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!
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20151222/d3653608/attachment-0001.html>
I spent some time reading docs and such change is not documented, so this is bug. I'll open issue... 22.12.2015 10:53, Dmitry Melekhov ?????:> 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! >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151222/b5242c2e/attachment.html>
On Tue, Dec 22, 2015 at 1:47 AM, Dmitry Melekhov <dm at belkam.com> wrote:> I spent some time reading docs and such change is not documented, so this > is bug. > I'll open issue... > >Not necessarily. Certain aspects of features was definitely changed in 13, and may require the use of a pre-dial handler now. Please provide the full context of the call in Asterisk 13, including where you set the __GOTO_ON_BLINDXFER variable. What you've included below does not show enough information.> 22.12.2015 10:53, Dmitry Melekhov ?????: > > 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! > > > > -- > _____________________________________________________________________ > -- 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 >-- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151222/7b318224/attachment.html>