How to call “virtual extensions” using AMI?.
Action: Originate
Channel: SIP/422
Context:from-internal
Exten: 832323232323
Priority: 1
What do I need to write in “Channel”?
How to call “virtual extensions” using AMI?.
Action: Originate
Channel: SIP/422
Context:from-internal
Exten: 832323232323
Priority: 1
What do I need to write in “Channel”?
local/422@from-internal
local/422@from-internal - not work
what I see on the console after a call from a sip phone :
-- Executing [s@macro-dial:23] Dial("SIP/422-0000129b", "Local/FMPR-101@from-internal&Local/FMGL-2120169#@from-internal,22,HhTtrIM(auto-blkvm)b(func-apply-sipheaders^s^1),") in new stack
-- Local/FMPR-101@from-internal-000005b1;1 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("Local/FMPR-101@from-internal-000005b1;1", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@func-apply-sipheaders:2] NoOp("Local/FMPR-101@from-internal-000005b1;1", "Applying SIP Headers to channel Local/FMPR-101@from-internal-000005b1;1") in new stack
-- Executing [s@func-apply-sipheaders:3] Set("Local/FMPR-101@from-internal-000005b1;1", "TECH=Local") in new stack
-- Executing [s@func-apply-sipheaders:4] Set("Local/FMPR-101@from-internal-000005b1;1", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:5] While("Local/FMPR-101@from-internal-000005b1;1", "0") in new stack
-- Jumping to priority 13
-- Executing [s@func-apply-sipheaders:14] Return("Local/FMPR-101@from-internal-000005b1;1", "") in new stack
== Spawn extension (from-internal, 101, 1) exited non-zero on 'Local/FMPR-101@from-internal-000005b1;1'
-- Local/FMPR-101@from-internal-000005b1;1 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Local/FMGL-2120169#@from-internal-000005b2;1 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("Local/FMGL-2120169#@from-internal-000005b2;1", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack`
This is a partial log.
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII
Post a full call trace using pastebin.
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