[Asterisk-Dev] SIP INVITE vs TO URI

[Sherwood McGowan]

My apologies. I jumped because I had seen that problem before. I'm quite
sorry to have jumped the gun.

Cheers,
Sherwood

->-----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 4:03 PM
->To: Asterisk Developers Mailing List
->Subject: Re: [Asterisk-Dev] SIP INVITE vs TO URI
->
->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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