"timing test" does similar, it just doesn't do the automatic calculation.
Confbridge normally operates at a mixing interval of 20ms, which is 50 ticks
per second. That would be what you would want to test.
If you don't get 50 per second then that means ConfBridge will not provide a
steady
g the timerfd module.
Regards
Robert McGilvray
SS GlobeOp
Associate Director, IT Network Security
GlobeOp Financial Services | 1565 Front Street | Yorktown Hts NY 10598
t: +1 (914)-293-3584 | f: +1 (914)-293-3510
rmcgi...@globeop.com | www.ssctech.com<http://www.ssctech.com/> |
> Are you selectively loading modules? If so you need the new res_pjproject.so
> loaded.
Yes. That did it, thanks.
Bob
This email with all information contained herein or attached hereto may contain
confidential and/or privileged information intended for the addressee(s) only.
If you have
ad.so.0 (0x7fe0138ca000)
libc.so.6 => /lib64/libc.so.6 (0x7fe013509000)
libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x7fe0132f2000)
/lib64/ld-linux-x86-64.so.2 (0x7fe0169ef000)
Regards
Robert McGilvray
o: 914 293 3584
From: asterisk-users-boun...@lists.digium.com
Hello,
We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe
and chan_sip for conferences. I have been testing the new versions of Asterisk
with PJSIP and ConfBridge but have run into an issue which is preventing us
from moving forward. Everything works fine until a
MUST be marked as
sendonly or inactive in the answer.
"
Is this a bug or am I wrong in my interpretation of the dialog?
Thanks!
Robert McGilvray
o: 914 293 3584
From: Robert McGilvray
Sent: Thursday, March 17, 2016 12:55 PM
To: 'asterisk-users@lists.digium.com'
Subject: Hold/Resum
I do not use ConfBridge() in a large installation. I use MeetMe on
1.6.0.*
The timing is different for ConfBridge, as it does not require DAHDI.
If you have that good of an experience with 1.4, why change anything?
I like new things. ConfBridge eliminates the need for an external timing
I have an existing conference bridge running on Asterisk 1.4.2 using
MeetMe and it's been pretty much rock solid since it was installed. We
do around 460,000 minutes on it monthly and peak at about 150
simultaneous sip channels. I'm adding a second bridge for redundancy
purposes into another
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Kenner
Sent: Thursday, October 08, 2009 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MeetMe option question
We've started to use
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Sent: Friday, September 18, 2009 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR Records for MeetMe
Andy Rosen wrote:
...
Is there a built-in way of detecting fax tones, or a switch to T.38 on a
SIP channel? I need to periodically check some efax servers for
availability and figured the best way to ensure they are operational is
to check for tones. I've looked into Nvdetect but the company seems to
have gone out
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