[Freeswitch-users] SIP incoming call routing

[Anthony Minessale]

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 at 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 at gmail.com
>
> To: freeswitch-users at 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 at 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 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SIP incoming call routing
>
>
>
>
>
> On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal <
> saurabh_aggarwal at 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_call_from_a_Gateway
>
> ram
>
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> --
> Anthony Minessale II
>
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-- 
Anthony Minessale II

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