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
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