Re: [asterisk-users] Asterisk Fax detection *11.7

2014-02-01 Thread Jakob-Matthias Böttger 

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

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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-23 Thread Martin 



in the sip.conf i specified

[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes


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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[asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Jakob-Matthias Böttger 

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/123SIP/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@from-sip:1] Answer(SIP/abcde-0016, 
) in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source 
address to 123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016, 
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, 
) in new stack
-- Executing [12345678912@from-sip:4] 
Progress(SIP/abcde-0016, ) in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 
5) in new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016, 
SIP/123SIP/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/123-0018 connected line has changed. Saving it until 
answer for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until 
answer for SIP/abcde-0016

-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing


Any hints why thats not working?

Best Regards Jakob


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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Leandro Dardini 
It is really more interesting the receiving part. Can you paste here?

Leandro


2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de

 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/123SIP/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@from-sip:1] Answer(SIP/abcde-0016,
 ) in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address
 to 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5)
 in new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/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/123-0018 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 is ringing


 Any hints why thats not working?

 Best Regards Jakob


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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Jakob-Matthias Böttger 

Hi

The log i've posted

== Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
-- Executing [12345678912 tel:%5B12345678912@from-sip:1] 
Answer(SIP/abcde-0016, ) in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source 
address to 123.456.789.123:17108
-- Executing [12345678912 tel:%5B12345678912@from-sip:2] 
GotoIf(SIP/abcde-0016, 0?black,1) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:3] 
Ringing(SIP/abcde-0016, ) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:4] 
Progress(SIP/abcde-0016, ) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:5] 
Wait(SIP/abcde-0016, 5) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:6] 
Dial(SIP/abcde-0016, SIP/123SIP/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/123-0018 connected line has changed. Saving it until 
answer for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until 
answer for SIP/abcde-0016

-- SIP/123-0018 is ringing
-- SIP/456-0017 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
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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Leandro Dardini 
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 Böttger ja...@j-mb.de

  Hi

 The log i've posted


 == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
 ) in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address
 to 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5)
 in new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/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/123-0018 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 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

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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Jakob-Matthias Böttger 

i already added a Progess() and Wait(5) and it still does not detect faxes.


Am 21.01.2014 16:53, schrieb Leandro Dardini:
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 Böttger ja...@j-mb.de mailto:ja...@j-mb.de

Hi

The log i've posted


== Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
-- Executing [12345678912 tel:%5B12345678912@from-sip:1]
Answer(SIP/abcde-0016, ) in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source
address to 123.456.789.123:17108
-- Executing [12345678912 tel:%5B12345678912@from-sip:2]
GotoIf(SIP/abcde-0016, 0?black,1) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:3]
Ringing(SIP/abcde-0016, ) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:4]
Progress(SIP/abcde-0016, ) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:5]
Wait(SIP/abcde-0016, 5) in new stack
-- Executing [12345678912 tel:%5B12345678912@from-sip:6]
Dial(SIP/abcde-0016, SIP/123SIP/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/123-0018 connected line has changed. Saving it
until answer for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it
until answer for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 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

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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Leandro Dardini 
Please paste the actual code. First has to be the Wait and then any other
thing.

Leandro


2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de

  i already added a Progess() and Wait(5) and it still does not detect
 faxes.


 Am 21.01.2014 16:53, schrieb Leandro Dardini:

 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 Böttger ja...@j-mb.de

  Hi

 The log i've posted


 == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016,
 ) in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address
 to 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016,
 ) in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016,
 5) in new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/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/123-0018 connected line has changed. Saving it until
 answer for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until
 answer for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 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

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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Paul Belanger 
On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger ja...@j-mb.de 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/123SIP/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@from-sip:1] Answer(SIP/abcde-0016, )
 in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address to
 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
 in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
 in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
 new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/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/123-0018 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 is ringing


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.

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread Larry Moore 

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/123SIP/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@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
  0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/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/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing


Any hints why thats not working?

Best Regards Jakob




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Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-21 Thread 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/123SIP/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@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/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/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing


Any hints why thats not working?

Best Regards Jakob




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