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

[Andy Beak]

Hi,

I set my list to subscribe to digest and I can't see how to reply to 
your reply without starting a new thread.

There is no need for SIP username and password because the provider 
authenticates me on my IP address.

I thought that "host=192.168.34.1" would be the sip provider IP address.

At this point I don't need to accept incoming calls or place 
VOIP-to-VOIP.  All I need to do is connect to their PBX to place a call 
to a cellphone.

I reread all the documentation I could find and couldn't see where else 
in sip.conf I should set the provider IP.

Thanks for your reply,
   Andy

 > 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 <http://www.ilovetovoip.com>

> On 2010-07-20 5:09 AM, "Andy Beak" <andrewb at xxxxxxxxxxxxxxx 
> <mailto:andrewb at xxxxxxxxxxxxxxx>> 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
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided byhttp://www.api-digital.com  --
New to Asterisk? Join us for a live introductory webinar every Thurs:
                http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
    http://lists.digium.com/mailman/listinfo/asterisk-users



-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 5606 bytes
Desc: S/MIME Cryptographic Signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/dc0a2f3a/attachment.bin