[OpenSIPS-Users] Transfer issue

[Peter den Hartog]

Oke i feel so happy right now, i fixed it! it works! i can now create dials
over opensips, true asterisk, outside inside i can transfer, everything
works! damn i'm happy :D!
the answer was in my opensips.cfg and the routing back to asterisk, i've
created a routing script that just subscribe and trows the rest in to
asterisk.

i'm thinking of creating a big straightforward blogpost about this, how you
should do this, with what goes where and stuff like that.

I would like to thank everybody who replied on this issue, thanks alot.

On Tue, Oct 27, 2009 at 12:16 PM, Iñaki Baz Castillo <ibc at aliax.net> wrote:

> El Martes, 27 de Octubre de 2009, Peter den Hartog escribió:
> > 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%3A105 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<sip%3A0624469780 at 172.16.1.10>
> >;tag=as47c203e8.
> > To: <sip:0031851119105 at 172.16.1.14 <sip%3A0031851119105 at 172.16.1.14>
> >;tag=2AE312D6-A13BBC6D.
> > Contact: <sip:0624469780 at 172.16.1.10 <sip%3A0624469780 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%3A105 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<sip%3A0624469780 at 172.16.1.10>
> >;tag=as47c203e8.
> > To: <sip:0031851119105 at 172.16.1.14 <sip%3A0031851119105 at 172.16.1.14>
> >;tag=2AE312D6-A13BBC6D.
> > Contact: <sip:0624469780 at 172.16.1.10 <sip%3A0624469780 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.
>
>
> Are your client behind a router with SIP ALG?:
>  http://www.voip-info.org/wiki/view/Routers+SIP+ALG
>
>
> --
> Iñaki Baz Castillo <ibc at aliax.net>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Groet // Kind regards,
Peter den Hartog

Sent from Leidschendam, ZH, Netherlands
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