Hi list,
I face the following problem on incoming calls from my provider which
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
not sended to the context set in provider sip.conf definition, but are
going to the default context setted in [general].
Provider uses few IP's for incoming calls which are not the one used for
register.
Here are the logs:
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: --- (15 headers 22
lines) ---
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: Sending to
85.xx.xx.2:5060 (no NAT)
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: Using INVITE request as
basis request - 07403bb3412fc5206dec905b4eb26...@85.xx.xx.2
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: No matching peer for
'0033xxxxxxxxx' from '85.xx.xx.2:5060'
...
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: Peer audio RTP is at
port 85.x.xx.2:16566
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: Looking for 027xxxxxxin
default-guest (domain 217.yy.yy.yy)
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: list_route: hop:
<sip:0033xxxxxx...@85.xx.xx.2>
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: RDNIS for this call is
027xxxxxx (reason )
Our asterisk is registered with the provider, registerer IP from the
provider being 85.xx.xx.3:
Sip.conf
[general]
context=default-guest ;where incoming calls ended
...
register => 01234567:mysec...@sip.provider.net/01234567
[01234567]
type=peer
defaultuser=01234567
secret=mysecret
host=sip.provider.net
deny=0.0.0.0/0.0.0.0
permit=85.xx.xx.0/255.255.255.0
directmedia=no
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw,alaw
context=from-Provider
insecure=port,invite
fromdomain = sip.provider.net
fromuser=01234567
sendrpid = yes
nat=yes
What is wrong?
Thanks for any hint
--
Daniel
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