Displaying 20 results from an estimated 800 matches similar to: "Group Listen on SIP Phone"
2008 Apr 18
1
Polycom LDAP Corporate Directory
Anyone use the LDAP feature yet on the polycom phones? If so how well
does it work? How are you using it in your environment?
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati
ons/corporate_directory_access.html
Roy Anciso
Director of Technology
Manistee Intermediate School District
772 East Parkdale Avenue
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-398-3036
roy at
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instructions provided by Grandstream support but it
doesn't seems to be working...
2008 Mar 16
1
LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)
If you write a HowTo, would you please insert it into the wiki at
http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks.
On Sun, 2008-03-16 at 07:09 -0500,
asterisk-users-request at lists.digium.com wrote:
> Date: Sat, 15 Mar 2008 18:20:32 -0200
> From: "Gonzalo Servat" <gservat at gmail.com>
> Subject: Re: [asterisk-users] LDAP
> To: "Asterisk Users Mailing
2008 Apr 04
4
Advice on best operator phone (with attendant console)
One of our clients is using a Grandstream GXP2000 with an attendant
console. We have used the same phone with past clients successfully
however this particular operator processes around 200 calls a hours and
the GXP2000 for sure does not like the quick line shuffling and call
volume. We get the following problems randomly:
1. menu stops working
2. transfer key stops working
3. Line 1 LED gets
2008 May 28
3
Asterisk VoIP in Dubai/UAE?
Dear All,
We have a customer who is opening a new office in Dubai and we know
that VoIP is blocked over there.
Has anyone a solution to getting VoIP back out (we want interoffice
calls back to the UK)? We we're thinking of IAX trunking, but not sure
if that is blocked or just SIP etc.
A VPN works, but is not great. We have seen:
http://www.speed-voip.com/voiceguard.html
At the moment it
2008 Mar 14
3
Anyone know of a pass through ATA
Anyone know of a company that makes a pass through ATA?
By pass through I mean have an Ethernet switch built into the ATA, like most
desktop phones have.
All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN
ports.
I fooled around with DMZ etc...but it just doesn't work as well.
Thermal
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2008 Mar 08
2
Experiences with grandstream GXW 4024 FXS?
Dear all,
Just wanted to know if any one had deployed the Grandstream GXW 4024
yet. Wanted to hear any feedback and/or problems with this unit that
you may have experienced.
Thank you.
--
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
www.emergen.biz
2008 Apr 15
5
Conferencing..
I figured that asterisk can do conferencing if we have zap interface. The
BRI cards I have they do not have Zaptel. How do I enable conferencing on my
server?
Regards
Ajey
2008 Apr 04
2
Click to call
somebody knows some application web that allows me to call to my
internal extensions of my asterisk, example click to call.
I was proving the click to call of this example but it doesn't work
http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html
greeting
rickygm
2009 May 12
2
Is anyone keeping up with the versions?
We are still using 1.4 and were going to start testing with 1.6.0, but then
1.6.1 was released and now 1.6.2 is already in beta 2.
That seems like a lot of independent releases to maintain. I read about all
the regressions ans hurried dot releases, makes us nervous.
How is everyone doing their testing?
-Matt
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2008 Mar 17
1
ldap for sip users.
Hi,
I had asterisk 1.4.17 with the extensions which is
configured in the sip.conf it was working fine.
Now I am having the requirement to authenticate the
SIP users through the OpenLDAP not through the
sip.conf.
Steps I have done :
Did a check out by using the following command,
http://svn.digium.com/svn/asterisk/trunk. [^]
then given configure, make , make install. and taken
the sample ldap
2006 May 02
3
Queue reporting seems broken.
I am trying to figure out which one of our agents is answering the calls.
According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the
only time the queue_log puts the channel (agent) is during logoff & logon.
There is the connect & completeagent message, but it doesn't show which
channel (agent) answered the phone.
I can't even figure it our cross referencing the
2008 Apr 03
12
Web page to show online extensions?
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Thank you.
2008 Feb 02
2
Polycom - Buddy Watch not a choice when adding Speed Dial
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at "Auto Divert".....I know it
is the end of the list b/c the down arrow on the right side of the screen
disappears when I get to Auto Divert.
When I add <bw>1</bw> manually to the speed dial file it doesn't change
anything.
The buttons work well for
2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP trunk.
Something like this in sip.conf [general] would be in effect for all SIP
clients:
jbenable = yes
jbmaxsize = 150
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes
I only want
2008 Mar 17
2
php web chat + asterisk -> callcenter
Hello,
How can I make a live chat (mainly text, but with voice/video chat if
possible) interacting with asterisk?
Can asterisk control simultaneously the queue between people calling by
phone and people by web chat?
At my work, there is a call center using asterisk to control the queue of
the clients (by phone) already. This part is ok.
But now I need to make a chat room at the website
2008 Mar 06
14
FXS channel banks
Greetings list,
I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at.
If anyone's had experience using channel
2010 Apr 13
1
Interesting One Way Audio
I have an Asterisk box, 1.4.30 with a PRI.
A Mitel 3300 is connected to the Asterisk box via SIP trunking.
When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.
But - when I route the Mitel user to echo() it works, send and receive. The
Mitel user also can record and playback greetings.
One thing I have noticed is that when the Mitel user
2008 Apr 04
5
Ring back when free?
Has anyone here implemented "Ring back when free" in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2. A hears "The number you called is busy. To use ringback, press 5"
3. A presses 5, and hears "Your ringback request has been accepted".
4. A hangs up.
5. Later, B hangs up. The system then calls A (if A is now busy, it
2009 Feb 11
2
problem with 'which' and strings
I have written a function that goes through a database and calculates various
metric scores and allocates them to a data set. For 400 of the 500 sites
that I calculate these metrics for works fine and allocates the scores into
the appropriate column. For some reason, some sites I run into the below
problem:
Here is what I am doing:
> names(orig.metric)
[1] "BenInsect"