search for: bridge_native_rtp

Hi there; 2 linux boxes and Windows all report an error and the archive is not extractable. Wget reports the size as follows: 2017-02-14 08:36:21 (7.29 MB/s) - ?asterisk-14-current.tar.gz? saved [40653605/40653605] It starts un-tarring but then.... asterisk-14.3.0/bridges/bridge_native_rtp.c asterisk-14.3.0/sounds/ asterisk-14.3.0/sounds/asterisk-core-sounds-en-gsm-1.5.tar.gz gzip: stdin: invalid compressed data--format violated tar: Unexpected EOF in archive tar: Unexpected EOF in archive tar: Error is not recoverable: exiting now -------------- next part -------------- An HTML att...

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

...cipher but it is not valid (Reported by Joshua Colp) * ASTERISK-24262 - AMI CoreShowChannel missing several output fields and event documentation (Reported by Mitch Claborn) * ASTERISK-24356 - PJSIP: Directed pickup causes deadlock (Reported by Richard Mudgett) * ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge (Reported by Jonathan Rose) * ASTERISK-24384 - chan_motif: format capabilities leak on module load error (Reported by Corey Farrell) * ASTERISK-24385 - ch...

...cipher but it is not valid (Reported by Joshua Colp) * ASTERISK-24262 - AMI CoreShowChannel missing several output fields and event documentation (Reported by Mitch Claborn) * ASTERISK-24356 - PJSIP: Directed pickup causes deadlock (Reported by Richard Mudgett) * ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge (Reported by Jonathan Rose) * ASTERISK-24384 - chan_motif: format capabilities leak on module load error (Reported by Corey Farrell) * ASTERISK-24385 - ch...

Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was

...[Nov 21 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...

...narios. The bridging framework is smart enough to pick the best bridging technology available for the situation. If the situation changes during a call, the bridging framework can change the bridge technology to support the new situation. * bridge_simple is for normal two party communication. * bridge_native_rtp is a special case of two party bridge were both parties use RTP for media exchange. The native technology allows for direct media. * bridge_softmix is for multi-party bridges where you can have 1 to n users communicating in a conference. As you found out, bridge_softmix can be used as a fallback...

...anged to '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...

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++

...narios. The bridging framework is smart enough to pick the best bridging technology available for the situation. If the situation changes during a call, the bridging framework can change the bridge technology to support the new situation. * bridge_simple is for normal two party communication. * bridge_native_rtp is a special case of two party bridge were both parties use RTP for media exchange. The native technology allows for direct media. * bridge_softmix is for multi-party bridges where you can have 1 to n users communicating in a conference. As you found out, bridge_softmix can be used as a fallback...

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

...enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int (Reported by Walter Doekes) * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra channel (Reported by Niklas Larsson) * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible (Reported by Yaniv Simhi) * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' (Reported by...

...enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int (Reported by Walter Doekes) * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra channel (Reported by Niklas Larsson) * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible (Reported by Yaniv Simhi) * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' (Reported by...

...* ASTERISK-25201 - Crash in PJSIP distributor on already free'd threadpool (Reported by Matt Jordan) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully started when completing attended transfer (Reported by Joshua Colp) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_chec...

...Ramirez Norambuena) * ASTERISK-27264 - res_pjsip_session: Crashes after sending PRACK and receiving 200 OK (Reported by Daniel Heckl) * ASTERISK-27260 - [pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36 (Reported by Daniel Heckl) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-16898 - SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 (Reported by Marcello Ceschia) * ASTERISK-27274 - RTCP needs bette...

...(Reported by Ross Beer) * ASTERISK-27289 - A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability (Reported by Ross Beer) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call l...

...(Reported by Ross Beer) * ASTERISK-27289 - A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) * ASTERISK-27283 - Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-27257 - bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis - Prescom) * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability (Reported by Ross Beer) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call l...

...udgett) * ASTERISK-25183 - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel (Reported by Matt Jordan) * ASTERISK-25201 - Crash in PJSIP distributor on already free'd threadpool (Reported by Matt Jordan) * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully started when completing attended transfer (Reported by Joshua Colp) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-25146 - DNS: Create system level resolver (Reported by Joshua Colp) * ASTERISK...

...udgett) * ASTERISK-25183 - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel (Reported by Matt Jordan) * ASTERISK-25201 - Crash in PJSIP distributor on already free'd threadpool (Reported by Matt Jordan) * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully started when completing attended transfer (Reported by Joshua Colp) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_chec...