[asterisk-users] SIP URI Forwarding

Rizwan Hisham rizwanhasham at gmail.com
Wed Sep 17 09:25:00 CDT 2008


thats what i am passing

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

adf is username and 123 is the password



On Wed, Sep 17, 2008 at 6:01 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

>
> 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@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@tulip.axvoice.com:9060>)
> > exten=> 123,2,Hangup()
> >
> > exten=> 456,1,Dial(SIP/adf:123 at tulip.axvoice.com:9060
> > <http://adf:123@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
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best Regards
Rizwan Hisham
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080917/89cba836/attachment.htm 


More information about the asterisk-users mailing list