search for: simple_bridge

...RTP TOS bits 184 [Mar 26 12:11:04] == Using SIP RTP CoS mark 5 [Mar 26 12:11:04] -- Called SIP/7051 [Mar 26 12:11:05] -- SIP/7051-00000253 is ringing [Mar 26 12:11:11] -- SIP/7051-00000253 answered SIP/pabx-e1-00000252 [Mar 26 12:11:11] -- Channel SIP/pabx-e1-00000252 joined 'simple_bridge' basic-bridge <b1c97b75-bd5f-4762-96dd-7aa68c472827> [Mar 26 12:11:11] -- Channel SIP/7051-00000253 joined 'simple_bridge' basic-bridge <b1c97b75-bd5f-4762-96dd-7aa68c472827> Now, extension 7051 places the call on hold and calls 7003, who answers: [Mar 26 12:11:17]...

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017 -- Channel SIP/101-00000017...

.../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-3eb9af7c6cfd> Call ends : [Sep 17 11:34:36] VERBOSE[23420][C-00...

...t /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called PJSIP/99/sip:99 at 192.168.1.73:5060 -- PJSIP/99-00000023 is ringing -- PJSIP/99-00000023 answered PJSIP/304-00000022 -- Channel PJSIP/304-00000022 joined 'simple_bridge' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> -- Channel PJSIP/99-00000023 joined 'simple_bridge' basic-bridge <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> > Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from simple_bridge technology to native_rtp...

I have a hosted server. I have TWO different locations what have phones. Chicago and Indiana If I send audio direct from server to Chicago I hear it - same with indiana. But if indiana calls chicago - NO AUDIO. I see this in the CLI -- Channel SIP/63009-00000013 joined 'simple_bridge' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> -- Channel SIP/63000-00000012 joined 'simple_bridge' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> > Bridge 475050e7-9d99-43f0-a9bf-7aa581a97fd9: switching from simple_bridge technology to native_rtp...

...that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference 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...

...;PJSIP/sbcs-00000092", "PJSIP/+31xxxxxxxxx at uplink") in new stack -- Called PJSIP/+31xxxxxxxxx at uplink -- PJSIP/uplink-00000093 is making progress passing it to PJSIP/sbcs-00000092 -- PJSIP/uplink-00000093 answered PJSIP/sbcs-00000092 -- Channel PJSIP/uplink-00000093 joined 'simple_bridge' basic-bridge <3b25c543-13a3-4d74-b2fe-7122a1cfe4a4> -- Channel PJSIP/sbcs-00000092 joined 'simple_bridge' basic-bridge <3b25c543-13a3-4d74-b2fe-7122a1cfe4a4> > Bridge 3b25c543-13a3-4d74-b2fe-7122a1cfe4a4: switching from simple_bridge technology to native_rtp > R...

Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp in endpoints table ?direct_media? sets to ?no? on all endpoints but it doesn?t help. if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_moni...

...p_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-4f1b-9d79-11eb96f37260> [2016-09-08 21:00:28] VERBOSE[18771][C-0...

...uilding JSON from '{s: s, s: s}': Invalid UTF-8 string. [Mar 5 11:13:14] ERROR[3526]: json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. -- SIP/asterisk-0000000c answered OOH323/kanbaikal-6 -- Channel OOH323/kanbaikal-6 joined 'simple_bridge' basic-bridge <d751932e-2e26-4671-8fcc-048b20156ec9> -- Channel SIP/asterisk-0000000c joined 'simple_bridge' basic-bridge <d751932e-2e26-4671-8fcc-048b20156ec9> > 0x7fa9d800d610 -- Probation passed - setting RTP source address to 192.168.166.55:3098 [Mar...

...ordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries switch simple_bridge to native_rtp >> >> Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge >> technology to native_rtp >> >> in endpoints table ?direct_media? sets to ?no? on all endpoints but it >> doesn?t help. >> >> if native_rtp not work for...

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

ok switching to "Console/default" does show the text --- <("<) --- Call to device 'default' on console from 'default' <2564286000> --- (>")> --- --- <("<) --- Auto-answered --- (>")> --- However I don't hear any audio. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

...efault' on console from 'MyName Here' <2564286000> --- (>")> --- --- <("<) --- Auto-answered --- (>")> --- -- Called Console/default -- Console/default answered SIP/devgeis_to_nuc11cdev2-00000000 -- Channel Console/default joined 'simple_bridge' basic-bridge <e6f6e4e9-aa1f-452d-883d-65c4d93c59b1> [Sep 8 08:07:10] WARNING[282457][C-00000001]: chan_console.c:651 console_indicate: Don't know how to display condition 26 on Console/default -- Channel SIP/devgeis_to_nuc11cdev2-00000000 joined 'simple_bridge' basic-bri...

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

...s}': Invalid UTF-8 string. >> [Mar 5 11:13:14] ERROR[3526]: json.c:704 ast_json_vpack: Error >> building JSON from '{s: s, s: s}': Invalid UTF-8 string. >> -- SIP/asterisk-0000000c answered OOH323/kanbaikal-6 >> -- Channel OOH323/kanbaikal-6 joined 'simple_bridge' basic-bridge >> <d751932e-2e26-4671-8fcc-048b20156ec9> >> -- Channel SIP/asterisk-0000000c joined 'simple_bridge' >> basic-bridge <d751932e-2e26-4671-8fcc-048b20156ec9> >> > 0x7fa9d800d610 -- Probation passed - setting RTP source &gt...

...Script Executing Application: (Dial) Options: > (PJSIP/99/sip:99 at 192.168.1.73:5060,20) > -- Called PJSIP/99/sip:99 at 192.168.1.73:5060 > -- PJSIP/99-00000023 is ringing > -- PJSIP/99-00000023 answered PJSIP/304-00000022 > -- Channel PJSIP/304-00000022 joined 'simple_bridge' basic-bridge > <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> > -- Channel PJSIP/99-00000023 joined 'simple_bridge' basic-bridge > <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> > > Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from > simple_bridge tec...

...-- Called SIP/6003 -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 answered SIP/6004-00000000 -- Channel SIP/6004-00000000 joined 'simple_bridge' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07> -- Channel SIP/6003-00000001 joined 'simple_bridge' basic-bridge <2a01fb30-96e2-48b7-baaa-c2f172127c07> > Bridge 2a01fb30-96e2-48b7-baaa-c2f172127c07: switching from simple_bridge technology to native_rtp...

...ne channel: | -- Started music on hold, class 'default', on channel 'DAHDI/i1/6000-436' -- <SIP/5082-00000046> Playing 'pbx-transfer.ulaw' (language 'ru') -- Stopped music on hold on DAHDI/i1/6000-436 -- Channel DAHDI/i1/6000-436 left 'simple_bridge' basic-bridge <f5100b94-4c34-40af-9c92-7e129c2bdb00> -- Executing [0 at fromtransfer:1] NoOp("DAHDI/i1/6000-436", "") in new stack -- Executing [0 at fromtransfer:2] Gosub("DAHDI/i1/6000-436", "dynamic-nway,5082,1") in new stack --...

...ssing 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-3eb9af7c6cfd> > > Cal...