Hi Everyone.
I have a customer looking to deploy about 20 Grandstream GXP2140 phones.
Normally they would deploy Yealink brand phones but they want a phone with
gigabit pass through and the Yealinks with gigabit are too expensive for their
budget.
Does anyone on the list have experience
I have seen a similar problem occasionally. We will be doing maintenance on a
customer's server and they will have one or two ghost channels on their
machine hundreds of hours old but with no call associated with them. So far we
have just been rebooting their server or issuing a hangup command
) or softphones (Counterpath) embed a
module that metter MOS.
Regards
2015-03-25 14:21 GMT+01:00 Patrick Beaumont
p.beaum...@hatsoffsoftware.co.uk:
Hi everyone.
We regularly get customers complaining about call quality issues. Most
of
the time it turns out to be their own broadband. Very
Hi everyone.
We regularly get customers complaining about call quality issues. Most of
the time it turns out to be their own broadband. Very occasionally server
load. Does anyone have any advice or links to advice on measuring call
quality?
I’ve been playing around with “sip show channelstats”
I encountered a bug in some Polycom models where it would refuse to register to
a domain that started with a number (e.g 3something.voip.com). Could that be
applicable here?
Regards,
Patrick.
From: Jordan Cook - Gyron Networks
jordan.c...@gyron.netmailto:jordan.c...@gyron.net
Reply-To:
My suspicion would be that the line
o=Z 0 0 IN IP4 146.115.163.234
is causing the problem. Your SIP client is reporting it's external IP address
for the audio stream rather than it's internal one. I would look at the
settings in your sip client to see if it has any automatic NAT stuff (like
I believe the Unmonitored status is linked to the qualify setting for each
user. If they aren't set to qualify=yes then it won't check their status.
Regards,
Patrick.
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com
Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using “sip
info” for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.ukmailto:p.beaum...@hatsoffsoftware.co.uk
wrote:
Thanks Richard. This is exactly the answer I was looking for.
I'm now assuming that Asterisk 11 was using it's equivalent bridge_simple but
I was getting confused because the only bridge module I saw
Hi Everyone.
I was referred here by malcolmd of the Asterisk forums. What follows is a copy
of this question: http://forums.asterisk.org/viewtopic.php?f=1t=92007?
I've recently upgraded from Asterisk 11 to Asterisk 13.
Most of it went smoothly thanks to the documentation detailing how to
...@digium.com
Sent: 09 December 2014 20:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam
score:8%]
On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.ukmailto:p.beaum
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