk-users
mailto:asterisk-users-boun...@lists.digium.com>>
On Behalf Of George Joseph
Sent: Tuesday, August 2, 2022 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [External] [asterisk-users] Geo location 18.14.0-rc1 question
On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
I believe I have everything configured correctly, but Asterisk is complaining
about my configuration
It is complaining about confidence settings.
From the Asterisk Geolocation Implementation Wiki, I believe I have this set
correctly.
Sub-parameters:
* value: A percentage indicating the confidence or "unknown".
* pdf: "unknown", "normal" or "rectangular"
Example: confidence = value=80, pdf=unknown
If no confidence parameter is specified, the default is 95%.
See
RFC7459<https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459>
for the exact definition of this parameter.
[08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
suitable for category 'IS_loc_1' named 'confidence' at line 12 of
[08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an
object of type 'location' with id 'IS_loc_1' from configuration file
'geolocation.conf'
[IS_loc_1]
type = location
format = civicAddress
Re: [asterisk-users] [External] [External] Geo location 18.14.0-rc1 question
Thank you George From: asterisk-users On Behalf Of George Joseph Sent: Tuesday, August 2, 2022 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] Geo location 18.14.0-rc1 question On Tue, Aug 2, 2022 at 1:35 PM George Joseph mailto:gjos...@sangoma.com>> wrote: On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Is the allow_routing setting on the geolocation Wiki Profile also not fully implemented? Well, 99% of the code is there. The 1% is parsing the config option. Not sure how I missed that. I'll have a patch up first thing in the morning UTC-6. I'll call it "allow_use_for_routing" in profile. Actually just "allow_routing_use" In the code, I see geolocation_routing used instead of allow_routing. Tried both and Asterisk indicates it cannot find suitable setting so it doesn’t create the profile object. Dan From: Dan Cropp Sent: Tuesday, August 2, 2022 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: RE: [External] [asterisk-users] Geo location 18.14.0-rc1 question Thank you George. From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of George Joseph Sent: Tuesday, August 2, 2022 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [External] [asterisk-users] Geo location 18.14.0-rc1 question On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp mailto:d...@amtelco.com>> wrote: I believe I have everything configured correctly, but Asterisk is complaining about my configuration It is complaining about confidence settings. From the Asterisk Geolocation Implementation Wiki, I believe I have this set correctly. Sub-parameters: * value: A percentage indicating the confidence or "unknown". * pdf: "unknown", "normal" or "rectangular" Example: confidence = value=80, pdf=unknown If no confidence parameter is specified, the default is 95%. See RFC7459<https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459> for the exact definition of this parameter. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_loc_1' named 'confidence' at line 12 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'location' with id 'IS_loc_1' from configuration file 'geolocation.conf' [IS_loc_1] type = location format = civicAddress confidence = value=95, pdf=unknown location_info = country=US,A1=Wisconsin,A3=Madison location_info = HNO=4800,RD=Main,STS=Drive,PC=53704 Remove the confidence param for now.I documented it before I implemented it. :) Also seeing problems with location_refinement setting. Again, I believe my setting matches what is on the Asterisk Geolocation Implementation wiki. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'profile' with id 'IS_prof_20' from configuration file 'geolocation.conf' [IS_prof_20] type = profile profile_action = prefer_incoming pidf_element = person usage_rules = retransmission_allowed=no location_reference = IS_loc_22 location_refinement = ROOM=292 location_refinement = FLR=1 Pffft. I renamed this to "location_info_refinement" to better match the "location_info" parameter in the Location object. I forgot to rename it in the wiki documentation. If you just change the name it should work. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start he
Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question
Is the allow_routing setting on the geolocation Wiki Profile also not fully implemented? In the code, I see geolocation_routing used instead of allow_routing. Tried both and Asterisk indicates it cannot find suitable setting so it doesn’t create the profile object. Dan From: Dan Cropp Sent: Tuesday, August 2, 2022 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [External] [asterisk-users] Geo location 18.14.0-rc1 question Thank you George. From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of George Joseph Sent: Tuesday, August 2, 2022 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [External] [asterisk-users] Geo location 18.14.0-rc1 question On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp mailto:d...@amtelco.com>> wrote: I believe I have everything configured correctly, but Asterisk is complaining about my configuration It is complaining about confidence settings. From the Asterisk Geolocation Implementation Wiki, I believe I have this set correctly. Sub-parameters: * value: A percentage indicating the confidence or "unknown". * pdf: "unknown", "normal" or "rectangular" Example: confidence = value=80, pdf=unknown If no confidence parameter is specified, the default is 95%. See RFC7459<https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459> for the exact definition of this parameter. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_loc_1' named 'confidence' at line 12 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'location' with id 'IS_loc_1' from configuration file 'geolocation.conf' [IS_loc_1] type = location format = civicAddress confidence = value=95, pdf=unknown location_info = country=US,A1=Wisconsin,A3=Madison location_info = HNO=4800,RD=Main,STS=Drive,PC=53704 Remove the confidence param for now.I documented it before I implemented it. :) Also seeing problems with location_refinement setting. Again, I believe my setting matches what is on the Asterisk Geolocation Implementation wiki. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'profile' with id 'IS_prof_20' from configuration file 'geolocation.conf' [IS_prof_20] type = profile profile_action = prefer_incoming pidf_element = person usage_rules = retransmission_allowed=no location_reference = IS_loc_22 location_refinement = ROOM=292 location_refinement = FLR=1 Pffft. I renamed this to "location_info_refinement" to better match the "location_info" parameter in the Location object. I forgot to rename it in the wiki documentation. If you just change the name it should work. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Geo location 18.14.0-rc1 question
Thank you George. From: asterisk-users On Behalf Of George Joseph Sent: Tuesday, August 2, 2022 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Geo location 18.14.0-rc1 question On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp mailto:d...@amtelco.com>> wrote: I believe I have everything configured correctly, but Asterisk is complaining about my configuration It is complaining about confidence settings. From the Asterisk Geolocation Implementation Wiki, I believe I have this set correctly. Sub-parameters: * value: A percentage indicating the confidence or "unknown". * pdf: "unknown", "normal" or "rectangular" Example: confidence = value=80, pdf=unknown If no confidence parameter is specified, the default is 95%. See RFC7459<https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459> for the exact definition of this parameter. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_loc_1' named 'confidence' at line 12 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'location' with id 'IS_loc_1' from configuration file 'geolocation.conf' [IS_loc_1] type = location format = civicAddress confidence = value=95, pdf=unknown location_info = country=US,A1=Wisconsin,A3=Madison location_info = HNO=4800,RD=Main,STS=Drive,PC=53704 Remove the confidence param for now.I documented it before I implemented it. :) Also seeing problems with location_refinement setting. Again, I believe my setting matches what is on the Asterisk Geolocation Implementation wiki. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'profile' with id 'IS_prof_20' from configuration file 'geolocation.conf' [IS_prof_20] type = profile profile_action = prefer_incoming pidf_element = person usage_rules = retransmission_allowed=no location_reference = IS_loc_22 location_refinement = ROOM=292 location_refinement = FLR=1 Pffft. I renamed this to "location_info_refinement" to better match the "location_info" parameter in the Location object. I forgot to rename it in the wiki documentation. If you just change the name it should work. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Geo location 18.14.0-rc1 question
I believe I have everything configured correctly, but Asterisk is complaining about my configuration It is complaining about confidence settings. >From the Asterisk Geolocation Implementation Wiki, I believe I have this set >correctly. Sub-parameters: * value: A percentage indicating the confidence or "unknown". * pdf: "unknown", "normal" or "rectangular" Example: confidence = value=80, pdf=unknown If no confidence parameter is specified, the default is 95%. See RFC7459<https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459> for the exact definition of this parameter. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_loc_1' named 'confidence' at line 12 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'location' with id 'IS_loc_1' from configuration file 'geolocation.conf' [IS_loc_1] type = location format = civicAddress confidence = value=95, pdf=unknown location_info = country=US,A1=Wisconsin,A3=Madison location_info = HNO=4800,RD=Main,STS=Drive,PC=53704 Also seeing problems with location_refinement setting. Again, I believe my setting matches what is on the Asterisk Geolocation Implementation wiki. [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create an object of type 'profile' with id 'IS_prof_20' from configuration file 'geolocation.conf' [IS_prof_20] type = profile profile_action = prefer_incoming pidf_element = person usage_rules = retransmission_allowed=no location_reference = IS_loc_22 location_refinement = ROOM=292 location_refinement = FLR=1 Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Question about the Geo Location support being added
Thank you George.
Also, thank you and everyone else involved for the work adding Geo Location in
Asterisk.
Looking forward to the update.
In our case, all the agent messaging and several other features are handled by
a separate server.
This external server would be responsible for deciding which GEO Location Info
to use for the Originate.
* Agent has their own emergency, so use Agent’s GEO Location (require Agent
to verify every time they login since many work remotely and will work from
home or relatives homes).
* Agent monitoring video from several facilities around the country notices
a fire in California facility so the GEO Location would be the California
facility.
* Hospital Agent talking with patient in hospital room 1234 and patient has
heart attack. Agent has to be able to call 911 (GEO Location of hospital and
room 1234), but they also have to be able to toggle between 911 call and the
patients call.
It sounds like the inherited channel variables is exactly what I was looking
for. External application just passes the replacement values for these
variables (_HNO in the example you used, along with many others).
Have an awesome day!
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Wednesday, July 27, 2022 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Question about the Geo Location
support being added
On Wed, Jul 27, 2022 at 11:02 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Looking at the Asterisk wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation
Just FYI, I'm in the process of clarifying and adding more info. Should be
done Friday.
I see the dial plan support the GeolocProfileCreate and there is support for
GEOLOC_PROFILE settings to be set on the dial plan.
We currently use AMI Originate support. We may have dozens/hundreds of calls
in the system and external to Asterisk, someone executes a behavior where we
perform the Originate, if the party answers, we ConfBridge the necessary calls
together. It can be multiple calls and we never know when the total calls
bridged together will need to be increased. Because of the random increase in
calls, we can’t use the Dial to bridge the parties together.
The GEO Location information for the original caller can vary significantly
because they could be WebRTC. We are planning to require the setup of the Geo
Location for each call to be provided to us (either via the incoming call or it
may be provided from third party software). Either way, we will know what the
GEO Location to use for the Originate. Trying to wrap my head around the best
way to achieve this.
A real scenario to test!!! Thanks!
Using AMI Originate, is it possible to set the GEOLOC_PROFILE settings via the
Variable header?
I've not tested this but you don't need to do it at all...
My thought would be to configure an outgoing Geo Location profile for the PJSIP
endpoint, but it would have the minimum settings.
Actually it would have a template specifying replacement channel variables.
When sending the AMI Originate, provide all the adjustments to the
GEOLOC_PROFILE settings via the Variable.
Is this possible or might there be a better way to achieve this?
It's possible but probably not needed. Let's say you're using Civic Address
and a direct originate to the remote party via Dial. In the originate, you
can specify regular, inherited channel variables with the official Civic
Address parameters preceded by '_'. Let's use HNO (house number) as an
example. You'd set _HNO=1633 in the originate and since it has the '_' prefix
it's going to be inherited by the outgoing channel. In the outgoing channel's
profile/location, you'd set 'location_info = HNO=${_HNO}. Of course there'd be
more than just the HNO parameter set but it's the same technique. The outgoing
channel has a very generic location template populated with values received
from the incoming channel.
Now, this isn't going to work if you're originating both calls and adding them
to a bridge yourself but in this case, you have both channels at the same time
so you can just add the incoming channel's location info directly to the
outgoing channel's variables as you originate the outgoing call. Youdon';t
need to create a new GEOLOC_PROFILE for the outgoing channel.
All of this assumes that I actually understood your situation correctly. :)
How are you getting the caller's info in the first place?
Alternatively, I could generate an internal local channel, configure the
GeoLocProfile on it, configure all GEOLOC_PROFILE adjustments on it, then have
it perform the Dial. If the other end answers or not, treat it exactly as we
currently do using the Originate.
Sounds more complicated than it needs to be.
