I am able to register S1 as a peer in S2 and dial from S1 to S2, but this is not my requirement. I want to dial from C1 into S1 and S1 should redirect the call to S2.
I am trying to do a load balancing setup between S1 and S2. S1 will be primary server which accepts all calls and then based on some conditions redirect some calls to S2 On Thu, Oct 11, 2012 at 6:39 PM, Danny Nicholas <[email protected]> wrote: > So what happens when you dial directly from S1 to S2? > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Deepesh D > Sent: Thursday, October 11, 2012 5:44 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another > asterisk? > > 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 <[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 -- _____________________________________________________________________ -- 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
