Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
When somebody calls me on fxo4 port * sents that call to SIP 214 phone.
The problem is that when call ends and SIP user hangs up, the line stays
up. Now I don't use Congestion any more. Can sombody tell me do I
Tzafrir Cohen wrote:
On Thu, Dec 29, 2005 at 12:51:47PM +0100, Olle E Johansson wrote:
I usually do
asterisk -rvn | tee /tmp/sipdebug.txt
Then turn on sip debug on the cli. This captures everything.
You need to make sure that the debug output is sent to the console in
We are using perl for agi
I will try this command
Thank You
Rehan
What are you using for AGI
The correct command to send
Would be:
EXEC Set(${CALLERID(num)}=0005551212)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rehan
Douglas Garstang ha scritto:
The word from Kevin Fleming and Digium is that the use of realtime to
support multiple Asterisk boxes sharing sip is not supported or even
known to work at this point.
What about IAX ? If I connect two asterisk servers to a common mysql
backend (only iaxusers,
Hello all,
I have a curious issue, and I was hoping maybe somebody has an idea...
I have a Siemens DECT ISDN base connected to a HFC-PCI card in NT mode.
When I use it (or one of the connected DECT phones) pending outbound calls
are disconnected after *exactly* 30 seconds (if the call is
On 00:26, Fri 30 Dec 05, Robert La Ferla wrote:
William M. Sandiford wrote:
I just upgraded my system to the latest svn-trunk
I previously made extensive use of the SetAccount() function, but now
I'm getting the following error
Dec 29 20:54:08 WARNING[4925]: pbx.c:1679
Hi to All,Is anyone here has a settings on VM and asterisk for interconnection via SIP.ThanksLito
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Hello friends,
I wanted to ask if we can dial agents like the way we dial extensions. I
wanted to try this because the users can login and others can dial them. If a
person has not logged in, he isnt avalaible. I dont want to put people in a
queue. Has anyone tried this before? I was trying
There are options for queues.conf to not allow callers to join a queue
if no members are logged in, also you can 'call an agent' with the agent
channel, (IE agent/100)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday,
Hi,
I've just received a brand new td2400e , Where i can found some
documentation for this card?, Digium's site do not show very usefull.
I'd like to know how to configure zaptel.conf and zapata.conf basically.
Thanks, and Happy New Year to all.
--
Manuel Casal
[EMAIL PROTECTED]
[EMAIL
On 12/30/05, Manuel Casal [EMAIL PROTECTED] wrote:
Hi,
I've just received a brand new td2400e , Where i can found some
documentation for this card?, Digium's site do not show very usefull.
I'd like to know how to configure zaptel.conf and zapata.conf basically.
Thanks, and Happy New Year
Hello,
Do as with a TDM400P, but use the correct driver (modprobe wctdm24xxp).
You have only more channels, it's all !
Insert the quad modules starting from number 1 printed place on the PCB.
This card run well and echocancel is very good.
Good luck !
Francois BERGERET,
[EMAIL PROTECTED],
Anyone have any info on porting numbers away from a VoIP provider to a Ma
Bell or the like? Thanks!!
I had a friend port his from Bell -VOIP -VOIP. He had no trouble.
I would use a couple providers. So this way if one goes down there is a
backup.
In very general terms (at least in
BJ Weschke escribió:
On 12/30/05, Manuel Casal [EMAIL PROTECTED] wrote:
Hi,
I've just received a brand new td2400e , Where i can found some
documentation for this card?, Digium's site do not show very usefull.
I'd like to know how to configure zaptel.conf and zapata.conf basically.
On Fri, 30 Dec 2005 07:23:30 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
In very general terms (at least in the US), telephone numbers that are
considered portable can be moved from one itsp to another. However,
the move process generally involves a request for that move on the
part of the
Hi,
I am using the Queue application for 5 queues I have in my Call Center,
and will by the end of January, implement it for the rest of the company
(another 10 queues).
