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

[Vallimamod ABDULLAH]

Hi Schneur,

What do you mean precisely by never hitting the asterisk server ?
As your ngrep trace shows, both packets are sent over the wire to the exact same address (68.233.222.9:5060) so they should both reach Asterisk. But it's possible that the latter doesn't treat them the same way, depending on nat issues most of the time (Asterisk send replies to the contact header URI by default if I recall correctly...)

Try to make a ngrep trace on the iface attached to 68.233.222.9 and check if Asterisk does not send the answer to a different IP. Also enable the debug log on the asterisk console to spot any error / warning messages or sip retransmissions.

Hope this would help.

Regards,
-vma
.

On Sep 21, 2011, at 11:43 PM, Schneur Rosenberg wrote:

> 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.
>>> 
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> 
>> --
>> --
>> *--*--*--*--*--*
>> Duane
>> *--*--*--*--*--*
>> --
>> 
>> _______________________________________________
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>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
> 
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