[asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

Wed Aug 1 12:52:59 CDT 2012

This is /etc/dahdi/system.conf

fxsks=1-4
fxsks=5-8
echocanceller=mg2,1-8
loadzone = us
defaultzone=us

And /etc/asterisk/chan_dahdi.conf
language=en
context=fax-out
signalling=fxs_ks
faxdetect=both
echocancel=no
cancallforward=yes
canpark=yes
transfer=yes
echocancelwhenbridged=yes
group=3
callgroup=3
channel => 4-4

Is the actual hardware that I should change? 

Thanks,

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, August 01, 2012 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

Sounds like DAHDI/4 is a FXO port.  FXO ports are considered answered when
dialing is completed.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of motty.cruz
Sent: Wednesday, August 01, 2012 1:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

 that is correct! The reason I think is because when Dahdi "answered"
iaxmodem thinks is the remote fax machine that answered, but it reality it
keeps ringing, if the remote fax machine answer within the first ring then
iaxmodem connects but if not it does not detect a fax. 

Here is a sip extension dialing throught the same context. 

dxxx*CLI>
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Executing [xxx1463 at fax-out:1] Dial("SIP/507-00000000",
"dahdi/g3/wwxxx1463") in new stack
    -- Called g3/wwxxx1463
    -- DAHDI/4-1 answered SIP/507-00000000
    -- Hungup 'DAHDI/4-1'

Dahdi answered, but after dahdi answer it rang for 4 rings before remote fax
answered, if I would be the iaxmodem I would had given up by then, 

Do you see my problem? Anybody else experienced same issue? 

Thanks, 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Nelson
Sent: Wednesday, August 01, 2012 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

----- Original Message -----
> Thanks Tim,
> I tried your suggestion below the logs:
> 
>     -- Accepting AUTHENTICATED call from xxx.xx.xx.xx:
>        > requested format = ulaw,
>        > requested prefs = (),
>        > actual format = ulaw,
>        > host prefs = (ulaw),
>        > priority = mine
>     -- Executing [xxx1463 at fax-out:1] Dial("IAX2/503-7761",
> "dahdi/g3/wwxxx1463") in new stack
>     -- Called g3/wwxxx1463
>     -- DAHDI/4-1 answered IAX2/503-7761
>     -- Registered IAX2 '503' (AUTHENTICATED) at xxx.xx.xx.xx:4570 [Aug
> 1 09:04:59] NOTICE[3392]: chan_iax2.c:8486 update_registry:
> Restricting registration for peer '503' to 300 seconds (requested 60)
>     -- Registered IAX2 '503' (AUTHENTICATED) at xxx.xx.xx.xx:44145 
> [Aug  1 09:05:03] NOTICE[3391]: chan_iax2.c:8486 update_registry:
> Restricting registration for peer '503' to 300 seconds (requested 60)
>     -- Hungup 'DAHDI/4-1'
>   == Spawn extension (fax-out, xxx1463, 1) exited non-zero on 
> 'IAX2/503-7761'
>     -- Hungup 'IAX2/503-7761'
> 
> 
>  [root at drew home]# faxstat -s
> HylaFAX scheduler on host.xxxxx.com: Running Modem ttyIAX0
> (+1.xxx.8626): Running and idle
> 
> JID  Pri S  Owner Number       Pages Dials     TTS Status
> 9    126 S   root xxx1463       0:1   1:12   16:10 No carrier
> detected
> 

Your setup looks correct. Can you connect a normal analog phone to the POTS
line and dial that fax number directly? I just want to see if that is
successful or not, indicating if the problem is PSTN related (need to dial
10 digits, or 1+10 for example in the US).

The interesting thing is the result within Hylafax is 'No Carrier' which
means the call was indeed answered, but fax was not present on the other
side.

--Tim

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users