One of the main problems I face and my call center managers are worried
about is the fact that when an agent uses the DND
On 11:38, Fri 30 Dec 05, Dov Bigio wrote:
Hi,
I am using the Queue application for 5 queues I have in my Call Center, and
will by the end of January, implement it for the rest of the company (another
10 queues).
One of the main problems I face and my call center managers are worried
RTFM
ns2*CLI show application SetAccount
ns2*CLI
-= Info about application 'SetAccount' =-
[Synopsis]
Set the CDR Account Code
[Description]
SetAccount([account]): This application will set the channel account code for
billing purposes.
SetAccount has been deprecated in favor of the
I am installing an Asterisk box equipped with the Sangoma A102 card. The
telco
just tested the PRI interface and it is ll ok. I
now connect my Asterisk box and I can't get the D-Channel up. If I enable
intense pri debug I see messages like the following:
--SNIP START--
[ 02 01 7f ]
You can check status of Peer with Asterisk Management Interface (AMI)
www.voip-info.org/wiki-Asterisk+manager+API
Cheers,Giovanni Miano
2005/12/30, Dov Bigio [EMAIL PROTECTED]:
Hi,
I am using the Queue application for 5 queues I have in my Call Center,
and will by the end of January,
On 12/30/05, Manuel Casal [EMAIL PROTECTED] wrote:
I have a tdm2403e with 3 fxo modules plus echo cancelation. But in the
future u must add a fsx module so id like to learn to configure both...
Ok. So in /etc/zaptel.conf you add:
fxsks=13-24
And in /etc/asterisk/zapata.conf you add:
Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-
exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup
where agent 12 is configured as :-
agent = 12,12, vivek
After the agent is logged in on extension no12 as follows
Callback
Hello group members,
This is my first mail to this list. I am having one problem. When I dial a
number from zap channel, there's 5-6 seconds delay. Is there any way to
reduce/remove this delay?
Thanks
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I have just setup asterisk on a debian sarge box. I am
running Asterisk1.21 with AMP and chan_capi_cm 0.6.1 using a BT
Speedway (AVM Fritz)ISDN card, connected to a BT ISDN2e line. Currently we
have 6 extensions(SIP) configured all using CounterPath(Xten) eyebeam
softphone.After many hours
Try to recompile/reinstall (make clean; make install) zaptel after the
wanpipe installation to have the new 'patched' zaptel modules installed on
your system.
David Yat Sin
Sangoma Technologies
(905) 474-1990 x119
(800) 388-2475 x119
Fax: (905) 474 9223
MSN: [EMAIL PROTECTED]
Email: [EMAIL
I'm not sure if I'm right about this. But I think with a regular phone
connection. You first dial the number and send the digits to the PBX and
than the PBX has to redial the digits on the real phone line. Hence the
delay. I think you get that with all PBXs when dialing an outside line.
Hello,
I'm rather new to Asterisk so I'm in the wrong group for this issue please
let me know.
I'm using TDM400p with 2 FXO's on channel 3-4, Fedora Core4, 2.6, udev, and
I have the card on it's own IRQ
when I drop a .call file into /var/spool/asterisk/outgoing I see the
following on the CLI...
Hey all
I have a Gentoo system with Asterisk 1.2 installed. Its
been working great, for some reason the Zaptel module for my Wildcard TDM
(wctdm) seems to go missing anytime the server reboots, causing me to have to
go to the Zaptel source directory and do a quick make install. This
is
Since the last hurricane (that left me without phone for around 3 weeks
or so), I did the call forwarding (remote call forwarding in fact).
Lucky I was running in the cable modem in a couple of days (power restored).
I was planning in having two DIDs in distinct providers (I've been using
But a peer whose Softphone is on DND mode is still
considered available, isn't it?
- Original Message -
From:
Giovanni
Miano
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, December 30, 2005 12:21
PM
Subject: Re:
I have been experiencing a similar problem.
I have not yet been able to figure out
what the exact problem is but I know that the errors are inconsitant.
