Extension 200 is member of a queue. At night time, it is forwarded to a different number. Now when this extension is dialed directly, call forwarding works, but when a call comes into the queue, ext. 200 keeps on ringing and doesn't get forwarded. Why is that and how to fix it?
-- Zeeshan A
Hi,
Am Dienstag, 3. Oktober 2006 14:43 schrieb Greg Delgado:
I want to pop up a web page when a queue member phone
rings but, instead of displaying the clid, I want to
display the DID number the call came in. Any ideas how
to best implement this?
you can try FOP with such an entry in your
On 2006-10-02 13:55:15 -0700, Brian Candler [EMAIL PROTECTED] said:
On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote:
[invalid]
exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)
Are you sure that your invalid context is correctly written?
I've never heard about this
I've configured our PBX so that when a user dials 80 on the PBX
extension, it goes out an ISDN TE interface on the PBX and into an NT
interface on my asterisk machine, where it jumps into the 's' extension.
Asterisk then does a DISA(no-password|sip_provider_out) which allows the
call to go out
Hello,
I'm trying to get it to work but I can't find the right way. I would be glad
if the list could point me the right directions.
What I want: My Asterisk dialing out to a number (my mobile phone) and
playing an IVR to the called part saying press one to accept this call. If
the called
What do you mean by that it is forwarded. Is it set
on the phone or do you have it set in que memeber.
- Original Message -
From:
Zeeshan
Zakaria
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, October 04, 2006 8:56
AM
Subject:
Yes SDSL lines do hook up to the DSLAM. I do not know if the DSLAM itselfis a fear for you, but if it is, you can look at it in this way -All it does is aggregate signals for transmission through a switch and on to a high speed backbone. This, invariably, at some point or another, is the same
I pick up extension 200, dial *72 and forward it to another number. When a call comes in to the queue, it dials extension 200 along with the other extensions. I expect queue not to dial extension 200 but to dial the forwarded number which it doesn't do and keep ringing extension 200, and there is
Hi,I have postfix mail server. I get mail with tif attachments. So my question is , how to encode atachment, that I could to fax using spandsp, cause if I grab attachment from mail message to another file with name blabla.tif
, but this file is like a text file, not binary?Thanks
Shawn Kelley wrote:
Beware of the SPA-3000, we had a nightmare trying to get rid of echo issues
with it on the PSTN connection. We still haven't got it quite right even
after trying all kinds of settings and firmwares.
Mine, after about 5 days of use stops working until it is power-cycled.
Hello All!
Prompt, what it means?
*CLI -- Extension '' in context 'did-inbound' from '' does not
exist. Rejecting call on channel 1/1, span 1
This message gives out an asterisk at a call from internal number to
softphone.
Where it is necessary to adjust?
begin:vcard
fn:Eugeniy
Great,
I've been testing it for the last few days. Everything works fine
except the following:
Cisco7940 (SIP firmware) -- Asterisk -- C2600 -- POTS phone
The Cisco phone can hear the POTS phone, but the POTS phone cannot
hear the Cisco phone. If the call is set up the other way round:
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
How does one do this?
Just append anything you like to the URL, and it will end up in
the AGI variable agi_network_script.
e.g.
exten = _X.,1,AGI(agi://localhost:4573/begin/serv1)
When the FastAGI is called, you get the
Hello!
How can i change the verbose logging level to a file in 1.4?
In 1.2 i was used to set the verbose level via asterisk -Rx 'set
verbose 5' but in 1.4 it is always reset to OFF again, so (nearly)
nothing is logged to /var/lib/asterisk/verbose:
---test2 asterisk # asterisk -Rx 'core verbose
I don't think that that there's any way around this. At some point you
require human intervention.
Perhaps the only way to do it would be to set up some sort of timer.
After x seconds if you don't get a key press Asterisk moves the call to
it's own VM?
