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

[Schneur Rosenberg]

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 <duane.larson at gmail.com> 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
> <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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>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>
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