[OpenSIPS-Users] Sip invite sent, not reaching dest from certain phones

Schneur Rosenberg rosenberg11219 at gmail.com
Wed Sep 21 23:25:06 CEST 2011


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:61 at 68.233.222.9:5060 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:solhome5 at opensips.myserverip.com>;tag=833ac73613f3482o0.
To: <sip:61 at opensips.myserverip.com>.
Remote-Party-ID: solhome5
<sip:solhome5 at opensips.myserverip.com>;screen=yes;party=calling.
Call-ID: 78a92c07-62e399fe at 192.168.1.104.
CSeq: 102 INVITE.
Max-Forwards: 69.
Contact: solhome5 <sip:solhome5 at 173.220.6.65:5060;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:61 at 68.233.222.9:5060;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:test2 at opensips.myserverip.com:5060>;tag=ef646132b8.
To: <sip:61 at opensips.myserverip.com: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:test2 at 173.220.6.65:32857;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.



More information about the Users mailing list