Displaying 20 results from an estimated 1000 matches similar to: "Your thoughts.."
2003 Sep 08
1
SIP Status Codes
Can anyone give me a pointer to descriptions of the status codes my
Grandstream phone displays? I've looked on Google but can't find a
definitive listing of SIP codes.
--
JustThe.net Internet & Multimedia Services
22674 Motnocab Road * Apple Valley, CA 92307-1950
Steve Sobol, Proprietor
888.480.4NET (4638) * 248.724.4NET * sjsobol@JustThe.net
2003 Nov 02
3
PHP Manager examples
Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there.
Thanks,
Kevin
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2003 Sep 25
2
AGI: getting the return code from an exec()'d application?
So I hacked up the Dial app to return a numeric return code instead of
changing contexts based on a number being busy or unanswered. The purpose
for this modified dial app, which I call AGIDial, is to help me concoct a
"follow-me" type of application. The app returns -1 for a completed call,
0 for unanswered, or 1 for busy.
Well, I hooked the thing up to an AGI script that uses perl and
2003 Aug 05
1
So now I'm playing around with Queues....
and I found a reference to an AgentLogin.rtf. Looks great, except I can't
get it to work.
queues.conf:
[sjs-testq]
music = default
timeout = 1
retry = 1
maxlen = 0
member => Agent/10001
agents.conf:
agent => 10001,1234,Steve Sobol
extensions.conf:
(I have a phone line set up on which the main menu tells you
to press 1 to be added to queue. Pressing 1 lands you here)
exten =>
2003 Jul 31
1
PHP API for Manager - Plaintext auth needed?
Quick question: My PHP script is now able to connect to the manager port
and successfully authenticate using MD5. I would strongly prefer not to
do plaintext authentication at all. Would anyone object to plaintext
authentication being left out?
--
JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity]
22674 Motnocab Road * Apple Valley, CA 92307-1950
Steve
2003 Oct 14
3
My Grandstream works, but my X-Lite doesn't: no sound after 5sec
X-Lite build 1079 consistently chokes no matter which codec I use -
after five seconds I suddenly have no sound coming in and possibly no
sound going out too. Putting the line I'm on on hold and then switching
back to it gives me another five seconds of sound, then it dies, etc.
The Grandstream 101 I'm using is a piece of junk but I don't have the same
problem with it.
Not sure
2003 Aug 24
1
Any way to distinguish between...
a call on which caller ID is unavailable, and a call that's supposed
to be private?
As a side note, I have a phone on which I have caller ID blocked, but the
Asterisk server still ends up getting caller ID from that line anyway.
--
JustThe.net Internet & Multimedia Services
22674 Motnocab Road * Apple Valley, CA 92307-1950
Steve Sobol, Proprietor
888.480.4NET (4638) * 248.724.4NET *
2003 Jul 30
2
X-Lite and Call transfer using Asterisk
Hi,
Anyone succeed using call transfer function in X-Lite?
It is stated that this feature is available in the Lite version too, but for
me it doesn't work.
Clicking on Transfer button, then entering the number and then clicking
again on transfer doesn't work.
I miss something?
Thanks,
Dan
2003 Sep 12
3
7206 as SIP->PSTN Gateway?
All,
I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway.
Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know
which cards, if any, exist for a 7206VXR to act in a similar capacity,
either as a T1/PRI, DS3, or POTS FXO/FXS?
What other Cisco routers can act as SIP gateways today?
Thanks,
Dave
2003 Aug 31
5
Newbie IVR question
2003 Aug 05
5
(no subject)
Does anyone keep a known telemarketer caller id database? If not has anyone
proposed an Asterisk community project to share this information? Sort of a
nation wide blacklist so Asterisk'ers can cut down on the garbage calls...
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2003 Sep 12
3
say number question
I searched for "say number" in the * google archives and have not found reference to options for "say number". I would like to have * say digits instead of the hundreds and thousands. EG, "1234" would say one two three four.
2003 Nov 02
2
Clearing Queue Stats?
Is there a way to clear the Queue stats?
That is with out restarting *?
I'd like to reset them daily and don't see a way
to do that.
Unless the only way is maybe a cron restart asterisk
like every weekday @ 04:00?
2004 Sep 16
1
quality of musiconhold...
Hi everyone!
I was wondering... Does the musiconhold quality improve if the mpg123
processes run with a negative priority? If so, is there a way to make
them start like that, so I don't have to renice them?
Regards,
Evert
2003 Sep 08
2
live monitoring
Hello,
I've search through all of the lists and cannot find any descriptions of
live monitoring (monitoring a phone call going on between an extension and a
zaptel channel live from another extension while the monitoring phone is
muted). I am aware of the monitor function which is actually a call
recorder, but I'm looking for live monitoring from a muted extension. is
this easily
2006 Nov 22
2
How to park calls on a specific extension
Currently at our office, if I want someone else to pick up a call, I have
to transfer the call to them. So I'm looking into call parking, which is
ALMOST perfect.
The missing piece of the puzzle: I'm extension 203. I want any call I park
to get parked at extension 2203. I want a call my boss parks to park at
2205, since he's ext. 205. In other words, I want calls parked FROM
2003 Sep 05
4
app_queue input needed...
A friend and I have recently added the ability to announce the callers
position in the call queue every x seconds.. or even just inject an
anouncement every x seconds. All setup in queues.conf and can be setup
per queue.
My next project is to add the ability to announce the callers estimated
wait time. I want some feedback to see whats the best method to calculate
that? What do you want just
2003 Jul 21
4
Dynamically setting up/tearing down extensions
Hello, * newbie here,
I'm designing a setup that is to eventually be used in a production
virtual PBX/VoIP service.
Customers need to be able to change their setups over the web - I want
them to be able to do simple things like setting up call forwarding, as
well as more intricate stuff that will require me to re-generate their
dialplans.
Administration of the service is to be
2006 Nov 13
2
STUN with one public and one private IP?
I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I
thought Asterisk was cool by itself, but Trixbox has made just about
everything turnkey. Great stuff!
So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox,
which sits on our DMZ with a single public IP. I need the phones to work
from random places behind NAT, as well as in the office. I'm using
2003 Jul 30
3
Manager.pm port
For anyone that cares...
I am porting James Golovich's Manager.pm over to PHP. I plan on also
doing some documentation which will cover both the Perl and PHP APIs,
which will be almost identical (at least, to whatever extent is
practical).
Will let y'all know when I have some usable code to show you.
--
JustThe.net Internet & Multimedia Svcs. [The Fusion of Content &