Dan
--
_
-- Bandwidth and Colocation Provided b
[asterisk-users] Question about the Geo Location support being added
Looking at the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation I see the dial plan support the GeolocProfileCreate and there is support for GEOLOC_PROFILE settings to be set on the dial plan. We currently use AMI Originate support. We may have dozens/hundreds of calls in the system and external to Asterisk, someone executes a behavior where we perform the Originate, if the party answers, we ConfBridge the necessary calls together. It can be multiple calls and we never know when the total calls bridged together will need to be increased. Because of the random increase in calls, we can't use the Dial to bridge the parties together. The GEO Location information for the original caller can vary significantly because they could be WebRTC. We are planning to require the setup of the Geo Location for each call to be provided to us (either via the incoming call or it may be provided from third party software). Either way, we will know what the GEO Location to use for the Originate. Trying to wrap my head around the best way to achieve this. Using AMI Originate, is it possible to set the GEOLOC_PROFILE settings via the Variable header? My thought would be to configure an outgoing Geo Location profile for the PJSIP endpoint, but it would have the minimum settings. When sending the AMI Originate, provide all the adjustments to the GEOLOC_PROFILE settings via the Variable. Is this possible or might there be a better way to achieve this? Alternatively, I could generate an internal local channel, configure the GeoLocProfile on it, configure all GEOLOC_PROFILE adjustments on it, then have it perform the Dial. If the other end answers or not, treat it exactly as we currently do using the Originate. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] a couple of problems with confbridge
I believe the answer #2 depends on the user options for each participant. If all participants have user options with wait for marked set to true there will be no conference/recording until at least one marked user joins. If any participants have user options with wait for marked set to false, when they join the conference bridge it is actually going. Thus, if the bridge options had the record enabled it would start recording. If only marked user joins first, it's met the criteria and will conference and start recording. Dan -Original Message- From: asterisk-users On Behalf Of John Covici Sent: Tuesday, June 28, 2022 6:28 PM To: asterisk-users@lists.digium.com Subject: [External] [asterisk-users] a couple of problems with confbridge Hi. I have been using meetme for years, but I wanted to try confbridge as meetme is going away soon.I am having a few problems/questions doing this. 1. When I list the confbridge users in a bridge, I only get the caller id number -- I have a number of contacts in contact manager and I am using superfecta, but the name does not appear. I do need the name to see who is on there. 2. I will be using a conference with a marked user -- and I would like to record the conference -- when does the recording start -- when the first user comes on or when the marked user joins? 3. In the sample file it says you cannot have more than one user profile on a bridge, but I need two, one for the marked user and another one for regular users -- how do I work around this? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] Geolocation/E911
Thank you Joshua. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Monday, May 23, 2022 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] Geolocation/E911 On Mon, May 23, 2022 at 5:52 PM Dan Cropp mailto:d...@amtelco.com>> wrote: We have a customer who wants to perform both directions. For inbound calls, they have indicated their switch will provide Geolocation/E911 information using two different ways. Sounds like it depends on the equipment before the SIP proxy. It will either be in SIP headers (which we retrieve through PJSIP_HEADER in the dial plan). Trivial for us to do this approach. Second approacy is their SIP proxy will append information to the body with mime delimiters (indicating content type of pidf+xml). They want us to retrieve this information (via Asterisk) and store it into their database. When a 911 agent answers the call, they retrieve the data from the database and use it as needed for assisting the first responders. If Asterisk can’t do this and isn’t planned to be able to support this, we’re looking to make Kamailio able to process the INVITE in front of Asterisk. Idea being Kamailio could communicate the information our customer requires and we store it in the database. When Agent answers the call, use the Call-ID provided by Asterisk to match it with the Kamailio INVITE data’s Call-ID. We are still trying to gather details on how the customer wants the outbound calls to work. Both directions will be supported according to the specification and standard. The information will be accessible from the dialplan using a dialplan function, but not the pidf+xml itself. If they're doing something outside of the standard, then it likely wouldn't work. The code is not written for 911 PSAPs/handlers specifically, it's written for receiving the information from an endpoint, and for sending it out. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Geolocation/E911
Thank you Joshua. I will read both links you provided. As mentioned in another response. If what our customer requires doesn’t fit well with Asterisk development, we believe we can do this work with Kamailio front ending the calls. Something customer is already requiring us to do for high availability requirements (911 call center). Asterisk would be far cleaner since there would be no timing issues of call information from 2 boxes. Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Monday, May 23, 2022 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Geolocation/E911 On Mon, May 23, 2022 at 5:18 PM Sebastian Nielsen mailto:sebast...@sebbe.eu>> wrote: What are you talking about exactly? What I have understand, E911 geo information is sent “out of band” when 911 is called. So the question is, is you operating a 911 call centre – then you should have this information at hand already how to access this information. Or are it the opposite, you want to send geocoded information about SIP handset location to a E911 service when 911 is called? (that you want to embed in the outgoing body when 911 is called)? I think gaining access to these types of E911 API’s requires you being a SIP operator – and if you are a SIP operator with a number plan already, you should already have access to these types of API from the beginning and how to use them. There are specifications for conveying location information dynamically in the SIP INVITE to an upstream provider, which uses pidf+xml alongside SDP in the INVITE. This allows you to provide the address, floor, room, longitude/latitude dynamically with the call. You can see what it looks like on the Bandwidth site[1]. It is a standard though, not specific to them. [1] https://support.bandwidth.com/hc/en-us/articles/360006080074-E911-Dynamic-Location-Routing-integration-and-testing-guide -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Geolocation/E911
We have a customer who wants to perform both directions. For inbound calls, they have indicated their switch will provide Geolocation/E911 information using two different ways. Sounds like it depends on the equipment before the SIP proxy. It will either be in SIP headers (which we retrieve through PJSIP_HEADER in the dial plan). Trivial for us to do this approach. Second approacy is their SIP proxy will append information to the body with mime delimiters (indicating content type of pidf+xml). They want us to retrieve this information (via Asterisk) and store it into their database. When a 911 agent answers the call, they retrieve the data from the database and use it as needed for assisting the first responders. If Asterisk can’t do this and isn’t planned to be able to support this, we’re looking to make Kamailio able to process the INVITE in front of Asterisk. Idea being Kamailio could communicate the information our customer requires and we store it in the database. When Agent answers the call, use the Call-ID provided by Asterisk to match it with the Kamailio INVITE data’s Call-ID. We are still trying to gather details on how the customer wants the outbound calls to work. Dan From: asterisk-users On Behalf Of Sebastian Nielsen Sent: Monday, May 23, 2022 3:19 PM To: 'Mailing List' Subject: Re: [External] [asterisk-users] Geolocation/E911 What are you talking about exactly? What I have understand, E911 geo information is sent “out of band” when 911 is called. So the question is, is you operating a 911 call centre – then you should have this information at hand already how to access this information. Or are it the opposite, you want to send geocoded information about SIP handset location to a E911 service when 911 is called? (that you want to embed in the outgoing body when 911 is called)? I think gaining access to these types of E911 API’s requires you being a SIP operator – and if you are a SIP operator with a number plan already, you should already have access to these types of API from the beginning and how to use them. Från: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> mailto:asterisk-users-boun...@lists.digium.com>> För Dan Cropp Skickat: den 23 maj 2022 22:01 Till: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Ämne: [asterisk-users] Geolocation/E911 Out of curiosity, is there any documentation on what is planned for the Geolocation/E911? Is the plan for Asterisk to expose the SIP body and leave it to dial plan, ARI, AMI to process the data? For example, mime pidf+xml section? Or is there a different approach being worked on (or planned to be worked on)? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Geolocation/E911
Out of curiosity, is there any documentation on what is planned for the Geolocation/E911? Is the plan for Asterisk to expose the SIP body and leave it to dial plan, ARI, AMI to process the data? For example, mime pidf+xml section? Or is there a different approach being worked on (or planned to be worked on)? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] Asterisk PJSIP pidf+xml presence question
Thank you Joshua. Thank you to those working on this addition. Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, May 20, 2022 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] Asterisk PJSIP pidf+xml presence question On Fri, May 20, 2022 at 1:59 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Thank you Joshua. Any guess on a timeframe for geolocation/E911 support being part of an Asterisk version? A month or two? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Asterisk PJSIP pidf+xml presence question
Thank you Joshua. Any guess on a timeframe for geolocation/E911 support being part of an Asterisk version? From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, May 20, 2022 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] Asterisk PJSIP pidf+xml presence question On Fri, May 20, 2022 at 1:43 PM Dan Cropp mailto:d...@amtelco.com>> wrote: We have a customer where their switch sends pidf+xml presence information in the SIP INVITE message. Does Asterisk process this pidf+xml information? Does it store this in a channel variable that a dial plan could access? If not, does it store present this information to AMI/ARI applications in any way? At Astricon 2019, one of the presenters talking about Presence and the growing requirements for 911. Needing to know not just building, but where (floor, etc) in building. I don’t recall the full presentation, but I’m guessing pidf is what he was referring to. This is not currently supported. It is ignored if present in the INVITE, and can't be accessed anywhere. You also can't send it. Support for geolocation/E911 is being worked on. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PJSIP pidf+xml presence question
We have a customer where their switch sends pidf+xml presence information in the SIP INVITE message. Does Asterisk process this pidf+xml information? Does it store this in a channel variable that a dial plan could access? If not, does it store present this information to AMI/ARI applications in any way? At Astricon 2019, one of the presenters talking about Presence and the growing requirements for 911. Needing to know not just building, but where (floor, etc) in building. I don't recall the full presentation, but I'm guessing pidf is what he was referring to. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] [External] Asterisk 18.12.0 question
After further testing, not sure this is chan_sip related. I can disable chan_sip.so from loading in modules.conf and that does solve the startup/loading for res_pjsip_transport_websocket. However, there is some issue with the wss transport. Seeing this in both 16.26.0 (not in 16.25.0) and 18.12.0 (not in 18.11.2). REGISTER message comes in, is accepted. However, when it goes to send the OPTIONS, it’s outputting the Unsupported transport. [05/19 10:11:41.992] VERBOSE[2456] res_pjsip_logger.c: <--- Received SIP request (907 bytes) from WSS:192.168.32.27:56443 ---> REGISTER sip:mybox.mydomain.com SIP/2.0 Via: SIP/2.0/WSS c2537bthsnvo.invalid;branch=z9hG4bK2816987 Max-Forwards: 69 To: From: ;tag=24ipeon952 Call-ID: lshogr91tba8r5f335c1g5 CSeq: 2 REGISTER Authorization: Digest algorithm=MD5, username="1234", realm="asterisk", nonce="1652973101/72159fe10d9432b64a16fec84fc414e7", uri="sip:mybox.mydomain.com", response="f46f710af7db6e2e86ec2fabe38325e8", opaque="06a146a816d699e2", qop=auth, cnonce="meehpb38l93l", nc=0001 Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=600 Expires: 600 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: path,gruu,outbound User-Agent: JsSIP 3.3.6 Content-Length: 0 [05/19 10:11:41.993] VERBOSE[2456] res_pjsip_logger.c: <--- Transmitting SIP response (482 bytes) to WSS:192.168.32.27:56443 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS c2537bthsnvo.invalid;rport=56443;received=192.168.32.27;branch=z9hG4bK2816987 Call-ID: lshogr91tba8r5f335c1g5 From: ;tag=24ipeon952 To: ;tag=z9hG4bK2816987 CSeq: 2 REGISTER Date: Thu, 19 May 2022 15:11:41 GMT Contact: ;expires=599 Expires: 600 Server: Asterisk PBX 18.12.0 Content-Length: 0 [05/19 10:11:41.994] ERROR[2456] res_pjsip.c: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 1234 Identical behavior happening with Asterisk 16.26.0, but not on Asterisk 16.25.0 Configuration files are same for between Asterisk versions. [transport3] type = transport bind = 0.0.0.0 protocol = wss allow_reload = no [1234] type = aor max_contacts = 1 remove_existing = yes qualify_frequency = 60 [1234] type = auth auth_type = userpass username = 1234 password = mypassword [1234] type = endpoint context = IS auth = 1234 aors = 1234 dtmf_mode = rfc4733 webrtc = yes disallow = all allow = ulaw transport = transport3 acl = acl5 Might this be because PJSIP 2.12 changes to the “WebRTC updates with AEC3 & AGC2” From: Dan Cropp Sent: Friday, May 13, 2022 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question Thank you Joshua!!! Not loading chan_sip module resolved the problem. Hope you have an awesome weekend. From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of Joshua C. Colp Sent: Friday, May 13, 2022 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question On Fri, May 13, 2022 at 3:19 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Thanks Joshua. I didn’t describe that very well. When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes. My thought was I forgot some module that was required. I noticed I forgot to include these files, so I added them to the package. Rolled back the VM and re-installed. Didn’t make a difference whether they were present or not. /usr/lib/asterisk/modules/codec_g729a.* /usr/lib/asterisk/modules/codec_silk.* /usr/lib/asterisk/modules/codec_siren14.* /usr/lib/asterisk/modules/codec_siren7.* /usr/lib/asterisk/modules/format_ogg_opus.so Comparing the menuselect-tree between the two versions, only changes I see are func_evalexten res_aeap res_speech_aeap and four test_aeap_... added to the TEST_FRAMEWORK. Would it make sense for me to modify my bash script to disable those settings, compile, and try installing? Bash script configures the menuselect options and compiles asterisk. Seems like that would be a better apples to apples comparison. Eliminating the new features. You can. It would also make sense as a test to just not load chan_sip and see what happens. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] [External] Asterisk 18.12.0 question
