NO These are the invites going from the opensips to the asterisk NOT the ones from the phone, I did a ngrep on the asterisk box and the packet never reaches it, both opensips and asterisk are open no NAT, the phones are behind a nat as you can see in the sip packets
On Thu, Sep 22, 2011 at 12:37 AM, Duane Larson <[email protected]> wrote: > These are the INVITES that are coming from your Phones correct? These won't > help to troubleshoot I don't think. You will need to show the INVITES that > are leaving OpenSIPS and heading towards your Asterisk server. > > Honestly if your opensips.cfg does the exact same thing for linksys and > aastra phones I can't see it being an opensips issue. That's just a guess > since I don't have anything to go on. > > On Wed, Sep 21, 2011 at 4:25 PM, Schneur Rosenberg > <[email protected]> wrote: >> >> I'm pretty new to opensips, I'm having a interesting problem, I use my >> opensips for loadbalancing purposes I'm trying to place a call, and >> from My linksys phone everything works fine, call comes into opensips >> and opensips sends it to my asterisk system and call goes through >> properly, from other phone (Aastra) Opensips accept the call, it even >> sends it to the Asterisk but in never hits the asterisk server, can >> anyone please review the 2 invites and let me know why second invite >> gets lost, and how I can fix it >> >> Here is the invite from the Linksys that worked >> >> U 64.69.40.120:5060 -> 68.233.222.9:5060 >> INVITE sip:[email protected]:5060 SIP/2.0. >> Record-Route: <sip:64.69.40.120;lr=on>. >> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK9857.800b0681.0. >> Via: SIP/2.0/UDP >> >> 192.168.1.104:5060;rport=5060;received=173.220.6.65;branch=z9hG4bK-4194567e. >> From: solhome5 >> <sip:[email protected]>;tag=833ac73613f3482o0. >> To: <sip:[email protected]>. >> Remote-Party-ID: solhome5 >> <sip:[email protected]>;screen=yes;party=calling. >> Call-ID: [email protected]. >> CSeq: 102 INVITE. >> Max-Forwards: 69. >> Contact: solhome5 <sip:[email protected]:5060;nat=yes>. >> Expires: 240. >> User-Agent: Linksys/SPA2102-5.2.12. >> Content-Length: 446. >> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. >> Supported: x-sipura, replaces. >> Content-Type: application/sdp. >> >> Here is the invite of the Aastra that did not work >> >> U 64.69.40.120:5060 -> 68.233.222.9:5060 >> INVITE sip:[email protected]:5060;user=phone SIP/2.0. >> Record-Route: <sip:64.69.40.120;lr=on>. >> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK847d.93e9e112.0. >> Via: SIP/2.0/UDP >> >> 192.168.1.102;rport=32857;received=173.220.6.65;branch=z9hG4bK7d9e53a542fc6ebe6.6e57a6889920eccf1. >> Max-Forwards: 69. >> From: "test2" <sip:[email protected]:5060>;tag=ef646132b8. >> To: <sip:[email protected]:5060;user=phone>. >> Call-ID: f12b5324f31c0d30. >> CSeq: 20777 INVITE. >> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, >> PRACK, SUBSCRIBE, INFO. >> Allow-Events: talk, hold, conference, LocalModeStatus. >> Contact: "test2" >> >> <sip:[email protected]:32857;transport=udp;nat=yes>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2A6C9E>". >> Supported: path, 100rel, replaces. >> User-Agent: Aastra 57iCT/3.2.2.56. >> Content-Type: application/sdp. >> Content-Length: 630. >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > -- > *--*--*--*--*--* > Duane > *--*--*--*--*--* > -- > > _______________________________________________ > 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
