[asterisk-users] How to use 'Transfer' to send calls to another asterisk?

Deepesh D deep.d2010 at gmail.com
Thu Oct 11 05:44:28 CDT 2012


In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'.

When I directly dial from C1 into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.

On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy <seandarcy2 at gmail.com> wrote:
> On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D <deep.d2010 at gmail.com> wrote:
>> Hello,
>>
>> How do I use the asterisk application 'Transfer' to transfer a SIP
>> call from one asterisk to another?
>>
>> I have the following scenario. I have two asterisk servers S1 and S2.
>> There is a third asterisk server C1 which registers as a peer to S1.
>> From C1, I dial into S1 using 'Dial' command. What I want to do is,
>> use the Transfer command in S1 and transfer the call to S2.
>>
>> Dialplan on S1
>> [test_context]
>> exten => _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
>> exten => _X.,n,NoOp(${TRANSFERSTATUS})
>> exten => _X.,n,Hangup
>>
>> Dialplan on S2
>> [default]
>> exten => _X.,1,Playback(somemsg)
>> exten => _X.,n,Hangup
>>
>> [test_context]
>> exten => _X.,1,Answer
>> exten => _X.,n,Playback(msg)
>> exten => _X.,n,Hangup
>>
>> The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
>>
>> In C1, I have set 'promiscredir = yes' in sip.conf.
>>
>> When I dial from C1, the call is successfully transferred to S1 (I get
>> TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
>> S2). But the call does not get authenticated on S2 and goes into
>> default context instead of 'test_context'. How can I transfer the call
>> such that S2 authenticates the call and sends it to the required
>> context?
>>
>> Thanks
>>
>
> What happens when you dial into S2 from outside?
>
> Did you set a context in sip.conf on S2?
>
> sean
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list