Hi all
If I have the following in my dialplan:
exten=25001,hint,SIP/25001
Doing a
core show hint 25001
results in
25001@local : SIP/25001
State:IdleWatchers 0
1 hint matching extension 25001
in the Asterisk CLI.
What does the
Watchers 0
The Watchers value show how many other phones you have configured with BLF
(Busy Lamp Field) “pointing” to that equipment.
It means how many other phones you have watching that equipment state.
In your case, if you don’t use BLF on your phones, that value will always be 0,
i suppose.
--
On Sat, Jan 17, 2015 at 12:04 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello Asterisk Users,
I have been looking for similar auto dtmf mode implementation on pjsip,
but didn't see it coming, so I decided to give it a try.
My basic plan was to do it as simple as possible with minimum
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn´t get trough
support tells me it was my asterisk server, but does not really work
me and my internal calls are working
I would recommend capturing traffic outside your Asterisk server with
Wireshark, then running the Telephony/Rtp/Analysize Streams option to
determine if you have packet loss at that point in the network.
On Mon, Jan 19, 2015 at 1:00 PM, Todd R. tjrl...@live.com wrote:
Thanks but no Adtran here.
Thanks but no Adtran here.
I do think these stats are indicating an issue, I just don't know how to prove
it outside Asterisk.
From: ewiel...@nyigc.com
To: tjrl...@live.com; asterisk-users@lists.digium.com
Date: Mon, 19 Jan 2015 13:55:33 -0500
Subject: RE: [asterisk-users] sip show channelstats
Additional info:
At the moment I am running 1.8.x but the other day I was getting the same
results on 11.x
Here is a sample from show channelstats. I do think this command is showing
that there is trouble between specific IP's and my Asterisk box but I don't
know if the numbers are accurate and
I've seen something similar with Adtran SIP gateways.When a re-invite
happens the Adtran gets all confused about call stats and marks the
pre-reinvite leg of the call as losing large numbers of packets.BTW, IIRC
reinvites happen when a codec changes or the channel switches to T.38.
I am seeing lots of lost packets when running the command sip show channelstats
at the CLI.
There are issues across multiple Asterisk servers I am trying to diagnose but
everything I read seems to point to this command being pretty unreliable.
Can I trust the info this command shows?
I am
Hi, when I make an outgoing call sends me a busy here, and no one is making call
Contact: sip:984783842@50.X.X.X:5060
Content-Length: 0
-- Executing [984783842@to_pstn:1] Dial(SIP/101-004e,
SIP/5001/84783842@,40,rRT) in new stack
== Using SIP VIDEO TOS bits 136
== Using
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