On Sun, 2006-10-29 at 15:33 -0800, Tom Lynn wrote:
Without providing a link to the list, or citing your front-runners,
you can't really expect people to reply, can you?
http://www.voip-info.org/wiki-Asterisk+GUI
I am currently trying out VoiceOne.
On 10/27/06, Frédéric Blaise [EMAIL
I don't think it is a phone problem. I get a US ring tone on a PAP2,
SPA-942 and IDEFdisk softphone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, 30 October 2006 5:08 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi everyone, I'm working with Asterisk 1.4-beta3 and ARA for my
Univesity thesis, to enable jingle support into an administrative
framework for asterisk developed in our lab. It's possible to map
jabber's and gtalk's user from the ARA database, as I have already done
with sip and iax users? I need
Yes, same channel and same ESSID for all AP's.
Are you connecting each AP to the LAN? Or only one connected, and the others
as relay?
With WDS, you have to keep same channel and ESSID for a good roaming.
If connected to the lan, doing it worked really good for me, roaming was
working in the
this is really ugly workaround, because using r option in dial you
lose any other progress tones, including busy, congestion, and you will
always hear ring tone even in case of congestion...
PJ
Michiel van Baak wrote:
check your Dial call. You can add a r to the options. That
way it will
Does anybody use Intel S3000AHLX board with Digium TE110P E1 card? Have you
experienced any problems? I'm planning following configuration, so I would
appreciate any experience both positive and negative.
Best regards,
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.:
Sunday, October 29, 2006, 10:00:22 PM, Jose Limeres wrote:
I have a problem with the dialing tone in PAP2:
When making a call, I can hear the calling tone 5 times and then it
stops. The called party still hears the call but not the calling
party.
I've playing around with different
I've got an odd situation where callerid is only picked up every other call. Is there anything I can do so callerid works on all calls?I'm seeing the channel hangs up during a call == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1'Oct 30 10:20:04
Hi Dear
I want to use asterisk(1.2.7.1) as a router by caller
id.
I have only a DID number, I want to map this number to
some ip-phones , base on received Caller-id.
it is my database's view:
456 | DID | 14193016880 |2 | hangup |
|
455 | DID
Hi. Ever since I bought my Digium 400P with 1 FXS and 1FXO module,
once in a while I hear what sounds like a touchtone in my ear on a
phone hooked up to the FXS module. This was not heard by the other
side, and although it was annoying, it was not too much of a problem
till I was using the
Has far as I know, Asterisk doesn't support ex-girlfriend logic in
realtime extensions so far.
Regards,
Ricardo.
Pezhman Lali wrote:
Hi Dear
I want to use asterisk(1.2.7.1) as a router by caller
id.
I have only a DID number, I want to map this number to
some ip-phones , base on received
Hi All
If this is the incorrect place or people can suggest a
better forum to post or an Asterisk consulting service I would be most
grateful.
I am using an SBC with our * server. All end points register
via the SBC to the * server.
The sip_buddies table has an entry for user
Hi!
for those using vGSM cards, today we released version 0.18.0,
that fixes a lot of small things and implements a lot
of new features, also to improve performances!
In the weekend on a test customer the channel driver
made 67000+ dials without a glitch!
Users are encouraged to upgrade!
Please
Hi all,Suppose I have a simple conference configuration as below ---meetme.conf[general][rooms]conf = 0041435215311-and I have a dial plan like this
Ehsan Khosrowshahi wrote:
Hi all,
Suppose I have a simple conference configuration as below --
-
meetme.conf
[general]
[rooms]
conf = 0041435215311
-
and I have a dial plan like this --
Hi,
We are successfuly using TAPI with Asterisk in order to provide a
generic and fairly well supported interface from Windows desktops to
Asterisk - This allows caller-id popping and click-to-dial from TAPI
aware environments.
Is there an equivalent telephony interface available for Mac OS X,
Hi Dear Users,
I am new to Asterisk and had a query which is probably primitive. I
wanted to know whether I can use the Digium Hardware and receive and
establish connection to a host SIP Server which is totally a different
platform.
Let me explain -
Usually there is a E1-VoIP gateway
Hello, Asterisk.
