Hi Mark, Glad it helped. I actually did not know about chan_sip option supportpath because I was using additional opensips for wws to webrtc. Thanks for the details.
Good luck On Thu, Jul 23, 2020 at 9:19 AM Mark Allen <m...@allenclan.co.uk> wrote: > > [SOLVED] > > Thanks Stas. Your workaround did solve the problem and I see that with 3.1 > path support is baked into mid-registrar module as options to > mid_registrar_save(). > > Once we added in the path module functionality, at first it didn't work. > Looking at sngrep traces we could see that the path information was appended > on the inbound route from OpenSIPS to Asterisk, but when Asterisk made the > call to the outbound destination it failed to include it as route info. This > was resolved by setting supportpath="yes" in sip.conf and worked with > CHAN_SIP. We tried to get it working with PJSIP without any joy but it's not > a priority for us at the moment so we'll have to investigate the cause later. > > Thanks for your help Stas, and thanks also to the others who took the time to > reply. > > cheers, > > Mark > > > On Tue, 14 Jul 2020 at 16:23, Stas Kobzar <staskob...@gmail.com> wrote: >> >> Hello Mark, >> >> I had a similar challenge. Using "path" module on both opensips helps >> to overcome this problem. >> https://opensips.org/docs/modules/3.2.x/path.html >> >> In your mid-registerer you need to enable path support. See "save" >> function params p0 and v. >> in your webrtc opensips use path module and function add_path_received >> >> On Tue, Jul 14, 2020 at 11:14 AM Mark Allen <m...@allenclan.co.uk> wrote: >> > >> > I'm new to OpenSIPS and I've hit a problem I can't find a way past >> > >> > We have a test setup with an OpenSIPS mid-registrar in front of an >> > Asterisk PBX. Mid-registrar is currently in mode 1 (registration >> > throttling). We have SIP and WebRTC endpoints that we want to use. >> > >> > Current state is: >> > >> > REGISTER: WebRTC webphone (Mizutech) -> OpenSIPS Mid-registrar -> >> > Asterisk = success >> > REGISTER: SIP softphone (LinPhone) -> OpenSIPS Mid-registrar -> >> > Asterisk = success >> > >> > INVITE: SIP softphone -> OpenSIPS -> Asterisk -> OpenSIPS -> SIP >> > softphone = success, call connects with audio both ways >> > INVITE: WebRTC webphone -> OpenSIPS -> Asterisk -> OpenSIPS -> SIP >> > softphone = success, call connects with audio both ways >> > INVITE: SIP softphone -> OpenSIPS -> Asterisk -> OpenSIPS -> WebRTC >> > webphone = fails with "476 Unresolvable destination" >> > >> > syslog messages: >> > ERROR:core:sip_resolvehost: forced proto 6 not matching sips uri >> > CRITICAL:core:mk_proxy: could not resolve hostname: "4xp44jxl0qq0.invalid" >> > ERROR:tm:uri2proxy: bad host name in URI >> > <sips:11001@4xp44jxl0qq0.invalid;rtcweb-breaker=yes;transport=wss> >> > ERROR:tm:t_forward_nonack: failure to add branches >> > >> > >> > Following past reports that I've found with a similar error, >> > fix_nated_contact() is run on INVITE messages just before rtpengine flags >> > are set and the t_relay() command, but it doesn't appear to make any >> > difference. If I change the t_relay() to t_relay(0x04,) to disable DNS >> > Failover, I still see the same errors in the log file. I've also checked >> > the record in the OpenSIPS DB "location" table and it seems to me that it >> > has the correct contact_id and contact info for the destination... >> > >> > contact_id: 2004383309156582802 >> > contact: >> > sips:11001@4xp44jxl0qq0.invalid;rtcweb-breaker=yes;transport=wss >> > >> > I'm stuck on where I can go from here - any help very much appreciated! >> > >> > thx >> > >> > Mark >> > >> > >> > Setup: >> > OpenSIPS 3.0.2 on Debian Buster >> > RTPEngine Version: 8.4.0.0+0~mr8.4.0.0 >> > >> > INVITE: >> > 2020/07/14 14:22:06.176544 192.168.50.185:5060 -> 192.168.50.69:5060 >> > INVITE sip:11001@192.168.50.69:5060;ctid=2004383309156582802 SIP/2.0 >> > Via: SIP/2.0/UDP >> > 192.168.50.185:5060;rport;branch=z9hG4bKPj3e87a449-f4cc-4128-abbe-95706a1a44a0 >> > From: "11002" >> > <sip:11002@192.168.50.185>;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 >> > To: <sip:11001@192.168.50.69;ctid=2004383309156582802> >> > Contact: <sip:asterisk@192.168.50.185:5060> >> > Call-ID: d1524788-cac2-4bea-a905-4e17ba006688 >> > CSeq: 24456 INVITE >> > Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, >> > CANCEL, UPDATE, PRACK, MESSAGE, REFER >> > Supported: 100rel, timer, replaces, norefersub >> > Session-Expires: 1800 >> > Min-SE: 90 >> > P-Asserted-Identity: "11002" <sip:11002@192.168.50.185> >> > Max-Forwards: 70 >> > User-Agent: FPBX-15.0.16.63(16.9.0) >> > Content-Type: application/sdp >> > Content-Length: 411 >> > >> > v=0 >> > o=- 263255642 263255642 IN IP4 192.168.50.185 >> > s=Asterisk >> > c=IN IP4 192.168.50.185 >> > t=0 0 >> > m=audio 10292 RTP/AVPF 9 107 8 0 3 111 101 >> > a=rtpmap:9 G722/8000 >> > a=rtpmap:107 opus/48000/2 >> > a=fmtp:107 useinbandfec=1 >> > a=rtpmap:8 PCMA/8000 >> > a=rtpmap:0 PCMU/8000 >> > a=rtpmap:3 GSM/8000 >> > a=rtpmap:111 G726-32/8000 >> > a=rtpmap:101 telephone-event/8000 >> > a=fmtp:101 0-16 >> > a=ptime:20 >> > a=maxptime:20 >> > a=sendrecv >> > a=rtcp-mux >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users@lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
