asterisk users - May 2007 - SIP Dial Command to a non-Asterisk url

Dear All,

I have a tiny dial plan like:

[testing]
exten => 454,s,Ringing()
exten => 454,n,Wait(4)
exten => 454,n,Dial(SIP/slee@192.168.45.183:5605,10)
exten => 454,n,Hangup


This connects fine when I dial 454 from any extension in my system,
but there is never any audio?

Where can I start to look for debugging this? It's all internal so no
NAT problems?

Thanks,

Gavin.

Gavin,

   Does the Asterisk server's route to 192.168.45.18 traverse a firewall or
router that may be blocking non-SIP ports that are dynamically allocated?

   SDP -- part of the SIP INVITE transaction payload -- negotiates arbitrary 
ports between the two endpoints for actually passing media.  If these are
being dropped somewhere along the way, you'll have no audio in one or
more directions of the call path.

   Best thing to do is to is a packet capture on the Asterisk server and
filter on 192.168.45.183 to verify that you're seeing bidirectional media,
from and to that host.  Chances are something will be missing.

   Of course, it could be a non-IP problem of some sort as well, perhaps
even something fairly obvious.

-- Alex

--
Alex Balashov   <sasha@presidium.org>

Does the non-Asterisk server _answer_ the line? :)

Gavin Henry wrote:
> Dear All,
> 
> I have a tiny dial plan like:
> 
> [testing]
> exten => 454,s,Ringing()
> exten => 454,n,Wait(4)
> exten => 454,n,Dial(SIP/slee@192.168.45.183:5605,10)
> exten => 454,n,Hangup
> 
> 
> This connects fine when I dial 454 from any extension in my system,
> but there is never any audio?
> 
> Where can I start to look for debugging this? It's all internal so no
> NAT problems?
> 
> Thanks,
> 
> Gavin.
> _______________________________________________
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> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

The 2 most common problems I've seen for "no audio" in one or both
directions is usually either a firewall (which you already said you don't
have) or a CODEC problem.

Make sure both sides are negotiating the same CODEC.  I've often seen
situations where something like the Asterisk server will allow gsm, g711,
etc. and the phone is set for g711, but because gsm was first in the list
on the asterisk side, asterisk was trying to do gsm and the phone wanted
g711 and they wouldn't sync up.  It wasn't until I did a:

	disallow=all
	allow=g711

in sip.conf that it finally started working for me.

That may not be your exact problem, but my guess would be a CODEC issue if
it's not your firewall.

-- Nick


On Wed, 23 May 2007, Gavin Henry wrote:
> Dear All,
> 
> I have a tiny dial plan like:
> 
> [testing]
> exten => 454,s,Ringing()
> exten => 454,n,Wait(4)
> exten => 454,n,Dial(SIP/slee@192.168.45.183:5605,10)
> exten => 454,n,Hangup
> 
> 
> This connects fine when I dial 454 from any extension in my system,
> but there is never any audio?
> 
> Where can I start to look for debugging this? It's all internal so no
> NAT problems?
> 
> Thanks,
> 
> Gavin.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
>