Sometimes nothing for 2 days and sometimes
5 times a day.
I thought about it a lot and I have found
only one thing in common.
The
Since the last hurricane (that left me without phone for around 3 weeks
or so), I did the call forwarding (remote call forwarding in fact).
Lucky I was running in the cable modem in a couple of days (power restored).
I was planning in having two DIDs in distinct providers (I've been using
Hello Ryan. Check out the file /etc/modules.conf, /etc/modules.d/zaptel
... if for some reason you have empty the modules.conf,
modules-update force will fix it, tough. In order to provide you with
further help, please provide more clues.
Best Regards
PD.
1. did you compiled the kernel
I restarted as you say.
PRI Debug bellow
[Kasterisk1*CLI
-- Executing
[1;36;40mMacro[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
[1;35;40mdialout-trunk|1|2514990|[0;37;40m) in new stack
[Kasterisk1*CLI
-- Executing
[1;36;40mGotoIf[0;37;40m([1;35;40mSIP/225-99e9[0;37;40m,
hu?On 12/30/05, Angelito Manansala [EMAIL PROTECTED] wrote:
Hi to All,Is anyone here has a settings on VM and asterisk for interconnection via SIP.ThanksLito
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I tend to be one of
those kind of guys that likes to eliminate all warnings. Although my
system is running just fine, I keep getting the following
message
Dec 30 10:39:51
WARNING[29172]: rtp.c:779 ast_rtp_make_compatible: Channel 'IAX2/11903-16385'
has no RTP, not doing anything
This
In addition to my earlier
message about an RTP warning, I'm also getting this one a lot. My system
is running just fine, I justkeep getting the following
warningmessage.
Dec 30
10:52:07 WARNING[16732]: app_addon_sql_mysql.c:316 aMYSQL_fetch:
ast_MYSQL_fetch: numFields=7
I really don't
Voicemaster is a commercial softswitch.
www.sysmaster.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: Friday, December 30, 2005
10:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Christian Stredicke writes:
Generally I think if people have a problem today they should move to
4.5. This version seems to be pretty stable, we did not get any
crash-complains or major problem reports from this version.
For those who want to move on (feature-wise), it is time to jump on
Hi,
I'm trying to get information on what the current status of TBCT support
in Asterisk is.
Thanks in advance,
Chris
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Has anyone used this feature?
I have been trying to find documentation on it but can't. I have tried the one
example shown but can't get it to work.
Anyone had success?
--
___
Play 100s of games for FREE! http://games.mail.com/
[EMAIL PROTECTED] wrote:
Hello group members,
This is my first mail to this list. I am having one problem. When I
dial a
number from zap channel, there's 5-6 seconds delay. Is there any way to
reduce/remove this delay?
First of all try to find where the delay stands.
Dial the number with
I am looking to be notified via email when a host fails it's qualify (is
unreachable). I found this patch
(http://bugs.digium.com/view.php?id=5372) but I wasn't sure if I could
get that from it.
Anyone else tried this?
-Jonathan
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Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and work with
asterisk+chan_capi?
I know they probably work fine with mutliHFC cards with either bristuff
of chan_misdn but since I have some spare Divas, I was curious
Posted this to -dev, but it may be more appropriate here as I haven't
released my patches for it...
I've run into a couple issues relating to RPID.
I have an Asterisk 1.2.1 installation doing SIP for SPA-2002 and PAP2-NA
ATA's. From the Asterisk box, we then do SIP to a VoIP provider who
This boils down pretty easily, how important is your phone service and how
reliable is your internet access? If you internet access is not 100% stable
and reliable then you will have problems using ITSP's. If your phone service
is critical to the operation of your business then you need to think
I'm searching around, but not finding definative info on this, is the
maximum number of extensions available in FOP limited to 100? if so, is
there another operator console (commercial or open source) that will allow
at least 200 max extensions? The docs for FOP seem to be quite breif on
changing
Its not that hard to modify the FOP settings but there is a limit to what
you can accomplish because of screen size vs readability. You can only make
buttons so small before the become unusable.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thanks, but I'm looking for information on porting numbers when the current
provider holding the numbers goes out of business and is unreachable. Can I
get the numbers? The business has had the same phone number for almost 30
years and definitely can't lose the number due to some provider's
Hello All,
I am looking for more thorough debug than
the one provided by the command "iax2 debug". Could anybody point me a good
documentation about this?