I am currently using Asterisk (asterisk-1.2.13) and asterisk-addons-1.2.3_1
on FreeBSD 6.1-RELEASE-p10
So, after setup asterisk for realtime extension:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = asterisk
dbpass = asterisk
dbport = 3306
dbsock =
Hi all,
when calling from the PSTN with Match As You Go Dialing (lift the handset
before start to dial) over a zap channel Asterisk simply takes the first
extension digit and tries to match it. Because no valid one-digit-extension
exists in my dialplan, matching fails and Asterisk says that a
Yes, you just need to setup asterisk with Digium board on the same
server of your sipserver, and then you must establish a trunk between
your sip server and asterisk.
Then you must route calls using asterisk dialplan as well as your sip
server dialplan.
Be aware that if you have both on same
hi all, in console mode how i can display the logged users?
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in sip
sip show peers (or users)
in iax
iax2 show peers (or users)
if u want more datailled view of the 208 user
sip show peer 208
Pablo Allietti a écrit :
hi all, in console mode how i can display the logged users?
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Take a look at smartCID (found at www.generationd.com)
You can take actions such as block/limit call times/accept based caller
number. It will also fill in the missing CID name based on database lookup
(or 411 reverse lookup).
MD
-Original Message-
From: [EMAIL PROTECTED]
I used some of the ideas found here:http://www.voip-info.org/wiki/view/Asterisk+manager+ExamplesOn 10/30/06,
Steve Davies [EMAIL PROTECTED] wrote:
Hi,We are successfuly using TAPI with Asterisk in order to provide ageneric and fairly well supported interface from Windows desktops toAsterisk -
On 10/30/06, Tom Vile [EMAIL PROTECTED] wrote:
I used some of the ideas found here:
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
Okay, so I didn't think to search from the AstManager perspective :)
Thanks for the pointer.
Steve
___
Hi,
I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings?
Thanks--
Dear Michiel and Supporters!
Thank you for your reply!
Here is the complete code for my monitor.php
html
head
titleAsterisk Status/title
link href=common/classic.css rel=stylesheet type=text/css
/head
body
h2Asterisk Status: ? echo `hostname`; echo
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi people,
pls does anybody know what (T) and (D) letter means?
server3*CLI iax2 show peers
Name/UsernameHost Mask Port Status
SERVER1 xxx.xxx.xxx.xxx (D)
Zeeshan Zakaria wrote:
Hi,
I'd set the daylight saving option to yes on all the GXP-2000 phones,
but apparantly it doesn't move it an hour back on last sunday of
October. So
I don't specify it on the phone. My Asterisk server changes it's time
and all of the phones pick it up.
Doug
On Sun, Oct 29, 2006 at 03:09:42PM +, Conrad Wood wrote:
On 29 Oct 2006, at 11:02, Matthew Thompson wrote:
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think*
Zeeshan Zakaria wrote:
Hi,
I'd set the daylight saving option to yes on all the GXP-2000 phones,
but apparantly it doesn't move it an hour back on last sunday of
October. So now I am stuck will all the phones showing the wrong time.
Isn't there an option so that it'll automatically update
If you have automated the configuration process, all you have to do is:1) Set option P75 (Daylight savings time) to 0 or 1 accordingly in the configuration file.2) Regenerate the compiled configuration file(s).3) Make sure they are in the TFTP or Web server.4) Reboot the phones so they read the
John Novack wrote:
And with the change in the US next year, adding three weeks in the
Spring and one week in the Fall, where will ALL those smart DST
devices be??
I was thinking that this weekend as well. What a waste.
Doug
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DL == Doug Lytle [EMAIL PROTECTED] writes:
DL I don't specify it on the phone. My Asterisk server changes it's
DL time and all of the phones pick it up.
The phones get their time from Asterisk? Which protocol do they use
for that?
/Benny
___
We are hosting multiple companies with Asterisk.
For a high degree of control, each company has many contexts that are included
from a main context. I had wanted to use realtime, but realised very soon that
it didn't scale. For each context that you put a realtime switch statement in,
Alban ha scritto:
Yes, same channel and same ESSID for all AP's.
Are you connecting each AP to the LAN? Or only one connected, and the others
as relay?
With WDS, you have to keep same channel and ESSID for a good roaming.
If connected to the lan, doing it worked really good for me, roaming
Hi,
I think the (T) is for Trunk.
Regards
Fred
___
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina
Sent: lundi, 30. octobre 2006 15:34
To: Asterisk Users Mailing List - Non-Commercial
Hi.
Again one big mysterious problem I hope some good guy can help me solve.
I'm trying to connect some Siemens C450 SIP
IP Dect phones to asterisk (1.2.13)
(I have actually 3 handsets + 3 ip base).
