search for: res_pjsip_session

...8e0' [Jun 8 12:28:09] DEBUG[4180] res_srtp.c: local_key64 2Rbo7TRiuRAnS0IYJeSn0ELEYAVnkOVCUwou7pxO len 40 [Jun 8 12:28:09] DEBUG[4180] res_pjsip_sdp_rtp.c: Stream msid: 0x7f0578077610 audio 23eb03ca-f0ee-406a-b7cd-5fb19fc33fa2 ddca7927-ff8d-45ab-a61f-9474f8b7a9df [Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: Method is INVITE [Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '10.215.144.48' [Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_resolver.c: Transport type for target '10.215.144.48' is 'ws' [Jun 8 12:28:09] DEBUG[4180] res...

...t=0 0 m=audio 60132 RTP/AVP 8 101 c=IN IP4 83.143.192.141 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcpping:F:1253794:125379478 m=video 60136 RTP/AVP 100 c=IN IP4 83.143.192.141 a=inactive a=rtcpping:F:1253795:125379578 [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The state change pertains to the endpoint 'srv_d15140(PJSIP/srv_d15140-00000255)' [2016-02-10 22:58:17] DEBUG[27034] res_pjsip_session.c: The inv session still has an i...

Asterisk Project Security Advisory - AST-2020-001 Product Asterisk Summary Remote crash in res_pjsip_session Nature of Advisory Denial of service Susceptibility Remote authenticated sessions Severity Moderate Exploits Known No...

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

...ridge.c:1046 smart_bridge_operation: Bridge f8e63423-8fc7-44e4-a33d-c55b7d87d30f is already using the new technology. [2017-06-15 07:43:57] DEBUG[25171]: pjproject:0 <?>: endpoint .Request msg INVITE/cseq=24421 (tdta0x7f5f180a98f8) created. [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip_session.c:971 ast_sip_session_refresh: Sending session refresh SDP via re-INVITE to 91 [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip_session.c:2501 handle_outgoing_request: Method is INVITE [2017-06-15 07:43:57] DEBUG[25171]: pjproject:0 <?>: inv0x7f5f18019fc8 .Sending Request msg INVITE/cseq...

...=> +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from 'from-twilio' (UDP:mysillyApp.pstn.twilio.com:5060) to extension '+17775551212' rejected because extension not found in context 'from-twilio-remove-plus'. [Dec 2 15:09:29] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from 'from-t...

...my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' (UDP:10.10.10.80:5062) to extension '<xxxxxxxxxx>' rejected because extension not found in context 'gv-voice'. I thought "s" (as in start ?) would match any extension sent to that context. sean

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 >> +++

...e are any updates since 18.17.1 to see if I can spot any fixes or recent commits. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000055e7c091ed95 in __ao2_ref (user_data=user_data at entry=0x1, delta=delta at entry=1, tag=tag at entry=0x0, file=file at entry=0x7f773800e012 "res_pjsip_session.c", line=line at entry=3639, func=func at entry=0x7f7738011d20 <__PRETTY_FUNCTION__.38105> "ast_sip_dialog_get_session") at astobj2.c:501 501 astobj2.c: No such file or directory. [Current thread is 1 (Thread 0x7f772c1a1700 (LWP 124120))] (gdb) bt #0 0x000055e7c091ed95...

...today to the res_pjsip_nat.c file, but not sure if that would apply to the issue. Any suggestions for where I should look or ask? (gdb) bt #0 0x000055e7c091ed95 in __ao2_ref (user_data=user_data at entry=0x1, delta=delta at entry=1, tag=tag at entry=0x0, file=file at entry=0x7f773800e012 "res_pjsip_session.c", line=line at entry=3639, func=func at entry=0x7f7738011d20 <__PRETTY_FUNCTION__.38105> "ast_sip_dialog_get_session") at astobj2.c:501 #1 0x00007f773800a0da in ast_sip_dialog_get_session (dlg=dlg at entry=0x7f777415de48) at res_pjsip_session.c:3639 #2 0x00007f773800d3e...