Thank you Joshua!!! Not loading chan_sip module resolved the problem. Hope you have an awesome weekend. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, May 13, 2022 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 question On Fri, May 13, 2022 at 3:19 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Thanks Joshua. I didn’t describe that very well. When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes. My thought was I forgot some module that was required. I noticed I forgot to include these files, so I added them to the package. Rolled back the VM and re-installed. Didn’t make a difference whether they were present or not. /usr/lib/asterisk/modules/codec_g729a.* /usr/lib/asterisk/modules/codec_silk.* /usr/lib/asterisk/modules/codec_siren14.* /usr/lib/asterisk/modules/codec_siren7.* /usr/lib/asterisk/modules/format_ogg_opus.so Comparing the menuselect-tree between the two versions, only changes I see are func_evalexten res_aeap res_speech_aeap and four test_aeap_... added to the TEST_FRAMEWORK. Would it make sense for me to modify my bash script to disable those settings, compile, and try installing? Bash script configures the menuselect options and compiles asterisk. Seems like that would be a better apples to apples comparison. Eliminating the new features. You can. It would also make sense as a test to just not load chan_sip and see what happens. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] [External] Asterisk 18.12.0 question
Thanks Joshua. I didn’t describe that very well. When I first noticed the res_http_transport_websocket wasn’t loading on that box, I compared the modules folder on both boxes. My thought was I forgot some module that was required. I noticed I forgot to include these files, so I added them to the package. Rolled back the VM and re-installed. Didn’t make a difference whether they were present or not. /usr/lib/asterisk/modules/codec_g729a.* /usr/lib/asterisk/modules/codec_silk.* /usr/lib/asterisk/modules/codec_siren14.* /usr/lib/asterisk/modules/codec_siren7.* /usr/lib/asterisk/modules/format_ogg_opus.so Comparing the menuselect-tree between the two versions, only changes I see are func_evalexten res_aeap res_speech_aeap and four test_aeap_... added to the TEST_FRAMEWORK. Would it make sense for me to modify my bash script to disable those settings, compile, and try installing? Bash script configures the menuselect options and compiles asterisk. Seems like that would be a better apples to apples comparison. Eliminating the new features. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, May 13, 2022 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [External] [asterisk-users] [External] Asterisk 18.12.0 question On Fri, May 13, 2022 at 2:43 PM Dan Cropp mailto:d...@amtelco.com>> wrote: Hi Joshua, Thank you for helping me diagnose this. Interesting that they are the exact same between versions. File sizes are slightly different between the two when I compile them. 18.12.0 would have any new default configurations that were not part of 18.11.2, such as aeap. Some codecs also seem to be defaulted now. Not sure if that makes a difference. After this e-mail, I will try disabling some of the new additions. Maybe that will resolve things. 18.11.2 408136 18.12.0 411856 The source files are different, but the resulting binaries can differ between versions if headers/other things change. You'd also need to specify what "Some codecs also seem to be defaulted now.", unless things are deprecated or binary then the default is to have them enabled. Here is output as it loads the res_pjsip_transport_websocket.so being loaded on 18.11.2… Loading app_stack.so. == AGI Command 'gosub' registered == Registered application 'StackPop' == Registered application 'Return' == Registered application 'GosubIf' == Registered application 'Gosub' == Registered custom function 'LOCAL' == Registered custom function 'LOCAL_PEEK' == Registered custom function 'STACK_PEEK' == app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) Loading res_pjsip_path.so. == res_pjsip_path.so => (PJSIP Path Header Support) Loading res_pjsip_transport_websocket.so. == WebSocket registered sub-protocol 'sip' == res_pjsip_transport_websocket.so => (PJSIP WebSocket Transport Support) Loading res_stasis_recording.so. == res_stasis_recording.so => (Stasis application recording support) Loading res_pjsip_nat.so. == res_pjsip_nat.so => (PJSIP NAT Support) Loading res_pjsip_diversion.so. == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support) Here is what logs show on 18.12.0…. Loading app_stack.so. == AGI Command 'gosub' registered == Registered application 'StackPop' == Registered application 'Return' == Registered application 'GosubIf' == Registered application 'Gosub' == Registered custom function 'LOCAL' == Registered custom function 'LOCAL_PEEK' == Registered custom function 'STACK_PEEK' == app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) Loading res_pjsip_path.so. == res_pjsip_path.so => (PJSIP Path Header Support) Loading res_pjsip_transport_websocket.so. Loading res_stasis_recording.so. == res_stasis_recording.so => (Stasis application recording support) Loading res_pjsip_nat.so. == res_pjsip_nat.so => (PJSIP NAT Support) Loading res_pjsip_diversion.so. == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support) … [May 13 12:34:08] WARNING[1400]: loader.c:2393 load_modules: Some non-required modules failed to load. [May 13 12:34:08] WARNING[1400]: loader.c:2487 load_modules: Module 'chan_sip' has been loaded but was deprecated in Asterisk version 17 and will be removed in Asterisk version 21. [May 13 12:34:08] ERROR[1400]: loader.c:2508 load_modules: Error loading module 'app_queue.so', missing dependency: res_monitor [May 13 12:34:08] ERROR[1400]: loader.c:2508 load_modules: res_pjsip_transport_websocket declined to load. [May 13 12:34:08] WARNING[1407]: chan_sip.c:35470 deprecation_notice: chan_sip has no official maintainer and is deprecated. Migration to [May 13 12:34:08] WARNING[1407]: chan_sip.c:35471 deprecation_notice: chan_pjsip is recommended. See guides at the Asterisk Wiki: [May 13 12:34:08] WARNING[1407]: chan_sip.c:35472 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_
Re: [asterisk-users] [External] Asterisk 18.12.0 question
Hi Joshua, Thank you for helping me diagnose this. Interesting that they are the exact same between versions. File sizes are slightly different between the two when I compile them. 18.12.0 would have any new default configurations that were not part of 18.11.2, such as aeap. Some codecs also seem to be defaulted now. Not sure if that makes a difference. After this e-mail, I will try disabling some of the new additions. Maybe that will resolve things. 18.11.2 408136 18.12.0 411856 Here is output as it loads the res_pjsip_transport_websocket.so being loaded on 18.11.2… Loading app_stack.so. == AGI Command 'gosub' registered == Registered application 'StackPop' == Registered application 'Return' == Registered application 'GosubIf' == Registered application 'Gosub' == Registered custom function 'LOCAL' == Registered custom function 'LOCAL_PEEK' == Registered custom function 'STACK_PEEK' == app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) Loading res_pjsip_path.so. == res_pjsip_path.so => (PJSIP Path Header Support) Loading res_pjsip_transport_websocket.so. == WebSocket registered sub-protocol 'sip' == res_pjsip_transport_websocket.so => (PJSIP WebSocket Transport Support) Loading res_stasis_recording.so. == res_stasis_recording.so => (Stasis application recording support) Loading res_pjsip_nat.so. == res_pjsip_nat.so => (PJSIP NAT Support) Loading res_pjsip_diversion.so. == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support) Here is what logs show on 18.12.0…. Loading app_stack.so. == AGI Command 'gosub' registered == Registered application 'StackPop' == Registered application 'Return' == Registered application 'GosubIf' == Registered application 'Gosub' == Registered custom function 'LOCAL' == Registered custom function 'LOCAL_PEEK' == Registered custom function 'STACK_PEEK' == app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) Loading res_pjsip_path.so. == res_pjsip_path.so => (PJSIP Path Header Support) Loading res_pjsip_transport_websocket.so. Loading res_stasis_recording.so. == res_stasis_recording.so => (Stasis application recording support) Loading res_pjsip_nat.so. == res_pjsip_nat.so => (PJSIP NAT Support) Loading res_pjsip_diversion.so. == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support) … [May 13 12:34:08] WARNING[1400]: loader.c:2393 load_modules: Some non-required modules failed to load. [May 13 12:34:08] WARNING[1400]: loader.c:2487 load_modules: Module 'chan_sip' has been loaded but was deprecated in Asterisk version 17 and will be removed in Asterisk version 21. [May 13 12:34:08] ERROR[1400]: loader.c:2508 load_modules: Error loading module 'app_queue.so', missing dependency: res_monitor [May 13 12:34:08] ERROR[1400]: loader.c:2508 load_modules: res_pjsip_transport_websocket declined to load. [May 13 12:34:08] WARNING[1407]: chan_sip.c:35470 deprecation_notice: chan_sip has no official maintainer and is deprecated. Migration to [May 13 12:34:08] WARNING[1407]: chan_sip.c:35471 deprecation_notice: chan_pjsip is recommended. See guides at the Asterisk Wiki: [May 13 12:34:08] WARNING[1407]: chan_sip.c:35472 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip [May 13 12:34:08] WARNING[1407]: chan_sip.c:35473 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip Because some of our customers have refused to migrate to PJSIP, we still compile support for chan_sip. However, we make sure to disable the chan_sip web support in the configuration files. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 18.12.0 question
I have been using Asterisk 18.11.2. Just tried Asterisk 18.12.0 and am running into a problem with the res_pjsip_transport_websocket. Using Ubuntu 20 I use a bash shell script to compile Asterisk with settings. I didn't modify any settings from Asterisk 18.11.2 build that works. After compiling, I make an install .deb pkg that wraps all the Asterisk support and use this dpkg to install on other boxes. Eliminates the need to compile and everything on customer boxes. All this process works great for Asterisk 18.11.2 (and did so for several other Asterisk versions prior to 18.11.2). I can run Asterisk on the box I compile on. However, the boxes I install the .deb pkg, Asterisk always complains that it cannot load the res_pjsip_transport_websocket module. I have looked at all the dependencies and confirmed I am copying all the .so from my compile box to this new box I install on. Did something change with the res_pjsip_transport_websocket where it requires something new? Does PJSIP 2.12 require something new that the previous PJSIP version Asterisk used did not require? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem
As far as I'm aware Josh, it doesnt stop a call from happening - I've had the same "errors" pop up when using Twilio and Simwood but calls continue just fine. On Thu, Dec 2, 2021 at 2:30 PM Joshua C. Colp wrote: > On Thu, Dec 2, 2021 at 10:18 AM James Cloos wrote: > >> > "KT" == Kingsley Tart writes: >> >> KT> I can't get Asterisk to send a SIP call to Twilio over TLS >> KT> because it complains about Twilio's wildcard certificate. >> >> the sip rfc claims that wildcard certs should be invalid for sip. >> >> digium insisted on following that advise as set in stone, and so >> asterisk refuses such certs. i doubt that stance is different >> under sangoma. >> >> the only workaround is to remind twil of the rfc and get them to >> replace the wildcard with an rfc-copliant cert. at least for the >> sip ports. >> > > To be specific, this is in PJSIP land. There was no insisting or anything > and it wasn't a decision we originally made. It's the way that Teluu > implemented the TLS transport in PJSIP and since we use PJSIP then it > applies to us. If someone contributed a change to Asterisk to make it > configurable in some way, then we'd certainly review it. At this point > though noone has done such a thing. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem
It shouldnt stop the call from happening. It will be something else... up your debugging level and see what else you get Lots of providers go against this part of the spec but I've run Asterisk 18 with twilio over sip over tls and everything worked, it just spat out the error line On Thu, Dec 2, 2021 at 12:21 AM Kingsley Tart wrote: > On Wed, 2021-12-01 at 22:54 +0100, Antony Stone wrote: > > So, https://datatracker.ietf.org/doc/html/rfc5922#section-7.2 does seem > pretty > > clear about this. "Implementations MUST NOT match any form of wildcard" > > > > Have you contacted the provider who is using a wildcard certificate in > this way > > and referred them to the RFC? > > No I haven't, but if I did I suspect they would take no notice. Twilio > is a big provider who do what they do because they can. > > And I can see why they do this, because customers can set up their own > SIP trunks on their system with their unique hostname, so it makes > sense for them to have a wildcard cert, whether in violation of the RFC > or not. > > -- > Cheers, > Kingsley. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Re: Is app_queue going to stay around or is it being deprecated (uses res_monitor)
