asterisk users - Nov 2006 - Asterisk servers being greedy and not letting go of the media path. (using IAX2 channels)

Evening everyone (obviously depends on when you're readin this, but hey).

I'm trying to set up a multi * server situation, and am falling over at the
second server, and after a day of google etc, have come up against somewhat
of a brick wall.

I can make calls each way between the two servers no problem, and can
include the required extension at the remote * server as part of my main
incoming dialplan. My problem comes with * attempting to pass the media path
to the other server.

What is happening is:

Incoming call from iax2 provider to main * server
  --> dial sip extension on main * server
  --> setup IAX2 channel to remote * server (which then rings extension)

Pickup call on extension on remote * server
  --> main server sip extension stops ringing
  --> ast console on main server I get : 

-------------------------------------------------
   -- Attempting native bridge of IAX2/voipprovider/6 and
IAX2/remote*server/7
    -- Channel 'IAX2/voipprovider/6' unable to transfer
    -- Channel 'IAX2/remote*server/7' unable to transfer
-------------------------------------------------

In the user/friend declarations (user for incoming voip provider, friend for
remote * server) in the two iax.conf files I have notransfer=no, and also up
in the [general] section of the iax.conf.

The problem is that when remote * user answers the phone, and then transfers
the call to an extension on the main * server, there is massive (ie 2
seconds) delay, and using IAX2 show channels at the two consoles, the call
is doing the following:

PSTN -> VOIP PROVIDER -> main * server -> remote * server -> main *
server
-> SIP extension on main * server.

Anyone have any ideas on how to make the * servers give up the media path?

Cheers

Mat

For the benefit of the archives, my problem was a simple one.

I hadn't forwarded the IAX port on the router of the remote * server
connection, so when voip provider was trying to connect directly to the
remote * server, it couldn't.

Hurray for wasting an entire day over a simple silly little thing ;-)

Mat
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> Mat Stace
> Sent: 06 November 2006 17:42
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Asterisk servers being greedy and 
> not letting goof the media path. (using IAX2 channels)
> 
> 
> Evening everyone (obviously depends on when you're readin 
> this, but hey).
> 
> I'm trying to set up a multi * server situation, and am 
> falling over at the second server, and after a day of google 
> etc, have come up against somewhat of a brick wall.
> 
> I can make calls each way between the two servers no problem, 
> and can include the required extension at the remote * server 
> as part of my main incoming dialplan. My problem comes with * 
> attempting to pass the media path to the other server.
> 
> What is happening is:
> 
> Incoming call from iax2 provider to main * server
>   --> dial sip extension on main * server
>   --> setup IAX2 channel to remote * server (which then rings 
> extension)
> 
> Pickup call on extension on remote * server
>   --> main server sip extension stops ringing
>   --> ast console on main server I get : 
> 
> -------------------------------------------------
>    -- Attempting native bridge of IAX2/voipprovider/6 and 
> IAX2/remote*server/7
>     -- Channel 'IAX2/voipprovider/6' unable to transfer
>     -- Channel 'IAX2/remote*server/7' unable to transfer
> -------------------------------------------------
> 
> In the user/friend declarations (user for incoming voip 
> provider, friend for remote * server) in the two iax.conf 
> files I have notransfer=no, and also up in the [general] 
> section of the iax.conf.
> 
> The problem is that when remote * user answers the phone, and 
> then transfers the call to an extension on the main * server, 
> there is massive (ie 2
> seconds) delay, and using IAX2 show channels at the two 
> consoles, the call is doing the following:
> 
> PSTN -> VOIP PROVIDER -> main * server -> remote * server -> 
> main * server
> -> SIP extension on main * server.
> 
> Anyone have any ideas on how to make the * servers give up 
> the media path?
> 
> Cheers
> 
> Mat
> 
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