Hello ALL,
Anybody performed ASTERISK Testing for RFC 3261 Compliance?
If Yes,
Please share Result.
Best Regards,Sakharam Thorat. --
_
-- Bandwidth and Colocation Provided by
Hi,
We have a system with both ISDN trunks and SIP. We receive incoming calls on
both but always dial out via SIP.
When dialing out the caller id is set like this:
exten = _X.,1,Set(CALLERID(num)=${CC_ORIGNUM})
exten = _X.,n,Set(CALLERID(name)=${CC_ORIGNAME})
exten =
Hello list,
My system behaves in an odd manner, and I can't find why.
When users leave a message on the voicemail, once the message is
recorded and the user hangs up, Asterisk crashes.
I can't figure out when it started to behave like this.
Here is the extract of the dialplan where it occurs :
Hi Henrik.
You might want to read
http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header
and
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
On Mon, Oct 28, 2013 at 11:04 AM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
Hi,
We have a system
Hello
i check the dahdi-channels.conf
in span 1 when i use it like below i can do my outband calls without issue
; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel = 17-31
context = default
group = 63
but when i
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs are
detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk server from VM machine to dedicated machine
More
asterisk-users-boun...@lists.digium.com wrote on 10/28/2013 01:29:13 PM:
From: Eddie Mikell emik...@rimmkaufman.com
To: asterisk-users@lists.digium.com,
Date: 10/28/2013 01:29 PM
Subject: [asterisk-users] Tired of dropouts and garbled phone calls
- where to go next?
Sent by:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the
Asterisk - No analogue.
We have a very lightly loaded 60 Mbs cable link to the Internet that
tests pretty close to that most of the
On 10/28/2013 3:59 PM, Ron Wheeler said:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No
analogue.
I don't have any problems with IAX, but I hear some do.
We have a
Does using SIP to your ITSP make any difference?I stopped using IAX2 and
switched to SIP around 2003 when I experienced similar problems, never looked
back. If you insist on using IAX2, then Google for iax2 audio problems
-Original Message-
From:
Asterisk is a swiss army knife, you should either know how to use it or
rely on ready made software which control routing of calls through variable
bit rates (skype does that very effectively)
So the key here for you to research upon from those several hundred results
is variable bit rate codec
On Mon, 28 Oct 2013, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone service
that is being provided. The choppy phone
calls, and drop outs are detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk
Ron Wheeler писал 28.10.2013 21:59:
I have not found any
good tools to track down the causes of poor voice quality.
In my case,
I have good incoming quality and terrible quality going out.
That is,
I can hear people perfectly well but they complain that my voice drops
out and is garbled
On 29/10/2013, at 9:55 am, Mike mike...@microdel.org wrote:
On Mon, 28 Oct 2013, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone service
that is being provided. The choppy phone
calls, and drop outs are detrimental to our sales force.
I've
iperf is great. Another essential troubleshooting tool is nfsen/nfdump
(or any netflow/sflow monitoring utility that shows DSCP/TOS tag values).
On 10/28/2013 01:55 PM, Mike wrote:
As stated in previous replies if you haven't already I would certainly
try to isolate the problem, e.g., are
On 10/28/2013 07:29 PM, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs
are detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk
On Mon, 28 Oct 2013, Mike wrote:
I found iperf (http://iperf.sourceforge.net/) to be a free and easy
starting point, which actually turned out to be all I needed.
I've used iperf to check bandwidth before, but never looked deeper into
it's features. Thanks for the nudge. Maybe you can help
Steve Edwards wrote:
What? Why did my bandwidth dive from 800 Mbits/sec to 1 Mbits/sec?
--help shows:
Client specific:
-b, --bandwidth #[KM]for UDP, bandwidth to send at in bits/sec
(default 1 Mbit/sec, implies -u)
Doug
--
Ben Franklin quote:
Those who would
On 13-10-28 06:03 PM, Patrick Lists wrote:
On 10/28/2013 07:29 PM, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs
are detrimental to our sales force.
I've tried about
In my case, I have good incoming quality and terrible quality going out.
That is, I can hear people perfectly well but they complain that my
voice drops out and is garbled regardless of who places the call.
This suggests to me that you may have congestion problems in your
upstream traffic
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