Asterisk Fax Detection *11.7
Hello everybody
I’m trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with FAX Statistics:
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Hello everybody
I’m trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with FAX Statistics:
————-
10 thoughts on - Asterisk Fax Detection *11.7
It is really more interesting the receiving part. Can you paste here?
Leandro
2014/1/21 Jakob-Matthias B
Hi
The log i’ve posted
== Using SIP VIDEO CoS mark 6@from-sip:1]@from-sip:2]@from-sip:3]@from-sip:4]@from-sip:5]@from-sip:6]
== Using SIP RTP CoS mark 5
— Executing [12345678912
Answer(“SIP/abcde-00000016”, “”) in new stack
> 0x7fd11404cd00 — Probation passed – setting RTP source address to 123.456.789.123:17108
— Executing [12345678912
GotoIf(“SIP/abcde-00000016”, “0?black,1”) in new stack
— Executing [12345678912
Ringing(“SIP/abcde-00000016”, “”) in new stack
— Executing [12345678912
Progress(“SIP/abcde-00000016”, “”) in new stack
— Executing [12345678912
Wait(“SIP/abcde-00000016”, “5”) in new stack
— Executing [12345678912
Dial(“SIP/abcde-00000016”, “SIP/123&SIP/456,30,oxX”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
— Called SIP/200
— Called SIP/201
— SIP/123-00000018 connected line has changed. Saving it until answer for SIP/abcde-00000016
— SIP/456-00000017 connected line has changed. Saving it until answer for SIP/abcde-00000016
— SIP/123-00000018 is ringing
— SIP/456-00000017 is ringing
is that what asterisk is showing during an incoming fax call. It looks like the faxdetection is not working but why?
Regards Jakob
I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it.
Leandro
2014/1/21 Jakob-Matthias B
i already added a Progess() and Wait(5) and it still does not detect faxes.
Am 21.01.2014 16:53, schrieb Leandro Dardini:
Please paste the actual code. First has to be the Wait and then any other thing.
Leandro
2014/1/21 Jakob-Matthias B
Don’t expect T.30 over SIP to be reliable. If you need fax, you should be using T.38. Your codec is likely the issue.
Hello,
Perhaps you need to have directmedia=no set for the channel, the call doesn’t appear to have been answered hence asterisk won’t be able to hear any tones to determine for itself if the call is an incoming fax.
Larry.
Sorry, I missed the line showing the call had been answered.
There is a typo in the last line above. Should be “canreinvite”. AFAIK it’s obsoleted in favor of directmedia. BTW, try to set it to NO. BTW, what is the codec order? Fax detection doesn’t work reliably over compressed codecs (g729 etc…), in my case didn’t work at all… try to add:
directmedia=no disallow=all allow=ulaw allow=alaw
to your peer definition.
Martin
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Hello,
now i added
directmedia=no disallow=all allow=ulaw allow=alaw
and i changed the caninvite part to canreinvite. Now the faxdetection is working well. But now, after the faxsession has started, i’m getting
res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short
as error.
Regards Jakob