[asterisk-users] Call not going through and failing because "never answered"

[Zeeshan Zakaria]

In your sip.conf, there is no mention of your sip provider's IP address,
username and secret (password). Even if the provider doesn't have username
and secret requirements, there should at least be his IP address somewhere
in your sip.conf. Do they require registration? You should ask them what sip
credentials you need to have on your system.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-07-20 5:09 AM, "Andy Beak" <andrewb at cellsmart.co.za> wrote:

Hi,

I'm trying to use Asterisk to place Automated Voice Calls.

A verbose log from Asterisk CLI taken when I place a call in the spool
directory looks like this:

   -- Attempting call on SIP/MTN-NEW/my-number for application
MP3Player(/myfile) (Retry 1)
 == Using SIP RTP CoS mark 5
> Channel SIP/MTN-NEW-00000005 was never answered.
[Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339 attempt_thread: Call failed
to go through, reason (8) Congestion (circuits busy)

My sip.conf looks like this:

[MTN-NEW]
host=192.168.34.1
disallow=all
allow=ilbc
allow=gsm
allow=g729
allow=g723
allow=ulaw
allow=g729
type=peer

My SIP provider says that no traffic is picked up at their SBC or on the WAN
gateway port assigned to us.

I've just done a fresh reinstall of Asterisk and am using sample
configurations for all other conf files.  I am using an open source g729
codec and have tried shuffling the gsm up above it in case it doesn't work
properly (to no avail).

Can anybody help me on this?  My boss is breathing down my neck and I've
never worked with Asterisk before.

Thanks,
 Andy


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