Re: [OpenSIPS-Users] ws and hold problem
Hi Mikhail, When using WS(S), for the messages coming from the WS(S) endpoint ( with Contact with "invalid") you need to do "fix_nated_contact()", to transform that contact into a routable SIP URI. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 05/25/2019 03:35 AM, Mikhail wrote: 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 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:jsip2@192.168.18.78 SIP/2.0 Contact: server resends to jsip2 invite and replaces Contact with the real ip of jsip1: INVITE sip:60s8k9ep@192.168.21.117:49882;transport=ws SIP/2.0 Contact: When jssip1 place call on hold or unhold, it sends invite to server with INVITE sip:60s8k9ep@192.168.21.117:51630;transport=ws SIP/2.0 Contact: server resends to jsip2 invite and do not changes Contact: INVITE sip:60s8k9ep@192.168.21.117:51630;transport=ws SIP/2.0 Contact: Next, when jssip2 places call on hold, it sends invite: INVITE sip:lp7tmvt6@djgppiddv7t0.invalid;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 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
Re: [OpenSIPS-Users] ws and hold problem
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:users-boun...@lists.opensips.org] On Behalf Of Mikhail Sent: Saturday, 25 May 2019 10:06 AM To: users@lists.opensips.org 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 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:jsip2@192.168.18.78 SIP/2.0 Contact: server resends to jsip2 invite and replaces Contact with the real ip of jsip1: INVITE sip:60s8k9ep@192.168.21.117:49882;transport=ws SIP/2.0 Contact: When jssip1 place call on hold or unhold, it sends invite to server with INVITE sip:60s8k9ep@192.168.21.117:51630;transport=ws SIP/2.0 Contact: server resends to jsip2 invite and do not changes Contact: INVITE sip:60s8k9ep@192.168.21.117:51630;transport=ws SIP/2.0 Contact: Next, when jssip2 places call on hold, it sends invite: INVITE sip:lp7tmvt6@djgppiddv7t0.invalid;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 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
[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 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:jsip2@192.168.18.78 SIP/2.0 Contact: server resends to jsip2 invite and replaces Contact with the real ip of jsip1: INVITE sip:60s8k9ep@192.168.21.117:49882;transport=ws SIP/2.0 Contact: When jssip1 place call on hold or unhold, it sends invite to server with INVITE sip:60s8k9ep@192.168.21.117:51630;transport=ws SIP/2.0 Contact: server resends to jsip2 invite and do not changes Contact: INVITE sip:60s8k9ep@192.168.21.117:51630;transport=ws SIP/2.0 Contact: Next, when jssip2 places call on hold, it sends invite: INVITE sip:lp7tmvt6@djgppiddv7t0.invalid;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 Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
