Hello Mikhail, What exactly have you got there. It looks like a closed test setup. Is it just Three routers in an upstream loop and Three computers.
Are you just trying to make some basic voip audio only phone calls. Is there a more specific project you are trying to have help develop. I am not familiar with these other technologies but it somehow seems you have not provided enough information. Alex -----Original Message----- From: Users [mailto:[email protected]] On Behalf Of Mikhail Sent: Saturday, 25 May 2019 10:06 AM To: [email protected] Subject: [OpenSIPS-Users] ws and hold problem Hi opensips 2.4 and rtpengine. webrtc SIP clients based on jsSIP 3.3.6 to accounts jssip1 and jssip2 jssip1 calls jssip2 or jssip2 calls jssip1 - call established jssip1 place call on hold and then unhold - no problem now if jssip2 place call on hold it receives SIP/2.0 476 Unresolvable destination (476/TM) from opensips and call brakes. also in opensips.log there are a messages like this: May 25 02:42:41 opensips-01 /usr/sbin/opensips[19704]: CRITICAL:core:mk_proxy: could not resolve hostname: "djgppiddv7t0.invalid" May 25 02:42:41 opensips-01 /usr/sbin/opensips[19704]: ERROR:tm:uri2proxy: bad host name in URI <sip:[email protected];transport=ws;ob> May 25 02:42:41 opensips-01 /usr/sbin/opensips[19704]: ERROR:tm:t_forward_nonack: failure to add branches what i found: While call setup jsip1 sends initial invite: INVITE sip:[email protected] SIP/2.0 Contact: <sip:[email protected];transport=ws;ob> server resends to jsip2 invite and replaces Contact with the real ip of jsip1: INVITE sip:[email protected]:49882;transport=ws SIP/2.0 Contact: <sip:[email protected]:46099;transport=ws;ob> When jssip1 place call on hold or unhold, it sends invite to server with INVITE sip:[email protected]:51630;transport=ws SIP/2.0 Contact: <sip:[email protected];transport=ws;ob> server resends to jsip2 invite and do not changes Contact: INVITE sip:[email protected]:51630;transport=ws SIP/2.0 Contact: <sip:[email protected];transport=ws;ob> Next, when jssip2 places call on hold, it sends invite: INVITE sip:[email protected];transport=ws;ob SIP/2.0 and server can't resolve djgppiddv7t0.invalid, it expects real address here does anybody have an idea, who is responsible for the problem - jssip, opensips or rtpengine ? Laba Mikhail _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
