The "treat long distance prefix as optional" is indeed checked (and the
dialplan has been activated).  
 
That said, I think my core question is, according the applicable SIP
RFCs, is the "To:" field in the INVITE packet of SIP "allowed" to be
different than the argument of the INVITE field?  

If they can be different, then I have a gateway issue because it's
clearly using the "To:" field to form its dialstring.  If not, then we
have a sipX issue as its only modifying/updating the INVITE argument
(not the "To:" field).  

I have verified (via tcpdump/wireshark) that the gateway receives from
sipX a SIP packet of this form (i.e. INVITE line with digit '1'
correctly prepended to dialstring, "To:" field without the digit '1'
prepended to dialstring), so it's apparent what's occurring--I'm seeking
which is deemed correct behavior so I can followup and hopefully get it
resolved.

Thx/all thoughts welcomed-

sdm
________________________________

        From: Tony Graziano [mailto:[EMAIL PROTECTED] 
        Sent: Thursday, September 25, 2008 3:35 PM
        To: [email protected]; Stephen D. Miller
        Subject: Re: [sipx-users] Basic SIP INVITE question...
        
        
        Good question.
         
        I don't think your issue is a gateway issue, rather go to your
dialplan in questyion, click ADVANCED and see if the 
        option:
         
        Treat long distance prefix as optional
         
        is checked, if not CHECK IT, apply, activate, try. Perhaps leave
your proxy log set to debug and tail the log file with:
         
        tail /var/log/sipxpbx/sipXproxy.log -f
         
        Post your results.

        >>> "Stephen D. Miller" <[EMAIL PROTECTED]> 09/25/08
03:18PM >>>
        
        In my long distance dial plan, I have sipX (v3.10) prepend the
necessary
        1 for dialing outbound long distance calls over the PSTN.  Using
a
        network sniff, I have verified that sipX does indeed do this and
        creates/sends the following INVITE to the gateway:
        
                INVITE sip:[EMAIL PROTECTED] SIP/2.0
                Record-Route:
        
<sip:172.20.1.101:5060;lr;sipXecs-rs=%2Afrom%7EYmIxY2ZiMjA%60.400_authru
        les%2Aauth%7E%21eae23d5a72f9f0d65973ce6895b2cf13>
                Max-Forwards: 18
                Contact: <sip:[EMAIL PROTECTED]:25948>
                To: "9495551212"<sip:[EMAIL PROTECTED]>
                From: "Garage"<sip:[EMAIL PROTECTED]>;tag=bb1cfb20
                Call-Id: MDVlNDM1NmU3ODE5ZmI0NjhmNGQzMzdiODJmNzg2ZDQ.
                Cseq: 2 INVITE
        
            ...
        
        However, the gateway (with no other dialstring-altering dialplan
of its
        own) stubbornly dials the number in the "To:" field--not the
target in
        the INVITE line of the packet.  This call then fails in the PSTN
since
        it's missing the leading digit '1'.
        
        Am I correct in assuming this is the gateway's problem--and not
one
        where sipX should be altering the "To:" field of the INVITE?
        
        Any info/help is appreciated/
        
        sdm
        
        (Gateway=Grandstream GXW4104)
        
        
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