search for: res_pjsip_transport_websocket

On Tue, Oct 23, 2018 at 1:35 PM Dan Cropp <dan at amtelco.com> wrote: > The res_pjsip_transport_websocket failing to load seems to be a conflict > with the chan_sip.so loading. > > When I make the chan_sip.so not load, res_pjsip_transport_websocket.so > does load. > > We have customers who need chan_sip and chan_pjsip, so we need to load > both. Is there a way to make the res_pjsi...

I just noticed this upon startup since updating from 15.6.1 to 16.0.0 - do any of these matter? [Oct 18 12:12:18] WARNING[4489]: loader.c:2228 load_modules: Some non-required modules failed to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: res_pjsip_transport_websocket declined to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: cdr_sqlite3_custom declined to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: cel_sqlite3_custom declined to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modules: pbx_ael declined to load. I d...

...specific protocol. You can have multiple. > And how is this accomplished? If I put an extra transport section in pjsip.conf it does not show with "pjsip show transports", I only see the first. I also get an error in the log file: [Feb 15 12:42:36] VERBOSE[4377] loader.c: Loading res_pjsip_transport_websocket.so. [Feb 15 12:42:36] VERBOSE[4377] loader.c: res_pjsip_transport_websocket.so => (PJSIP WebSocket Transport Support) [Feb 15 12:52:42] DEBUG[5793] res_sorcery_config.c: Using bucket size of '1' for objects of type 'transport' from 'pjsip.conf' [Feb 15 12:52:42] ERROR[5...

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]:

I'm learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side. I noticed that the Asterisk wiki mentions the need for res_pjsip_transport_websocket, so does that mean Asterisk requires the signaling to occur over a websocket? If I used a SIPJS fork which places the signaling over UDP (eg https://github.com/cwysong85/sipjs-udp) will it just be a regular SIP client and I shouldn't have to configure anything special in Asterisk, just re...

...[Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'res_pjsip_dlg_options.so': /usr/lib64/asterisk/modules/res_pjsip_dlg_options.so: undefined symbol: ast_sip_add_header [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'res_pjsip_transport_websocket.so': /usr/lib64/asterisk/modules/res_pjsip_transport_websocket.so: undefined symbol: ast_sip_create_serializer [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_contact.so': /usr/lib64/asterisk/modules/func_pjsip_contact.so: undefined symb...

...e transaction state change event is RX_MSG [Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: The current inv state is CALLING [Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: Received response [Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: Response is 100 Trying [Jun 8 12:28:09] DEBUG[4180] res_pjsip_transport_websocket.c: Response msg 180/INVITE/cseq=30101 (rdata0x7f057801c9e0) re-writing Contact URI from 192.0.2.235:0;transport=wss to 10.215.144.48:64842;transport=ws [Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_distributor.c: Searching for serializer associated with dialog dlg0x7f0578069118 for Response msg 180...

...res_pjsip_pidf.so load => res_pjsip_pubsub.so noload => res_pjsip_refer.so noload => res_pjsip_registrar_expire.so noload => res_pjsip_registrar.so load => res_pjsip_rfc3326.so load => res_pjsip_sdp_rtp.so load => res_pjsip_session.so noload => res_pjsip_t38.so noload => res_pjsip_transport_websocket.so load => res_pjsip_acl.so I need to identify the caller, and since there is no way to see the IP address of the caller in the dial plan, at least I need to force Asterisk to match it to a peer. Any ideas?

...Hajek Michal) * ASTERISK-28032 - Realtime queuemembers are not updated during retry phase (Reported by lvl) * ASTERISK-27988 - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) * ASTERISK-28022 - res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) Improvements made in this release: ----------------------------------- * ASTERI...

...deric LE FOLL) * ASTERISK-28032 - Realtime queuemembers are not updated during retry phase (Reported by lvl) * ASTERISK-27988 - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) Improvements made in this release: ----------------------------------- * ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI (Reported by Alex...

...uration file exists (Reported by Ronald Raikes) * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by J??rgen H) * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) * ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by To...

...gistration (Reported by Frederic LE FOLL) * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by J??rgen H) * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) * ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by To...

...sip_publish_asterisk.so res_pjsip_pubsub.so res_pjsip_refer.so res_pjsip_registrar_expire.so res_pjsip_registrar.so res_pjsip_rfc3326.so res_pjsip_sdp_rtp.so res_pjsip_send_to_voicemail.so res_pjsip_session.so res_pjsip_sips_contact.so res_pjsip.so res_pjsip_t38.so res_pjsip_transport_management.so res_pjsip_transport_websocket.so res_pjsip_xpidf_body_generator.so Asterisk CLI : [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error loading module 'res_pjsip_registrar.so': /usr/lib64/asterisk/modules/res_pjsip_registrar.so: undefined symbol: ast_sip_location_retrieve_aor_contacts_nolock [A...

...box basis (Reported by Mark Scholten) * ASTERISK-26598 - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) * ASTERISK-17067 - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by J??rgen H) * ASTERISK-25628 - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) * ASTERISK-26825 - pjsip.conf.sample: use...

...box basis (Reported by Mark Scholten) * ASTERISK-26598 - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) * ASTERISK-17067 - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by J??rgen H) * ASTERISK-25628 - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) * ASTERISK-26774 - core: Playback URL fai...

...Hajek Michal) * ASTERISK-28032 - Realtime queuemembers are not updated during retry phase (Reported by lvl) * ASTERISK-27988 - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentific...

...Hajek Michal) * ASTERISK-28032 - Realtime queuemembers are not updated during retry phase (Reported by lvl) * ASTERISK-27988 - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentific...

...ts/" subfolder (Reported by Jonathan Harris) - [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>] - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) - [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>] - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by J?rgen H) - [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/ASTERISK-25628>] - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagi...

...ts/" subfolder (Reported by Jonathan Harris) - [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>] - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) - [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>] - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by J?rgen H) - [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/ASTERISK-25628>] - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagi...