[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

I just checked a bit better and noticed this error while transfering:

U 172.16.1.10:5060 -> 172.16.1.14:5060
NOTIFY sip:105 at 172.16.0.24 SIP/2.0.
Via: SIP/2.0/UDP 172.16.1.10:5060;branch=z9hG4bK23a1000e;rport.
Route: <sip:172.16.1.14;lr=on>.
From: "0624469780" <sip:0624469780 at 172.16.1.10>;tag=as47c203e8.
To: <sip:0031851119105 at 172.16.1.14>;tag=2AE312D6-A13BBC6D.
Contact: <sip:0624469780 at 172.16.1.10>.
Call-ID: 05aedaab03eadeca3b42d0b84d880efb at 172.16.1.10.
CSeq: 103 NOTIFY.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Event: refer;id=2.
Subscription-state: terminated;reason=noresource.
Content-Type: message/sipfrag;version=2.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Length: 49.
.
SIP/2.0 481 Call leg/transaction does not exist.


U 172.16.1.14:5060 -> 172.16.0.24:5060
NOTIFY sip:105 at 172.16.0.24 SIP/2.0.
Via: SIP/2.0/UDP 172.16.1.14;branch=z9hG4bK1df2.c4df24a7.0.
Via: SIP/2.0/UDP
172.16.1.10:5060;received=172.16.1.10;branch=z9hG4bK23a1000e;rport=5060.
From: "0624469780" <sip:0624469780 at 172.16.1.10>;tag=as47c203e8.
To: <sip:0031851119105 at 172.16.1.14>;tag=2AE312D6-A13BBC6D.
Contact: <sip:0624469780 at 172.16.1.10>.
Call-ID: 05aedaab03eadeca3b42d0b84d880efb at 172.16.1.10.
CSeq: 103 NOTIFY.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Event: refer;id=2.
Subscription-state: terminated;reason=noresource.
Content-Type: message/sipfrag;version=2.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Content-Length: 49.
.
SIP/2.0 481 Call leg/transaction does not exist.

That is the message that apears when pressing the transfer button.


Iñaki Baz Castillo wrote:
> 
> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió:
>> Well yes, it does work for the internal calls, but
>> when a call comes in true asterisk to an opensips extention i CAN'T
>>  transfer it :-), i get transfer failed in my screen of my phone, and the
>>  call stays on the original called extention. This is only for announced
>>  transfers, unannounced works fine.
>> 
>> Flavio post stated something about routing your REFER's back to asterisk,
>>  so it should work.. but i don't know how to route these calls back to
>> the
>>  asterisk.
> 
> Please, you *already* have the answer. When a phone is speaking with
> Asterisk 
> (through OpenSIPS) you must route REFER to Asterisk as *any* other
> in-dialog 
> request, this is, the *same* as when a phone is speaking with other phone 
> directly (through OpenSIPS).
> 
> If the REFER fails this is because Asterisk is rejecting it !!!
> 
> I already suggested you to do a SIP capture (using ngrep) to inspect which 
> error replies Asterisk when the REFER arrives to it. Please do it and
> paste it 
> here (I expect a 403 or 404, so it means a wrong configuration in you 
> Asterisk, no more).
> 
> And please, forget anything about exotic routing of the REFER.
> 
> 
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

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