Thank you Joshua From: asterisk-users On Behalf Of Joshua C. Colp Sent: Thursday, November 18, 2021 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [External] Re: [asterisk-users] Is app_queue going to stay around or is it being deprecated (uses res_monitor) On Thu, Nov 18, 2021 at 4:34 PM Dan Cropp mailto:d...@amtelco.com>> wrote: We currently use the Queue. Under app_queue, it uses module res_monitor (which is on the to be deprecated list). Is it safe to continue using Queue (app_queue)? The app_queue module is not on the deprecated list, therefore it is not being deprecated. The app_queue module has both Monitor and MixMonitor support. When res_monitor is removed, then the Monitor support in app_queue would go as well. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is app_queue going to stay around or is it being deprecated (uses res_monitor)
We currently use the Queue. Under app_queue, it uses module res_monitor (which is on the to be deprecated list). Is it safe to continue using Queue (app_queue)? Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Re: Question on ExternalMedia and the codec
Thank you George. Using slin16 instead of generic slin resolved the issue. Have a good day! Dan From: asterisk-users On Behalf Of George Joseph Sent: Wednesday, October 13, 2021 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [External] Re: [asterisk-users] Question on ExternalMedia and the codec On Tue, Oct 12, 2021 at 2:54 PM Dan Cropp mailto:d...@amtelco.com>> wrote: We tell asterisk to use the slin format for ExternalMedia. However, the unicast channel is selecting ulaw formatand the RTP data is indicating it’s ulaw format. Anyone know why ulaw format would be on chosen? What do your ARI requests look like? Are you just requesting "slin" or one of the specific variants? [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is /ari/channels/externalMedia?app=a2519b4b-4d90-4d18-906b-717d02f8d569_host=192.168.32.148:8080=slin [10/12 16:13:39.396] DEBUG[1665] http.c: match request [ari/channels/externalMedia] with handler [static] len 6 [10/12 16:13:39.396] DEBUG[1665] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [10/12 16:13:39.396] DEBUG[1665] http.c: Match made with [ari] [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels/externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari deviceStates: Didn't match channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari applications: Didn't match channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari channels: Explicit match with channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels create: Didn't match externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels channelId: Matched wildcard. [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our source address is '192.168.33.34'. [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fef60018320' [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) RTP allocated port 12226 [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE creating session 192.168.33.34:12226<http://192.168.33.34:12226> (12226) [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE create [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add system candidates [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add candidate: 192.168.33.34:12226<http://192.168.33.34:12226>, 2130706431 [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: RTP instance '0x7fef60018320' is setup and ready to go [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: : Formats: (none) [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: Channel is being initialized or destroyed [10/12 16:13:39.396] DEBUG[1665] stasis.c: Creating topic. name: channel:1634055219.4, detail: [10/12 16:13:39.396] DEBUG[1665] stasis.c: Topic 'channel:1634055219.4': 0x7fef6008d170 created [10/12 16:13:39.396] DEBUG[1665] channel.c: Channel 0x7fef6008a910 'UnicastRTP/192.168.32.148:8080-0x7fef60018320' allocated [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our source address is '192.168.33.34'. [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: UnicastRTP/192.168.32.148:8080-0x7fef60018320: Formats: (ulaw) [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: New topology set [10/12 16:13:39.396] DEBUG[1665] res_stasis.c: a2519b4b-4d90-4d18-906b-717d02f8d569: Subscribing to 1634055219.4 [10/12 16:13:39.396] DEBUG[1665] stasis/app.c: Channel '1634055219.4' is 1 interested in a2519b4b-4d90-4d18-906b-717d02f8d569 [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP keeping session open. status_code:200 [10/12 16:13:39.396] DEBUG[1666] stasis/app.c: Channel '1634055219.4' is 2 interested in a2519b4b-4d90-4d18-906b-717d02f8d569 Have a good day! Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here:
[asterisk-users] Question on ExternalMedia and the codec
We tell asterisk to use the slin format for ExternalMedia. However, the unicast channel is selecting ulaw formatand the RTP data is indicating it's ulaw format. Anyone know why ulaw format would be on chosen? [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is /ari/channels/externalMedia?app=a2519b4b-4d90-4d18-906b-717d02f8d569_host=192.168.32.148:8080=slin [10/12 16:13:39.396] DEBUG[1665] http.c: match request [ari/channels/externalMedia] with handler [static] len 6 [10/12 16:13:39.396] DEBUG[1665] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [10/12 16:13:39.396] DEBUG[1665] http.c: Match made with [ari] [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels/externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari deviceStates: Didn't match channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari applications: Didn't match channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari channels: Explicit match with channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels create: Didn't match externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels channelId: Matched wildcard. [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our source address is '192.168.33.34'. [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fef60018320' [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) RTP allocated port 12226 [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE creating session 192.168.33.34:12226 (12226) [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE create [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add system candidates [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add candidate: 192.168.33.34:12226, 2130706431 [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: RTP instance '0x7fef60018320' is setup and ready to go [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: : Formats: (none) [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: Channel is being initialized or destroyed [10/12 16:13:39.396] DEBUG[1665] stasis.c: Creating topic. name: channel:1634055219.4, detail: [10/12 16:13:39.396] DEBUG[1665] stasis.c: Topic 'channel:1634055219.4': 0x7fef6008d170 created [10/12 16:13:39.396] DEBUG[1665] channel.c: Channel 0x7fef6008a910 'UnicastRTP/192.168.32.148:8080-0x7fef60018320' allocated [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our source address is '192.168.33.34'. [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: UnicastRTP/192.168.32.148:8080-0x7fef60018320: Formats: (ulaw) [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: New topology set [10/12 16:13:39.396] DEBUG[1665] res_stasis.c: a2519b4b-4d90-4d18-906b-717d02f8d569: Subscribing to 1634055219.4 [10/12 16:13:39.396] DEBUG[1665] stasis/app.c: Channel '1634055219.4' is 1 interested in a2519b4b-4d90-4d18-906b-717d02f8d569 [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP keeping session open. status_code:200 [10/12 16:13:39.396] DEBUG[1666] stasis/app.c: Channel '1634055219.4' is 2 interested in a2519b4b-4d90-4d18-906b-717d02f8d569 Have a good day! Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] External media codec question
When we perform ExternalMedia with the slin format, we are still receiving ulaw rtp packets. Asterisk logs show it's selecting ulaw. I'm guessing we are missing a menuselect or configuration setting. Anyone have any suggestions for the possible cause and what to look at? Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge recording "Failed to get 160 samples from read factory" and "Read factory ... and write factory ... both fail to provide 160 samples"
We are running Asterisk 16.17.0 and discovered what we think is an issue. We have a single call in a ConfBridge. Tell the ConfBridge to start recording. We see non-stop audiohook.c 160 samples failures. As soon as we stop recording (AMI ConfBridgeStopRecord) these failures stop. [10/04 14:57:57.109] DEBUG[2039] manager.c: Running action 'ConfbridgeStartRecord' [10/04 14:57:57.109] DEBUG[2039] stasis.c: Creating topic. name: channel:1633377477.19, detail: [10/04 14:57:57.109] DEBUG[2039] stasis.c: Topic 'channel:1633377477.19': 0x7fcc68016c00 created [10/04 14:57:57.109] DEBUG[2039] stasis.c: Creating topic. name: cache:59/channel:1633377477.19, detail: [10/04 14:57:57.109] DEBUG[2039] stasis.c: Topic 'cache:59/channel:1633377477.19': 0x7fcc68063e90 created [10/04 14:57:57.109] DEBUG[2039] channel.c: Channel 0x7fcc680977a0 'CBRec/IS__OpBridge1-0004' allocated [10/04 14:57:57.109] DEBUG[2039] bridge_roles.c: Set role 'recorder' [10/04 14:57:57.109] DEBUG[2039] autochan.c: Created autochan 0x7fcc6801bfa0 to hold channel CBRec/IS__OpBridge1-0004 (0x7fcc680977a0) [10/04 14:57:57.109] DEBUG[2530] bridge_channel.c: Bridge 44a7691f-bd48-40cd-9d05-4c2c3f482087: 0x7fcc68099d40(CBRec/IS__OpBridge1-0004) is joining [10/04 14:57:57.109] DEBUG[2530] bridge_channel.c: Bridge 44a7691f-bd48-40cd-9d05-4c2c3f482087: pushing 0x7fcc68099d40(CBRec/IS__OpBridge1-0004) [10/04 14:57:57.109] DEBUG[2530] bridge_roles.c: Set role 'recorder' [10/04 14:57:57.109] DEBUG[2530] bridge.c: Bridge 44a7691f-bd48-40cd-9d05-4c2c3f482087: 0x7fcc68099d40(CBRec/IS__OpBridge1-0004) is joining softmix technology [10/04 14:57:57.109] DEBUG[2530] bridge_softmix.c: CBRec/IS__OpBridge1-0004: [10/04 14:57:57.109] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.109] DEBUG[2530] channel.c: Channel CBRec/IS__OpBridge1-0004 setting write format path: slin -> slin [10/04 14:57:57.109] DEBUG[2530] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [10/04 14:57:57.109] DEBUG[2530] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [10/04 14:57:57.109] DEBUG[2530] bridge_softmix.c: [10/04 14:57:57.109] DEBUG[2530] bridge_softmix.c: CBRec/IS__OpBridge1-0004: [10/04 14:57:57.109] DEBUG[2530] bridge_softmix.c: CBRec/IS__OpBridge1-0004: Not in SFU mode [10/04 14:57:57.109] DEBUG[2039] manager.c: Examining AMI event: Event: VarSet^M Privilege: dialplan,all^M Channel: PJSIP/1003-^M ChannelState: 6^M ChannelStateDesc: Up^M CallerIDNum: 1003^M CallerIDName: 1003^M ConnectedLineNum: ^M ConnectedLineName: ^M Language: en^M AccountCode: 2^M Context: IS^M Exten: 333^M Priority: 18^M Uniqueid: 1633376893.0^M Linkedid: 1633376893.0^M Variable: BRIDGEPEER^M Value: CBAnn/IS__OpBridge1-0005;2,CBRec/IS__OpBridge1-0004^M ^M Event: ConfbridgeRecord^M Privilege: call,all^M Conference: IS__OpBridge1^M BridgeUniqueid: 44a7691f-bd48-40cd-9d05-4c2c3f482087^M BridgeType: base^M BridgeTechnology: softmix^M BridgeCreator: ConfBridge^M BridgeName: IS__OpBridge1^M BridgeNumChannels: 3^M BridgeVideoSourceMode: none^M ^M [10/04 14:57:57.128] DEBUG[2530] audiohook.c: Flushing audiohook 0x7fcc680f3c00 so it remains in sync [10/04 14:57:57.128] DEBUG[2530] audiohook.c: Audiohook 0x7fcc680f3c00 has stale audio in its factories. Flushing them both [10/04 14:57:57.128] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory 0x7fcc680f3c88 [10/04 14:57:57.128] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.148] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory 0x7fcc680f3c88 [10/04 14:57:57.148] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.168] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory 0x7fcc680f3c88 [10/04 14:57:57.168] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.188] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory 0x7fcc680f3c88 [10/04 14:57:57.188] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.208] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory 0x7fcc680f3c88 [10/04 14:57:57.208] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.228] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory 0x7fcc680f3c88 [10/04 14:57:57.228] DEBUG[2529] audiohook.c: Read factory 0x7fcc680f3c88 and write factory 0x7fcc680f46c8 both fail to provide 160 samples [10/04 14:57:57.248] DEBUG[2529] audiohook.c: Failed to get 160 samples from read factory
Re: [asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK refcount related messages
] libc.so.6 clone.S:97 clone() [09/28 09:31:55.638] WARNING[2872][C-0189] channel.c: Exceptionally long voice queue length queuing to CBAnn/IS__b6b99750-c9aa-420c-9e9f-fa41e9a99fa9-000d;1 [09/28 09:31:55.706] WARNING[3092][C-01ca] channel.c: Exceptionally long voice queue length queuing to CBAnn/IS__43f4512f-4009-448f-9ccf-4d3ada975f74-0054;1 [09/28 09:31:55.764] WARNING[3231][C-0161] channel.c: Exceptionally long voice queue length queuing to CBAnn/IS__57f2e3a7-6c86-423f-b35f-25048ef68591-007d;1 [09/28 09:31:55.771] ERROR[3208] : Got 8 backtrace records # 0: [0x55d428652177] asterisk utils.c:2454 __ast_assert_failed() # 1: [0x55d4284bb742] asterisk astobj2.c:588 __ao2_ref() # 2: [0x7faa79aa75c0] bridge_softmix.so bridge_softmix.c:257 softmix_process_write_audio() # 3: [0x7faa79aabef7] bridge_softmix.so bridge_softmix.c:1934 softmix_mixing_loop() # 4: [0x7faa79aac54d] bridge_softmix.so bridge_softmix.c:2047 softmix_mixing_thread() # 5: [0x55d42864ef9b] asterisk utils.c:1299 dummy_start() # 6: [0x7fab064b96db] libpthread.so.0 pthread_create.c:463 start_thread() # 7: [0x7fab056ab71f] libc.so.6 clone.S:97 clone() From: asterisk-users On Behalf Of Joshua C. Colp Sent: Thursday, September 23, 2021 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK refcount related messages On Thu, Sep 23, 2021 at 1:59 PM Dan Cropp mailto:d...@amtelco.com>> wrote: We have an extremely busy/large customer. They run fine most of the time, but periodically asterisk will output FRACK refcount related messages. It doesn’t seem to be related to the volume, because it’s not breaking during their peak times. When this happens, the system becomes unstable and they have to restart to get things resolved. To give an idea of the instability, we have seen INVITE/Trying responses in SIP messaging logs. We tell Asterisk to answer via AMI, but Asterisk never sends the OK (even 24 seconds later it hasn’t sent). Eventually the other send CANCEL of the call. We’ve now captured 4 different days where something like the following occurs. 1) Is there a good way to tell if this may be fixed in Asterisk 16.20.0 (short of upgrading)? You could examine the changes between the two and see if any issues seem relevant. 2) Would this be something I should submit as an asterisk issue? Unfortunately, site is so busy capturing the debug will be very difficult (if not impossible) due to amount of data. Without a backtrace, or debug logs it'd be really hard to look at. The message just means that an object was referenced a lot. Why that is - who knows. It could be that the system can't keep up for some reason, or there's an off-nominal path, or something is deadlocked, or something else. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any thoughts on how to resolve "Exceptionally long voice queue length queueing to CBAnn"
