Hello everybody,
I'm using villagetelco firmware
GA01.1/SECN/;
after configuration, SIP registers and calls are received; but, when trying to call out via my SIP provider I get
- Starting simple switch on 'DAHDI/1-1'
-- Executing [02511259@incoming-local:1] Dial("DAHDI/1-1", "SIP/02511259@sipaccount,120,r") in new stack
[Oct 15 17:38:55] WARNING[2483][C-00000002]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'DAHDI/1-1' status is 'CHANUNAVAIL'
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
Trunk is not recognized;
after checking last working configuration, I identified a /etc/config/voip file which is missing in the filesystem
installing this file and restarting asterisk (/etc/init.d/asterisk restart) make things work.
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [02511259@incoming-local:1] Dial("DAHDI/1-1", "SIP/02511259@sipaccount,120,r") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/02511259@sipaccount
-- SIP/sipaccount-00000000 is making progress passing it to DAHDI/1-1
-- SIP/sipaccount-00000000 answered DAHDI/1-1
> 0x72d9a0 -- Probation passed - setting RTP source address to
62.94.199.39:52700 == Spawn extension (incoming-local, 02511259, 1) exited non-zero on 'DAHDI/1-1'
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
So this file looks like needed to have SIP calls work;