[Freeswitch-users] domain-wise context

[Anthony Minessale]

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension

The first one transfers the call to the desired exten/dialplan/context and
the 2nd one executes the specified extension in a similar manner and returns
to the same point in the dp.

You make one inbound context and use routing logic from there to decide
which context to transfer to.

Next time, please have a little more patience, I don't like it when people
reply to themselves on the list asking why nobody answered when their
question is only unanswered for 2 days especially during a holiday weekend.



On Sat, Apr 3, 2010 at 1:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> I expected at east one reply saying that the question is stupid, and the
> solution is simple !!
> Any folks who can help me understand only how to achieve this in FS which
> is acheivable in asterisk as follows:
> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
> the call goes to context "mydomain" in dialplan)
> domain = yourdomain.com, yourdomain (if any call has domain as "
> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>
> These calls can come from anywhere, in my case it comes from an Opensips
> instance !!
>
> Thanks for any replies :)
>
> --- Jayesh
>
>
> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi All,
>> I am quite very new to freeswitch and I am kind of playing with it to
>> understand it better.
>> I am primarily using Opensips as registrar and SIP Proxy and intend to use
>> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
>> Opensips is a multi-domain setup and I wish to have all the configuration of
>> media-capabilties segregated domain-wise in the FS too.
>>
>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>> redirect that call to FS IP address keeping the URI intact. I add the
>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>> go to voicemail I do the same thing.
>> The requirement is I want to maintain the dialplans for each domains
>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>> the call should go to dialplan context domain1 and similarly if call from
>> Opensips comes to FS with domain2 the call handling should be mentioned in
>> the domain2 context.
>>
>> The problem is; I am not able to send the calls to respective contexts
>> according to their domains when they come from Opensips. I've read the
>> examples on multi-domain setup and have tried taking some help from that
>> example, but whenever the call comes from Opensips to FS, it tries to go
>> into the context that is defined in the SIP Profile. If i don't mention
>> anything in the SIP Profile, it tries to search for default context.
>> I have tried the following:
>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>> 2) mentioned parameters in domain1.xml as follows:
>> <include>
>> <domain name = "domain1.com"
>> <params>
>> <param name="user_context" value "domain1.com"/>
>> </params>
>> </domain>
>> </include>
>> 3) Similarly done for file domain2.xml.
>>
>> But I am just not able to get the calls to the required context according
>> to the domain value in the r-uri. In asterisk something like this in
>> sip.conf worked fine for me:
>> domain=domain1.com, domain1.com
>> domain=domain2.com, domain2.com
>>  Can someone please help me understanding where I am going wrong or have I
>> mis-understood something?
>>
>> Thanks in advance !!
>>
>> --- Jayesh
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100403/37494126/attachment.html