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[asterisk-users] Asterisk 11.8.0 and 11.8.1


 
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geisj at pagestation.com
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PostPosted: Mon Mar 24, 2014 6:58 pm    Post subject: [asterisk-users] Asterisk 11.8.0 and 11.8.1 Reply with quote

I have used every asterisk 11.8.X version.Have not had an issue till 11.8.0 and 11.8.1
When I use ConfBridge I am attempting to put all 
participants in MUTE mode and just one talker...


However, since 11.8.0 I am hearing feedback in the 
announcement like the channel is not really muted.
I dropped back to 11.7.0 and I hear no feedback.


Has something changed - or - am I not correctly setting 
up the confbridge? How do I tell whats up?


Thanks


Jerry
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geisj at pagestation.com
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PostPosted: Tue Mar 25, 2014 10:14 am    Post subject: [asterisk-users] Asterisk 11.8.0 and 11.8.1 Reply with quote

OK back in the office so I can get some files....


in my confbridge.conf file
[MessageNetConfUserMuted]
type=user
quiet=yes
startmuted=yes
announce_only_user=no
announce_user_count_all=no
announce_join_leave=no


This is from the console so 410 and 411 are joining as MUTED.


^[[Kdevgeis*CLI> ^M^[[0K       > Channel SIP/411-0000001c was answered
^M^[[Kdevgeis*CLI> ^M^[[0K    -- Executing [smvoice_pa_app_confbridge_intercom@smvoice-transfers:1] ConfBridge("SIP/411-0000001c", "PA0014,MessageNetConfBridge,MessageNetConfUserMuted") in new s
^M^[[Kdevgeis*CLI> ^M^[[0K       > 0x7f343c0137c0 -- Probation passed - setting RTP source address to 192.168.1.38:8000
^M^[[Kdevgeis*CLI> ^M^[[0K       > Channel SIP/410-0000001d was answered
^M^[[Kdevgeis*CLI> ^M^[[0K    -- Executing [smvoice_pa_app_confbridge_intercom@smvoice-transfers:1] ConfBridge("SIP/410-0000001d", "PA0014,MessageNetConfBridge,MessageNetConfUserMuted") in new s
^M^[[Kdevgeis*CLI> ^M^[[0K       > 0x7f345c013c30 -- Probation passed - setting RTP source address to 192.168.1.39:8000
^M^[[Kdevgeis*CLI> ^M^[


yet  I hear feedback on the speaker.


I start the call by calling a Local channel, with variables of what file (wave) to play and what devices 
to bring in conference.


This all worked fine 11.0 -> 11.7. I have only encountered problems with 11.8+
dropping back to 11.7 works again.


So how do I found out if its something I have wrong or was this introduced in 11.8+




Thanks,


Jerry



On Mon, Mar 24, 2014 at 7:58 PM, Jerry Geis <geisj@pagestation.com (geisj@pagestation.com)> wrote:
Quote:
I have used every asterisk 11.8.X version.Have not had an issue till 11.8.0 and 11.8.1
When I use ConfBridge I am attempting to put all 
participants in MUTE mode and just one talker...


However, since 11.8.0 I am hearing feedback in the 
announcement like the channel is not really muted.
I dropped back to 11.7.0 and I hear no feedback.


Has something changed - or - am I not correctly setting 
up the confbridge? How do I tell whats up?


Thanks


Jerry

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rnewton at digium.com
Guest





PostPosted: Tue Mar 25, 2014 10:27 am    Post subject: [asterisk-users] Asterisk 11.8.0 and 11.8.1 Reply with quote

On Tue, Mar 25, 2014 at 10:13 AM, Jerry Geis <geisj@pagestation.com> wrote:
Quote:
OK back in the office so I can get some files....

in my confbridge.conf file
[MessageNetConfUserMuted]
type=user
quiet=yes
startmuted=yes
announce_only_user=no
announce_user_count_all=no
announce_join_leave=no

This is from the console so 410 and 411 are joining as MUTED.
<snip>
Quote:
This all worked fine 11.0 -> 11.7. I have only encountered problems with
11.8+
dropping back to 11.7 works again.

So how do I found out if its something I have wrong or was this introduced
in 11.8+

Hey Jerry, when something like this occurs suddenly between minor
release versions, you can always check issues.asterisk.org/jira to see
if it has been reported. A search for the words muted and confbridge,
then ordering the results by creation date will show this issue:

https://issues.asterisk.org/jira/browse/ASTERISK-23461

Which looks like the same issue that you are having. If you click on
the source tab you can see the commits it was fixed in. Looks like it
was after 11.8.1, so you'll have to wait until the next release, or
grab 11 from SVN.

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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