[Freeswitch-users] Incoming call from registered gateway with DID in To: not URI

[Steven Ayre]

<param name="extension" value="auto_to_user"/> was what did the trick
though. Thanks! :)

There's a minor issue from it where the trunk shows unregistered on
their website (they expect the Contact header to be username at ip") but
incoming calls are still sent through fine so it doesn't actually
matter at all. They can still see FS registered on their switch.

I've added the configuration to the SIP Provider Examples section of the Wiki.

One of their guys said by email they tested on Git within the last few
weeks and found calls were hanging up after 2m40s due to something
about how we handle timers... I'm going to test that next. Were there
any versions in Git recently that might have had problems?

-Steve




On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
> Also if you set the extension to auto_to_user it'll do it also.
>
> /b
>
> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>
>> I'm currently handling it with:
>>
>>  <param name="context" value="numbergroup"/>
>>
>> and:
>>
>>  <context name="numbergroup">
>>    <extension name="numbergroup">
>>      <condition field="destination_number" expression="^USERNAME$">
>>        <action application="transfer" data="${sip_to_user} XML default"/>
>>      </condition>
>>    </extension>
>>  </context>
>>
>> Is there a better way of handling it? Such as a param that tells Sofia
>> to use the To not the URI to populate destination_number?
>>
>> -Steve
>
>
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