I have a issue with IAX connection.
Sometimes it stucked.If so, I have to restart my asterisk through CLI
command"restart now".
Just a curiosity really. Anyone know how I can do this?
exten = page,1,SetVar(_ALERT_INFO=ring-answer)
exten = page,2,Page(SIP/a00090101SIP/a00090301)
exten = page,3,Playback(tt-weasels)
ie Play back the sound file after the phones receiving the page have answered?
I know page is really
Reposting. I forgot to change the subject. Oops.
Just a curiosity really. Anyone know how I can do this?
exten = page,1,SetVar(_ALERT_INFO=ring-answer)
exten = page,2,Page(SIP/a00090101SIP/a00090301)
exten = page,3,Playback(tt-weasels)
ie Play back the sound file after the phones receiving the
Q.931 (8) len=13
Call Ref: len= 2 (reference 4/0x4) (Terminator)
Message type: DISCONNECT (69)
[08 02 80 81]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: User (0)
Ext: 1 Cause: Unallocated (unassigned) number (1),
class = Normal Event (0) ]
So I guess I'm unclear on who 'owns' the number? If my ITSP goes bust, and
I want to port my number to another phone service provider (ITSP or a Ma
Bell or something), does the porting process REQUIRE an acknowledgement from
the ITSP that is out of business? Or, because I was the one who ported
Hello,
I wish to thank people who have called me to test my
config .
I have to test an IVR menu recorded in french so if
you call press * .
Thanks again
Harry
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Nouveau : téléphonez
Has anyone sucessfully placed a call to a vonage user using one of the sip
peering networks.
I am trying to use sipbroker and use
exten = number,1,Dial(Sip/*472number@sipbroker.com)
i have even tried calling: number@sphone.vopr.vonage.net
I get the same return message from sipbroker as i do
Does anyone have a snippet of extensions.conf to share where
they call a number to open or close a queue? Thanks.
-- Bud
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Hello
I know there are 4 well known ENUM trees: e164.arpa , e164.org ,
e164.info and enum.org
Now... to which of these should I redirect my ENUM querys?
I read that e164.org is a free public ENUM root that works in a donation
based system and is free for the public at large to use.
Shouldnt
On Fri, 2005-12-30 at 18:26 +, Joao Pereira wrote:
Hello
I know there are 4 well known ENUM trees: e164.arpa , e164.org ,
e164.info and enum.org
Now... to which of these should I redirect my ENUM querys?
I read that e164.org is a free public ENUM root that works in a donation
based
I'm not sure it it's going to help for you, but try playing around
with the Local channels. and use that local channel as one of the
called devices in the page app.
On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
Reposting. I forgot to change the subject. Oops.
Just a curiosity really.
Thanks for the info everyone. I think I'll just keep my numbers at my telco
and forward it to Teliax or another ITSP. Sounds like that's the safest
thing to do.
Thx again!
-Ross
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port your number (there
are some potential issues that cna prevent that but genereally that is a
true statement).
An interesting wrinkle I'm running against is that you cannot port
Hello Steve,
It's not incoming, its outgoing when I am experiencing delay.I can give
you the snippet of log if you wish.
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I'm probably mistaken and unaware of a feature, but I thought the
concept of dialing an agent does not exist. An agent is not a channel,
but rather, someone who associates themself with a station from which
they service a queue.
