asterisk users - Oct 2005 - overlap zaphfc - dialtone

Hello all,

 

I have a problem with overlap dialing and don't know how to get rid of it.

 

My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D channels), SIP
phones (I just removed TDM400P with 4 FXS)

 

 

I created test extension 222 which goes directly to g1. In Zapata.conf
overlapdial is set to yes.

 

First I created this extension:

 

exten => 222,1,Dial(zap/g1,100,tc)

 

and channel got hangup every time.. So I even saw bug 4913
http://bugs2.digium.com/view.php?id=4913
<http://bugs2.digium.com/view.php?id=4913&nbn=1> &nbn=1 and bug
4771.. But
that wasn't my problem. my problem is that I didn't included / after
g1..
So, I changed that and now my extension look like:

 

exten => 222,1,Dial(zap/g1/,100,tc)

 

This solved the problem with line being hangup-ed like in bug 4913, and I am
getting the telco dialtone.

 

So, when dialing 222 I get:

 

A12*CLI>

    -- Executing Dial("SIP/200-dd52", "zap/g1/|100|tc") in
new stack

    -- Requested transfer capability: 0x00 - SPEECH

    -- Called g1/

    

After that, I can even dial from this dialtone, but when called party rings
I get the following message and auto hangup:

 

    -- Zap/1-1 is making progress passing it to SIP/200-dd52

    -- Zap/1-1 is ringing, hanging up.

    -- Hungup 'Zap/1-1'

 

 

The called phone stops ringing, Zap channel hangs up, And SIP phone is still
"on the air" without anybody.

 

Thank You all for help,

Goran

 

 

 

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>My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D 
>channels), SIP phones (I just removed TDM400P with 4 FXS)
>
>I created test extension 222 which goes directly to g1. In 
>Zapata.conf overlapdial is set to yes.
>
>First I created this extension:
>
>exten => 222,1,Dial(zap/g1,100,tc)
>
>and channel got hangup every time.. So I even saw bug 4913 
>http://bugs2.digium.com/view.php?id=4913&nbn=1 and bug 4771.. 
>But that wasn't my problem. my problem is that I didn't 
>included / after g1.. So, I changed that and now my extension 
>look like:
>
>exten => 222,1,Dial(zap/g1/,100,tc)
>
>This solved the problem with line being hangup-ed like in bug 
>4913, and I am getting the telco dialtone.
>
>So, when dialing 222 I get:
>
>    -- Executing Dial("SIP/200-dd52", "zap/g1/|100|tc")
in new stack
>    -- Requested transfer capability: 0x00 - SPEECH
>    -- Called g1/
>    -- Zap/1-1 is making progress passing it to SIP/200-dd52
>    -- Zap/1-1 is ringing, hanging up.
>    -- Hungup 'Zap/1-1'

It seems that when it detects Ringing, Asterisk executes Hangup in the same
time.... But why?