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!<br>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.<br>
<br>i'm thinking of creating a big straightforward blogpost about this, how you should do this, with what goes where and stuff like that.<br><br>I would like to thank everybody who replied on this issue, thanks alot.<br>
<br><div class="gmail_quote">On Tue, Oct 27, 2009 at 12:16 PM, Iñaki Baz Castillo <span dir="ltr"><<a href="mailto:ibc@aliax.net">ibc@aliax.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
El Martes, 27 de Octubre de 2009, Peter den Hartog escribió:<br>
<div><div></div><div class="h5">> I just checked a bit better and noticed this error while transfering:<br>
><br>
> U <a href="http://172.16.1.10:5060" target="_blank">172.16.1.10:5060</a> -> <a href="http://172.16.1.14:5060" target="_blank">172.16.1.14:5060</a><br>
> NOTIFY <a href="mailto:sip%3A105@172.16.0.24">sip:105@172.16.0.24</a> SIP/2.0.<br>
> Via: SIP/2.0/UDP 172.16.1.10:5060;branch=z9hG4bK23a1000e;rport.<br>
> Route: <sip:172.16.1.14;lr=on>.<br>
> From: "0624469780" <<a href="mailto:sip%3A0624469780@172.16.1.10">sip:0624469780@172.16.1.10</a>>;tag=as47c203e8.<br>
> To: <<a href="mailto:sip%3A0031851119105@172.16.1.14">sip:0031851119105@172.16.1.14</a>>;tag=2AE312D6-A13BBC6D.<br>
> Contact: <<a href="mailto:sip%3A0624469780@172.16.1.10">sip:0624469780@172.16.1.10</a>>.<br>
> Call-ID: <a href="mailto:05aedaab03eadeca3b42d0b84d880efb@172.16.1.10">05aedaab03eadeca3b42d0b84d880efb@172.16.1.10</a>.<br>
> CSeq: 103 NOTIFY.<br>
> User-Agent: Asterisk PBX.<br>
> Max-Forwards: 70.<br>
> Event: refer;id=2.<br>
> Subscription-state: terminated;reason=noresource.<br>
> Content-Type: message/sipfrag;version=2.0.<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>
> Supported: replaces.<br>
> Content-Length: 49.<br>
> .<br>
> SIP/2.0 481 Call leg/transaction does not exist.<br>
><br>
><br>
> U <a href="http://172.16.1.14:5060" target="_blank">172.16.1.14:5060</a> -> <a href="http://172.16.0.24:5060" target="_blank">172.16.0.24:5060</a><br>
> NOTIFY <a href="mailto:sip%3A105@172.16.0.24">sip:105@172.16.0.24</a> SIP/2.0.<br>
> Via: SIP/2.0/UDP 172.16.1.14;branch=z9hG4bK1df2.c4df24a7.0.<br>
> Via: SIP/2.0/UDP<br>
> 172.16.1.10:5060;received=172.16.1.10;branch=z9hG4bK23a1000e;rport=5060.<br>
> From: "0624469780" <<a href="mailto:sip%3A0624469780@172.16.1.10">sip:0624469780@172.16.1.10</a>>;tag=as47c203e8.<br>
> To: <<a href="mailto:sip%3A0031851119105@172.16.1.14">sip:0031851119105@172.16.1.14</a>>;tag=2AE312D6-A13BBC6D.<br>
> Contact: <<a href="mailto:sip%3A0624469780@172.16.1.10">sip:0624469780@172.16.1.10</a>>.<br>
> Call-ID: <a href="mailto:05aedaab03eadeca3b42d0b84d880efb@172.16.1.10">05aedaab03eadeca3b42d0b84d880efb@172.16.1.10</a>.<br>
> CSeq: 103 NOTIFY.<br>
> User-Agent: Asterisk PBX.<br>
> Max-Forwards: 69.<br>
> Event: refer;id=2.<br>
> Subscription-state: terminated;reason=noresource.<br>
> Content-Type: message/sipfrag;version=2.0.<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.<br>
> Supported: replaces.<br>
> Content-Length: 49.<br>
> .<br>
> SIP/2.0 481 Call leg/transaction does not exist.<br>
><br>
> That is the message that apears when pressing the transfer button.<br>
<br>
<br>
</div></div>Are your client behind a router with SIP ALG?:<br>
<a href="http://www.voip-info.org/wiki/view/Routers+SIP+ALG" target="_blank">http://www.voip-info.org/wiki/view/Routers+SIP+ALG</a><br>
<div><div></div><div class="h5"><br>
<br>
--<br>
Iñaki Baz Castillo <<a href="mailto:ibc@aliax.net">ibc@aliax.net</a>><br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Groet // Kind regards,<br>Peter den Hartog<br><br>Sent from Leidschendam, ZH, Netherlands