After configuring them and rebooting,
all of them register properly on asterisk,
then, after
Zeeshan Zakaria wrote:
Hi,
I'd set the daylight saving option to yes on all the GXP-2000 phones,
but apparantly it doesn't move it an hour back on last sunday of
October. So now I am stuck will all the phones showing the wrong time.
Isn't there an option so that it'll automatically update
Zeeshan Zakaria wrote:
Hi,
I'd set the daylight saving option to yes on all the GXP-2000 phones,
but apparantly it doesn't move it an hour back on last sunday of
October. So now I am stuck will all the phones showing the wrong time.
Isn't there an option so that it'll automatically update
Hello, Moses!
At 09:20 PM 27/10/2006, you wrote:
what about
exten = h,n,System(mycommand /some/file /some/other/dir/)
Where mycommand is your custom shell script to sleep before moving the file.
That would work, but I'm trying to avoid kludges like that. Hence my
question about doing it
Hi people,
I would like to read your suggestions as to where the issue might be.
ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port.
TDM04B= 4 FXO signal fxls
There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
will not make mention of it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jim Rice
Sent: Wednesday, 25 October 2006 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Looking for Wireless Heaset for Polycom
501
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG If you include 10 contexts, and each one of those has a realtime
DG switch, than that's 10 times that Asterisk has to query the
DG database, for a single call.
Not that I would make extensions.conf realtime, but...
One trick to avoid includes
Hi,
My users are currently using an operator console interface like this:
see it at: http://www.whssf.org/interface.jpg
which came with a Praxon PDX we got about 5 years ago, which is now
unsupported,
it works very good and converts any analog phone plugged into the
system into a powerful
On 10/30/06, Dean Collins [EMAIL PROTECTED] wrote:
We've used the Plantronics CS50 wireless Headset with the HL10 Handset
Lifter. About $240.
The handset lifter leaves a lot to be desired with the 501.
It lifts the handset off the cradle, but doesn't completely hang it up
properly.
Benny Amorsen wrote:
all of the phones pick it up.
The phones get their time from Asterisk? Which protocol do they use
for that?
The Polycom's use NTP. And I point the NTP to the Asterisk server.
Doug
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Hi
what is the best billing solution for Asterisk ?
With WWW manager interface for user can see the real invoice...
Thanks bye
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After John Todd's talk at Astricon about the ISN project, I spent much
of the weekend playing around with it.
I have discovered that the default dialplans on my Sipura gear, as well
as my Grandstream phones, intercept the * key that is a required part
of ISN numbers and interpret it as a
Try www.asterisk2billing.org
Noc Phibee escribió:
Hi
what is the best billing solution for Asterisk ?
With WWW manager interface for user can see the real invoice...
Thanks bye
___
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Sorry to re-post, but as noone has answeredme ...
Maybe somebody will this time :o)
Thanks !
JM
On 10/29/06, Jean-Marc Salsa wrote:
Hi
I was trying to have realtime voicemail working with ODBC Driver.Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as
On Mon, 2006-10-30 at 18:31 +0100, Noc Phibee wrote:
Hi
what is the best billing solution for Asterisk ?
With WWW manager interface for user can see the real invoice...
I'm using a2billing and works like a charm for me :)
Regards,
Thanks bye
Asterisk B: Create an extension (just as you would if you want to connect a SIP client)Asterisk A: Have this guy register using the extension (just as you would using a SIP client like SJPhone) - You will probably have to use type=peer though.
Make sure you take care of NAT and stuff like that if
...but I'll need to give the users a good mean to see
what's going on,
who is busy,
easy transfer with click here and there,
easy conference,
easy queue handler,
easy way to see/use many lines at the same time
is there any best console they can use?
Have a look at FOP : http://www.asternic.org/
Is the Wildcard X100P still supported? I have one sitting around that I
bought 3+ years ago and never used it. I need the functionality now.
Before I run off and buy something new, I'm curious if this will just
work.
I also have an old TDM400P with 2 FXS modules that I bought at the same
Hi David,
It can be set on the Sipura / Linksys devices. Look under Admin, Advanced,
Regional, Call Progress Tones. There is a link floating around on Whirlpool
forums to a page and auto provision file containing the correct settings to
produce Australian tones. It also depends on whether
If Someone did that, How I connect extensions.conf with this type of Hybrid
system to work with asterisk inside this schema:
PSTN---PANASONIC KX -- Asterisk
|
|-send internal call
Thanks.