Running Asterisk 16.17.0 We combine calls into a ConfBridge using AMI with AsynAGI. Executing actions to ConfBridge channels into the same ConfBridge. It's a very large and busy system, so there are dozens of these that may happen during a second (different channels), so thousands in an hour. "WARNING[20710][C-33a6] channel.c: Exceptionally long voice queue length queuing to CBAnn/" Is this expected? Any thoughts on how to resolve this? Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK refcount related messages
/bridge_softmix.so(+0x40af) [0x7fc15362f0af] # 4: /usr/lib/asterisk/modules/bridge_softmix.so(+0x560a) [0x7fc15363060a] # 5: /usr/sbin/asterisk(+0x1db41f) [0x5637ec06041f] # 6: /lib/x86_64-linux-gnu/libpthread.so.0(+0x76db) [0x7fc1e9ebf6db] # 7: /lib/x86_64-linux-gnu/libc.so.6(clone+0x3f) [0x7fc1e93f971f] Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large system seeing single CPU core spiking
Anyone know what the pjsip/distributor tasks are doing? Customer site had issue this morning and I was able to capture part of the taskprocessors. This is the first time I have noticed any of our systems or load tests with pjsip/distributor Max Depth reaching double digits. Right now, their system has more active channels and active calls in the system than when they experienced problems, but the pjsip/distributors all have a MaxDepth of 2 or 3. Any recommendations or thoughts on what may cause pjsip/distributor tasks to see MaxDepth increase? Could this cause jitter and PJSIP packet processing to be slow? (Other end drops offline because OPTIONS reply not received or maybe not processed in time) pjsip/distributor-0ae9 3010 0 13450500 pjsip/distributor-0aea 2978 0 30450500 pjsip/distributor-0aeb 3132 0 11450500 pjsip/distributor-0aec 2757 0 6450500 pjsip/distributor-0aed 3336 0 15450500 pjsip/distributor-0aee 2946 0 14450500 pjsip/distributor-0aef 2966 0 8450500 pjsip/distributor-0af0 2805 0 18450500 pjsip/distributor-0af1 3341 0 12450500 pjsip/distributor-0af2 3282 0 9450500 pjsip/distributor-0af3 3228 0 11450500 pjsip/distributor-0af4 2866 0 9450500 pjsip/distributor-0af5 2902 0 10450500 pjsip/distributor-0af6 3032 0 24450500 pjsip/distributor-0af7 2872 0 8450500 pjsip/distributor-0af8 2881 0 9450500 pjsip/distributor-0af9 2933 0 12450500 pjsip/distributor-0afa 3002 0 8450500 pjsip/distributor-0afb 2873 0 6450500 pjsip/distributor-0afc 0 5450500 pjsip/distributor-0afd 2947 0 10450500 pjsip/distributor-0afe 3075 0 8450500 pjsip/distributor-0aff 3204 0 37450500 pjsip/distributor-0b00 3230 0 8450500 pjsip/distributor-0b01 3041 0 6450500 pjsip/distributor-0b02 2943 0 10450500 pjsip/distributor-0b03 3056 0 14450500 pjsip/distributor-0b04 2756 0 8450500 pjsip/distributor-0b05 2848 0 9450500 pjsip/distributor-0b06 2965 0 10450500 pjsip/distributor-0b07 2902 0 8450500 Any thoughts? Dan From: Dan Cropp Sent: Tuesday, September 14, 2021 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Large system seeing single CPU core spiking Thank you George. That’s good advice on the realtime mode. I can have them change this so realtime mode later this week
Re: [asterisk-users] Large system seeing single CPU core spiking
Thank you George. That’s good advice on the realtime mode. I can have them change this so realtime mode later this week. Customer’s taskprocessor list is very large. There are a large number of entries from core show taskprocessors data The are one a few that are showing the Max Depth of 10 or more. Only including those and the stasis/pool Processor Processed In Queue Max Depth Low water High water pjsip/pool-control 501599 0 89450500 stasis/m:cache_pattern:0/endpoint:all-15f0 383224 0 21450500 stasis/m:devicestate:all-0002 233836 0 28450500 stasis/m:manager:core-0006 4649316 0 69 2700 3000 stasis/pool 11670 0 2450500 stasis/pool-control 23505 0 75450500 5922 taskprocessors We do use AMI for a significant amount of communication (action/events). Might this be a singleton that could explain the high use for a single asterisk process id? NOTE: working on migrating to ARI which I know will help in the call control. Dan From: asterisk-users On Behalf Of George Joseph Sent: Tuesday, September 14, 2021 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Large system seeing single CPU core spiking On Tue, Sep 14, 2021 at 9:19 AM Dan Cropp mailto:d...@amtelco.com>> wrote: Thank you George. It is using local file based configuration files. Well, that's good at least. It eliminates the database layer which can be troublesome in virtualized environments, especially if a SAN and/or a remote database server is used. Other factors. We run Asterisk in realtime mode to allow it to run as fast as possible. Running at "realtime" level is usually NOT a good thing for Asterisk and rarely needed when there are adequate resources. Let's say you have a local DNS resolver running. If the system is stressed, Asterisk could actually starve the resolver of resources, which then causes Asterisk to back up waiting for DNS resolution to complete. We've seen this happen when using a database backend for configuration. Someone thinks "I'll just give Asterisk more resources" forgetting that Asterisk needs the database engine to run. I just learned customer upgraded to 24 CPU cores. Although, I’m not sure they actually assigned 24 physical cores to this machine or just increasing Hyper-V values. How is this VM's priority versus other VMs on the same cluster? Just because it has 24 threads doesn't mean it's got 24 threads dedicated. Does using a realtime priority in the VM trickle down to Hyper-V's hypervisor's resource management algorithms? I will monitor for additional information and see if the customer will allow me to capture a coredump when problems are happening. Waiting for them to report an incident. Here is a small sample of the system right now (24 cores), to the best of my knowledge it’s running fine. top -p 1509 -n 1 -H -b top - 15:06:32 up 9:06, 2 users, load average: 6.02, 5.59, 5.26 Threads: 1709 total, 8 running, 1701 sleeping, 0 stopped, 0 zombie %Cpu(s): 3.1 us, 2.5 sy, 0.0 ni, 94.3 id, 0.0 wa, 0.0 hi, 0.1 si, 0.0 st KiB Mem : 32143072 total, 29750072 free, 1016132 used, 1376868 buff/cache KiB Swap: 8388604 total, 8388604 free,0 used. 30697060 avail Mem PID USER PR NIVIRTRESSHR S %CPU %MEM TIME+ COMMAND 1830 root -11 0 13.741g 493680 28828 R 99.9 1.5 174:13.39 asterisk 1541 root -11 0 13.741g 493680 28828 R 14.3 1.5 20:03.30 asterisk 33601 root -11 0 13.741g 493680 28828 S 9.5 1.5 0:16.30 asterisk 46605 root -11 0 13.741g 493680 28828 S 9.5 1.5 0:30.06 asterisk 2295 root -11 0 13.741g 493680 28828 S 4.8 1.5 12:25.50 asterisk 2297 root -11 0 13.741g 493680 28828 S 4.8 1.5 1:10.59 asterisk There's definitely one thread that's pegging a CPU. If that thread is one of the few "singleton" threads, that can be an issue. What does "core show taskprocessors" indicate? Are there any that are hitting their limits? From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of George Joseph Sent: Tuesday, September 14, 2021 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [asterisk-users] Large system seeing single CPU core spiking On Tue, Sep 14, 2021 at 8:07 AM Dan Cropp mailto:d...@am
Re: [asterisk-users] Large system seeing single CPU core spiking
Thank you George. It is using local file based configuration files. Other factors. We run Asterisk in realtime mode to allow it to run as fast as possible. I just learned customer upgraded to 24 CPU cores. Although, I’m not sure they actually assigned 24 physical cores to this machine or just increasing Hyper-V values. I will monitor for additional information and see if the customer will allow me to capture a coredump when problems are happening. Waiting for them to report an incident. Here is a small sample of the system right now (24 cores), to the best of my knowledge it’s running fine. top -p 1509 -n 1 -H -b top - 15:06:32 up 9:06, 2 users, load average: 6.02, 5.59, 5.26 Threads: 1709 total, 8 running, 1701 sleeping, 0 stopped, 0 zombie %Cpu(s): 3.1 us, 2.5 sy, 0.0 ni, 94.3 id, 0.0 wa, 0.0 hi, 0.1 si, 0.0 st KiB Mem : 32143072 total, 29750072 free, 1016132 used, 1376868 buff/cache KiB Swap: 8388604 total, 8388604 free,0 used. 30697060 avail Mem PID USER PR NIVIRTRESSHR S %CPU %MEM TIME+ COMMAND 1830 root -11 0 13.741g 493680 28828 R 99.9 1.5 174:13.39 asterisk 1541 root -11 0 13.741g 493680 28828 R 14.3 1.5 20:03.30 asterisk 33601 root -11 0 13.741g 493680 28828 S 9.5 1.5 0:16.30 asterisk 46605 root -11 0 13.741g 493680 28828 S 9.5 1.5 0:30.06 asterisk 2295 root -11 0 13.741g 493680 28828 S 4.8 1.5 12:25.50 asterisk 2297 root -11 0 13.741g 493680 28828 S 4.8 1.5 1:10.59 asterisk From: asterisk-users On Behalf Of George Joseph Sent: Tuesday, September 14, 2021 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Large system seeing single CPU core spiking On Tue, Sep 14, 2021 at 8:07 AM Dan Cropp mailto:d...@amtelco.com>> wrote: I am working with a very large customer running Asterisk with PJSIP. Systems total channels have been over 2500 (which includes hundreds of local channels and ConfBridges) when the issues occur. It’s running on a Hyper-V VM with 12 CPU cores. Things work fine most of the time. They periodically see 10-30 minute periods where audio starts sounding like jitter buffer type issues. Can literally have someone spelling their name and a ConfBridge recording of it shows the audio is missing a letter or two. The odd part is another system (not running Asterisk) was handling these calls previously. The overall network has plenty of bandwidth (as evidenced by another system able to handle the call volume) One area that has perplexed us is when using htop, we see a single CPU core will spike to 100%. Which core does keep changing. Asterisk is definitely the process using the vast majority of the CPU cycles. We recently found a setting on Hyper-V networking SR-IOV which improved things. Prior to changing this setting, we were seeing SIP OPTIONS packets/responses would occasionally take more than 3 seconds causing devices to drop and come back online. We have configured a similar system running at Amazon handling far more traffic and can’t get the single CPU core to spike. Only small static pops during the calls. The sheer scale of the system is making it hard to diagnose the problem. Any thoughts on how to diagnose what is causing the single CPU core to spike? Any thoughts on how to diagnose the problem? Any other thoughts/comments? The first thing I'd do is see where the CPU is spending time: userspace, system, nice, wait, etc. Is it actually the asterisk process consuming the CPU? Is Asterisk running with local file-based configs, local database, remote database, etc? If call quality is really bad already and your customer agrees, you could try the following the next time it happens... 1. Run "top -p `pidof asterisk` -n 1 -H -b" to get a list of all of Asterisk's threads and their CPU utilization. 2. Run ast_coredumper with the --RUNNING option. This will pause Asterisk while the dump is being generated! 3. See if you can correlate the high cpu thread IDs from the top output to the threads listed in the coredumper's -brief.txt file. That _may_ give you an idea of where to look. Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+S
[asterisk-users] Large system seeing single CPU core spiking
I am working with a very large customer running Asterisk with PJSIP. Systems total channels have been over 2500 (which includes hundreds of local channels and ConfBridges) when the issues occur. It's running on a Hyper-V VM with 12 CPU cores. Things work fine most of the time. They periodically see 10-30 minute periods where audio starts sounding like jitter buffer type issues. Can literally have someone spelling their name and a ConfBridge recording of it shows the audio is missing a letter or two. The odd part is another system (not running Asterisk) was handling these calls previously. The overall network has plenty of bandwidth (as evidenced by another system able to handle the call volume) One area that has perplexed us is when using htop, we see a single CPU core will spike to 100%. Which core does keep changing. Asterisk is definitely the process using the vast majority of the CPU cycles. We recently found a setting on Hyper-V networking SR-IOV which improved things. Prior to changing this setting, we were seeing SIP OPTIONS packets/responses would occasionally take more than 3 seconds causing devices to drop and come back online. We have configured a similar system running at Amazon handling far more traffic and can't get the single CPU core to spike. Only small static pops during the calls. The sheer scale of the system is making it hard to diagnose the problem. Any thoughts on how to diagnose what is causing the single CPU core to spike? Any thoughts on how to diagnose the problem? Any other thoughts/comments? Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Hold / Transfer via AMI
We found no way to do this from AMI.
Very tied up on other projects, but if another developer wanted to look into
adding support for it, I believe it would be something along these lines….
int action_hold(struct mansession *s, const struct message *m)
{
const char *channelarg = astman_get_header(m, "Channel");
struct ast_channel *chan = NULL;
int res = -1;
if (ast_strlen_zero(channelarg))
{
astman_send_error(s, m, "No channel specified");
return 0;
}
chan = ast_channel_get_by_name(channelarg);
if (chan)
{
ast_log(LOG_DEBUG, "Putting channel %s on hold (0x%p)\n",
ast_channel_name(chan), chan);
if ((res = ast_indicate(chan, AST_CONTROL_HOLD)))
{
astman_send_error(s, m, "Failed to put channel on hold");
}
ast_channel_unref(chan);
}
else astman_send_error(s, m, "No such channel");
if (!res) astman_send_ack(s, m, "Hold");
return 0;
}
int action_unhold(struct mansession *s, const struct message *m)
{
const char *channelarg = astman_get_header(m, "Channel");
struct ast_channel *chan = NULL;
int res = -1;
if (ast_strlen_zero(channelarg))
{
astman_send_error(s, m, "No channel specified");
return 0;
}
chan = ast_channel_get_by_name(channelarg);
if (chan)
{
if ((res = ast_indicate(chan, AST_CONTROL_UNHOLD)))
{
astman_send_error(s, m, "Failed to remove channel from hold");
}
ast_channel_unref(chan);
}
else astman_send_error(s, m, "No such channel");
if (!res) astman_send_ack(s, m, "Unhold");
return 0;
}
const char *hold_synopsis = "Place Channel on Hold";
const char *hold_desc = "Place the channel specified by the 'Channel' argument
on Hold";
const char *unhold_synopsis = "Remove Channel from Hold";
const char *unhold_desc = "Remove the channel specified by the 'Channel'
argument from Hold";
ast_manager_register2("Hold", EVENT_FLAG_CALL, action_hold, AST_MODULE_SELF,
hold_synopsis, hold_desc);
ast_manager_register2("UnHold", EVENT_FLAG_CALL, action_unhold,
AST_MODULE_SELF, unhold_synopsis, unhold_desc);
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Wednesday, July 21, 2021 5:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Hold / Transfer via AMI
On Wed, Jul 21, 2021 at 7:39 AM Antony Stone
mailto:antony.st...@asterisk.open.source.it>>
wrote:
Hi.
