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 Post subject: Can not call
PostPosted: Wed Jan 31, 2007 7:27 pm 
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Joined: Wed Jan 31, 2007 6:58 pm
Posts: 1
Hello all,

When I tried to make a call with an X-Lite, a voice tell me :

1. You have 150.27cents please enter the number that you to call and press the pound key

(150.27 is the balance of my customer)

2. When I enter the number wich i would call (ie:00330673569244) the same voice come.

Some one have an idea ?

Thanks to all.

Asterisk used : Trixbo v1.2.3 / FreePBX 2.1.1
Softphone : X-Lite v3.0


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 Post subject:
PostPosted: Wed Jan 31, 2007 8:06 pm 
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Joined: Tue Jun 06, 2006 12:14 pm
Posts: 685
Location: florida
You have to read your CLI and see what's happening. Whether its your DTMF digits not coming in, or maybe its finding no route, or it thinks that's an invalid number for example.


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 Post subject:
PostPosted: Wed Jan 31, 2007 10:51 pm 
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Joined: Mon May 29, 2006 7:07 pm
Posts: 287
Location: Denver
Yea i agree it's probably DTMF not set right in additional_a2billing_sip.conf file - it should be dtmfmode=rfc2833 in most cases.


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 Post subject:
PostPosted: Wed Jan 31, 2007 11:07 pm 
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Joined: Fri Apr 28, 2006 5:11 am
Posts: 426
i'm not agree. ulaw and alaw can not be as rfc2833. so, for most cases it's should be as
dtmfmode=auto
in case when ulaw/alaw it's will work as =inbound, but when will any other codecs - will be =rfc2833 in auto switch by asterisk.
this is good documented feature.


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 Post subject:
PostPosted: Mon Mar 19, 2007 11:21 am 
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Joined: Sat Apr 15, 2006 2:26 pm
Posts: 87
Location: Germany
anest wrote:
i'm not agree. ulaw and alaw can not be as rfc2833.

Sorry, why not?
http://www.ietf.org/rfc/rfc2833.txt
This RFC states a method of encoding DTMF in RTP packets. There is nothing to do with codecs.
The fact is, using ulaw/alaw it's possible to pass DTMF in-band. But it doesn't mean you can't use rfc2833.


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 Post subject:
PostPosted: Wed Mar 28, 2007 11:21 am 
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Joined: Fri Apr 28, 2006 5:11 am
Posts: 426
this does not matter. now you can set it as you wish. before this was not work.
if you dont trust me - please read digium bugtrack and voip-info.org before criticize me ;)


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