hi
i have configured sip trunk with our asterisk server. it was registered and working fine and i was able to send and receive calls until yesterday, but today when I tried to start asterisk on my PC, strange behavior is seen. it shows reachable and registered status on my asterisk client but on asterisk server it does not show any thing with command ‘sip show registry’ and calls are not being routed to my asterisk client.
on asterisk server i see this error on cli
Got SIP response 489 “Bad event” back from 115.101.10.27(asterisk client’s router ip)
I see contact field in sip register message is wrong. (sip:s@192.168.0.20)
what parameters should i use that will change contact header field to valid address?
see the trace and config below
REGISTER sip:my.ast-server.com:5678 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6f76c341;rport
From: sip:abc@my.ast-server.com;tag=as23f4bafa
To: sip:abc@my.ast-server.com
Call-ID: 129974637fd74fdf561b1fd8402c9da9@192.168.0.20
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“abc”, realm=“asterisk”, algorithm=MD5, uri=“sip:my.ast-server.com:5678”, nonce=“63c5e348”, response=“282d7bb441a5c0170628aa529a77ef0d”, opaque=""
Expires: 120
Contact: sip:s@192.168.0.20
Event: registration
Content-Length: 0
here is my register statement and trunk config.
register => abc:secret@my.ast-server.com:5678
[abc]
type=peer
username=abc
authname=abc
secret=secret
port=5678
host=my.ast-server.com
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes
kindly help. also plz give me address of some good documentation regarding sip header manipulation via sip conf parameters.
thanks
Nasir Javaid