On Wed, 2006-10-04 at 07:00 -0200, Daniel
Hello Users...Can any one help on Asterisk with MySqLI don't want to use ODBC+MySqL. for RealTime...Just need the MySql and Asterisk integration..On That i need extension.conf ,sip.conf
,and voicemail.conf,meetme.conf,musiconhold.conf are in MySql Databases accesingIn Flaf files its
Check out the MySQL realtime module. It is in asterisk-addons. You can
read more about this at:
http://www.voip-info.org/wiki/view/Asterisk+RealTime
You will need to compile the add-ons yourself though (unless your
distribution includes a package for them).
--Brian
On Wed, Oct 04, 2006 at
How is the best way to add,clear mailboxes and change
passwords for voicemail. I am guessing you need to remove the conf entries for
the mailbox restart asterisk and then add them back in and restart asterisk. Is
there a better way?___
--Bandwidth and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Have a number of asterisk servers and want to get some good stats tracking
going (with Asterisk-Stat) -- but this requires cdr logging to mysql, apache
and the stats software running on each server.
Or does it? Of course, I can either
My
sip.conf file simply has:
[2111]username=2111type=friendsecret=2111qualify=noport=5060nat=never[EMAIL PROTECTED]
host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid="Joe
U." 2111
Scott HigginbothamSystems / Network Operations
Manager215.259.2185 or 1.800.835.5710
build libpri.
On 10/3/06, Eugeniy Khvastunov [EMAIL PROTECTED] wrote:
yusuf пишет:
Eugeniy Khvastunov wrote:
Hello!
Why Asterisk tell: Unknown signalling method 'pri_cpe'
Why the asterisk does not know such signaling method?
[chan_zap.so] = (Zapata Telephony)
Oct 3 13:04:02
Hi Paul -
It'd be great if I didn't have to enter the
digits and press the Park button again.
If you're interested in easier parking you might want to check out the patch at:
http://bugs.digium.com/bug_view_page.php?bug_id=7090
You can do one-touch parking with it.
When my users had to
On Tue, Oct 03, 2006 at 02:09:08PM +0300, Eugeniy Khvastunov wrote:
After installation libpri I need to reinstall asterisk?
More specifically: You need to build asterisk after libpri (of a
matching version) is installed.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
You don't need to restart Asterisk. Just do a reload app_voicemail.so
On Wed, 2006-10-04 at 06:45 -0500, Jordan Novak wrote:
How is the best way to add,clear mailboxes and change passwords for
voicemail. I am guessing you need to remove the conf entries for the
mailbox restart asterisk and
Hi, Now I'm testing faxes with spandsp. I have problems that spandsp do not add headers to fax page: LOCALHEADERINFO.Please help me.
___
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To UNSUBSCRIBE or update options
Hi all.
I'm confused. On http://www.voip-info.org/wiki-Asterisk+manager+API it
says that the PlayDTMF command is available since version 1.2.8. I
upgraded to version 1.2.12.1 but I cant find it if I type in show
manager commands there is no PlayDTMF command. According to resources
on the
You are just not loading the module. Connect to Asterisk terminal
# asterisk -vr
and load the module
CLI load app_senddtmf.so
Best Regards.
On 10/4/06, Jan du Toit [EMAIL PROTECTED] wrote:
Hi all.
I'm confused. On http://www.voip-info.org/wiki-Asterisk+manager+API it
says that the
Crazy Boy wrote:
Hi,
Sorry to post this in this forum.
I am new to Trixbox. When I am trying to install Trixbox, I am facing
this problem. First I have installed Trixbox ISO image file from a CD.
When its rebooting and Asterisk is installing, it is got stucked near
this below point:
This parking patch looks like a good idea. I applied the patch but it
doesn't seem to work. The patch install was successful and I modified my
features.conf like the features.conf.sample suggested. I don't see any
mention of the k or K in the 'show application dial'. Any ideas? Did I miss
a
I've used the prompt pls-wait-connect-call to give my users a cue to cool
their heels for a second or two in circumstances like this, and no one has
complained. That's probably the most useful prompt in Asterisk!
