The extension param only influences the username portion of the contact address?. If they require a certain contact address username they are insane. We also have extra params like from-domain, and caller-id-in-from to compensate for other foolish broken sip services who use pointless unnatural requirements on the contents of your invite for security. If you must set the contact to your username, you would have to match on something else in the invite. It would be up to them to send something significant in the invite that you could match against to tell what number they are calling. destination_number is not the only thing you can regex for, route all calls to the "info" app and call the various did and look for a variable that will tell you the info you need. On Wed, Nov 5, 2008 at 1:36 AM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
Thanks - that does work to an extent.
Now the problem is that not all gateways would allow "arbitrary" extensions. E.g. AIM Callout - it *requires* that the extension/caller-id be your aim username.
-Saurabh
------------------------------
Date: Wed, 29 Oct 2008 12:46:44 -0500 From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP incoming call routing
whatever you put in the "extension" param in the gateway should control what destination_number it has in the inbound call. you can also do your regex in your dialplan on any of the info in the sip packet besides destination number if you wish.
On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register.
-Saurabh
------------------------------
Date: Wed, 29 Oct 2008 15:12:28 +0530 From: talk2ram@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP incoming call routing
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway.
Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem?
Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping?
-Saurabh
have you looked at this example
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c...
ram
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