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 *--*--*--*--*--* --
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