-Original Message-
From: James Harper [mailto:[EMAIL PROTECTED]
Sent:
Hi Kevin -
This parking patch looks like a good idea. I applied the patch but it
doesn't seem to work. The patch install was successful and I modified my
features.conf like the features.conf.sample suggested. I don't see any
mention of the k or K in the 'show application dial'. Any ideas?
Hi all,
first of all sorry for the question. I know there is
an asterisk-java mailinglist but i am not subscribed
to this list and i am sure there are asterisk-java
guru on this list who can help me.
I am trying to get the status of a peer using
SipShowPeerAction. Unfortunately the getStatus
Hi Zeeshan -
Is there any better and receptionist friendly IP phone, with just one
button parking option, and maybe somebetter option for paging as well.
You might play with the ParkAndAnnounce() application which parks a call
and then plays the resultant parking slot number to a channel of
I am using FOP .27 and I have Zap IAX trunks. Although the IAX trunks
do show and appear registered (not dimmed) on the display, they show no
activity while in use. Any ideas??
Segments of op_buttons.cfg iax.conf are included:
op_buttons.cfg
[Zap/1]
Position=23
Label=Cook (Main)%0a(905)
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 4:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Passing Arguments to FastAGI
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Look at using the 'M' flag for the dial command - if you set up a
macro that requires a keypress, and call that macro from the Dial
command, you can force asterisk to only bridge the two call legs if
you hit something on your phone
see here, and pay attention to the 'Dial macros' blurb.
Giedrius Augys wrote:
Hi,
Now I'm testing faxes with spandsp. I have problems that spandsp do
not add headers to fax page: LOCALHEADERINFO.
Please help me.
There is a bug in adding page header with spandsp-0.0.2pre26. I have
fixed this in the development code, but I haven't yet put the
Im running Asterisk/Zaptel on a Fedora Core 4
machine. The software runs ok with one exception. Zaptel appears to
load OK on bootup, but when you check it on login, zttool still shows red/nop alarms
on the T1 lines. I have to manually start it again for the alarms to
disappear and the T1
Hi,
I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.
I read http://www.voip-info.org/wiki/view/Intercepting+SIP+Calls and i
got a clue about
Had
the same problem in fc2. Solution was to chkconfig zaptel off chkconfig
asterisk off then in rc.local modprobewct1xxp (i think) then
ztcfgthen start safe_asterisk. Dunno why.
Hey,
is OnStar using Asterisk? Details, please.
-Original Message-From: Shea, Matt
I'm trying to get moh working correctly in Asterisk 1.4. A complete lack of
documentation isn't helping much.
I have this in sip.conf:
[3254101]
type=friend
...
mohsuggest=class1
[3254102]
type=friend
...
mohsuggest=class2
A call is bridged between the two extensions. When 3254102 puts
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Is ztcfg running at boot after the zaptel modules have been loaded?
What's the output of ztcfg?
Shea, Matt wrote:
I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The software
runs ok with one exception. Zaptel appears to load OK on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Do you need to actively intercept the call (i.e. participate in the
conversation) or just listen in the channel? For the latter you can just
use the ChanSpy application.
Delca wrote:
Hi,
I'm deploying an asterisk PBX for a Call Center and i was
On Tue, Oct 03, 2006 at 09:41:04PM -0600, Joseph wrote:
Since you are just planning it, keep in mind to select something that
will be IPv6 ready.
I don't know that this is necessary, actually.
If I understood the OP correctly, he's terminating line/trunk
appearances which arrive at his switch
Hi
list,
I'm
testingtransfer withsip re-inviteand
bristuff-0.0.8-RCnusing anHFC pci card connetced directly to telco;
this is what happen:
1.SIP phone
calls a mobile phone (or another residential phone)
2. The called party
answers the call
3. Now the sip phone
puts on hold the calland
hi,
what is mean by partially incompatible in
http://www.digium.com/en/docs/misc/compatibility_notes.php
i have server with E7221+te110p mobo and i think i dont have any problems
thanks
---
Marek Cervenka
===
On Wed, Oct 04, 2006 at 04:31:51PM +, Delca wrote:
I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.