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hican you please post some user or config guide.thanks in advancearunOn 10/24/06, Ed Greenberg
[EMAIL PROTECTED] wrote:I will sign in with good experiences with MP124 and Mediant 1000. I have an
MP202 under test.--On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas[EMAIL PROTECTED] wrote: I
Hi, before you start throwing shoes to me, i know there are a lot of
Asterisk Billing plataforms, but actually no one seems to accomplish
what i need. They are to complex (a2billing) or doesn't have too much
documentation (astbill and mcc) or are poorly developed (trabas).
What i was looking for
I am looking for phones that work well (or at all) when outside of the
network and behind a router, such as at someone's home or in a hotel.
My Polycom IP601s do not seem to be up to the task, so I am hoping that
there is a good alternative for my outside sales people to use to talk
to my
Hi Groupies,I am sort of new to the whole asterisk thing, especially when it comes to the Digium TE110P card. Does anyone have experience setting this up? If so can you help me out? The provider for the PRI is going to be ATT/SBC.ThanksJulian
___
Thanks all for your answer ;=) i start test this week a2billing
Noc Phibee a écrit :
Hi
what is the best billing solution for Asterisk ?
With WWW manager interface for user can see the real invoice...
Thanks bye
___
Hi
do you know if they have external Box (not internal card) for
connect Analog Line and Pri/Isdn to asterisk for incomming and
outgoing calls ...
Thanks
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To
Does anyone know of a really lightweight web interface that
allows users to log in and modify attributes of their extension only?
Thanks
Curt
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I was wondering if anyone has implemented a feature that would allow a user to record a phone call and once the call has ended, the call is forwarded to his voicemail?ThanksTom Vile
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Hi,
Is there a way to create trunk groups while asterisk is running.
For exemple let's say that zapata.conf defines g0 as channels 1-23
I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23
Any hints appreciated.
Andre Courchesne
Is the Wildcard X100P still supported? I have one sitting around that I
bought 3+ years ago and never used it. I need the functionality now.
Before I run off and buy something new, I'm curious if this will just
work.
It still works with the latest Zaptel (1.2.10)
I also have an old TDM400P
Use an analog extension port on the Panasonic to a FXO port on Asterisk.
On 10/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If Someone did that, How I connect extensions.conf with this type of Hybrid
system to work with asterisk inside this schema:
PSTN---PANASONIC KX -- Asterisk
I have looked on this list but may have missed it. I am having
problems patching the makefile in the asterisk channels source tree.
I keep getting:
--
patching file Makefile
Hunk #1 FAILED at 72.
Hunk #2 FAILED at 143.
Hunk #3 FAILED at 178.
On 29 Oct 2006, at 11:02, Matthew Thompson wrote:
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from adept
My advice is to first make some tests to see if a reload is enough for Asterisk to read any group definitions change in zapata.conf, otherwise no on-the-fly change will workAlyed
Return-Path: [EMAIL PROTECTED] Mon Oct 30 13:23:36 2006Received: from
Isn't your problem more about NAT traversal rather than the phones themselves?if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phonesof course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster.Alyed
Check out the SPA-3000 from Sipura (www.sipura.com).
On 10/30/06, Noc Phibee [EMAIL PROTECTED] wrote:
Hi
do you know if they have external Box (not internal card) for
connect Analog Line and Pri/Isdn to asterisk for incomming and
outgoing calls ...
Thanks
We have a number of clients who will be needing a server to host
Asterisk on. Many of these clients use analog (FXO) lines that will
need to be connected to Asterisk via Sangoma cards. Can anyone
recommend an industry-standard server (like IBM, Dell, HP, etc.) that
has enough open PCI slots
Dear all,
I've recently installed Asterisk and am trying to understand where exactly
Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy.
I am specifically interested in SIP. Could anyone perhaps point me out to a
diagram with SIP users and Asterisk to better understand
How many analog lines are you looking at? Hundreds?
PaulH
On Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote:
We have a number of clients who will be needing a server to host
Asterisk on. Many of these clients use analog (FXO) lines that will
need to be connected to Asterisk via Sangoma
Learn how a patchfile/Makefile works, and fix the patch. Actually the
Makefile patch never has applied cleanly in my experience, so always a
few fixes are needed.
On 10/30/06, Christian Jensen [EMAIL PROTECTED] wrote:
I have looked on this list but may have missed it. I am having
problems
Greetings,
If somebody knows how to concatenate several .gsm files in one or create a
macro and use with background() please reply.
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Sounds like some nice work - we are currently finishing off a 200 seat
call centrewhich has just been _hard_ work.