From the lack of response to my question, I'm assuming that either:
a) putting a call on hold is not possible via AMI
or
b) everyone thinks it's so obvious that I should be able to see it for myself
Can anyone confirm one way or the other?
If it simply isn't possible, I'd like to put my efforts into exploring
alternative solutions instead of spending more time on AMI.
From a surface level I know of no way from AMI to control putting a call on and
off hold.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and
www.asterisk.org
This email is intended only for the use of the party to which it is addressed
and may contain information that is privileged, confidential, or protected by
law. If you are not the intended recipient you are hereby notified that any
dissemination, copying or distribution of this email or its contents is
strictly prohibited. If you have received this message in error, please notify
us immediately by replying to the message and deleting it from your computer.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Audio Sockets and media conversions
Has anyone used Audio Sockets with Amazon Transcribe? I'm still very new to the Audio Socket and have only just started looking at Amazon's Transcribe documentation so there may be something I am missing. I'm looking for live transcription of the call as opposed to post call transcription. Is there an easy way to convert the media from signed linear, 16-bit, 8kHz, mono PCM (little-endian) to one of the media formats Amazon Transcribe supports (mp3, mp4, wav, flac, ogg, amr, webm)? Obviously, if I write the data to a file, I can use sox to convert and then stream the conversion, but that is not a very nice scheme for live transcription. Any other suggestions for a better way to do this? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CPU spike
Running Asterisk 16.17.0 We have an interesting scenario where we see Asterisk CPU usage spike to the point the entire system is maxed out. There is a specific scenario where we have two ConfBridges and they are connected via a local channel. Everything is fine here. Callers <-> ConfBridge A <-> local channel <-> ConfBridge B <-> Callers We may be recording both ConfBridges for different groups. Again, all is fine. Where we run into a problem is when someone decides they need to record a response. We connect another local channel to the ConfBridge. All is fine. Callers <-> ConfBridge A <-> local channel <-> ConfBridge B <-> Callers <-> Local Channel Then, we initiate a record on the other end of the local channel. Callers <-> ConfBridge A <-> local channel <-> ConfBridge B <-> Callers <-> Local Channel (start recording) Suddenly, the CPU usage spikes to the point the system is starving and gradually gets worse. Looking at the debug log for Asterisk, we are seeing thousands of messages like this (roughly 8-10 sets every millisecond) [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007: Indicated Video Update [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007 [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007: Indicated Video Update [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007 [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007: Indicated Video Update [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007 [05/05 10:25:28.541] DEBUG[2981][C-0038] chan_pjsip.c: PJSIP/1003-0007: Indicated Video Update After this occurs, the messages in the debug output can also be a bit out of order for the date/time listed in the file. An interesting scenario, if we first start recording on the Local channel, then add it to the ConfBridge this CPU spike does not occur. We realize this is a bit of an extreme scenario, two ConfBridges connected with a local channel, and needing to connect another local channel to one of the ConfBridges and start recording. Unfortunately, it's a scenario we are stuck with due to a unique customer requirement. Any thoughts or suggestions? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Are there any settings for DTMF detection?
One of my co-workers just migrated from a Samsung phone to a Pixel 5 phone. An app on the phone dials into our asterisk. He has the same app installed on both and can move the SIM card between them. Call is answered and a prompt plays to collect digits. The app dials a number followed by a pound (example 1234#). Asterisk and our AMI based application sees this and everything is good so far. Another play begins to ask for another set of DTMFs. Milliseconds after asterisk starts playing, it is reporting the DTMF # ended a second time. The RTP is showing no new # being sent (1-2 seconds later phone dials digits for this second query) What I determined when listening to a comparison of a tcpdump and the RTP streams: Samsung has an interdigit timeout to the point you can hear a delay between the digits. Just from listening, probably 150-200 ms between digits. When listening to his Pixel 5 phone's RTP, the digits are speed dialed. From listening, either there is no pause between digits or it is extremely small. This is PJSIP and DTMFs are inband. Are there any configuration setting in asterisk to adjust the amount of time between digits? Any suggestions that I can try? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARI record question
Please disregard. I found my problem. We use a unique folder for the spool. Once I created the recording folder in our directory everything worked as expected. Dan From: asterisk-users On Behalf Of Dan Cropp Sent: Tuesday, August 11, 2020 9:24 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] ARI record question I'm attempting to run a test of the ARI recording of audio from the channel. When I send the record command, it's failing. curl -v -u asterisk:asterisk -X POST "http://locahost:8088/ari/channels/mychanntest.1/record?name=mytest=WAV=300=3; [08/11 09:14:13.290] WARNING[23806]: ari/resource_channels.c:812 ast_ari_channels_record: Unrecognized recording error: No such file or directory Based on this link, it sounds like the recording folder was the issue. I changed it to be writeable by anyone to see if that would resolve the issue but it has not. https://asterisk-app-dev.digium.narkive.com/oC8mEKF8/call-recording-using-ari My /var/spool/asterisk/recording folder is configured to allow writes sudo ls -l /var/spool/asterisk/ total 28 drwxr-xr-x 2 root root 4096 Sep 13 2019 dictate drwxr-xr-x 2 root root 4096 Sep 13 2019 meetme drwxr-xr-x 2 root root 4096 Sep 13 2019 monitor drwxrwxrwx 2 root root 4096 Sep 13 2019 recording drwxr-xr-x 2 root root 4096 Sep 13 2019 system drwxr-xr-x 2 root root 4096 Sep 13 2019 tmp drwxr-xr-x 2 root root 4096 Sep 13 2019 voicemail Any suggestions on what I am doing wrong? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ARI record question
I'm attempting to run a test of the ARI recording of audio from the channel. When I send the record command, it's failing. curl -v -u asterisk:asterisk -X POST "http://locahost:8088/ari/channels/mychanntest.1/record?name=mytest=WAV=300=3; [08/11 09:14:13.290] WARNING[23806]: ari/resource_channels.c:812 ast_ari_channels_record: Unrecognized recording error: No such file or directory Based on this link, it sounds like the recording folder was the issue. I changed it to be writeable by anyone to see if that would resolve the issue but it has not. https://asterisk-app-dev.digium.narkive.com/oC8mEKF8/call-recording-using-ari My /var/spool/asterisk/recording folder is configured to allow writes sudo ls -l /var/spool/asterisk/ total 28 drwxr-xr-x 2 root root 4096 Sep 13 2019 dictate drwxr-xr-x 2 root root 4096 Sep 13 2019 meetme drwxr-xr-x 2 root root 4096 Sep 13 2019 monitor drwxrwxrwx 2 root root 4096 Sep 13 2019 recording drwxr-xr-x 2 root root 4096 Sep 13 2019 system drwxr-xr-x 2 root root 4096 Sep 13 2019 tmp drwxr-xr-x 2 root root 4096 Sep 13 2019 voicemail Any suggestions on what I am doing wrong? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ARI Stop Playback
Thank you Joshua. That matches what I experienced last week.
I will build the string to play the number prompts using individual sound
prompts instead of using number.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, August 10, 2020 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ARI Stop Playback
On Thu, Aug 6, 2020 at 7:28 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a
number is being played?
Here is a test I am running. I am playing multiple portions (sounds and
numbers).
curl -v -u asterisk:asterisk -X POST
http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789
If I attempt to stop while any of the numbers portions is being played, it does
not stop the playback. I hear a skip or hiccup, but it continues playing.
When I send the same exact stop playback a few seconds later (after the 553
portion playing the sound portion), it does stop the playback. I have run this
test several times and it’s failing.
curl -v -u asterisk:asterisk -X DELETE
http://localhost:8088/ari/playbacks/3b38730d-7954-4c50-9cdc-e4643ffc8c62
I am using 16.12.0
Is there some reason a DELETE playback doesn’t work during a number portion or
is this a bug?
They are two separate paths and implementations, and I don't think the number
part has implemented the handling of stop and such.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com<http://www.sangoma.com> and
www.asterisk.org<http://www.asterisk.org>
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Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Thank you Jöran
That did the trick.
I had been trying to figure out how to do this without the json content and
couldn’t figure out how to do it.
Dan
From: asterisk-users On Behalf Of
Jöran Vinzens
Sent: Monday, August 10, 2020 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a
call and pass both the caller id name and number?
Hi Dan,
i did it wrong, sorry:
curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST
"http://localhost:8088/ari/channels/newChannelId;<http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world>
--data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,
"PJSIP_HEADER(add,P-Asserted-Identity)":"foobar"} }'
there was a bracket missing after the function of PJSIP_HEADER
BR
On Mon, Aug 10, 2020 at 3:57 PM Jöran Vinzens
mailto:vinz...@sipgate.de>> wrote:
Hi Dan,
i would do something like this (it is not a copy of what we are doing but an
example of how i would do it)
Important here is the "--data" and "-H" Option as well as the "variables"
section within the Body. I added the callerid function here as well as it is
the sample in the asterisk wiki.
curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST
"http://localhost:8088/ari/channels/newChannelId;<http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world>
--data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,
"PJSIP_HEADER(add,P-Asserted-Identity":"foobar"} }'
BR
Jöran
On Mon, Aug 10, 2020 at 3:43 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Hi Jöran,
Would it be possible to see an example using curl of how you are passing the
PAI Header through ARI create?
Dan
From: asterisk-users
mailto:asterisk-users-boun...@lists.digium.com>>
On Behalf Of Jöran Vinzens
Sent: Friday, August 7, 2020 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a
call and pass both the caller id name and number?
Hi Dan,
as far as PPI and PAI Header, we use the channel Vars in order to do that. In
Latest Asterisk you can set Channel vars within the create command in the Body.
Just set the PJSIP(add,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
BR
Jöran
On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
An additional follow-up question, if I need to set the P-Asserted-Identity on
the create (originate), is there a way to do this with ARI?
From: asterisk-users
mailto:asterisk-users-boun...@lists.digium.com>>
On Behalf Of Dan Cropp
Sent: Friday, August 7, 2020 11:51 AM
To: 'asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>'
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] With ARI, is it possible to create (originate) a call
and pass both the caller id name and number?
I’m trying to transition from AMI to ARI.
Running into a small hiccup when I try to create (originate a call) with the
caller id name and number
I can pass the Name and Number if the name has no spaces in it and it shows up
in my PhonerLite application.
curl -v -u asterisk:asterisk -X POST
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291<http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan%3c291>>
However, when the caller id name has a space in it, I can’t figure out how to
pass the name and number successfully. The following only displays asterisk
for the number and Dan for the name
curl -v -u asterisk:asterisk -X POST
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan
Cropp<291>
Here is an example of how we do this with AMI successfully.
Action: Originate
ActionID: S40
Channel: PJSIP/1003@1003
Exten: createcall
Context: IS
Priority: 1
Timeout: 6
CallerID: Dan Cropp <291>
Variable:
CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2
Async: true
Dan
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Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp mailto:d...@amtelco.com>> wrote: An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>' mailto:asterisk-users@lists.digium.com>> Subject: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? I’m trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291<http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan%3c291>> However, when the caller id name has a space in it, I can’t figure out how to pass the name and number successfully. The following only displays asterisk for the number and Dan for the name curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003@1003 Exten: createcall Context: IS Priority: 1 Timeout: 6 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jöran Vinzens - vinz...@sipgate.de<mailto:vinz...@sipgate.de> Telefon: +49 211-63 55 56-21 Telefax: +49 211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de<http://www.sipgate.de> - www.sipgate.co.uk<http://www.sipgate.co.uk> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Thank you Jöran I also figured out my problem with the caller id name/number. In case anyone else encounters the caller id name issue, replace the spaces in the name with control sequence for a space %20 From: asterisk-users On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp mailto:d...@amtelco.com>> wrote: An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>' mailto:asterisk-users@lists.digium.com>> Subject: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? I’m trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291<http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan%3c291>> However, when the caller id name has a space in it, I can’t figure out how to pass the name and number successfully. The following only displays asterisk for the number and Dan for the name curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003@1003 Exten: createcall Context: IS Priority: 1 Timeout: 6 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jöran Vinzens - vinz...@sipgate.de<mailto:vinz...@sipgate.de> Telefon: +49 211-63 55 56-21 Telefax: +49 211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de<http://www.sipgate.de> - www.sipgate.co.uk<http://www.sipgate.co.uk> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number? I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291<http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan%3c291>> However, when the caller id name has a space in it, I can't figure out how to pass the name and number successfully. The following only displays asterisk for the number and Dan for the name curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003@1003 Exten: createcall Context: IS Priority: 1 Timeout: 6 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291> However, when the caller id name has a space in it, I can't figure out how to pass the name and number successfully. The following only displays asterisk for the number and Dan for the name curl -v -u asterisk:asterisk -X POST http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003@1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ARI Stop Playback
Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a
number is being played?