...f0b02334 in negotiate_incoming_sdp_stream (session=0x7fba3c031200, session_media=<value optimized out>, sdp=<value optimized out>, stream=<value optimized out>) at res_pjsip_t38.c:703 #2 0x00007fba0499ccf6 in handle_incoming_sdp (session=0x7fba3c031200, sdp=0x7fba3c0adfb8) at res_pjsip_session.c:243 #3 0x00007fba0499e650 in session_inv_on_rx_offer (inv=0x7fba3c0504e8, offer=0x7fba3c0adfb8) at res_pjsip_session.c:3009 #4 0x00007fba44b1b501 in inv_check_sdp_in_incoming_msg (inv=0x7fba3c0504e8, tsx=0x7fba08006878, rdata=0x7fba3c0b00a8) at ../src/pjsip-ua/sip_inv.c:2110 #5 0x00007fba44b20...

...g to SIP/IAX and packets now arrive and go where they should. Two problems remain. 1. Still can't register my phone The username and password are correct. I don't know what else to try. 2. Asterisk can't find the extension in my inbound context. [May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite:  voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because extension not found in context 'voipms-inbound'. I changed the name of the context in pjsip's  to 'voipms-inbound' and removed reference to '[mycontext]' from pjsip.c...

On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp <dcropp at amtelco.com> wrote: > I have a customer who just encountered a crash while running Asterisk > 18.17.1 version. > > > > I’m trying to adapt to the changes so not sure where best to look or how > to possibly report this. > > > > I started by going through >

...ound = yes endpoint/trust_id_outbound = yes endpoint/trust_connected_line = no endpoint/send_connected_line = no endpoint/context = trunkhandler_pbx-sip-t1 Attached sip sessions and debug log... the only thing I found interesting was finding a lack of a log item We SHOULD be seeing: DEBUG[XXXXX] res_pjsip_session.c: (null session): Setting external media address to 152.X.Y.Z This message is clearly lacking from the debug session where the incorrect media address is sent.  But there's not enough detail in the debugs to see why this decision was not made to use external_media_address By default we use...

...www.asterisk.org <http://www.asterisk.org/> Hello how can I disable all modules related to pjsip in modules.conf ?? I have now : [modules] autoload=yes preload => res_config_mysql.so noload => pbx_gtkconsole.so noload => res_pjsip.so noload => res_pjsip_pubsub.so noload => res_pjsip_session.so noload => chan_pjsip.so noload => res_pjsip_exten_state.so noload => res_pjsip_log_forwarder.so load => res_musiconhold.so noload => chan_alsa.so noload => chan_oss.so noload => chan_console.so This does not make the CLI erros go away. I still have the idea that pjsip is...

...ch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) Bugs fixed in this release: ----------------------------------- * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 (Reported by Ross Beer) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-29043 - app_queue: Leave empty sometimes...

...s (Reported by Nick French) * ASTERISK-29136 - config: Sample features.conf incorrectly includes " around sound files (Reported by Benjamin M.) * ASTERISK-29123 - logger.conf.sample missing comment mark on line 115 (Reported by Andrew Siplas) * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 (Reported by Ross Beer) * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF (Reported by under) * ASTERISK-29108 - resource_endpoints.c : Memory...

.... (Reported by Alexander Traud) * ASTERISK-27957 - PJSIP proposes ICE candidates on answer even if not in offer (Reported by Torrey Searle) * ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared. (Reported by Alexander Traud) * ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE attribute truncated (Reported by Kevin Harwell) * ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire (Reported by Alexei Gradinari) * ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason head...

...orted by Ronald Raikes) * ASTERISK-27372 - ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. (Reported by dtryba) * ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs (Reported by dtryba) * ASTERISK-27270 - cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured (Reported by Tzafrir Cohen) * ASTERISK-25266 - Application Originate returns SUCCESS to...

...offer (Reported by Torrey Searle) * ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS keep-alives. (Reported by Alexander Traud) * ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST' undeclared. (Reported by Alexander Traud) * ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE attribute truncated (Reported by Kevin Harwell) * ASTERISK-27956 - res_pjsip_pubsub: segfault in function publish_expire (Reported by Alexei Gradinari) * ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason head...