To be sure you could setup a soft phone and see if the caller ID name comes in
correctly.
On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks
jordan.c...@gyron.net wrote:
Hi,
In my dialplan I have the following line.
same = n,Set(CALLERID(name)=Support)
I am expecting
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd R.
Sent: Monday, January 19, 2015 1:45 PM
To: Asterisk-Users List
Subject: Re: [asterisk-users] sip show channelstats reliable? Additional info:
At the moment I am running 1.8.x but the other day I was getting the same
results on 11.x
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 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
Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP
trunks and our customer PBXs.
I love and understand Asterisk but the company I am working for is looking for
a more Commercial type solution where we can go to a vendor for support etc.
I know, we can get Asterisk
variable.
Date: Mon, 27 Oct 2014 08:51:42 -0500
Subject: Re: [asterisk-users] Setting Music on Hold with the Manager Interface
From: mjor...@digium.com
To: tjrl...@live.com; asterisk-users@lists.digium.com
On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote:
Does anyone know how
Does anyone know how to set the music on hold class with the Manager Interface
in 1.8?
Here is what I am using but I end up just getting no music when I put this in
place, when I remove it the default is back.
The classes I am setting work elsewhere just fine.
I did not include the opening of
OK, been messing with Asterisk for a long time and I have my opinion on where
the issues lies but sometimes it's just nice to see what others think that can
relate :-)
Here goes..
Inbound calls flow like this:Tier 1 Provider (SIP) Asterisk 1.8 Name Brand
PBX - Calls work fine
Outbound calls
Been working with Asterisk for a long time but this is the first time I have
dealt with this issue.
I am setting up an Asterisk box (FreePBX not my choice) to interface with an
e911 provider.
They say their switches only listen for RTP on ports 2-21001 which is
outside the normal range
Search google for Asterisk Manager Interface.
For the most part, if you have raw Asterisk installed then that's what you get
and have to build what you want on top of it or hire a developer to do it.
Date: Fri, 10 Jan 2014 12:12:47 -0500
From: szilvertho...@gmail.com
To:
Luminvox is one.. There are others out there..
Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448
From: jpra...@gmail.com
Date: Fri, 10 Jan 2014 12:16:43 -0800
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Text to Speech Engine
Hello,
Anyone know
Actually, scratch that.. Luminvox is not text to speech it's speech recognition
software. Got this mixed up and turned around :-) Anyhow, see the link I posted
earlier, it's got some good info to get you started.
From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Fri, 10 Jan 2014
May not be what you are looking for exactly but search Google for Nerd Vittles
BeagleBone. I am not suggesting you use that exact solution but, reading the
article with give you all sorts if ideas about what you could use in your
situation.
The BeagleBone is a small form factor computer like
I do this by writing custom CDR. I write the agents extension write into the
CDR records. This makes is easy to just parse through the CDR and get all the
info you need about the call.
Google something like asterisk custom CDR
On Nov 29, 2013, at 11:36 AM, Leandro Dardini
Did you have the externalip setting in sip.conf set to the Elastic IP?
Date: Sat, 23 Nov 2013 23:42:36 -0500
From: ja...@fivecats.org
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby
system?
On 11/22/2013 12:52 PM, Todd R
Just checking one more time to see if anyone has an opinion on this. I am
primarily interested in using a cloud type setup such as Amazon AWS for the
redundancy, easy backup and recovery options. It's not about price but the idea
that it will be very hard for a single piece of hardware to ruin
thinktwice about Amazon -- and virtual in general is not a good
idea for this sort of thing. I have seen messages about bad results
with amazon specifically.
Todd R. tjrl...@live.com wrote:
Just checking one more time to see if anyone has an opinion on this. I am
primarily interested
.
Todd R. tjrl...@live.com wrote:
Just checking one more time to see if anyone has an opinion on this. I am
primarily interested in using a cloud type setup such as Amazon AWS for the
redundancy, easy backup and recovery options. It's not about price but the
idea that it will be very
Took me a while but I have finally embraced cloud computing and all the
benefits.
The only thing I have yet to feel comfortable about putting in the cloud is
real live Asterisk boxes to be used in production. I know it's being done
because as far as I know Twilio is using Amazon for their
What do you want to happen once the call is made?
You can choose to fire the call off using the originate command with the
Asterisk Manager Interface from a PHP page or some other similar language. No
need for Perl on the Asterisk box at all really unless you need it for
something else.
Just put a new phone in place with the latest firmware from Cisco. This is the
first SPA501G we have with this firmware.
In the Asterisk CLI we are now seeing the error message below about once every
second. When we unplug the phone, the messages quit.
NOTICE[15539]: chan_sip.c:9602
Trying to figure out the best way to pull an active call out of a queue by
unique id and put it on hold. I don't want to put it on hold on the agent's
phone but I want it to be pulled away from the agent's phone and into Asterisk
limbo somewhere.
Shortly after I want to pull the same call out
I currently have all agents/members logged in with local channels. When a call
is sent to one of the agents, then the agent transfers the call out the line
frees up on their phone but still shows in-use until the call that was
transferred is hung up.
How can I free up the agent/local channel
From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Thu, 1 Aug 2013 12:50:32 -0500
Subject: [asterisk-users] Local agent/member in-use after transfer
I currently have all agents/members logged in with local channels. When a call
is sent to one of the agents, then the agent
It was the darn restart, thanks!
I wasn't aware that musiconhold.conf was only read at start but now I
remember that Asterisk only reads certain files upon reload.
Thx
On Jul 31, 2009, at 2:03 PM, Danny Nicholas da...@debsinc.com wrote:
Are your permissions ok (files and directories)? Did
Can Asterisk act as a SIP conference bridge? Looking through the source I
notice that it's required to have a Zaptel interface installed. Why is this
a requirement? Can you not mesh the VoIP streams together?
Thanks,
T..S
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