Here is a test I am running. I am playing multiple portions (sounds and
numbers).
curl -v -u asterisk:asterisk -X POST
http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,sound:demo-instruct,number:123,number:456,number:789
If I attempt to stop while any of the numbers portions is being played, it does
not stop the playback. I hear a skip or hiccup, but it continues playing.
When I send the same exact stop playback a few seconds later (after the 553
portion playing the sound portion), it does stop the playback. I have run this
test several times and it's failing.
curl -v -u asterisk:asterisk -X DELETE
http://localhost:8088/ari/playbacks/3b38730d-7954-4c50-9cdc-e4643ffc8c62
I am using 16.12.0
Is there some reason a DELETE playback doesn't work during a number portion or
is this a bug?
Dan
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Re: [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?
Hi Joshua, Thanks for responding. Please disregard, I just figured out the using the originate approach solved my problem. curl -v -u asterisk:asterisk -X POST "http://asterisk:asterisk@localhost:8088/ari/channels/mycallerid.1?endpoint=local/1000@mycontext=hello-world2=1000=mycontext=1=mycallerid.1=mycallerid.2=ulaw=30; Have a good day! Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Thursday, August 6, 2020 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI? On Thu, Aug 6, 2020 at 1:52 PM Dan Cropp mailto:d...@amtelco.com>> wrote: I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. What does the console show is actually happening with the channel? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?
I understand how to control the first local channel, but an having trouble
getting the second local channel to enter stasis.
I setup have the following extensions.conf to handle 1000 (basically had it
setup so if first stasis not there try second, but believe second channel never
processes the dial plan so even if second line was hello-world2 it would not
matter.
[mycontext]
exten => 1000,1,NoOp()
same => n,Stasis(hello-world)
same => n,GotoIf($[${STASISSTATUS}=FAILED]?IS_hello_world2:stasis_done)
same => n(IS_hello_world2),Stasis(hello-world2)
same => n(stasis_done),Hangup()
For testing, I am using curl
curl -v -u asterisk:asterisk -X POST
http://asterisk:asterisk@localhost:8088/ari/channels/create?endpoint=local/1000@mycontext
=hello-world2=mycontext=mycallerid.1=mycallerid.2=ulaw
I see the primary call on my hello-world2 stasis, but I never see anything for
the second leg of the local channel. I have an AMI connection and see both
channels there, but the send local channel seems to not process the dial plan.
Can someone tell me if this is possible and if so what am I doing wrong?
Here is the AMI for the two local channel
Event: Newchannel
Privilege: call,all
Channel: Local/1000@mycontext-000b;1
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: mycontext
Exten: 1000
Priority: 1
Uniqueid: mycallerid.1
Linkedid: mycallerid.1
Event: Newchannel
Privilege: call,all
Channel: Local/1000@mycontext-000b;2
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum:
CallerIDName:
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: mycontext
Exten: 1000
Priority: 1
Uniqueid: mycallerid.2
Linkedid: mycallerid.1
< {
"type": "ChannelDialplan",
"timestamp": "2020-08-06T11:37:50.531-0500",
"dialplan_app": "Stasis",
"dialplan_app_data": "hello-world2",
"channel": {
"id": "mycallerid.1",
"name": "Local/1000@mycontext-000b;1",
"state": "Down",
"caller": {
"name": "",
"number": ""
},
"connected": {
"name": "",
"number": ""
},
"accountcode": "",
"dialplan": {
"context": "mycontext",
"exten": "1000",
"priority": 1,
"app_name": "Stasis",
"app_data": "hello-world2"
},
"creationtime": "2020-08-06T11:37:50.531-0500",
"language": "en"
},
"asterisk_id": "00:15:5d:8e:01:38",
"application": "hello-world2"
}
< {
"variable": "STASISSTATUS",
"value": "",
"type": "ChannelVarset",
"timestamp": "2020-08-06T11:37:50.531-0500",
"channel": {
"id": "mycallerid.1",
"name": "Local/1000@mycontext-000b;1",
"state": "Down",
"caller": {
"name": "",
"number": ""
},
"connected": {
"name": "",
"number": ""
},
"accountcode": "",
"dialplan": {
"context": "mycontext",
"exten": "1000",
"priority": 1,
"app_name": "Stasis",
"app_data": "hello-world2"
},
"creationtime": "2020-08-06T11:37:50.531-0500",
"language": "en"
},
"asterisk_id": "00:15:5d:8e:01:38",
"application": "hello-world2"
}
< {
"type": "StasisStart",
"timestamp": "2020-08-06T11:37:50.531-0500",
"args": [],
"channel": {
"id": "mycallerid.1",
"name": "Local/1000@mycontext-000b;1",
"state": "Down",
"caller": {
"name": "",
"number": ""
},
"connected": {
"name": "",
"number": ""
},
"accountcode": "",
"dialplan": {
"context": "mycontext",
"exten": "1000",
"priority": 1,
"app_name": "Stasis",
"app_data": "hello-world2"
},
"creationtime": "2020-08-06T11:37:50.531-0500",
"language": "en"
},
"asterisk_id": "00:15:5d:8e:01:38",
"application": "hello-world2"
}
Here are the AMI
Event: Newchannel
Privilege: call,all
Channel: Local/1000@mycontext-000b;1
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: mycontext
Exten: 1000
Priority: 1
Uniqueid: mycallerid.1
Linkedid: mycallerid.1
Event: Newchannel
Privilege: call,all
Channel: Local/1000@mycontext-0007;2
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum:
CallerIDName:
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: mycontext
Exten: 1000
Priority: 1
Uniqueid: mycallerid.2
Linkedid: mycallerid.1
Dan
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Re: [asterisk-users] Stir Shaken is upon us
Thanks Josh for clarifying! I'd assumed it would be backported but didnt want to just assume :) Thanks Matt for doing the video! (hint hint theres a load of good content THIS WEEK over on the commcon youtube channel but that's all I'll say about that here) On Mon, Jul 13, 2020 at 9:55 AM Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier > wrote: > >> On 13.07.20 at 00:17 Joshua C. Colp wrote: >> > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote: >> > >> >> Asterisk 18 will have support based on this asterisk update Matt F did >> for >> >> CommCon's sponsor slots >> >> >> >> https://youtu.be/eas1csaX-wc >> >> >> >> >> > As well support will go into Asterisk 16 and 17 as well. It's just been >> > under active development so we've been waiting for that to finish before >> > bringing it back into those versions. >> >> One more question, >> what about the pjsip pcap support? Will it be backported to Asterisk 16, >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain. >> > > It can't be backported ... because it already is. :D This support is > actually in the latest releases of 13, 16, and 17. > > >> >> BTW: what about the (encrypted) RTP packets? Will they be dumped, too? >> > > Not yet supported but certainly something we'd like to see as well as the > RTCP, ICE, STUN, TURN, and DTLS packets. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken is upon us
Asterisk 18 will have support based on this asterisk update Matt F did for CommCon's sponsor slots https://youtu.be/eas1csaX-wc On Sun, 12 Jul 2020, 22:44 Steve Edwards, wrote: > On Sun, 12 Jul 2020, Saint Michael wrote: > > > WORLDWIDE EMERGENCY > > Again? > > > The code below needs to be executed before any SIP or PJSIP call > > destined to the US network, or soon no call will terminate. This is > > called Stir-Shaken, a new law from the FCC. If this is not working the > > whole Asterisk industry will crash, vanish, be gone. > > Seen any little chickens lately? > > According to 'https://www.fcc.gov/call-authentication': > > "In March 2020, the Commission adopted new rules requiring all originating > and terminating voice service providers to implement caller ID > authentication using STIR/SHAKEN technological standards in the Internet > Protocol (IP) portions of their networks by June 30, 2021." > > So this is a provider issue, not an end user issue and 'June 30, 2021' > doesn't sound like 'soon.' If this is legit, why haven't my providers said > squat? > > > Server = 208.73.232.47 > > So why do you want everybody to send you their call metadata? What's your > endgame? Generate leads to call to pitch your service? Poach clients? > > Sorry if I sound cynical. It's 2020 and I'm fresh out of "F's." > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts?
Thank you Joshua. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Wednesday, May 27, 2020 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts? On Wed, May 27, 2020 at 5:30 PM Dan Cropp mailto:d...@amtelco.com>> wrote: I have an endpoint with multiple phones registered as aor contacts. When I attempt to originate a call it will only ring one of the phones. Is it possible to ring multiple phones as a single endpoint. First phone to answer wins the call and all others terminated? Again, using AMI. No, you have to dial a Local channel that then uses PJSIP_DIAL_CONTACTS. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to have a single AMI originate ring multiple contacts?
I have an endpoint with multiple phones registered as aor contacts. When I attempt to originate a call it will only ring one of the phones. Is it possible to ring multiple phones as a single endpoint. First phone to answer wins the call and all others terminated? Again, using AMI. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ?
its fairly painless now https://github.com/nimbleape/asterisk-dialogflow-rtp-audioserver https://github.com/nimbleape/asterisk-dialogflow-ari-bridge Theres 2 repos - one for the ari bridge - 1:1 call -> external media and another for talking to dialogflow but theres no reason that couldnt go out to a radio station stream for example.. On Wed, May 6, 2020 at 8:55 PM Jonathan H wrote: > Thanks Dan - might have to scratch my head over that one for a while! > The phrase "you make your own RTP server" has made me all twitchy ;) > > Jonathan > > On Wed, 6 May 2020 at 07:21, Dan Jenkins wrote: > >> Hi Jonathan, >> >> I'd probably go down the external media route in the ARI now - you make >> your own RTP server and provide your own RTP back to asterisk >> >> On Sun, 3 May 2020, 13:07 Jonathan H, wrote: >> >>> Way back in 2016 the only way to allow callers to listen in to a stream >>> "at will" was to do the following: >>> >>> moh.conf >>> >>> [radio] >>> mode=custom >>> application=/usr/bin/mplayer https://example.com/stream.mp3 -quiet -ao >>> pcm:file=/dev/stdout -af volume=5,resample=8000,channels=1,format=alaw >>> >>> extensions.conf >>> >>> exten => radio,1,Verbose(1, Entered radio context) >>> same => n,Set(VOLUME(TX)=1) >>> same => n,WaitExten(27006,m(radio)) >>> same => n,Goto(#,1) >>> >>> It kind of works, but two problems here: >>> It's pulling data 24x7, giving the radio host artificial stats - all >>> rather needless as maybe one or two people might listen for 10 mins each in >>> a day. >>> And even though mplayer seems to stay up and running all the time, >>> sometimes Asterisk will stop listening on that pipe and everything needs a >>> restart (random, less than once a week). >>> >>> Is there a more modern/sensible way of achieving the same, just ensuring >>> that stream plays if someone listens, isn't playing when no-one is >>> listening, and listening can be exited with a specified key? >>> >>> Thanks! >>> >> -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ?
Hi Jonathan, I'd probably go down the external media route in the ARI now - you make your own RTP server and provide your own RTP back to asterisk On Sun, 3 May 2020, 13:07 Jonathan H, wrote: > Way back in 2016 the only way to allow callers to listen in to a stream > "at will" was to do the following: > > moh.conf > > [radio] > mode=custom > application=/usr/bin/mplayer https://example.com/stream.mp3 -quiet -ao > pcm:file=/dev/stdout -af volume=5,resample=8000,channels=1,format=alaw > > extensions.conf > > exten => radio,1,Verbose(1, Entered radio context) > same => n,Set(VOLUME(TX)=1) > same => n,WaitExten(27006,m(radio)) > same => n,Goto(#,1) > > It kind of works, but two problems here: > It's pulling data 24x7, giving the radio host artificial stats - all > rather needless as maybe one or two people might listen for 10 mins each in > a day. > And even though mplayer seems to stay up and running all the time, > sometimes Asterisk will stop listening on that pipe and everything needs a > restart (random, less than once a week). > > Is there a more modern/sensible way of achieving the same, just ensuring > that stream plays if someone listens, isn't playing when no-one is > listening, and listening can be exited with a specified key? > > Thanks! > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Webrtc and iOS devices
I honestly couldn't tell you if it would resolve it but there aren't many people going to be willing to help problem solve anything if you're running 13 - you'll get more support on 17 for example. Very easy to bring up a new instance or VM in the grand scheme of things to test the theory and get it working on most recent version of Asterisk On Tue, Apr 28, 2020 at 11:37 AM Teijo wrote: > Hello, > > > Currently audio conference. Should upgrading Asterisk from 13 to newer > version resolve webrtc/iOS problem? > > > Best regards, > > > Teijo > > Dan Jenkins kirjoitti 28.4.2020 klo 12.18: > > First things first, upgrade from 13 - WebRTC has moved a long a lot since > then. If you can't upgrade everything to 13 then run another asterisk > specifically for WebRTC and bridge to your other Asterisk > > Is this just an audio conference? > > On Sun, Apr 26, 2020 at 10:21 PM Teijo > wrote: > > > Hello, > > > Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc > and iOS (version 13.4.1) with Safari or any other browser working > properly in confbridge conference calls? I hope my Asterisk webrtc > related settings are not totally wrong, because several other browsers > from Windows seem to work. > > > Best regards, > > > Teijo > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at:https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Webrtc and iOS devices
First things first, upgrade from 13 - WebRTC has moved a long a lot since then. If you can't upgrade everything to 13 then run another asterisk specifically for WebRTC and bridge to your other Asterisk Is this just an audio conference? On Sun, Apr 26, 2020 at 10:21 PM Teijo wrote: > Hello, > > > Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc > and iOS (version 13.4.1) with Safari or any other browser working > properly in confbridge conference calls? I hope my Asterisk webrtc > related settings are not totally wrong, because several other browsers > from Windows seem to work. > > > Best regards, > > > Teijo > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max calls per box
I've heard of people having thousands of channels on a box Dovid. Not how I would personally do it myself but if you've already got the hardware. And think about if you can simplify the deployment by accessing the sound files via http so you only have them in one place On Thu, 19 Mar 2020, 00:50 Dovid Bender, wrote: > Hi, > > We have a requirement to build a cluster that can handle 30k calls. The > system is going to play one of 15,000 sound files. In the past we had no > issue with Asterisk doing a few hundred calls. When we went above that > Asterisk melted (this is going back quite a few years). FreeSwitch ended up > doing the trick. Does anyone have experience with recent versions of > Asterisk and if it's capable of handling 2k calls per box where it has to > play one of 15k files? I know that hardware plays a factor but we know we > can throw more cores at it. > > TIA. > > Regards, > > Dovid > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can an ARI Bridge support more than 2 channels the way a ConfBridge can?
