The problem I am running into is that when the call to my cellphone is
made, it appears as though the call "completes" so it never rolls to
asterisk voicemail.<br><br>Here is my current config:<br>exten => 102,1,Dial(${sipura},10,)<br>exten => 102,n,playback(pls-wait-connect-call)<br>exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r)<br>exten => 102,n,VoiceMail(
u102@default)<br>exten => 102,107,VoiceMail(b102@default)<br><br>Here is the log from asterisk:<br> -- Executing [102@internal:2] Playback("SIP/101-0a1178c0", "pls-wait-connect-call") in new stack
<br> -- Playing 'pls-wait-connect-call' (language 'en')<br> -- Executing [102@internal:3] Dial("SIP/101-0a1178c0", "IAX2/asterisk1/9139275900|10|r") in new stack<br> -- Called asterisk1/9139275900
<br> -- Call accepted by <a href="http://192.168.1.2">192.168.1.2</a> (format ulaw)<br> -- Format for call is ulaw<br> -- IAX2/asterisk1-7 answered SIP/101-0a1178c0<br> -- Hungup 'IAX2/asterisk1-7'<br><br>The one thing I will note is that there is not an analog trunk in this server. It hands off the outbound call to trixbox running on another server, which I fear may be my problem. Having said that, I will also note that I have had the same challenge trying to get follow-me set up on trixbox as well.
<br><br>Thanks<br><br>