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[asterisk-users] asterisk 13 n-way call problem

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dm at belkam.com
Guest

Posted: Tue Dec 22, 2015 1:54 am Post subject: [asterisk-users] asterisk 13 n-way call problem

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@fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack
    -- Executing [0@fromtransfer:1] NoOp("SIP/6052-00000ab6", "") in new stack
    -- Executing [0@fromtransfer:2] Gosub("SIP/6052-00000ab6", "dynamic-nway,6052,1") in new stack
    -- Executing [0@fromtransfer:2] Gosub("OOH323/7272-6385", "dynamic-nway,6052,1") in new stack
    -- Executing [6052@dynamic-nway:1] NoOp("OOH323/7272-6385", "") in new stack
    -- Executing [6052@dynamic-nway:1] NoOp("SIP/6052-00000ab6", "") in new stack
    -- Executing [6052@dynamic-nway:2] Answer("OOH323/7272-6385", "") in new stack
    -- Executing [6052@dynamic-nway:2] Answer("SIP/6052-00000ab6", "") in new stack
    -- Executing [6052@dynamic-nway:3] Set("OOH323/7272-6385", "CONFNO=6052") in new stack
    -- Executing [6052@dynamic-nway:3] Set("SIP/6052-00000ab6", "CONFNO=6052") in new stack
    -- Executing [6052@dynamic-nway:4] Set("OOH323/7272-6385", "MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
    -- Executing [6052@dynamic-nway:4] Set("SIP/6052-00000ab6", "MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
    -- Executing [6052@dynamic-nway:5] Set("OOH323/7272-6385", "DYNAMIC_FEATURES=") in new stack
    -- Executing [6052@dynamic-nway:5] Set("SIP/6052-00000ab6", "DYNAMIC_FEATURES=") in new stack
    -- Executing [6052@dynamic-nway:6] MeetMe("SIP/6052-00000ab6", "6052,1pdMXq") in new stack
    -- Executing [6052@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

Quote:

__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@fromtransfer:1] NoOp("DAHDI/i1/6000-436", "") in new stack
    -- Executing [0@fromtransfer:2] Gosub("DAHDI/i1/6000-436", "dynamic-nway,5082,1") in new stack
    -- Executing [5082@dynamic-nway:1] NoOp("DAHDI/i1/6000-436", "") in new stack
    -- Executing [5082@dynamic-nway:2] Answer("DAHDI/i1/6000-436", "") in new stack
    -- Executing [5082@dynamic-nway:3] Set("DAHDI/i1/6000-436", "CONFNO=5082") in new stack
    -- Executing [5082@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@dynamic-nway:5] Set("DAHDI/i1/6000-436", "DYNAMIC_FEATURES=") in new stack
    -- Executing [5082@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!

dm at belkam.com
Guest

Posted: Tue Dec 22, 2015 2:47 am Post subject: [asterisk-users] asterisk 13 n-way call problem

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 пишет:

Quote:

mjordan at digium.com
Guest

Posted: Tue Dec 22, 2015 5:40 pm Post subject: [asterisk-users] asterisk 13 n-way call problem

On Tue, Dec 22, 2015 at 1:47 AM, Dmitry Melekhov <dm@belkam.com (dm@belkam.com)> wrote:

Quote:

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.

Quote:

22.12.2015 10:53, Dmitry Melekhov пишет:

Quote:

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