[asterisk-users] asterisk 13 n-way call problem
Matthew Jordan
mjordan at digium.com
Tue Dec 22 16:39:37 CST 2015
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 пишет:
>
> 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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--
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
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