[asterisk-users] SIP URI Forwarding

[Rizwan Hisham]

Hi all,
I am having a problem with sip uri incoming calls. I have 2 asterisk servers
both are 1.4.2. i dial sip uri from one asterisk server which sends the call
to the other asterisk server by seeing its domain name in the uri. Invite
reaches the recieving asterist server but the call is not autenticated.
Everytime i see the following NOTICE on the asterisk server (caller end)

[Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite:
Failed to authenticate on INVITE to '"rizwan" <sip:abc at 192.168.0.7:9860
>;tag=as089d4adb'

My dialplan on caller end is:

[directcall]
exten=> 123,1,Dial(SIP/abc:0786 at tulip.axvoice.com:9060)
exten=> 123,2,Hangup()

exten=> 456,1,Dial(SIP/adf:123 at tulip.axvoice.com:9060)
exten=> 456,2,Hangup()

SIP general settings on receiving end are:

[general]
context=uricall-incoming
allowoverlap=no
bindport=9060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
relaxdtmf=yes
useragent=Asterisk PBX
dtmfmode = rfc2833
nat=no
canreinvite=yes

peer settings on receiving end:

[adf]
username=adf
type=friend
secret=XXX
qualify=25000
nat=yes
insecure=port,invite
host=dynamic
dtmfmode=rfc2833
context=sipuri-incoming
canreinvite=yes
callerid="adf xyz" <123>
accountcode=6:0:adf
amaflags=default
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm

am i doing something wrong here?


-- 
Best Regards
Rizwan Hisham
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