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

[Duane Larson]

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 <rosenberg11219 at gmail.com
> 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: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.
>
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>



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Duane
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