search for: bridge_channel

...[Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined 'simple_bridge' basic-bridge <ab233c52-249f-4370-bcdd-3eb9af7c6cfd> [Sep 17 11:30:05] VERBOSE[23420][C-00000051] bridge_channel.c: Channel SIP/mysippeer-00000108 joined 'simple_bridge' basic-bridge <ab233c52-249f-4370-bcdd-3eb9a...

...600,SIP/100) In the new Asterisk-version, the ParkAndAnnounce application gets called, but the call isn't parked. The only error I can see in the messages file is a DEBUG entry saying that the channel "failed to join Bridge", like this: [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge_channel.c:1994 bridge_channel_internal_join: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: 0x16e3768(SIP/vpn-sbc-00000001) failed to join Bridge Anyone else that has tried to convert old parking functionality into Asterisk 12.0.0 ? features.conf: parkswitch => *#,callee/caller,Macro(parkswitch)...

...in new stack [2016-09-08 21:00:25] VERBOSE[18771][C-0000066c] app_dial.c: Called SIP/0021628990XXX at SBC002_VirginMedia [2016-09-08 21:00:27] VERBOSE[18771][C-0000066c] app_dial.c: SIP/SBC002_VirginMedia-00000f67 answered SIP/201-boxoffice-00000f66 [2016-09-08 21:00:27] VERBOSE[18771][C-0000066c] bridge_channel.c: Channel SIP/201-boxoffice-00000f66 joined 'simple_bridge' basic-bridge <00bd58c3-3bce-4f1b-9d79-11eb96f37260> [2016-09-08 21:00:27] VERBOSE[18779][C-0000066c] bridge_channel.c: Channel SIP/SBC002_VirginMedia-00000f67 joined 'simple_bridge' basic-bridge <00bd58c3-3bce-4f1...

...annel.c: DTMF end '#' received on SIP/xxx-00000004, duration 257 ms [Dec 13 11:56:16] DTMF[18193][C-00000005] channel.c: DTMF begin emulation of '#' with duration 257 queued on SIP/xxx-00000004 *--- **SIP/xxx-00000004 **is hanged up:* [Dec 13 11:56:19] VERBOSE[18193][C-00000005] bridge_channel.c: Channel SIP/xxx-00000004 left 'native_rtp' basic-bridge <4a5905ac-29f8-41c5-9981-e9d0f4966c56> [Dec 13 11:56:19] DTMF[18193][C-00000005] bridge_channel.c: DTMF end '#' simulated to bridge 4a5905ac-29f8-41c5-9981-e9d0f4966c56 because SIP/xxx-00000004 left.? Duration 3012...

No, there is no Music On Hold starting and the bad thing is the call duration reported by asterisk was just few seconds while the call duration reported by the provider was few thousand seconds, the max allowed. So they will be able to terminate the call on the asterisk side and have it run on the provider side. Leandro 2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at

...rence I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do: [2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c: Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge <626258fc-0649-45c7-b0d3-630a06d2c91b>     Why are some calls using the simple bridge and others switch to the native_rtp bridge?  Could this be a codec problem?  How can I prevent the switch? -- Telecomunicaci...

...r 'mysip692' is occupied in another bridge (call) ? This I see in the logs just before the Call Queue starts calling the queue member : [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged 'SIP/mysip-00004e6a' and 'SIP/incoming-00004e63' in stack [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-00004e63 left 'native_rtp' basic-bridge <fed056d3-669a-493d-a4bd-f0d9ab0102a7> [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-00004e6a left 'native_rtp' basic-bridge <fed056d3-669a-493d-a4bd-f0d9ab0102a7> A bit too coincidal, no ? So t...

...051] app_dial.c: > SIP/myprovider-0000010b is making progress passing it to > SIP/mysippeer-00000108 > [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: > SIP/myprovider-0000010b answered SIP/mysippeer-00000108 > [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: > Channel SIP/myprovider-0000010b joined 'simple_bridge' > basic-bridge <ab233c52-249f-4370-bcdd-3eb9af7c6cfd> > [Sep 17 11:30:05] VERBOSE[23420][C-00000051] bridge_channel.c: > Channel SIP/mysippeer-00000108 joined 'simple_bridge' basic-bridge &...

