[Asterisk-Dev] SIP INVITE vs TO URI

[William Lloyd]

Yea, but like Kevin mentioned this isn;t the issue.

My original request might not have been that clear..

I can;t create a second register because there is only 1 register  
account for a bunch of DID's.

If the fix was to append the extension to the end of registration  
this wouldn;t be a -dev issue.

Thanks.

-bill


On 14-Sep-05, at 3:47 PM, Sherwood McGowan wrote:

> Also, create a second register for the other number, or numbers and  
> then you
> can route accordingly anyway.
>
>
>
> ->-----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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