If there is a secret= on the receiving peer, the sending peer needs to
provide that secret. Along with a username.
Rizwan Hisham wrote:
> Hi all,
> I am having a problem with sip uri incoming calls. I have 2 asterisk
> servers both are 1.4.2. <http://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
> <http://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
> <http://abc:0786 at tulip.axvoice.com:9060>)
> exten=> 123,2,Hangup()
>
> exten=> 456,1,Dial(SIP/adf:123 at tulip.axvoice.com:9060
> <http://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 <http://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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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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