[Asterisk-Users] An interesting call path observation..

[asterisk at lists.styx.org]

On Wed, Nov 26, 2003 at 06:18:00PM +0100, Philipp von Klitzing wrote:
> 
> The theory - as far as I was able to find out - involves:
> 
> transfer=yes/no in iax.conf
> canreinvite=yes/no in sip.conf

I have run into a related problem, and can't figure my way
out of it. This involves two asterisk servers. A and B.
A is here with my phone attached to it, B is remote and
unattended for the moment and undergoing some testing.

A has, in extensions.conf something like:

exten => 1,1,Dial,IAX2/B/1|60|tT
exten => 2,1,Dial,Zap/0/5145551212

B has,

exten => 1,1,DISA,no-password|default
exten => 2,1,Dial,IAX2/A/2|60|tT

Both also have some zaptel hardware installed and
trunking enabled.

So from my SIP phone I dial 1 and get the dialtone from B.
So far so good. Then I dial 2, and B dutifully calls 
A, my PSTN phone rings and then hands off the call and takes
itself out of the loop once I have answered.

But A keeps both the IAX2 channels up and no audio
is passed. 'show channels' on A looks something like:

CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
        Zap/1-1  (default	s	  1   )      Up Bridged Call  IAX2[B at B]/16385
IAX2[B at B]/16385  (default       s         1   )      Up Dial          Zap/0/5145551212
IAX2[B]/16384    (default       s         1   )      Up Bridged Call  SIP/ParcPhone-3ad1
SIP/ParcPhone-3ad1  (default    s         1   )      Up Dial          IAX2/B/1

Zap/1-1 should get bridged to SIP/ParcPhone-3ad1 and the IAX2
channels torn down, but that does not happen. Is it because of
the mismatch -- 'B' vs. 'B at B' that the calls are not percieved to
be hairpining on the same trunk? Or...

hmmm...

-w

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