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Hi<br>
<br>
The log i've posted<br>
<br>
== Using SIP VIDEO CoS mark 6<br>
== Using SIP RTP CoS mark 5<br>
-- Executing <a href="tel:%5B12345678912" value="+12345678912"
target="_blank">[12345678912</a>@from-sip:1]
Answer("SIP/abcde-00000016", "") in new stack<br>
> 0x7fd11404cd00 -- Probation passed - setting RTP source
address to 123.456.789.123:17108<br>
-- Executing <a href="tel:%5B12345678912" value="+12345678912"
target="_blank">[12345678912</a>@from-sip:2]
GotoIf("SIP/abcde-00000016", "0?black,1") in new stack<br>
-- Executing <a href="tel:%5B12345678912" value="+12345678912"
target="_blank">[12345678912</a>@from-sip:3]
Ringing("SIP/abcde-00000016", "") in new stack<br>
-- Executing <a href="tel:%5B12345678912" value="+12345678912"
target="_blank">[12345678912</a>@from-sip:4]
Progress("SIP/abcde-00000016", "") in new stack<br>
-- Executing <a href="tel:%5B12345678912" value="+12345678912"
target="_blank">[12345678912</a>@from-sip:5]
Wait("SIP/abcde-00000016", "5") in new stack<br>
-- Executing <a href="tel:%5B12345678912" value="+12345678912"
target="_blank">[12345678912</a>@from-sip:6]
Dial("SIP/abcde-00000016", "SIP/123&SIP/456,30,oxX") in new
stack<br>
== Using SIP RTP CoS mark 5<br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/200<br>
-- Called SIP/201<br>
-- SIP/123-00000018 connected line has changed. Saving it until
answer for SIP/abcde-00000016<br>
-- SIP/456-00000017 connected line has changed. Saving it until
answer for SIP/abcde-00000016<br>
-- SIP/123-00000018 is ringing<br>
-- SIP/456-00000017 is ringing<br>
<br>
is that what asterisk is showing during an incoming fax call. It
looks like the faxdetection is not working but why?<br>
<br>
Regards Jakob<br>
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