The call center bix calls that Service
Hi,
My setup is the
following: Voip provider---(SIP DID)---Asterisk box(SIP
through a termination provider)---multiple cell
phones.
The cell phones each
have their extension (201,202,203,204) and I'd like to be able to have them
transfer a call to somebody
I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.
have a look at these :
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
Hi list,
today I have been teaching a class on * and have found that many students
find it quite hard to understand how setting up IAX peering between two
servers may work. So I prepared a little step by step tutorial hoping it
might be useful to someone in the future.
See it at
Check out meetme.
We create a meetme conference for each agent when the agent logs in. As
customer's call in, the call is matched (by DNIS and IVR) to the longest
idle agent with the required skill (or any agent if no agent with the
matching skill is available).
The supervisors can join any
Hi,
I'd like to make record button working on snom 320/360 + asterisk.
As I learned from wireshark output, the phone produces SIP info
message Record: on, while record button pressed.
Can anybody give me an advice, how to teach asterisk to understand
that SIP info message and start recording
If they are just trying to listen in you can use zapbarge
- Original Message -
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 04, 2006 12:57 PM
Subject: Re: [asterisk-users] Call Interception
On Wed, Oct 04, 2006 at 04:31:51PM
Hello,
I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.
Thanks!
Joe
___
--Bandwidth and
Moises, do you know if the DTMF event in bug 6082 made it into version 1.4?
When I last tried to compile that branch it needed the latest version
of make 3.81, which trunk did not, and caused me to wonder if it had
been committed to trunk.
The DTMF detection events in trunk did not also
To: Whom it may Concern:
Well, it hit me last night as I was falling asleep... Asterisk (in the
app Zapateller)
can emit the tri-tone (you know beep-Beep-BEEP... The number you have
dialed is no longer
in service. Please check the number and...blah, blah)
Well, it occurred to me that, for the
Mark Farver wrote:
Nathan Bell wrote:
extensions.conf:
[from-ptsn]
exten = s,1,Answer()
exten = s,2,Playback(vm-goodbye)
exten = s,3,Hangup()
You might try adding a wait(3) command after the answer. Some
analog lines do not pass audio immediately after being answered.
(Something to do
Hello,
I had recently install an Asterisk PBX into a brand new PC: Intel
Pentium D 3.4GHz Dual-Core + P5LD2 motherboard + SATA HDD.
I´m planning to handle one E1 with a TE110P interface and I want to know
the compatibility between TE110P and Intel 945P chipset. I already buy
the hardware and
How about preparing a step by step guide to DUNDi? Good luck with that though
because base DUNDi docs are rarer than periodic element #114 in the known
universe.
Doug.
-Original Message-
From: lenz [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 11:11 AM
To:
Hello,
I need an USA DID + 15 b-channels. The only option I already have is
OpenVox and I want to see some alternatives. Sound quality is my
priority. Thanks in advance.
R.R Libera
___
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Sorry, when I said OpenVox I should say VoxBone.
Regards,
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I could be wrong here, but I think that you're looking for SendDTMF and
not PlayDTMF. getting it confuddled with PlayTones?
Jan du Toit wrote:
Hi all.
I'm confused. On http://www.voip-info.org/wiki-Asterisk+manager+API it
says that the PlayDTMF command is available since version 1.2.8. I
Hi Marek -
what is mean by partially incompatible in
http://www.digium.com/en/docs/misc/compatibility_notes.php
i have server with E7221+te110p mobo and i think i dont have any problems
You might want to ask Digium directly, but this generally means that
there's something on the
Hi Jordan -
On Wed, 2006-10-04 at 06:45 -0500, Jordan Novak wrote:
How is the best way to add,clear mailboxes and change passwords for
voicemail. I am guessing you need to remove the conf entries for the
mailbox restart asterisk and then add them back in and restart
asterisk. Is there a
Just a thought I had.
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
Doug.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi Folks,
I'm reading about meetme feature, but in accordance to voip-info it
say: A zaptel interface must be installed for conferencing to work.
