[Asterisk-Dev] SIP INVITE vs TO URI

[Steve]

What if he wants to route each number accordingly. I.E. 1212 to a different
context and 1213 to another context.

----- Original Message ----- 
From: "Sherwood McGowan" <madprofzero at yahoo.com>
To: "'Asterisk Developers Mailing List'" <asterisk-dev at lists.digium.com>
Sent: Wednesday, September 14, 2005 12:46 PM
Subject: RE: [Asterisk-Dev] SIP INVITE vs TO URI


> It's because your system is passing 's' as it's number, you have to append
> /1XXXYYYZZZZ at the end of your register statement. We come across this a
> lot at my company when our customers try to connect asterisk to us.
>
> Cheers and good luck
>
> ->-----Original Message-----
> ->From: asterisk-dev-bounces at lists.digium.com
> ->[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of
> ->William Lloyd
> ->Sent: Wednesday, September 14, 2005 2:56 PM
> ->To: asterisk-dev at lists.digium.com
> ->Subject: [Asterisk-Dev] SIP INVITE vs TO URI
> ->
> ->I'm trying to integrate Asterisk to work with a SIP provider.
> ->Outbound calls are not an issue.
> ->
> ->It's inbound with DID's I'm looking at.
> ->
> ->In sip.conf when I register the provider without an
> ->extensions like register => 1XXXYYYZZZZ:password at sip.provider.com:5060
> ->
> ->Asterisk registers at the provider with the 's' extension.
> ->
> ->the problem comes with the provider and the way they setup
> ->the user accounts.  They consider you to only have 1 account,
> ->not 2.  So the pilot number of 15145551212 is the only one
> ->that it accepts registrations from.
> ->
> ->For example say you have 2 DID's with this provider
> ->15145551212 and
> ->15145551213
> ->
> ->you would register with
> ->register => 15145551212:password at sip.provider.com:5060
> ->
> ->but when there is an incoming call from the provider for
> ->15145551213 the provider passes the destination in the TO
> ->field of the SIP header not in INVITE.
> ->
> ->The provider sends the destination as part of the TO field.
> ->In the example the asterisk host is behind NAT.
> ->
> ->For example
> ->INVITE sip:s at 192.168.100.20
> ->TO: <sip:15145551213 at 69.23.45.200>
> ->
> ->or
> ->
> ->INVITE sip:s at 192.168.100.20
> ->TO: <sip:15145551212 at 69.23.45.200>
> ->
> ->Both get handled by Asterisk s destination.
> ->
> ->I've been looking at the get_destination and
> ->handle_request_invite code in chan_sip.c to modify this
> ->behavior when the INVITE URI and TO URI do not match.
> ->
> ->In a quick read of the SIP RFC's this providers
> ->implementation seems to be valid.
> ->
> ->How would be the best way to handle this addition in the
> ->asterisk code?
> ->
> ->Ideally I'd like to approach it in a way that would be
> ->accepted back into the base asterisk code.
> ->
> ->thought?  suggestions?
> ->
> ->-bill
> ->wlloyd at slap.net
> ->
> ->
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>
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