search for: native_rtp

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_monitor it works,...

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168....

...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 some reason I...

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 joined 'simple_bridge...

On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <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 some reason I have oneway audio. how can I fix...

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launched AGI Script /pbx/agi.php > -- AGI Script Executing Application: (Dial) Options:...

...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 switch to the native_rtp...

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com> wrote: > Well, it breaks audio for all NAT endpoints, how can I fix this? > Local (packet to packet) bridging should not do that. Remote (direct media) can do that. Can you confirm - by looking at a verbose level 4 log - how Asterisk is bridging the two channels? -- Matthew Jordan Digium, Inc. | Director of Technology

Le 2017-07-05 18:51, Joshua Colp a ?crit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of type softmix even if two channels only are involved). I can do this with ARI of course, but handlin...

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of type softmix even if two channels only are involved). I can do this with ARI of course, but ha...

...;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 > Remotely bridged 'SIP/6003-00000001' and 'SIP/6004-00000000' - media will flow directly between them > Remotely bridged 'SIP/6003-00000001' and 'SIP/6004-00000000' - media will flow directly between them > 0x7f5968006760 -- Probation p...

...ot 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 an...

...16:23:56] app_queue.c: Local/mysip692 at CallFromQueue-0000081a;1 is ringing Could it be that it is because my Queue member '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/my...

...00004, 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 ms. Do you think it is a bug ? I would tend to s...

...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 > Remotely bridged 'SIP/63000-00000012' and 'SIP/63009-00000013' - media will flow directly between them I added in general section of sip.conf (chan_sip in use) directrtpsetup=no directmedia=no but yet I still see "media will flow directly between them". HOW d...

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

...y softmix does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 find_best_technology: Bridge technology simple_bridge does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 find_best_technology: Bridge technology native_rtp does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:505 find_best_technology: Chose bridge technology holding_bridge [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:771 bridge_base_init: Bridge 9f437397-4864-4351-bf29-b37e6ccacf12: calling holding_bri...

...o '2' from '0' on channel 'PJSIP/easybellPJSIP-00000009' [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip_t38.c:673 defer_incoming_sdp_stream: Deferring incoming SDP stream on PJSIP/easybellPJSIP-00000009 for peer re-invite [2017-06-15 07:43:57] DEBUG[25198][C-00000004]: bridge_native_rtp.c:348 native_rtp_bridge_compatible_check: Bridge 'f8e63423-8fc7-44e4-a33d-c55b7d87d30f' can not use native RTP bridge as it was forbidden while getting details [2017-06-15 07:43:57] DEBUG[25198][C-00000004]: bridge.c:506 find_best_technology: Bridge technology native_rtp is not compatible w...

...SK-21205 - [patch] dundi_read_result crash due to negative number (Reported by Jaco Kroon) * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI...

...SK-21205 - [patch] dundi_read_result crash due to negative number (Reported by Jaco Kroon) * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI...