Marek Červenka
2015-Aug-13 19:48 UTC
[asterisk-users] simultaneous use of chan_sip/chan_pjsip
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 hardphones/softphones (sip udp 5060) > - chan_pjsip - for webrtc > > > You can use both.. you will want to make sure your bind addresses and > ports don't conflict. > > Why not use chan_pjsip for all SIP connectivity?because it's BIG change for production environment we have own web gui for config generation and we need move to chan_pjsip safely -- --------------------------------------- Marek Cervenka ====================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150813/486e6b15/attachment.html>
Marek Červenka
2015-Aug-27 10:33 UTC
[asterisk-users] simultaneous use of chan_sip/chan_pjsip
Dne 13.8.2015 v 21:48 Marek ?ervenka napsal(a):> 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 hardphones/softphones (sip udp 5060) >> - chan_pjsip - for webrtc >> >> >> You can use both.. you will want to make sure your bind addresses and >> ports don't conflict. >> >> Why not use chan_pjsip for all SIP connectivity? > > because it's BIG change for production environment > we have own web gui for config generation and we need move to > chan_pjsip safelyfor the record it looks like the simultaneous use is not possible with this configuration sip.conf [general] transport=udp ... pjsip.conf [global] [transport-wss] type=transport protocol=wss bind=0.0.0.0 ... module res_pjsip_transport_websocket.so is not loaded and load fails *CLI> module load res_pjsip_transport_websocket.so [Aug 27 12:31:23] DEBUG[13977]: res_pjsip.c:1918 register_service_noref: Registered SIP service WebSocket Transport Module (0xb51353e0) [Aug 27 12:31:23] DEBUG[13977]: res_pjsip.c:1950 unregister_service_noref: Unregistered SIP service WebSocket Transport Module Unable to load module res_pjsip_transport_websocket.so Command 'module load res_pjsip_transport_websocket.so' failed. *CLI> module show like websoc Module Description Use Count Status Support Level res_http_websocket.so HTTP WebSocket Support 2 Running extended res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support 0 Not Running core -- --------------------------------------- Marek Cervenka ====================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150827/266115c6/attachment.html>
Joshua Colp
2015-Aug-27 10:37 UTC
[asterisk-users] simultaneous use of chan_sip/chan_pjsip
On 15-08-27 07:33 AM, Marek ?ervenka wrote:> Dne 13.8.2015 v 21:48 Marek ?ervenka napsal(a): >> Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): >>> On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka >>> <<mailto:cervajs at fpf.slu.cz>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 hardphones/softphones (sip udp 5060) >>> - chan_pjsip - for webrtc >>> >>> >>> You can use both.. you will want to make sure your bind addresses and >>> ports don't conflict. >>> >>> Why not use chan_pjsip for all SIP connectivity? >> >> because it's BIG change for production environment >> we have own web gui for config generation and we need move to >> chan_pjsip safely > > for the record > > it looks like the simultaneous use is not possibleSimultaneous use of everything but the websocket support is possible. There is an issue open[1] to make that configurable but noone has done it as of this time. [1] https://issues.asterisk.org/jira/browse/ASTERISK-24106 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org