Hi,
You also can see example script for create cluster.
https://github.com/netaskd/AFDINbeat
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Dmitry Burilov
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent:
Hi list,
I've always about 50 concurrent SIP callers listening to several MOH
streams (fed via mpg123) on Asterisk 11.4.0, 4x 2.2 GHz and the CPU
usage is always at about 250% causing stuttering of the streams and
delays when using Read() and Playback() (overall everything is just
really
I've been spending some time looking at some of the significant
changes Apple has made to Facetime in iOS 7. I'm far from an Apple
fanboy but some of them are pretty interesting:
- multiplexing everything over a single UDP port
- deflate compression with SIP
- various /slight/ protocol
Sometimes I see something similar on my systems when several (3 or more) MOH streams are fed by
mpg123. Suddenly some or all streams are stuttering, but the CPU load doesn't seem to go up
significantly.
Have you looked at what is happening with the receive queue of mpg123 (pgrep mpg123 and
Probably worth noting that sipgate will close (at least in the U.S.) on Oct.
31:
http://www.besttechie.com/2013/09/13/voip-provider-sipgate-will-close-oct-31/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Am 20.09.2013 15:17, schrieb jg:
Sometimes I see something similar on my systems when several (3 or more)
MOH streams are fed by mpg123. Suddenly some or all streams are
stuttering, but the CPU load doesn't seem to go up significantly.
My CPU load is permanent at 200-250%. I have 7 active
Hello,
I have set the direct media to be off, but still doesn't work. I am not sure
about NAT configuration!
SIP.conf, [general]
Hi Kristian,
On 09/20/2013 03:17 PM, Kristian Kielhofner wrote:
I've been spending some time looking at some of the significant
changes Apple has made to Facetime in iOS 7. I'm far from an Apple
fanboy but some of them are pretty interesting:
- multiplexing everything over a single UDP port
-
Hello,
If Asterisk version is 1.6 use nat=force_rport,comedia
On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I have set the direct media to be off, but still doesn't work. I am not
sure about NAT configuration!
SIP.conf, [general] section
Hello,
I have Asterisk 1.8.10.1Moving to nat=force_rport,comedia hasn't solved the
problem. Still having the same error!
I am not sure if this is related to the problem here, but I was trying to test
my voicemail and got this error No audio available).[Sep 20 14:05:41]
WARNING[11424]:
My CPU load is permanent at 200-250%. I have 7 active mpg123 streams.
I forgot something. Even if your 7 streams are mp3 streams they cannot consume the CPU power you
are seeing.
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Hello,
i think your logic is wrong please explain me what are you trying to do?
[internal]
exten = 7002,1,Answer()
exten = 7002,n,Playback(vm-nobodyavail)
exten = 7002,n,Hangup()
exten = 7001,1,Dial(SIP/7001,60)
exten = 7001,n,Hangup()
try this dial 7002 and you should listen vm-nobodyavail or
Hi list,
I know it's a bit OT, but for those who will be at the Astricon, we
are organizing a very informal meeting (maybe in front of a pint or
two) to talk about Asterisk for call-centers. No marketing or anything
- just a way to exchange ideas and meet f2f.
I created a facebook group to
Asmaa,
You're getting ahead of yourself. How do you expect audio to work if
your firewall/NAT settings aren't even configured correctly to
establish SIP sessions?
Go back and read the message that I sent yesterday. Fix the SIP
three-way handshake problem. That is step 1 and you'll know you
Hello,
Here is my extension context,
[internal]exten = 7001,1,Answer()exten = 7001,2,Dial(SIP/7001,60)exten =
7001,3,Playback(vm-nobodyavail)exten = 7001,4,VoiceMail(7001@main) ;forward to
voicemail mailboxexten = 7001,5,Hangup()
exten = 7002,1,Answer()exten = 7002,2,Dial(SIP/7002,60)exten =
My CPU load is permanent at 200-250%. I have 7 active mpg123 streams.
My understanding is that the mpg123 stream gets decoded only a single time regardless of the
number of listeners.
netstat -t -p shows me each of the 7 IP addresses and Recv-Q has some bytes listed (55000 -
113000) for each
Hello,
paste you extension context.
On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I have Asterisk 1.8.10.1
Moving to nat=force_rport,comedia hasn't solved the problem. Still having
the same error!
I am not sure if this is related to the problem here,
Hi Matthew,
Indeed I missed your previous message!After changing the externip, it worked
successfully... The sip session is established with the complete three-way
handshake, and the voice packet is exchanged with no problem!
Many thanks.
Date: Fri, 20 Sep 2013 10:01:52 -0500
From:
Asmaa Ahmed wrote:
Indeed I missed your previous message!
After changing the externip, it worked successfully... The sip
session is established with the complete three-way handshake, and
the voice packet is exchanged with no problem!
Many thanks.
Asmaa,
That's great news!! I guess
Am 20.09.2013 16:37, schrieb jg:
My CPU load is permanent at 200-250%. I have 7 active mpg123 streams.
I forgot something. Even if your 7 streams are mp3 streams they cannot
consume the CPU power you are seeing.
I thought so. So, I need to dig deeper... but how?
Some command to show the
Any takers?
astdb is based off of version 1 BerkeleyDB. Googling shows:
http://www.voip-info.org/wiki/view/Asterisk+database
It has a section on basic replication.
Doug
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- Original Message -
Is anyone aware of a way to replicate parts of the AstDB to another
Asterisk install?
For example, to export all CF entries on a FreePBX based system to
another system running FreePBX, I might do:
asterisk -rx 'database show' | grep CF
This gives me a list
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