[HELP] Problem with the ring duration while calling via SIP

#1

Hello,

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.

my sip.conf:

[general]
srvlookup=yes
language=de
disallow=all
allow=alaw

register => foo:bar@sip1.sipdiscount.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…

Thanks!