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" <andr...@xxxxxxxxxxxxxxx <mailto:andr...@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=peerMy 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
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