[asterisk-users] SIP header challenge

[Mr. Jones]

Thanks Rich -

Maybe I'll try the dev mailing list.

I'm not that familiar with the protocol level as well.

I'm thinking its related to one of those two items (user=phone or the
Contact: being blank).

I've looked through all the configs and I don't seem to see any way to
have the Contact fall back to parsing the SIP To: field.

For now I have a macro which does this "ok" but I can't use and of the
web interface tools to manage these extensions as I basically have to
bypass the extension pass the "extension" in to a 2nd macro for
dialing, voicemail, ec.

RJ

On 8/13/06, Rich Adamson <radamson at routers.com> wrote:
> >
> > I'm using a wholesale voip origination provider - they don't deal with
> > end users. As such they have statically defined my Asterisk box on
> > their end - there's no registration or authentication by my system
> > with theirs - other than them hardcoding the destination IP of my
> > server in their system.
> >
> > As such I don't have a username to "register" with them. Additionally
> > they will be originating 100s of DIDs for me at the end of the day, so
> > it would be horrible to have to register all these in some way.
> >
> > Also, I don't have any "register" statements with provider A, and
> > somehow that seems to work out just fine as well.
> >
> > It seems to me like Asterisk doesn't like the format of what's coming
> > from their system. Perhaps the addition of: ";user=phone" is confusing
> > Asterisk?
> >
> > That's the only obvious difference, that and the Contact: header being
> > set to "s".
> >
> > Any other ideas?
>
> Nope, other then if I were trying to troubleshoot the issue, I'd get an
> ethereal trace of both A and B along with a matching sip debug. By
> comparing the two trace methods one should be able to rule out at least
> some issues and narrow the possible root cause.
>
> I don't consider myself a sip protocol expert, but I'd have to guess
> that Olle, Kevin, and few others on the list would be able to help
> interpret that output if its condensed to some reasonable size, and not
> summarized by selected copy/paste of portions of the traces that might
> miss important details.
>
> Given the level of detail and amount of analysis time needed to fully
> understand the issue, posting all of that to the list probably isn't
> going to get the wanted result. Maybe a private email to one of the more
> knowledgeable sip folks requesting their assistance might be helpful.
>
> If the problem turns out to be something like the ";user=phone"
> mentioned, I'd have to wonder if the wholesale provider would actually
> do anything about it though.
>
> R.
>
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