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<TITLE>Re: [asterisk-users] Follow-me challenge</TITLE>
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<FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:12.0px'>Is the problem just when you don’t answer the cell phone? Many cell phones go to a voice announcement when they’re turned off or not answered, and Asterisk thinks the call has been answered. The other issue could be that your gateway (asterisk1) is answering the call before the outbound leg is answered. One workaround would be to use a macro that requires you to press a key to accept the call on your cell. (See the M option to the dial command and <a href="http://www.voip-info.org/wiki/view/Asterisk+tips+findme)">http://www.voip-info.org/wiki/view/Asterisk+tips+findme)</a><BR>
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Also, I see that you’re using the r option — you might want to drop that.<BR>
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I’m also not convinced that it will ever find 102,107 in your dialplan. You might want to look at using ${DIALSTATUS} and making it a bit more explicit.<BR>
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Cheers,<BR>
Eric<BR>
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On 2006-12-18 23:09, "Chris Johnson" <masterof1@gmail.com> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:12.0px'>The problem I am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail.<BR>
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Here is my current config:<BR>
exten => 102,1,Dial(${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>
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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 192.168.1.2 <a href="http://192.168.1.2"><http://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>
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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>
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Thanks<BR>
<BR>
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</SPAN></FONT></FONT><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:12.0px'>-- <BR>
Eric Jacksch<BR>
+1 613 860-0964 Ottawa<BR>
+1 647 722-3544 Toronto<BR>
+1 514 907-0031 Montreal<BR>
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"They who would give up an essential liberty for temporary security deserve neither liberty or security." -- Benjamin Franklin<BR>
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