You dial the queue with queue()
Message: 8
Date: Fri,
Ross C wrote:
Thanks, but I'm looking for information on porting numbers when the current
provider holding the numbers goes out of business and is unreachable. Can I
get the numbers? The business has had the same phone number for almost 30
years and definitely can't lose the number due to
Can anyone guide me enabling fax support in asterisk. I tried spandsp
patch but was unsuccessful. Because patch for chan_sip.c was not proper
for asterisk's version 1.2.1. Can anyone help me adding fax support in
asterisk 1.2.1.
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Thanks Matt.
Are there limitations with call forwarding? For example, with Teliax's pay
as you go plan you can have a whole bunch of simultaneous calls (we had 12
going the other day). So say we get 10 or 12 calls on our telco number that
forwards to Teliax, is there a limit to the number of
Depending on the forward type. You could put conditional or un-conditional
forwarding. As far as I know some telcos are placing restrictions on
conditional forwarding (and that depends on a case by case basis) but for
un-conditional forwarding I don't see why there could be a limitation.
Bogdan
There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).
However, I'm not sure which to use. A lot seem to be fairly
comprehensive... but until I kick the tires, it's trial-and-error. And
that would be a *lot* of trial-and-error.
So, here's what I'm
On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port your number (there
are some potential issues that cna prevent that but genereally that is a
true statement).
Matt Riddell wrote:
So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
the VoIP DID, change the forwarding to another number.
I thought my local telco told me that if I were to do that, I would have
to pay them LD charges for each call that came in to that number.
On Fri, 2005-12-30 at 13:33 -0600, Ross C wrote:
Thanks Matt.
Are there limitations with call forwarding? For example, with Teliax's pay
as you go plan you can have a whole bunch of simultaneous calls (we had 12
going the other day). So say we get 10 or 12 calls on our telco number that
On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote:
Depending on the forward type. You could put conditional or un-conditional
forwarding. As far as I know some telcos are placing restrictions on
conditional forwarding (and that depends on a case by case basis) but for
un-conditional
On Dec 30, 2005, at 1:48 PM, trixter aka Bret McDanel wrote:
On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port your number
(there
are some potential issues that
Of course most carriers these days charge extra per call path. So how
many simultaneous calls do you really need up?
On Dec 30, 2005, at 1:55 PM, trixter aka Bret McDanel wrote:
On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote:
Depending on the forward type. You could put conditional
Ken D'Ambrosio wrote:
There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).
However, I'm not sure which to use. A lot seem to be fairly
comprehensive... but until I kick the tires, it's trial-and-error.
And that would be a *lot* of trial-and-error.
trixter aka Bret McDanel wrote:
On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port your number (there are
some potential issues that cna prevent that but
This is a possible scenario indeed. But this scenario should be handled by
the switches of the telco...
bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
Sent: Friday, December 30, 2005 9:55 PM
To: Asterisk Users Mailing List
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Novack wrote:
trixter aka Bret McDanel wrote:
On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port
Logs are always helpful. Are you using AMP?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, December 30, 2005 2:10 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Problem on ZAP channel
Hello Steve,
It's not incoming, its
This is more of a curiosity and a thought than serious issue. But, I
wonder if I can get my Asterisk server to authenticate to my provider by
throwing the authentication requests to the SIP analog-adapter they
shipped me? (And I can't get in and see the authentication credentials
in the
There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).
However, I'm not sure which to use
Other then writing your own the best one I have found so far is AMP
AMP is great if the way it does things is the way you want to do it. And
for
On Fri, Dec 30, 2005 at 08:22:27PM +, Ron Wellsted wrote:
Within the UK, Number Portability between providers of the same type of
service is a legal requirement. Since we charge differently for calls
on landlines and mobiles, you cannot port mobile numbers to landlines or
landlines to
I've been searching for clever ways to add a wireless phone to our asterisk
install, I could setup ATAs on each station, but I'm wondering if something
like the SE-B2K (as seen at http://www.skype-phone.net/) can be configured
to work w/asterisk something like SJPhone. Anyone ever played with any
Hi,
I am in the process of selecting an Opteron based server and am looking
for other's experience with various motherboards and the TE411P 4-port T1.