Unfortunately I don't have a Zaptel Card, I'm using Asterisk like SIP
server then I would like to implement meetme function. What can I do?
Is
I'm no java or Asterisk guru, but, if what you have below is the exact
syntax you are using you might want to look at your capitulation of your
statements. I see sipShowPeerAction as well as SipShowPeerAction.
If this is of no value please ignore.
Zack
-Original Message-
From: [EMAIL
Hmmm,
It appears ztcfg is not being run. Any ideas why?
Matt
313-667-0970
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bernardo
Vieira
Sent: Wednesday, October 04, 2006 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I wonder if it has already been done somewhere?
http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect
It's not quite Tri-tone detection, and it's not done by the Dial()
commanda, but should yield the same result.
-BEGIN PGP
There's been a couple of those posted on this list already:http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/
http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdfSure they're for AAH/Trixbox, but the dialplan will work fine with vanilla Asterisk installs.Alex
On
R.R. Libera wrote:
Hello,
I need an USA DID + 15 b-channels. The only option I already have is
OpenVox and I want to see some alternatives. Sound quality is my
priority. Thanks in advance.
R.R Libera
___
--Bandwidth and Colocation provided by
Mike wrote:
Hi,
My setup is the following: Voip provider---(SIP
DID)---Asterisk box(SIP through a termination
provider)---multiple cell phones.
The cell phones each have their extension (201,202,203,204) and I'd
like to be able to have them transfer a call
Is anybody using the Intrado
V9-1-1 service with asterisk?
Could you share some info, setup information if so.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
Douglas Garstang wrote:
Just a thought I had.
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
Doug.
Doug,
It would be cool, but for now you can use System() and logger. If you
need to get something done quickly...
--
Kristian Kielhofner
You could uses System() and the Logger command. Wouldn't be hard.
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialplan Syslog
Just a
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
That could be usefull, but what is wrong with : System(logger Asterisk
can use syslog) ?
___
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Hello,
Can someone help me with this please?
Attached is the log file.
thank you
Original Message
Subject:[Fwd: asterisk-users Digest, Vol 26, Issue 166]
Date: Fri, 29 Sep 2006 10:31:21 -0400
From: asterisk-user [EMAIL PROTECTED]
To:
Alex,
Those
examples elaborate on the examples supplied with Asterisk, and that's about it.
I tried to build a tiered DUNDI model with upstream DUNDi servers that served
requests to downstream DUNDi servers that acted as registration servers and used
the 'precache' option to send the
When you Dial() the cell, are you passing the 't' parameter? Also: When the
call hits the cell, is Asterisk still in the media stream? canreinvite=no
should be explicitly specified in the SIP accounts of your providers in
sip.conf. One more thing: Do you know for a fact that inband DTMF is being
2006/10/4, Steve Underwood [EMAIL PROTECTED]:
Giedrius Augys wrote: Hi,Now I'm testing faxes with spandsp. I have problems that spandsp do not add headers to fax page: LOCALHEADERINFO. Please help me.There is a bug in adding page header with
spandsp-0.0.2pre26. I havefixed this in the development
Has anyone created a GUI for this. I would like to implement a server
specifically for Voicemail using out of band signalling tied to a PBX. I
fear the management will be exhaustive though.
___
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Anyone have happen know how to reset the password on a TNT Max?
Thanks.
Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
___
--Bandwidth and Colocation
Ah. I'd like to know what others think, but if you're right than it's a
lost cause.
I thought Asterisk kept some sort of control over the call.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Glaus
Sent: October 4, 2006 3:24 PM
To:
I found a workaround, inspired by Colin's suggestion to move the startup
to the rc.local file. It turned that his exact suggestion didn't work
in my situation. I subsequently discovered, though, that after Zaptel
and Asterisk started in the boot sequence in the usual way, all I had to
do for the
You didn't say, but my guess is you are using either a 4-port or 2-port
Digium card, right?
What do the contents of /etc/modprobe.d/zaptel look like?
