[asterisk-users] Asterisk Fax detection *11.7

[Larry Moore]

Sorry, I missed the line showing the call had been answered.

On 22/01/2014 8:11 AM, Larry Moore wrote:
> 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.
>
> On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:
>> 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:
>> ---------------
>>
>> Current Sessions : 0
>> Reserved Sessions : 0
>> Transmit Attempts : 0
>> Receive Attempts : 1
>> Completed FAXes : 1
>> Failed FAXes : 1
>>
>> Digium G.711
>> Licensed Channels : 1
>> Max Concurrent : 0
>> Success : 0
>> Switched to T.38 : 0
>> Canceled : 0
>> No FAX : 0
>> Partial : 0
>> Negotiation Failed : 0
>> Train Failure : 0
>> Protocol Error : 0
>> IO Partial : 0
>> IO Fail : 0
>>
>> Digium T.38
>> Licensed Channels : 1
>> Max Concurrent : 1
>> Success : 0
>> Canceled : 0
>> No FAX : 0
>> Partial : 0
>> Negotiation Failed : 0
>> Train Failure : 1
>> Protocol Error : 0
>> IO Partial : 0
>> IO Fail : 0
>>
>> so that should be ok.
>>
>> The corresponding dialplan section starts with
>>
>>
>> [from-sip]
>> include => inbound
>>
>> [inbound]
>> exten => _X.,1,Answer()
>> exten => _X.,n,GotoIf(${BLACKLIST()}?black,1)
>> exten => _X.,n,Ringing
>> exten => _X.,n,Progress()
>> exten => _X.,n,Wait(5)
>> exten => _X.,n,Dial(SIP/123&SIP/456,30,oxX)
>> ...
>> exten => fax,1,NoOp(**** FAX DETECTED ****)
>> exten => fax,n,Goto(fax-rx,receive,1)
>>
>> in the sip.conf i specified
>>
>> [general]
>> sendrpid=rpid
>> trustrpid=yes
>> language=de
>> videosupport=yes
>> callevents=yes
>> caninvite=yes
>> qualify=yes
>> nat=force_rport,comedia
>> faxdetect=yes
>> t38pt_udptl=yes
>>
>> ...
>>
>> [abcde]
>> type=peer
>> insecure=invite
>> defaultuser=12345678912
>> fromuser=12345678912
>> fromdomain=abcde.ab
>> secret=guess-what
>> host=abcde.ab
>> qualify=yes
>> context=from-sip
>> dtmfmode=rfc2833
>> callbackextension=12345678912
>>
>>
>> but all i can see if i try to send a testfax is
>>
>> == Using SIP VIDEO CoS mark 6
>> == Using SIP RTP CoS mark 5
>> -- Executing [12345678912 at from-sip:1] Answer("SIP/abcde-00000016", "")
>> in new stack
>> > 0x7fd11404cd00 -- Probation passed - setting RTP source address to
>> 123.456.789.123:17108
>> -- Executing [12345678912 at from-sip:2] GotoIf("SIP/abcde-00000016",
>> "0?black,1") in new stack
>> -- Executing [12345678912 at from-sip:3] Ringing("SIP/abcde-00000016", "")
>> in new stack
>> -- Executing [12345678912 at from-sip:4] Progress("SIP/abcde-00000016", "")
>> in new stack
>> -- Executing [12345678912 at from-sip:5] Wait("SIP/abcde-00000016", "5") in
>> new stack
>> -- Executing [12345678912 at from-sip:6] 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
>>
>>
>> Any hints why thats not working?
>>
>> Best Regards Jakob
>>
>>