Right now I'm looking at a Supermicro H8DA8 based 2x dual-core Opteron
system.
http://www.antonline.com/custom_CSE822T-H8DAR-SATA-_59.htm
Hi:
i have compiled Asterisk 1.2.1 without any problems
,But when i've tried to load the zaptel modules by
making modprobe zaptel this message shown:
FATAL: Module zaptel not found.
Regards;
jonny
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Tired of spam? Yahoo! Mail has
Help! Ive searched through the archives and Im
spinning my wheels. Im trying to get a new PRI working with
Asterisk 1.2.1. Im getting this kind of notice from the console
whenever I dial out:
-- Executing
Dial(SIP/Mikey-3b78, Zap/g2/5551212) in new stack
Dec 30 13:09:03 NOTICE[5657]:
Please don't post your question on different mailinglist seperately.
I already answered that one on the isdn4linux list.
Armin
On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port
Can you tell me how agent 12 is logging in, Zap, Iax, SIP???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 30, 2005 9:35 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Can we dial
Is it possible to record calls for specific ACD Agents?
From looking at queues.conf and agents.conf, it appears that all calls for a
specific queue can be record, or all calls for all agents can be recorded.
I'd like to be able to specify that calls for a _specific_ agent are recorded.
Case in
mm and sure you have compiled the zaptel packages and make install ?On 12/30/05, jonny hashem [EMAIL PROTECTED]
wrote:Hi:i have compiled Asterisk 1.2.1 without any problems,But when i've tried to load the zaptel modules by
making modprobe zaptel this message shown:FATAL: Module zaptel not
Okay, since isdn4linux.de seems to block mails from schlund.de...
On Fri, 30 Dec 2005, Armin Schindler wrote:
On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and
You can do something like this:
exten = pagenplay,1,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
[page]
exten = _X.,1,Set(TIMEOUT(absolute)=180) ; Three Minutes
exten = _X.,2,SetVar(_ALERT_INFO=ring-answer)
exten = _X.,3,Dial(SIP/${EXTEN}||A(tt-weasels))
-Original
So use call forwarding from the Telco, forward it to a VoIP DID, if you lose
the VoIP DID, change the forwarding to another number.
I thought my local telco told me that if I were to do that, I would have
to pay them LD charges for each call that came in to that number.
Or am I
Hi all,
Our asterisk servers do voicemail for our customers.
We never store them on the server but mail them to the
customer. Now every .wav file is sent as MSG0.WAV
Is it possible to give the wav file a more meaningfull name
like 20051230224900.wav (year month day hour minute
second)??
Hello,
Anyone know of a program that can analyse the RTP media stream and then output a
human readable graph or other file? I'd like to be able to show jitter,
difference, and if possible, echoes and other articfacts within a file of some
sort. Ethereal can show you a graph, but cannot save it as
Title: Message
If
yes, search if the modules are not in an any incorrect kernel branch if you have
several :
/lib/modules/2.6.12-1-686/zaptel/zaptel.ko
May be
it is in another branch as :
/lib/modules/2.6.12-1-386/zaptel/zaptel.ko
If
yes, check your configuration (headers, kernel),
On Fri, 2005-12-30 at 15:41 -0600, Rich Adamson wrote:
So use call forwarding from the Telco, forward it to a VoIP DID, if you
lose
the VoIP DID, change the forwarding to another number.
I thought my local telco told me that if I were to do that, I would have
to pay them LD
On 12/30/05, S McGowan [EMAIL PROTECTED] wrote:
Hello,
Anyone know of a program that can analyse the RTP media stream and then
output a
human readable graph or other file? I'd like to be able to show jitter,
difference, and if possible, echoes and other articfacts within a file of some
On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to record calls for specific ACD Agents?
From looking at queues.conf and agents.conf, it appears that all calls for a
specific queue can be record, or all calls for all agents can be recorded.
I'd like to be able to
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