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<TITLE>RE : [Asterisk-Users] Voicemail + SIP Message header</TITLE>
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<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">Hi Deepak, </FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">I had a</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"> <FONT SIZE=2 FACE="Courier New">similar</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New"> setup, However I was able to configure the softswitch to send a prefix followed by</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"> <FONT SIZE=2 FACE="Courier New">the number</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"> <FONT SIZE=2 FACE="Courier New">dialled</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New"> in the to field. This way I could route the call to the right mailbox and be able to play the right busy or away message. </FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">Can you say what softs</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">witch you are using ?</FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">Umar</FONT></SPAN><SPAN LANG="en-gb"></SPAN><SPAN LANG="en-gb"></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">I am trying to use Asterisk as a "pure" voicemail system and have the following</FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">setup:</FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the "To:" field, but the INVITE is still being sent to the number asterisk is configured for. </FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">Is there any way that I can configure asterisk to "read" the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number? </FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">Thanks</FONT></SPAN></P>
<P ALIGN=LEFT><SPAN LANG="en-gb"><FONT SIZE=2 FACE="Courier New">Deepak </FONT></SPAN></P>
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