Have you tried Dial instead of Transfer? -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Deepesh D Sent: Thursday, October 11, 2012 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
>From C1 when I directly dial 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. In all my peer definitions on S1 and S2 I define the context as 'test_context' and the default context is 'default' On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy <[email protected]> wrote: > On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D <[email protected]> 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 -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
