Displaying 20 results from an estimated 400 matches similar to: "Click to call"
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
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 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
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...
2010 Nov 29
3
How to initiate a two-party call from within Asterisk
The desired result is that user A's phone rings; when he picks it up,
user B is dialled, and user A's channel is connected to that. (This is
to be a back-end for a web-based address book.)
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 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
2006 Mar 31
5
Dial from php
Hi all,
Here is the situation. Linux workstation access a web page on a web
server (not the one running Asterisk). From that web page, we need to
initiate a dial-out on the Asterisk server and once the call is
connected, it must ring on the agent's hard phone.
Any pointers about how to initiale an Asterisk call from a remove web
server?
Thanks,
Andre Courchesne
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
2009 Jan 22
1
oslec + dahdi
Hi list, I install dahdi-linux successfully with the module of oslec
for the echo, but when I specify it in the system.conf the echo
canceller oslec it shows me errors:
DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22)
I see that the echo cancellers is supported: mg2, kb1, sec2, and sec
because oslec is not supported?, but he has support to compile it with
dahdi_linux!
best
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
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 Mar 14
1
Group Listen on SIP Phone
Anyone know of a SIP phone that supports group listen?
Group listen allow you use the handset but what the far end says comes out
the speaker...it is F802 on a Norstar.
Thermal
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2013 Jun 04
2
problem to install asterisk on vps digitalocean
Hi list, I try to install asterisk on vps server , but fails when I want to
install dahdi
[root at shark dahdi-linux-2.6.3-rc1]# make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware'
make[1]: Leaving directory
`/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware'
You do not appear to have the sources for
2009 Apr 13
2
retransmision error con asterisk 1.4.24.1
se?ores alguien le ha presentado este problema al acceder al voicemail
o al hacer una llamada a la pstn
1940> Playing 'vm-received' (language 'es')
-- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es')
-- <SIP/111-08d91940> Playing 'digits/at' (language 'es')
-- <SIP/111-08d91940> Playing
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
2007 Jun 19
2
PhpAgi call generation
hi
i'd like to write a simply application in php with phpAgi that:
- connect to Asterisk
- call an external number using a Zap channel
- play a message
here is some code:
<?php
$asm = new AGI_AsteriskManager();
if ($asm->connect()) {
$asm->Originate("Zap/g1/1","number","default","1");
/*
play message...
*/
} else {
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
2013 Nov 14
1
DAHDI with (CDR(userfield)
Hi list, I need some help to improve my cdr, now in my company are
asking me how
to know which of my phone numbers are most used when receiving calls from
the PSTN and incoming the IVR
was thinking about using userfield field, and I'm trying to do, I have at
the moment 4 channel DAHDI
; DAHDI CHANNEL 3=23XXXXX6
context=in
callerid=asreceived
group=1
signalling=fxs_ks
channel => 3
2008 Dec 16
1
problems of DNS
Hi list, I have for a year I have an account to call with broadvoice from
about 3 days beginning a not registered problem of, asterisk shows to a
message of error with the DNS, and my dns this working fine
WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for
registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying
REGISTER again (after 20 seconds)
[Dec 16