Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls,
when I call from cisco from, it work except hangup. when I call to cisco phone asterisk return congested debug of call <--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 ---> INVITE sip:111@192.168.1.61:51179;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.4:55246;rport;branch=z9hG4bKPjcb9ec9ba-0c77-4530-a3b7-44209357f3a0;alias From: <sip:502@192.168.1.4>;tag=abebd75c-501a-4b4f-ad69-ee98175b8dbd To: <sip:111@192.168.1.61> Contact: <sip:28552048-b20b-4e7c-8454-f7d1486fd8ef@192.168.1.4:55246;transport=TCP> Call-ID: bb515935-7292-47b4-890d-6f82eb335815 CSeq: 25333 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 283 v=0 o=- 1231372975 1231372975 IN IP4 192.168.1.4 s=Asterisk c=IN IP4 192.168.1.4 t=0 0 m=audio 17856 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [Feb 24 05:47:01] WARNING[16179]: pjsip:0 <?>: tsx0x7f1aa0157 Failed to send Request msg INVITE/cseq=12216 (tdta0x7f1aa00e41c0)! err=120111 (Connection refused) [Feb 24 05:47:01] ERROR[16179]: pjsip:0 <?>: tcpc0x7f1aa01c TCP connect() error: Connection refused [code=120111] [Feb 24 05:47:01] WARNING[16179]: pjsip:0 <?>: tsx0x7f1aa01c3 Failed to send Request msg INVITE/cseq=25333 (tdta0x7f1aa00ad810)! err=120111 (Connection refused) > On 24 Feb 2015, at 15:05, Joshua Colp <jc...@digium.com> wrote: > > Nick Awesome wrote: >> Hay guys, got trouble with registration with cisco 7975 > > The "force_rport" option is incompatible with Cisco, it needs to be > explicitly set to no in the endpoint. > > Cheers, > > -- > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
