search for: rtp_engin

...o us via a particular provider. The SIP traces look perfectly normal both on the ingress to us and egress to another telco. No additional sip messages after the call has been answered until the BYE is received. However in the asterisk logs I am getting this :- 2014-02-05 13:45:03 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44 2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging SIP/vmpubopensips1-001dfd...

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough

....c: Splitting '109.60.222.253' into... [2016-02-10 22:58:17] DEBUG[31024] netsock2.c: ...host '109.60.222.253' and port ''. [2016-02-10 22:58:17] DEBUG[31024] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffddc198f58' [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7ffdc23b7320 [2016-02-10 22:58:17] DEBUG[31024] rtp_engine.c: Don't have a default tx payload type 96 format for m type on 0x7ffdc23b73...

...tes in 1647238 allocations in file stasis_channels.c 1217035109 bytes in 10229320 allocations in file json.c 240064732 bytes in 1765287 allocations in file stasis_message.c 56402000 bytes in 1762250 allocations in file taskprocessor.c 26125344 bytes in 203171 allocations in file rtp_engine.c 17308827 bytes in 307848 allocations in file stasis_cache.c 9548128 bytes in 35482 allocations in file stasis_bridges.c 3923172 bytes in 92169 allocations in file res_rtp_asterisk.c 3099771 bytes in 29185 allocations in file strings.c Next minute it was already lik...

......host 'x.x.x.x' and port ''. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP c=IN IP4 x.x.x.x... OK. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_...

...invite: Initializing initreq for method INVITE - callid cf2990ba- 3f12-3d9e-adb6-52889c414ed3 Using INVITE request as basis request - cf2990ba-3f12-3d9e-adb6-52889c414ed3 Found peer 'vr1a882' for 'vr1a882' from 2.2.2.2:8558 [May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:421 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9072064' [May 19 16:47:43] DEBUG[14160][C-00000007]: res_rtp_asterisk.c:2437 ast_rtp_new: Allocated port 17304 for RTP instance '0x9072064' [May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:430...

...t/raw-udp"/><transport xmlns="http://www.google.com/transport/p2p"/></ses:session></iq> [Oct 24 21:18:49] DEBUG[2405] res_jabber.c: JABBER: Handling paktype IQ [Oct 24 21:18:49] DEBUG[2405] chan_gtalk.c: The client is guest for alloc [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] res_rtp_asterisk.c: Allocated port 11262 for RTP instance '0x1b86bc8' [Oct 24 21:18:49] DEBUG[2405] rtp_engine.c: RTP instance '0x1b86bc8' is setup and ready to go [Oct 24 21:18:49...

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp => debug,verbose(5) so i mean in https://github.com/asterisk/asterisk/blob/master/configs/samples/pjproject.conf.sample by few examples try to explain  what usefull info i can get set [startup] log_level=6 type=start...

...22:56] DEBUG[17418]: chan_sip.c:25057 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 4870fab953c16a611b9248584748fe59 at 127.0.0.1:0 - INVITE (No RTP) [Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9ded230' [Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 19718 for RTP instance '0x9ded230' [Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:353 ast_rtp_instance_new: RTP...

...nd port '5062'. [Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting '192.168.2.172' into... [Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport: ...host '192.168.2.172' and port ''. [Jan 28 23:03:32] DEBUG[1654]: rtp_engine.c:345 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xb7507ae8' [Jan 28 23:03:32] DEBUG[1654]: res_rtp_asterisk.c:499 ast_rtp_new: Allocated port 18530 for RTP instance '0xb7507ae8' [Jan 28 23:03:32] DEBUG[1654]: rtp_engine.c:354 ast_rtp_instance_new: RTP...

....o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a ->...

...679] pbx.c: -- Executing [21411615 at test_outgoing:1] Dial("Local/21411615 at test_outgoing-cb92;2", "khomp/gpstn/21411615,120,Ttr") in new stack [2012-08-01 14:29:44] DEBUG[6679] devicestate.c: device 'Khomp/B1C0' state '1' [2012-08-01 14:29:44] DEBUG[6679] rtp_engine.c: Can't find native functions for channel 'Khomp/B1C0-0.0' [2012-08-01 14:29:44] DEBUG[6679] rtp_engine.c: Seeded SDP of 'Khomp/B1C0-0.0' with that of 'Local/21411615 at test_outgoing-cb92;2' [2012-08-01 14:29:44] DEBUG[6679] channel.c: Not copying variable DIALEDTIME....

...p_dial.so load => func_callerid.so load => func_cut.so load => func_logic.so [global] Since Asterisk 16 (Debian Buster version) I have a dependency problem, where res_rtp_asterisk.so is dependent on res_pjproject.so: When I try to make a call: [Dec 22 22:00:55] ERROR[6093][C-00000001]: rtp_engine.c:474 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? mng*CLI> module load res_rtp_asterisk Unable to load module res_rtp_asterisk Command 'module load res_rtp_asterisk' failed. [Dec 22 22:03:39] ERROR[28261]: loader.c:170 module_load_error: res_rtp_asterisk loade...

...612089.24, detail: [Jun 8 12:28:09] DEBUG[4607][C-00000003] stasis.c: Topic 'cache:48/channel:1591612089.24': 0x7f056000dde0 created [Jun 8 12:28:09] DEBUG[4607][C-00000003] channel.c: Channel 0x7f056000ac40 'PJSIP/4053-00000002' allocated [Jun 8 12:28:09] DEBUG[4607][C-00000003] rtp_engine.c: Can't find native functions for channel 'IAX2/migration-8417' [Jun 8 12:28:09] DEBUG[4180] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f05780618e0' [Jun 8 12:28:09] VERBOSE[4607][C-00000003] app_dial.c: Called PJSIP/4053 [Jun 8 12:28:09] DEBUG[4180]...

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

....o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a ->...

...02] DEBUG[3582] chan_sip.c: Stopping retransmission on '09e2d5011044076a652a743a737d64a4 at 192.168.2.135:5060' of Request 103: Match Found [Nov 12 16:53:02] DEBUG[3582] chan_sip.c: Destroying SIP dialog 09e2d5011044076a652a743a737d64a4 at 192.168.2.135:5060 [Nov 12 16:53:02] DEBUG[3582] rtp_engine.c: Destroyed RTP instance '0x8985958' [Nov 12 16:53:02] DEBUG[3573] devicestate.c: No provider found, checking channel drivers for SIP - 1003 [Nov 12 16:53:02] DEBUG[3573] chan_sip.c: Checking device state for peer 1003 [Nov 12 16:53:02] DEBUG[3573] devicestate.c: Changing state for SIP...

...nto this revision. In my opinion there are the following alternatives in order to get the Fritz card running with Asterisk: A) Get chan-capi to compile: Unfortunately my C knowledge seems insufficent for this. I have found out that there is noch rtp.h in the asterisk source dir, only a file named rtp_engine.h. Changing the include accordingly unfortunately only fixes the very first error. I can't make any sense out of the second error (error: invalid operands to binary == (have "union <anonymous>" and "int")), as, in line 803, the left-hand operand of the == is no union,...

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at