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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Show us your sip.conf<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>michel freiha<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, January 28, 2009
9:30 AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] FAX</span></font><o:p></o:p></p>
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12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi all,<br>
<br>
When trying to send a FAX I got the following error:<br>
<br>
Executing [003228949469@micho:1] Dial("SIP/028949469-08466918",
"SIP/<a href="mailto:003228949469@80.169.210.181">003228949469@80.169.210.181</a>|60")
in new stack<br>
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
found to offer. Cancelling call to 003228949469<br>
-- Couldn't call 0032234534534@1.1.1.1.1<br>
<br>
Where I should define the codec other than the extension in order to succeed
the call?<br>
<br>
Regards<o:p></o:p></span></font></p>
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