Thanks Eric for the info. That's what I was looking for.
It's a "low-cost" gateway (Grandstream GXW-4104) so I guess I shouldn't
be that surprised. It works well for most configurations, but issues
like this make me wish I'd simply moved a bit further up-market to a
Patton or similar.
sdm
________________________________
From: eric aaron [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 25, 2008 4:24 PM
To: Stephen D. Miller
Subject: Re: [sipx-users] Basic SIP INVITE question...
On Thu, Sep 25, 2008 at 3:59 PM, Stephen D. Miller
<[EMAIL PROTECTED]> wrote:
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?
Yes, the TO-URI is allowed to be different and should be
different when a transformation (such as a prefix) takes place. The
TO-URI is supposed to reflect the actual dialstring sent by the client,
and is normally not changed.
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).
It does sound like a gateway issue. What type of gateway is it?
You should be able to tell it to route based on the R-URI.
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]
<mailto:[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]
<mailto:[EMAIL PROTECTED]> >
From: "Garage"<sip:[EMAIL PROTECTED]
<mailto:[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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