search for: res_pjsip_transport_websockets

...39;s sip.conf (websocket_enabled = no) or don't load the res_pjsip_transport_websocket.so module. >From sip.conf.sample: ;websocket_enabled = true ; Set to false to prevent chan_sip from listening to websockets. This ; is needed when using chan_sip and res_pjsip_transport_websockets on ; the same system. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181023/08249829/attachment.html>

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

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> 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 >

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

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

I have one single endpoint [inhouse](endpoint-basic) type=endpoint and one section like this [indentify] endpoint=inhouse match=X.Z.Y.X But when I call in from IP X.Z.Y.X, it does not match my peer. It tries to send the call the "default" context, and the number is not found. I know that I should have loaded this module, res_pjsip_endpoint_identifier_ip.so and it is loaded. In fact,

The Asterisk Development Team would like to announce the release of Asterisk 15.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.24.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem.

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release: