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<TITLE>RES: [asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?</TITLE>
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<P><FONT SIZE=2>Hello Philipp and All,</FONT>
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<P><FONT SIZE=2>My scenario is a bit different than the one I had explained before. I'm sorry. </FONT>
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<P><FONT SIZE=2>Let's suppose I have someone calling one of my Asterisk clients. This asterisk client has CFB (Call Forward Busy) activated. The forward number is a Voice Mail System, however is not a Asterisk's Voice Mail. </FONT></P>
<P><FONT SIZE=2>It is a third party Voice Mail System, that has a SIP Trunk with my Asterisk Box. </FONT>
</P>
<P><FONT SIZE=2>The test situation I have, demands having the same VMS Access number for both, leaving a message to the Client Mail Box, or for the subscriber to access its menu directory. </FONT></P>
<P><FONT SIZE=2>This VMS platform will differ these two type of calls, by some change on the invite message coming from the Asterisk. </FONT>
</P>
<P><FONT SIZE=2>I was thinking about using "SIP Response 181 (Call is being forwarded)" as an option to flag to the VMS letting it know, that it is supposed to treat as a call that was diverted to it. </FONT></P>
<P><FONT SIZE=2>But does any one have a suggestion, or real scenario similar to this that could help me??</FONT>
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<P><FONT SIZE=2>Thanks again,</FONT>
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<P><FONT SIZE=2>Marco</FONT>
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<P><FONT SIZE=2>-----Mensagem original-----</FONT>
<BR><FONT SIZE=2>De: asterisk-users-bounces@lists.digium.com [<A HREF="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</A>] Em nome de Philipp Kempgen</FONT>
<BR><FONT SIZE=2>Enviada em: terça-feira, 2 de junho de 2009 14:22</FONT>
<BR><FONT SIZE=2>Para: Asterisk Users Mailing List - Non-Commercial Discussion</FONT>
<BR><FONT SIZE=2>Assunto: Re: [asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?</FONT>
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<P><FONT SIZE=2>Marco Cordeiro schrieb:</FONT>
<BR><FONT SIZE=2>> So, what you are saying is that SIP trunks between 2 Asteriks might be able</FONT>
<BR><FONT SIZE=2>> to handle SIP Response 181 ?</FONT>
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<P><FONT SIZE=2>Looks like it, but I didn't test it.</FONT>
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<P><FONT SIZE=2>(Note to self: Here's the diff:</FONT>
<BR><FONT SIZE=2><A HREF="https://reviewboard.asterisk.org/r/201/diff/" TARGET="_blank">https://reviewboard.asterisk.org/r/201/diff/</A> )</FONT>
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<P><FONT SIZE=2>> -----Mensagem original-----</FONT>
<BR><FONT SIZE=2>> De: asterisk-users-bounces@lists.digium.com</FONT>
<BR><FONT SIZE=2>> [<A HREF="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</A>] Em nome de Philipp Kempgen</FONT>
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<P><FONT SIZE=2>> Philipp Kempgen schrieb:</FONT>
<BR><FONT SIZE=2>>> Marco Cordeiro schrieb:</FONT>
<BR><FONT SIZE=2>> </FONT>
<BR><FONT SIZE=2>>>> I have a situation that I have to notify the calling party that the call</FONT>
<BR><FONT SIZE=2>> is</FONT>
<BR><FONT SIZE=2>>>> being forwarded to another number. So far, in the tests that I made,</FONT>
<BR><FONT SIZE=2>> calling</FONT>
<BR><FONT SIZE=2>>>> from a SIP extension to another SIP extension with the forwarding</FONT>
<BR><FONT SIZE=2>> activated,</FONT>
<BR><FONT SIZE=2>>>> I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP</FONT>
<BR><FONT SIZE=2>>>> Response 181 CALL_IS_BEING_FORWARDED).</FONT>
<BR><FONT SIZE=2>>>> </FONT>
<BR><FONT SIZE=2>>>> The forwarding of the SIP extensions is being set with AstDB. </FONT>
<BR><FONT SIZE=2>>>> </FONT>
<BR><FONT SIZE=2>>>> My doubt is if, only a SIP Proxy would be able to trigger SIP Response</FONT>
<BR><FONT SIZE=2>> 181,</FONT>
<BR><FONT SIZE=2>>>> or if it would be possible with an Asterisk Server. </FONT>
<BR><FONT SIZE=2>>> </FONT>
<BR><FONT SIZE=2>>> IIRC Asterisk trunk can send and handle 181 Call is being forwarded.</FONT>
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<P><FONT SIZE=2> Philipp Kempgen</FONT>
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