Thank you Joshua. That’s awesome! Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Monday, February 24, 2020 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can an ARI Bridge support more than 2 channels the way a ConfBridge can? On Mon, Feb 24, 2020 at 8:07 PM Dan Cropp mailto:d...@amtelco.com>> wrote: We are looking to migrate from AMI to ARI. We currently rely heavily on ConfBridges for multiple party support. Is it possible to add more than 2 channels? If so, is there a limit? Or a way to configure the limit? Yes, you can add more than 2. The bridge will automatically transition to allowing it. There is no limit, it uses the same mixing as ConfBridge itself. Additional features - like enforcing a limit - would be up to the ARI application to implement. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can an ARI Bridge support more than 2 channels the way a ConfBridge can?
We are looking to migrate from AMI to ARI. We currently rely heavily on ConfBridges for multiple party support. Is it possible to add more than 2 channels? If so, is there a limit? Or a way to configure the limit? Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on pjsip.conf and aors
Thanks Joshua From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, February 14, 2020 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on pjsip.conf and aors On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp mailto:d...@amtelco.com>> wrote: I have the following configuration… [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1004 password = SuperSecretProbation [1004] type = endpoint context = IS transport = transport1 auth = auth3 aors = aor3 accountcode = 3 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no moh_passthrough = yes disallow = all allow = ulaw acl = acl1 When a register attempt is received, asterisk outputs… [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not found for endpoint '1004' If I change the aor3 to be 1004, everything works. As in [aor3] becomes [1004] and in the endpoint change aors = aor3 to be aors = 1004 Is there a setting I’m missing to allow the endpoint named 1004 to use an auth that doesn’t have the same 1004 name? There isn't a configuration option. AOR is a SIP concept, and in fact when you send a REGISTER you state which AOR you are registering to. Your REGISTER is therefore saying "add me to AOR 1004". Since it's not saying "add me to aor3" it doesn't work. Some devices allow you to specify while others just assume that everything uses your username. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on pjsip.conf and aors
I have the following configuration... [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1004 password = SuperSecretProbation [1004] type = endpoint context = IS transport = transport1 auth = auth3 aors = aor3 accountcode = 3 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no moh_passthrough = yes disallow = all allow = ulaw acl = acl1 When a register attempt is received, asterisk outputs... [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not found for endpoint '1004' If I change the aor3 to be 1004, everything works. As in [aor3] becomes [1004] and in the endpoint change aors = aor3 to be aors = 1004 Is there a setting I'm missing to allow the endpoint named 1004 to use an auth that doesn't have the same 1004 name? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character?
We think the reason hangup request does not work is because it's an alias. We are using the default cli_aliases.conf which supports hangup request calling channel request hangup. We think the alias detects the arguments correctly, but then passes the argument without the escaped character (or double quotes) to the real CLI command. We found that double quotes also work for channel request hangup "..." It would be nice if the CLI alias commands would pass arguments (or at least some arguments) with double quotes. That would allow the hangup request alias to work exactly as the channel request hangup that it calls. -Original Message- From: asterisk-users On Behalf Of Dan Cropp Sent: Thursday, January 16, 2020 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character? Thanks Doug. Turns out if using hangup request does not work with the escaped character CLI> hangup request PJSIP/1003\ a-0007 Usage: channel request hangup | Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel. If 'all' is specified instead of a channel name, all channels will see the hangup request. However, channel request hangup ... does support the escaped character. CLI> channel request hangup PJSIP/1003\ a-0007 Requested Hangup on channel 'PJSIP/1003 a-0007' Thank you for the help. Dan -Original Message- From: asterisk-users On Behalf Of Doug Lytle Sent: Thursday, January 16, 2020 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character? >>> Is there some control character(s) for the CLI to interpret everything in >>> between as a single argument? I think you can typically use tab completion when working with spaces or you can escape the space with a back slash For example Doug Lytle would be Doug\ Lytle Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character?
Thanks Doug. Turns out if using hangup request does not work with the escaped character CLI> hangup request PJSIP/1003\ a-0007 Usage: channel request hangup | Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel. If 'all' is specified instead of a channel name, all channels will see the hangup request. However, channel request hangup ... does support the escaped character. CLI> channel request hangup PJSIP/1003\ a-0007 Requested Hangup on channel 'PJSIP/1003 a-0007' Thank you for the help. Dan -Original Message- From: asterisk-users On Behalf Of Doug Lytle Sent: Thursday, January 16, 2020 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character? >>> Is there some control character(s) for the CLI to interpret everything in >>> between as a single argument? I think you can typically use tab completion when working with spaces or you can escape the space with a back slash For example Doug Lytle would be Doug\ Lytle Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] From the CLI, how can I hangup a channel name that includes a space character?
I have a customer who named their endpoint to include a space (example, 1003 a) >From the CLI, I want to hangup a channel on this endpoint >From core show channels concise, I see the channel name includes the space PJSIP/1003 a-0002 I realize the space is interpreted as an argument separator, so my first attempt below doesn't work. I have tried the following and all fail. hangup request PJSIP/1003 a-0002 hangup request 'PJSIP/1003 a-0002' hangup request "PJSIP/1003 a-0002" hangup request PJSIP/1003%20a-0002 hangup request PJSIP/1003 Is there some control character(s) for the CLI to interpret everything in between as a single argument? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk support one-legged transfers with external switches?
We have a customer using Avaya. Currently, they are using chan_sip. We are working to migrate them to PJSIP. I have not been filled in on the exact scenario. I suspect they have some auto forward feature on the number. Rather than their Avaya transferring internally, they tell Asterisk to transfer to a number (with the Asterisk IP). Doesn't seem correct to me, but it's pretty common for the major switch vendors to do things incorrectly. We originate from Asterisk to the Avaya endpoint. Send INVITE (with Allow REFER) Receive 100 Trying Receive 180 Ringing Receive 200 O Transmit ACK About a half second later, we receive a REFER from Avaya. I'm not sure if this is normal, but they send us a number to Refer-To @ asterisk IP address. From what I'm being told, they want a one-legged transfer where Asterisk would perform a transfer of this call to the number given @ the Avaya IP address. [11/12 15:09:00.680] VERBOSE[1338] chan_sip.c: SIP transfer to extension 12345@ABC by num...@xyz.org:5060 Transmit 202 Accepted Transmit NOTIFY "SIP/2.0 503 Service Unavailable (can't handle one-legged xfers)" 1) Is there a way to disable sending the REFER in the INVITE's Allow? (chan_sip currently). If not on chan_sip, how about PJSIP? Our theory is if Avaya doesn't receive the Allow: REFER they would do the transfer themselves. 2) Would PJSIP help in any way? Any other thoughts on how to solve this? My next inclination is something like Kamailio between Avaya and Asterisk. This type of transfer seems very insecure. It's basically, Avaya able to tell us to transfer to any number they want (without any restriction). Have a great day! Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?
Thank you Anthony I was not involved in the entire conference call with the customer. They plan to send the recordings to another company to anaylze the operators behavior. I believe you are correct about them left channel of the recording to be the operator and the right channel to be the 1-3 other participants (from ConfBridge). I will give the MixMonitor a try. Have a great day! Da -Original Message- From: asterisk-users On Behalf Of Antony Stone Sent: Friday, November 1, 2019 5:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono? On Friday 01 November 2019 at 22:29:28, Dan Cropp wrote: > We have a customer who wants us to record anywhere from 2-4 > participants on a call in stereo (as opposed to mono) quality audio. I'm assuming you mean you want to get one stereo recording for each participant, where the left channel is the participant and the right channel is the rest of the conference? If that's not correct, what do you want the two channels of a stero recording to contain? > We are using asterisk 16.6.1 > We are also currently using AMI/AsyncAGI and ConfBridge to bring the > parties together. I believe recording in the various file formats > (based on extension), it's always recording in mono quality. > > My one thought is to transition to using ARI Bridge (instead of > ConfBridge) and streamed audio using ExternalMedia. Then have a media > server capture the external media packets, stripping the payload > information and write directly to a file. Would that audio be of ulaw stereo > or mono? Suppose it *is* stereo - what would you expect the two channels to contain? It sounds like you want one single stereo recording of a conference with multiple participants. Are they all using stereo telephones and generating two-channel audio into the conference - or what?? > Any suggestions? How about a simple MixMonitor with btr options on each participant who dials in, before they get placed into the conference? Then the t channel should be the participant and the r channel should be the rest of the conference. Regards, Antony. -- What do you call a dinosaur with only one eye? A Doyouthinkesaurus. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?
We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to transition to using ARI Bridge (instead of ConfBridge) and streamed audio using ExternalMedia. Then have a media server capture the external media packets, stripping the payload information and write directly to a file. Would that audio be of ulaw stereo or mono? Any suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime PJSIP max_streams' issues
Thanks Joshua. This turned out to be my mistake. Quiet variable was enabled on the User and needed to be disabled. It's been at least a couple years since I wrote e-mails for my coworkers and forgot that setting. Have a great day! Dan -Original Message- From: asterisk-users On Behalf Of Joshua C. Colp Sent: Tuesday, October 22, 2019 4:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime PJSIP max_streams' issues On Tue, Oct 22, 2019, at 4:21 PM, Ahmed Chohan wrote: > Hi, > > I'm currently using Asterisk 16.4.0 cert version and working on webrtc. > For configuration perspective, I'm pretty much done with it but here > the real issue I'm currently facing i.e. when setting parameters > max_audio_streams & max_video_streams to any positive greater than 0 > integer value in realtime (DB) of any endpoints. After running command > "pjsip show endpoint 100101" it shows '0' but when setting as > 'NULL' in DB, showing output to 1 for both parameters. > > Furthermore, in AOR section, the max_connection is set to 1 for each > endpoints. The configuration option for there is max_contacts. > Please advise, for this issue. What database are you using? What type is the column? Do any other fields exhibit the problem? -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge and sound prompts
Just to add additional information, it seems this approach works with the CONFBRIDGE user variables just not the bridge variables... Action: SetVar^M ActionID: C81^M Channel: PJSIP/1003-0003^M Variable: CONFBRIDGE(user,announcement)^M Value: en/confbridge-join^M For the bridge variable, I have a second call in the same ConfBridge. I expected to hear the confbridge-join when I added channel PJSIP/1003-0003 to the ConfBridge, but there is no sound played to the other channel. I do end up in a ConfBridge with both calls able to hear each other so I know the confbridge is working. Did the naming for the CONFBRIDGE bridge variables changed? Dan From: asterisk-users On Behalf Of Dan Cropp Sent: Tuesday, October 22, 2019 3:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ConfBridge and sound prompts We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-0003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does anyone know if the ConfBridge sound variable setting approach changed? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge and sound prompts
We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-0003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does anyone know if the ConfBridge sound variable setting approach changed? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Async AGI seeing a big delay in events on 16.1.1 but not 16.3.0
We are using AsyncAGI with AMI. On a customer box running asterisk 16.1.1, we are seeing times where asterisk logs indicate it's started the agi:async extension. Event: Newexten ... Application: AGI AppData: agi:async It's taking 2 or more seconds before we see the Event: AsyncAGIStart I have not seen this before. Looking at boxes we have running inhouse (asterisk 16.3.0), I see this is taking about 7 millisecond. Anyone have any suggestions on why this may occur? Any chance what we are seeing on 16.1.1 but not 16.3.0 is because of Joshua's fix for 16.3.0? 2019-03-25 06:34 + [4d8cd2efbe] Joshua Colp * manager: Use separate lock for session event notification. When notifying a manager session that new events were available the same lock was used that was also held when doing things within the session (such as sending events out). If the manager session blocked for a period of time this would cause a back up of messages in Stasis and would also block any other sessions from receiving events. This change adds a separate lock to the manager session which is strictly used for notifying it that new events are available. ASTERISK-28350 Change-Id: Ifbcac007faca9ad0231640f5e82a6ca9228f261b -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
