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.<br>
<br>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.<br><br><div class="gmail_quote">
On Wed, Sep 21, 2011 at 4:25 PM, Schneur Rosenberg <span dir="ltr"><<a href="mailto:rosenberg11219@gmail.com">rosenberg11219@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
I'm pretty new to opensips, I'm having a interesting problem, I use my<br>
opensips for loadbalancing purposes I'm trying to place a call, and<br>
from My linksys phone everything works fine, call comes into opensips<br>
and opensips sends it to my asterisk system and call goes through<br>
properly, from other phone (Aastra) Opensips accept the call, it even<br>
sends it to the Asterisk but in never hits the asterisk server, can<br>
anyone please review the 2 invites and let me know why second invite<br>
gets lost, and how I can fix it<br>
<br>
Here is the invite from the Linksys that worked<br>
<br>
U <a href="http://64.69.40.120:5060" target="_blank">64.69.40.120:5060</a> -> <a href="http://68.233.222.9:5060" target="_blank">68.233.222.9:5060</a><br>
INVITE <a href="http://sip:61@68.233.222.9:5060" target="_blank">sip:61@68.233.222.9:5060</a> SIP/2.0.<br>
Record-Route: <sip:64.69.40.120;lr=on>.<br>
Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK9857.800b0681.0.<br>
Via: SIP/2.0/UDP<br>
192.168.1.104:5060;rport=5060;received=173.220.6.65;branch=z9hG4bK-4194567e.<br>
From: solhome5 <<a href="mailto:sip%3Asolhome5@opensips.myserverip.com">sip:solhome5@opensips.myserverip.com</a>>;tag=833ac73613f3482o0.<br>
To: <<a href="mailto:sip%3A61@opensips.myserverip.com">sip:61@opensips.myserverip.com</a>>.<br>
Remote-Party-ID: solhome5<br>
<<a href="mailto:sip%3Asolhome5@opensips.myserverip.com">sip:solhome5@opensips.myserverip.com</a>>;screen=yes;party=calling.<br>
Call-ID: <a href="mailto:78a92c07-62e399fe@192.168.1.104">78a92c07-62e399fe@192.168.1.104</a>.<br>
CSeq: 102 INVITE.<br>
Max-Forwards: 69.<br>
Contact: solhome5 <sip:solhome5@173.220.6.65:5060;nat=yes>.<br>
Expires: 240.<br>
User-Agent: Linksys/SPA2102-5.2.12.<br>
Content-Length: 446.<br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.<br>
Supported: x-sipura, replaces.<br>
Content-Type: application/sdp.<br>
<br>
Here is the invite of the Aastra that did not work<br>
<br>
U <a href="http://64.69.40.120:5060" target="_blank">64.69.40.120:5060</a> -> <a href="http://68.233.222.9:5060" target="_blank">68.233.222.9:5060</a><br>
INVITE sip:61@68.233.222.9:5060;user=phone SIP/2.0.<br>
Record-Route: <sip:64.69.40.120;lr=on>.<br>
Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK847d.93e9e112.0.<br>
Via: SIP/2.0/UDP<br>
192.168.1.102;rport=32857;received=173.220.6.65;branch=z9hG4bK7d9e53a542fc6ebe6.6e57a6889920eccf1.<br>
Max-Forwards: 69.<br>
From: "test2" <<a href="http://sip:test2@opensips.myserverip.com:5060" target="_blank">sip:test2@opensips.myserverip.com:5060</a>>;tag=ef646132b8.<br>
To: <sip:61@opensips.myserverip.com:5060;user=phone>.<br>
Call-ID: f12b5324f31c0d30.<br>
CSeq: 20777 INVITE.<br>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE,<br>
PRACK, SUBSCRIBE, INFO.<br>
Allow-Events: talk, hold, conference, LocalModeStatus.<br>
Contact: "test2"<br>
<sip:test2@173.220.6.65:32857;transport=udp;nat=yes>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2A6C9E>".<br>
Supported: path, 100rel, replaces.<br>
User-Agent: Aastra 57iCT/<a href="http://3.2.2.56" target="_blank">3.2.2.56</a>.<br>
Content-Type: application/sdp.<br>
Content-Length: 630.<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>