Hi
I use ADSL and SDSL on a lot of multi channel VoIP connections SIP and H323
and no real problems if you size the link correctly - this normally means
limiting no of calls to match available bandwidth.
Check out the upstream and downstream data rates and size on the smaller -
normally the
Hello, David.
Thank you for reply. But my problem is certainly in the size of JitterBuffer
of chan_local. I realy need to know how to change the size of JB (reduce).
BTW:
1. The file /etc/asterisk/dsp.conf doesn't exist in my Asterisk 1.6.0.6
(something wrong?).
2. VAD is already disable for all
Hi,
I have read the docs, and now I want to attempt to setup Asterisk 1.6. I
am not going to complicate it with load balancing, etc. The setup is just 1 SIP
line - no other in-house connections. All inbound traffic. I intend to keep
this simple. Imagine that I sell pies in my
Hi Everyone,
How is this possible? How can I go about debugging this? I think that the
sound chopping and choking is also related to this. I have never seen
Asterisk show 43% of cpu usuage when there is only one call going. It
actually flactuates down to 11% and up to 43%.
Please guide me as to
Please take a few minutes, and fill us in on a few things: Which version of
Asterisk, what codec if any,
server hardware ( Make (HP/Dell/IBM) Model, CPU ( single or multicore, speed ),
and any other pertinent info you can think of. Just trying to help you get a
more informed answer.
Good
I did use it for my first asterisk installation, but I moved to Debian
due to things I disliked in arch as a server distro.
I did not and still do not use Dahdi at all
(http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge).
I compiled everything from source without PKGBUILDS, since they
Hi,
On Sun, Apr 25, 2010 at 5:06 AM, mike mosier trixbo...@gmail.com wrote:
Howdy all
1. does anyone know a good voip / sip / qos monitoring tool?
Wireshark is quite good at it
http://wiki.wireshark.org/VoIP_calls
However I could only find it good for debugging, not monitoring
(tcpdump the
http://www.villasantilles.com/home.php
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Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and
try to make a call I get the following error message:
-- Executing [6781...@default:1] Dial(IAX2/iaxy-7477,
DAHDI/g1/96781948)
On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and
try to make a call I get the following error message:
--
On Sun, Apr 25, 2010 at 8:43 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk
and
try to make a call I get the following error message:
Hi,
I've noticed that one of my new servers (new mobo) if drifting slowly backwards
in time (in aprox. 24 hours, system time drifts back 5 minutes).
I have an ntpd process which is supposed to sync with a lan time server but
it's not quite working. So I'm launching a manual ntpdate or
Hi,
Are SIP gain parameters available in Asterisk 1.4/1.6?
I'm wondering if I can increase transmission gain on SIP channels.
Vieri
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On Sat, 24 Apr 2010, Michael Wilson wrote:
I think DSL is 1/2 duplex and in most cases way to slow on the way Up for
VOIP.
Not sure what country you're in, but DSL in the UK is full duplex and more
than capable of handling VoIP, as I'm sure the millions of users who use
it would attest. My
On Sun, 25 Apr 2010, Vieri wrote:
Hi,
I've noticed that one of my new servers (new mobo) if drifting slowly
backwards in time (in aprox. 24 hours, system time drifts back 5
minutes).
I have an ntpd process which is supposed to sync with a lan time server
but it's not quite working. So
I don't expect my SIP provider to provide useful Remote-Party-ID information.
Therefore, I am using CONNECTEDLINE(num)=xxx AND CONNECTEDLINE(name)=yyy to
populate remote party information from a local database.
I am also using the I (upper case i) option for Dial.
Generally I like to see to see
On Sat, Apr 24, 2010 at 7:01 AM, David White david.wh...@watchguard.com wrote:
call-id doesn't match?
SIP/2.0 200 OK
...
Call-ID: 2117388659-506...@82.158.83.xxx
...
ACK sip:6615xx...@130.117.xxx.xxx SIP/2.0
...
Call-ID: 2117388659-506...@192.168.1.100
...
I'm not sure, but I think
Hi,
I record an alaw file by asterisk's record monitor command, and i use
linux's file command to check it information.
file command recognized the alaw file as DATA, is it correct?
Quyps
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On Sunday 25 April 2010 23:22:21 Pham Quy wrote:
I record an alaw file by asterisk's record monitor command, and i use
linux's file command to check it information.
file command recognized the alaw file as DATA, is it correct?
The file command works by recognizing certain header data in a
Hi Tony.
Maybe you have already resolv this. I suppose the new phone is not
registering so the peers table isn't updated.
You may check if, when phone is turned on sends an registration request.
Best Regards
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
2010/4/22 Tony
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