PaulH
On Tue, 2006-10-31 at 02:32 +0800, Craig Guy wrote:
Hi David,
It can be set on the Sipura / Linksys devices. Look under Admin, Advanced,
Regional, Call Progress
Dear all,
I've recently installed Asterisk and am trying to understand where
exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work
as a proxy? (or only as a register server?) I am specifically interested in SIP. Could anyone perhaps
point me out to a diagram with SIP users
exten= X,1,Background(nameoffile)
exten= X,2,Background(nameoffile1)
For a macro, you need to be passing arguments... what are you trying
to pass?
You can also use audacity to concat .gsm files but you have to import
it from raw data. You will then have to downsample it from 41000 to
8000
I published a pretty detailed how to about this on the voip-info.org wiki
a couple years ago ... A lot of things have changed since then and I suspect
that some of the methods I used may be obsolete at this point ... But I
think it might be a good place for you to start ...
With my setup it
Hi all, i have an * version: Asterisk
SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and
when I make sip debug command i see this debug information:
-- SIP read from x.x.x.x:1024:
REGISTER sip:mysipserver.com
SIP/2.0
Via: SIP/2.0/UDP
Something like this?
PaulH
On Mon, 2006-10-30 at 16:08 -0800, je . wrote:
Dear all,
I've recently installed Asterisk and am trying to understand where
exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk
work as a proxy? (or only as a register server?) I am specifically
Hello,
One problem is solved and another appears... :(
I cannot receive incoming calls on trixbox. I defined one incoming
route (any DID/any CID) and forwading these calls to a SIP extension.
With capi and sip debug in asterisk -r console i dont detect any
incoming activity...
In xlog console i
/20061030/5e764423/asterisk-0001.jpeg___
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
I'm working on sip trunk between cme 4.x and asterisk (trixbox 1.2.3).
Well, the trunk is partially working, asterisk' extensions talk with
cme, but
- - when cme try to connect to asterisk' number, receives the number
dialed is not in
/mailman/listinfo/asterisk-users
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Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
What is the best way to do this? Thanks in advance
Here's a
Jeronimo Romero wrote:
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
I do the following:
I already have P246 = 04,01,7,02,00;10,-1,7,02,00;60 value set. I do have TFTP server and the phones read configuration from there when bootup.
Also I have:
# Daylight Savings Time: 0 - No, 1 - YesP75 = 1
All the phones have the latest firmware. But still they don't do automatic daylight saving.
Before I got down the path of converting a Cisco 7960 I haveover to SIP I wanted to try and set it up using Skinny.
Thephone registersok withAsterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call.
When I
Well, options b is unacceptable since I may be supplied a caller id name,
which I would want to pass on to my users.
Options a is a bit of a kludge, but I think I could make it work. I'd have a
cron job that updates the cdr.src field based on the cdr.user field. That
could work but I was
Erick Perez wrote:
PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13
Asterisk is being used as a meetme server for 8 more calls.
Everything works fine in terms of the asterisk/meetme. The issue
arises when the calls comes in via the ATA286 box and in any part of
the
It's noisy while talking.
Any idea?
Thanks in advance.
Jason
Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates
(http://voice.yahoo.com)
___
Thank you for the link. Chapter 8 was most useful in explaining the different types of connections (user/peer/friend) as well as the register function such that users may know how to contact another user. However I'm looking for something more specific. For instance, for a normal session
On 2006-10-29 01:35:46 -0800, Alberto Pastore [EMAIL PROTECTED] said:
Martin Joseph wrote:
I think it's cleary true that wiring WIFI infrastructure is easier and
more reliable then WDS.
On the other hand, I have been running my little network with WDS for
over three weeks now, and it has
Well, I've never actually been able to make chan_skinny work with 79xx
phones.
I found the chan_sccp to work quite well:
http://chan-sccp.berlios.de/
plus this patch for a problem on MeetMe (I don't remeber where I found
it, but it works!):
diff -uNr chan_sccp-20060408.org/sccp_pbx.c
je . wrote:
Thank you for the link. Chapter 8 was most useful in explaining the
different types of connections (user/peer/friend) as well as the
register function such that users may know how to contact another
user. However I'm looking for something more specific. For instance,
for a
Hi,
I have a requirement to limit the calls to our agents via a queue to 5
minutes. I had posted this to a previous thread by name Maximum
talktime in a queue? One work around that was suggested was to use
the S(x) in the dial command to the agents, so that all calls to that
extension would be
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