...94 430525994 IN IP4 52.22.22.22 s=Asterisk PBX 16.13.0 c=IN IP4 52.22.22.22 t=0 0 m=audio 17678 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:230 a=sendrecv <------------> [2020-09-19 23:42:22] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel IAX2/Downstream-26055 joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel SIP/Upstream-00021a0d joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c...

...ing enabled. The asterisk messages file has this... (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]....) [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f [08/12 15:30:55.881] VERBOSE[6921] bridge_channel.c: Channel CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f joined 'softmix' base-bridge <23340bca-6823-4c70-a395-e3b092aeb671>...

...94 430525994 IN IP4 52.22.22.22 s=Asterisk PBX 16.13.0 c=IN IP4 52.22.22.22 t=0 0 m=audio 17678 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:230 a=sendrecv <------------> [2020-09-19 23:42:22] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel IAX2/Downstream-26055 joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel SIP/Upstream-00021a0d joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c...

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on

...94 430525994 IN IP4 52.22.22.22 s=Asterisk PBX 16.13.0 c=IN IP4 52.22.22.22 t=0 0 m=audio 17678 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:230 a=sendrecv <------------> [2020-09-19 23:42:22] VERBOSE[15154][C-00021a1f] bridge_channel.c: Channel IAX2/Downstream-26055 joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c24e9> [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] bridge_channel.c: Channel SIP/Upstream-00021a0d joined 'simple_bridge' basic-bridge <0d377050-bca3-4db8-81e0-f677e37c...

...alls is the following line: > > [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge > 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge > technology to native_rtp > > Most calls just do: > > [2020-09-07 18:13:56] VERBOSE[15293][C-00000084] bridge_channel.c: > Channel PJSIP/1028-0000012a joined 'simple_bridge' basic-bridge > <626258fc-0649-45c7-b0d3-630a06d2c91b> > > Why are some calls using the simple bridge and others switch to the > native_rtp bridge? Could this be a codec problem? How can I prevent > the sw...

...libasteriskssl.so.1 -> libasteriskssl.so [LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o astobj2_container.o astobj2_hash.o astobj2_rbtree.o audiohook.o autochan.o autoservice.o backtrace.o bridge.o bridge_after.o bridge_basic.o bridge_channel.o bridge_roles.o bucket.o callerid.o ccss.o cdr.o cel.o channel.o channel_internal_api.o chanvars.o cli.o codec.o codec_builtin.o config.o config_options.o core_local.o core_unreal.o crypt.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o endpoints.o enum.o event.o features.o feat...

...ed by HZMI8gkCvPpom0tM) * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk (Reported by Kevin Harwell) * ASTERISK-24626 - Voicemail passwords not being stored in ARA (Reported by Paddy Grice) * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait in bridge_channel.c (Reported by George Joseph) * ASTERISK-24544 - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll (Reported by George Joseph) * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus (Reported by Mat...

...ed by HZMI8gkCvPpom0tM) * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk (Reported by Kevin Harwell) * ASTERISK-24626 - Voicemail passwords not being stored in ARA (Reported by Paddy Grice) * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait in bridge_channel.c (Reported by George Joseph) * ASTERISK-24544 - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll (Reported by George Joseph) * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus (Reported by Mat...

...ager show connected" output - #509: [bug]: res_pjsip: Crash when looking up transport state in use - #513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG - #520: [improvement]: menuselect: Use more specific error message. - #530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels - #539: [bug]: Existence of logger.xml causes linking failure

...ager show connected" output - #509: [bug]: res_pjsip: Crash when looking up transport state in use - #513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG - #520: [improvement]: menuselect: Use more specific error message. - #530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels - #539: [bug]: Existence of logger.xml causes linking failure