[asterisk-bugs] [JIRA] (ASTERISK-23497) chan_sip attended transfer, when completed always uses a simple bridge and often the completed transfer does not have audio

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23497?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-23497:
------------------------------------

    Attachment: call2_tcpdump
                call2_messages.txt
                call2_full.txt
                call1_tcpdump
                call1_messages.txt
                call1_full.txt

Attaching full, messages and pcaps from two attended transfers.

Call1 is an example of an attended transfer that completes, but doesn't have audio in either direction.

Call2 is an example of an attended transfer that complete, and has working audio.

Call 2 was performed right after Call 1.

The calls were performed as follows:

1. A calls B, B answers
2. B presses transfer button, (which places A on hold and prompts for dial)
3. B dials C by pressing dial softkey, C answers
4. B presses transfer button again without making any selection, it completes, A and C are connected
In both scenarios, the final bridge for A and C is a simple bridge and not native.  Sometimes, we see no audio in either direction.


> chan_sip attended transfer, when completed always uses a simple bridge and often the completed transfer does not have audio
> ---------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23497
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23497
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_simple, Channels/chan_sip/Transfers
>    Affects Versions: SVN, 12.1.1
>            Reporter: Etienne Lessard
>            Assignee: Rusty Newton
>            Severity: Minor
>         Attachments: call1_full.txt, call1_messages.txt, call1_tcpdump, call2_full.txt, call2_messages.txt, call2_tcpdump, failed-native-bridge-attended-xfer.txt
>
>
> Given I have 3 SIP phones that are registered with asterisk via chan_sip
> Given directmedia is activated in sip.conf
> When A calls B and B answers
> Then A and B are bridged using a native bridge
> When B puts A on hold, then calls C and C answers
> Then B and C are bridged using a native bridge
> When B finalize the transfer
> Then A and C are bridged using a "simple bridge", i.e. they are not bridged using a native bridge, which is what is expected
> More generally, direct media is always working fine in a "direct call" scenario, i.e. if A calls B or B calls C or A calls C, direct media is working fine. But after an attended transfer, direct media doesn't seem to work.



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