i’ve got a problem with the ringduration of sip calls.
The called phone rings only about 20 sec and then the call is aborted.
[general] srvlookup=yes language=de disallow=all allow=alaw register => foo:email@example.com/exten-foobar [sipsipdiscount] type=peer host=sip1.sipdiscount.com dtmfmode=rfc2833 canreinvite=yes fromdomain=sip1.sipdiscount.com username=foo fromuser=foo secret=foobar qualify=100
I also tried other sip provider but this doesnt change anything.
Here’s my extensions.conf entry for dialing:
[extern] exten => 321,1,Dial(SIP/somenumber@sipsipdiscount)
Here’s a log from such a unanswerd call:
-- Accepting overlap voice call from '123' to '321' on channel 0/1, span 1 -- Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1", "SIP/somenumber@sipsipdiscount|60") in new stack -- Called somenumber@sipsipdiscount -- SIP/sipsipdiscount-a150 is making progress passing it to Zap/1-1 -- Channel 0/1, span 1 got hangup request == Spawn extension (extern, 321, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1'
If the call gets answerd in the ring-duration everything works as it should.
I’m using asterisk 1.2.6 with bristuff 0.3.0-PRE-1n.
I hope someone can give me a hint to solve this problem…