On 13-11-27 07:35 AM, Jonas Kellens wrote:
Server specs :
XEON E3-1220V2
4 GB RAM
2 x 500GB HD (RAID0)
1 U
HOT-PLUG PSU
Linux sip.server.tld 2.6.32-358.18.1.el6.x86_64 #1 SMP Wed Aug 28
17:19:38 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux
There is no transcoding. Calls are using G711a.
Maybe the
On 13-11-27 04:57 PM, Salaheddine Elharit wrote:
hello list
i have an IVR menu in asterisk 1.4
like below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_p
Go inside the time machine and come back to 2013!
Use a newer version asterisk and you will get help. There are a lot of changes
and a lot of bugs solved.
Best regards.
Emiliano
Enviado desde mi BlackBerry de Personal (http://www.personal.com.ar/)
-Original Message-
From: Salaheddi
On 13-11-27 06:48 AM, Jonas Kellens wrote:
On 27-11-13 12:26, Jonas Kellens wrote:
Hello,
Using asterisk 1.8.24 on CentOS 6.4
I notice that the asterisk process is using between 105 en 110 % CPU :
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
1765 root 20 0 2508
hello list
i have an IVR menu in asterisk 1.4
like below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}music1)
exten => s,n,Background(${sounds_path}m
On 27/11/13 14:12, James Bensley wrote:
What is the maximum delay RTP will tolerate one way (Does Asterisk
have a limit too)?
Can this be tuned (increased or decreased) within Asterisk (I'm
thinking of DSL customers where we may have this issue between our
PBXs and the customer)?
There isnt o
Hello--
Boy, it's been a long time since I posted to the user mailing list!
Pardon me, I've posted this to dev also, but I thought the general users
should also be aware of this.
I'd like to announce a proposal to the Asterisk Community, that I
introduced at Astridevcon last month. It is a new
Hi All,
I have some questions regarding RTP and Asterisk;
I am trialling a new SIP upstream provider. We connect to them over
the Internet at present which I know is not ideal, but we are just
testing at present. During the trials we have had an issue where we
have had one way audio between us an
No, there are no visible errors. We performed several simulation tests in
our lab to debug the problem. Every time there was jitter in the simulated
network the card was dropping audio samples and the voice was illegible.
I am wondering if anyone tested Digium in such conditions.
--
__
Server specs :
XEON E3-1220V2
4 GB RAM
2 x 500GB HD (RAID0)
1 U
HOT-PLUG PSU
Linux sip.server.tld 2.6.32-358.18.1.el6.x86_64 #1 SMP Wed Aug 28
17:19:38 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux
There is no transcoding. Calls are using G711a.
Maybe there is some trancoding when using voicemail
Are you transcoding?
What is your server spec?
Regards
Andrew Colin-mobile
Vsave(PTY)Ltd
Original message
From: Jonas Kellens
Date:27/11/2013 13:48 (GMT+02:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk uses 105
On 27-11-13 12:26, Jonas Kellens wrote:
Hello,
Using asterisk 1.8.24 on CentOS 6.4
I notice that the asterisk process is using between 105 en 110 % CPU :
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk
Hello,
Using asterisk 1.8.24 on CentOS 6.4
I notice that the asterisk process is using between 105 en 110 % CPU :
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk
2682 mysql 20 0 627m 29m 6204 S
Hi List,
We have a major issue while broadcasting DTMF using meetme application. We are
sending DTMF to asterisk using SIP INFO message with duration 160.
INFO sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx:5060
From: ;tag=43
To: ;tag=9753.0207
Call-ID: xxx@xxx
CSeq: 25634 INFO
Content-Length: 26
Cont
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