You will probably find that there isn't an entry like:
install wct4xxp /sbin/modprobe --ignore-install wct4xxp $CMDLINE_OPTS
/sbin/ztcfg
I put
Hi List;
I would like to know where I can find the IP Phones
that can be used with Asterisk? Is there any link?
Regards
Bilal Ghayad
Mobile: 00965 9849460
Office: 00965 2623174
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
I tried unsuccessfully to get this to work. I am using AAH 2.7 which has
asterisk 1.2.5.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On 10/4/06, asterisk-user [EMAIL PROTECTED] wrote:
Hello,
Can someone help me with this please?
Attached is the log file.
thank you
Original Message
Subject:[Fwd: asterisk-users Digest, Vol 26, Issue 166]
Date: Fri, 29 Sep 2006 10:31:21 -0400
From: asterisk-user
yep,
# modprobe ztdummy
You need some special routines compiled in the kernel, google around a
bit to find wich ones.
Other solution may be use app_conference, is not included in asterisk
sources, that app does not require zaptel timing.
Regards
On 10/4/06, omar parihuana [EMAIL PROTECTED]
I could be wrong here, but I think that you're looking for SendDTMF and
not PlayDTMF. getting it confuddled with PlayTones?
He is not confused. PlayDTMF is a manager command, not an dial plan
application, but included in the same module that SendDTMF
(app_senddtmf.so). I dont think is available
omar parihuana wrote:
Is possible use meetme feature without Zaptel card? (ztdummy will be
the solution? )
Yup. :P
Thanks in advanced..
--
Mojo [EMAIL PROTECTED]
Office Manager, Horan Company, LLC
(907) 747- x112
___
--Bandwidth and
Well, System(logger) is going to be resource intenstive as it has to spawn a
process. I actually just emulated the behaviour with FastAGI. My client side
looks something like:
// -
//
// SysLogger:
//
//
??? I do it with a Zap channel no problem. In my case,
1. Call comes in from PSTN (Zap channel)
2. Call is routed back out a Zap channel using the Dial() command with the
't' option
3. Asterisk is still in the media stream, so it listens for inband DTMF
4. User presses Hash, Asterisk says
Do you know for a fact that inband DTMF is being
procesed by Asterisk when the call hits the cell?
Well it seems I'm wrong but how do you setup asterisk to process inband
dtmf? I dial the cell phone with the 't' in the dial string.
When I hit anything from my cell phone, asterisk doesn't
Hi List;
We need to apply Video conference, can asterisk
support this? What I need for that?
Regards
Bilal Ghayad
IP Telephony Engineer
Mobile: 00965 9849460
Office: 00965 2623174
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At 15:34 10/4/2006, bilal ghayyad wrote:
Hi List;
I would like to know where I can find the IP Phones
that can be used with Asterisk? Is there any link?
Regards
Bilal Ghayad
Mobile: 00965 9849460
Office: 00965 2623174
http://www.voip-info.org//tiki-pagehistory.php?page=Asterisk+phonesdiff2=38
On 4 Oct 2006, at 18:35, Joe wrote:
Hello,
I'm looking for a SIP client that work with Asterisk that will run on
Linux or Solaris and will work with X Windows. I know X won't all the
media to work but I'm really only interested in SIP signaling.
What are you running your X displays on ?
You
bilal ghayyad wrote:
Hi List;
I would like to know where I can find the IP Phones
that can be used with Asterisk? Is there any link?
Regards
Bilal Ghayad
Mobile: 00965 9849460
Office: 00965 2623174
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Hi AllI am using trixbox asterisk 1.2I have enabled canreinvite=yes and no tT in the dialplan as it has been described in the various forums.Still the voice call goes thru the asterisk server.
How can i really make the call between 2 grandstream devices( i am using HT 488, HT286 and SIP
Hi all,
Is it possible to connect a modem to a remote service through asterisk ?
Basicly to ilustrate : Accounting department need to connect with analog
modem to their bank to order some wire transfert.
Modem - Chanel Bank FXS - Asterisk - TDM2400 FXO - Modem in
remote site.
Actulaly, the
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