When I place an outbound call from asterisk 1.6.1.12 to a FXO port on
my Cisco 1760V 12.4, the channel changes - seemingly incrementing:
e.g., in the first call, below, the channel name is
"SIP/vgw1-00000075" -- the second call (on the same FXO port after a
soft hangup on the CLI) is "SIP/vgw1-00000077"
How can I extract this information in the dialplan so that I can use
the SoftHangup app in asterisk to disrupt an existing call ?
pbx1*CLI> soft hangup
SIP/vgw1-00000075 SIP/141-00000074
pbx1*CLI> soft hangup SIP/vgw1-00000075
Requested Hangup on channel 'SIP/vgw1-00000075'
-- Executing [h at extensions:1] Hangup("SIP/141-00000074",
"") in
new stack
== Spawn extension (extensions, h, 1) exited non-zero on
'SIP/141-00000074'
== Spawn extension (extensions, 09930267XXX0000, 1) exited
non-zero on 'SIP/141-00000074'
-- Executing [h at extensions:1] Hangup("SIP/141-00000074",
"") in
new stack
== Spawn extension (extensions, h, 1) exited non-zero on
'SIP/141-00000074'
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [09930267XXX0000 at extensions:1]
Dial("SIP/141-00000076", "SIP/9930267XXX0000 at vgw1") in
new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called 9930267XXX0000 at vgw1
-- SIP/vgw1-00000077 is making progress passing it to
SIP/141-00000076
-- SIP/vgw1-00000077 answered SIP/141-00000076
pbx1*CLI> soft hangup
SIP/vgw1-00000077 SIP/141-00000076
pbx1*CLI> soft hangup SIP/vgw1-00000077
Requested Hangup on channel 'SIP/vgw1-00000077'
-- Executing [h at extensions:1] Hangup("SIP/141-00000076",
"") in
new stack
== Spawn extension (extensions, h, 1) exited non-zero on
'SIP/141-00000076'
== Spawn extension (extensions, 09930267XXX0000, 1) exited
non-zero on 'SIP/141-00000076'
-- Executing [h at extensions:1] Hangup("SIP/141-00000076",
"") in
new stack
== Spawn extension (extensions, h, 1) exited non-zero on
'SIP/141-00000076'
--
Jeremy Kister
http://jeremy.kister.net./