Asterisk dial plan to move calls to other servers

Hi folks
i want to make static dial plan in a case some call a number ,i want asterisk to send the call to remote server in the way of 100@ as an example .

as example if i dial 200 , i want the call go to other server to the uri 100@

i tried below :
exten => _X.,1,NoOp(“The Context is :” ${CONTEXT} “and destination is”${EXTEN})
same => n,Dial(SIP/100@,30)
same => n,Hangup()

here is a test

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [222@xp90:1] NoOp(“SIP/00000-0001e9e8”, "“The Context is :” xp90 “and destination is"222”) in new stack
– Executing [222@xp90:2] Dial(“SIP/00000-0001e9e8”, “SIP/100@,30”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/100@
[2017-05-19 23:04:16] NOTICE[5622][C-0004b16f]: chan_sip.c:23281 handle_response_invite: Failed to authenticate on INVITE to ‘sip:00000@;tag=as4de5a082’
– SIP/ is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [222@xp90:3] Hangup(“SIP/00000-0001e9e8”, “”) in new stack
== Spawn extension (xp90, 222, 3) exited non-zero on ‘SIP/00000-0001e9e8’

the other side will be allowing all incoming request from the sender ip

hope to help
cheers should be, if your specification is correct. wants you to provide a password, which you can either do in the dial string, or in a sip.conf entry that you use instead of the IP address.

Note this requirement is the standard outgoing call through ITSP scenario, so should be trivial.

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