[asterisk-users] asterisk 13 n-way call problem
Dmitry Melekhov
dm at belkam.com
Tue Dec 22 00:53:39 CST 2015
Hello!
I need to use n-way call as it described here:
http://habrahabr.ru/sandbox/52259/
It is in russian, but dial plan is quite clear.
It works in asterisk 11:
-- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer)
priority 1
-- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new
stack
-- Executing [0 at fromtransfer:1] NoOp("SIP/6052-00000ab6", "") in
new stack
-- Executing [0 at fromtransfer:2] Gosub("SIP/6052-00000ab6",
"dynamic-nway,6052,1") in new stack
-- Executing [0 at fromtransfer:2] Gosub("OOH323/7272-6385",
"dynamic-nway,6052,1") in new stack
-- Executing [6052 at dynamic-nway:1] NoOp("OOH323/7272-6385", "") in
new stack
-- Executing [6052 at dynamic-nway:1] NoOp("SIP/6052-00000ab6", "") in
new stack
-- Executing [6052 at dynamic-nway:2] Answer("OOH323/7272-6385", "")
in new stack
-- Executing [6052 at dynamic-nway:2] Answer("SIP/6052-00000ab6", "")
in new stack
-- Executing [6052 at dynamic-nway:3] Set("OOH323/7272-6385",
"CONFNO=6052") in new stack
-- Executing [6052 at dynamic-nway:3] Set("SIP/6052-00000ab6",
"CONFNO=6052") in new stack
-- Executing [6052 at dynamic-nway:4] Set("OOH323/7272-6385",
"MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
-- Executing [6052 at dynamic-nway:4] Set("SIP/6052-00000ab6",
"MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
-- Executing [6052 at dynamic-nway:5] Set("OOH323/7272-6385",
"DYNAMIC_FEATURES=") in new stack
-- Executing [6052 at dynamic-nway:5] Set("SIP/6052-00000ab6",
"DYNAMIC_FEATURES=") in new stack
-- Executing [6052 at dynamic-nway:6] MeetMe("SIP/6052-00000ab6",
"6052,1pdMXq") in new stack
-- Executing [6052 at dynamic-nway:6] MeetMe("OOH323/7272-6385",
"6052,1pdMXq") in new stack
-- Created MeetMe conference 1023 for conference '6052'
== Spawn extension (sipphones, 7272, 3) exited non-zero on
'SIP/6052-00000ab6<ZOMBIE>'
As you can see both channels are passed to macro defined in
|__GOTO_ON_BLINDXFR=fromtransfer and everything works as expected. But I
have problem I know that macros are deprecated, but, problem here is
that in asterisk 13 |||GOTO_ON_BLINDXFR| is executed only for one channel: |
-- Started music on hold, class 'default', on channel
'DAHDI/i1/6000-436'
-- <SIP/5082-00000046> Playing 'pbx-transfer.ulaw' (language 'ru')
-- Stopped music on hold on DAHDI/i1/6000-436
-- Channel DAHDI/i1/6000-436 left 'simple_bridge' basic-bridge
<f5100b94-4c34-40af-9c92-7e129c2bdb00>
-- Executing [0 at fromtransfer:1] NoOp("DAHDI/i1/6000-436", "") in
new stack
-- Executing [0 at fromtransfer:2] Gosub("DAHDI/i1/6000-436",
"dynamic-nway,5082,1") in new stack
-- Executing [5082 at dynamic-nway:1] NoOp("DAHDI/i1/6000-436", "") in
new stack
-- Executing [5082 at dynamic-nway:2] Answer("DAHDI/i1/6000-436", "")
in new stack
-- Executing [5082 at dynamic-nway:3] Set("DAHDI/i1/6000-436",
"CONFNO=5082") in new stack
-- Executing [5082 at dynamic-nway:4] Set("DAHDI/i1/6000-436",
"MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
-- Channel SIP/5082-00000046 left 'simple_bridge' basic-bridge
<f5100b94-4c34-40af-9c92-7e129c2bdb00>
-- Executing [5082 at dynamic-nway:5] Set("DAHDI/i1/6000-436",
"DYNAMIC_FEATURES=") in new stack
-- Executing [5082 at dynamic-nway:6] MeetMe("DAHDI/i1/6000-436",
"5082,1pdMXq") in new stack
== Spawn extension (sipphonesconf, 6000, 4) exited non-zero on
'SIP/5082-00000046'
Is this expected or, may be, this is bug?
So,as you can see, macro is not executed for Channel SIP/5082 , so this
channel is not connected to conference.
Could you tell me how can I get n-way call using asterisk 13?
Thank you!
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