Transfer Calls not working after FreePBX upgrade

[filippov]

Hello, transfer is now working, but no sound in listen/whisper mode. If cal to 555 (freepbx spy channel) + number extension - all good.

[nicolas]

asterisk -rx "core show channels concise"

When you are listening to a call, look for the chanspy session and the parameter delimiter, is it a comma or a pipe, if a pipe, in fop2.cfg set force_parameter_delimiter=','

[filippov]

All the same, there is no sound. Output for asterisk -rx "core show channels concise" with and without force_parameter_delimiter=','

DAHDI/pseudo-922601214!default!s!1!Rsrvd!(None)!!!!!3!47!!1418758260.206057
DAHDI/i2/889658064166-31d!from-internal!STARTMEETME!4!Up!MeetMe!201,oTMr,!89658064166!!!3!30!!1418758277.206075
DAHDI/pseudo-2103507193!default!s!1!Rsrvd!(None)!!!!!3!47!!1418758260.206059
DAHDI/i2/88162280360-31c!from-internal!STARTMEETME!4!Up!MeetMe!201,oTMr,!88162280360!1418758252.206045!1418758252.206045!3!56!!1418758252.206045
SIP/1002-00000317!from-internal-increase-vol!!1!Up!ChanSpy!CONFERENCE/201,dq!1002!!!3!12!!1418758295.206085

[nicolas]

You cannot spy on conferences? Only extensions. If you want to hear a conference, just join it.

Best regards,

[filippov]

No sounds in spy mode (listen / whisper) only in conferences. Button listen / whisper work well with other internal numbers. If i call 555 + number conference sounds is good, but it is inconvenient for the user.

[nicolas]

You cannot spy on a conference in fop2, you can join the conference and listen in (same effect).

[filippov]

About spy in conference understand.
I have error with blind transfer from user in queues to conference - after transfer incoming call to conference, sip line of user form queues is not released. The line of user remains busy until incoming call not complet.

[nicolas]

I am not sure if I understand correctly the "flow" of the problem. Not sure who are you transferring nor how into the conference?

A line being busy after a transfer is made, and remaining busy until the transferred call finishes, is a classic asterisk "feature" when using Local/xxxx@yyyy/n channels. It is not a FOP2 problem, is a byproduct of /n in a Local channel inside Asterisk, and as such, cannot be fixed within FOP2, nor it is a FOP2 bug.

However, I am not sure if your description of the problem has to do with that or not.

Best regards,

[filippov]

Incomming call (from dahdi) -> direct to SIP extension -> blind transfer to conference = all ok - external call in conference and SIP extension line released.

Incomming call (from dahdi) -> queue -> SIP extension -> blind transfer to conference = problem - external call in conference but SIP extension not released line.

Screen in attach.

[nicolas]

Your description fits almost exactly on the case I described above. How do your queue members look like? Are you using Local/xxx@yyy/n devices?

asterisk -rx "queue show"

Best regards,