thank you for your reply. I'm French.
I added the field operator, and nothing.
when I call sda, it does not work.
I bought numbers sda.
I have a voip access.
Does the operator configure sip accounts?
Does the operator configures the corresponding sip sda?
Regards.
Ludo
2009/4/1 Carlos Rojas
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
On Wed, 1 Apr 2009, Erick Perez wrote:
We are planning to run an outbound only campaign. A 20-second voice message
will be played to callers and our dialer on machine1 will send to
machine2-asterisk (1.4) instructions to dial 400 calls, play the message and
hang up. This will be done for
Hello, all.
This is just an email to inform you I have added a SIP header in Asterisk SIP
message that is handled by the proxy:
On Asterisk extensions.conf:
SIPAddHeader(X-number-to-dial: ${NUMBERTOREACH})
Dial(SIP/${MAINPEER}|100|t)
and on OpenSIPS:
if (is_present_hf(X-number-to-dial)) {
I had the exact same problem and errors some time ago (search the
archives for PRI dropping #2) using Asterisk 1.4.18, Zaptel and a
Digium TE121. I tried all kind of things, had telco technicians come
out and whatnot. The solution was two-folded - 1) I reinstalled my
server, 2) I updated to
Hi!
I used a work around to the problem.
I added a Playback(silence/1) quite after the Answer() and now everything is
working fine again.
100, 1, Answer()
100, 2, Playback(silence/1)
100, 3, Dial(SIP/XX,,r)
Hope this helps,
Alex
Alex Mosburger
-Original Message-
From:
Hi Henrik,
I would like to do the same thing you are doing here. I want to implement an
external queue functionality so I need to stop a play file launched previously
with an async agi command on caller's channel, sending the call to agent's
extension.
I'm redirecting caller's channel with
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent:
Hi All,
At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael
Robertson to join the discussion to filed questions about OpenSky and
Gizmo5. I have been testing all of these Skype to X methods except SIP
for Skype since I have no word from them. I can tell you that we've
had good
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your
Hi,
I am trying to connect a doorbell to a Xorcom device. And the setup is
quite simple. But when I push the doorbell all I see on the asterisk cli
is:
-- Starting simple switch on 'Zap/11-1'
[Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough
digits (and no ambiguous match)...
Hi,
sorry for joining the discussion so lately. I'd like to ask you to check
http://bugs.digium.com/view.php?id=14810. The patch tries to address the
issue using channel-variables to propagate the hangup-cause to the calling
channel.
Best regards, Marcus
On Fri, Jan 23, 2009 at 3:08 PM,
Formula here: http://www.nessoft.com/kb/50 has jitter in it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: Marc Leurent [mailto:lf...@leurent.eu]
Sent: 2009 m. balandžio 2 d. 13:56
To: asterisk-users@lists.digium.com
Cc:
Dear All,
Thanks for taking the time to read this. I have been presented with a massive
task. I'm not an asterisk expert, but I do know my way around a linux server
and infrastructure, and I know when things are not done correctly. A large
number of minutes are routed every month, (1m+) and I
Loic Didelot wrote:
Hi,
I am trying to connect a doorbell to a Xorcom device. And the setup is
quite simple. But when I push the doorbell all I see on the asterisk cli
is:
-- Starting simple switch on 'Zap/11-1'
[Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough
digits
On Thu, Apr 02, 2009 at 01:06:04PM +0200, Loic Didelot wrote:
Hi,
I am trying to connect a doorbell to a Xorcom device. And the setup is
quite simple. But when I push the doorbell all I see on the asterisk cli
is:
-- Starting simple switch on 'Zap/11-1'
[Apr 2 13:00:40] DEBUG[8771]:
On Thu, Apr 02, 2009 at 09:33:49AM +0100, Gordon Henderson wrote:
As for the server - get *everything* in RAM. At least with no disk IO,
This is true with respect to e.g. recordings.
But most other operations won't bother the disk much. If you have 400
channels doing roughly the same things,
Martin wrote:
I wonder why people don't get it ? X100P is a winmodem was and always will be.
What makes you think anyone doesn't understand that? The problem is the
chip on the X100P isn't made any more, and X100P cards are no longer so
plentiful. You'll notice the price is going up. They
On Thu, Apr 02, 2009 at 08:28:44PM +0800, Steve Underwood wrote:
Several Winmodem chips are still readily available, and so are cards
containing them. What is missing is someone putting the effort into
making drivers for them.
Can you list, off the top of your head, modems for which the
Furthermore, the following two IETF documents address the need to both
signal the hold and provide the music:
1. RFC 5359 (Session Initiation Protocol Service Examples)
2. draft-worley-service-example-03 (Session Initiation Protocol
Service Example -- Music on Hold)
Unfortunately they both
Tzafrir Cohen wrote:
On Thu, Apr 02, 2009 at 08:28:44PM +0800, Steve Underwood wrote:
Several Winmodem chips are still readily available, and so are cards
containing them. What is missing is someone putting the effort into
making drivers for them.
Can you list, off the top of
Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something, do you still have the patch around?
Moy
On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote:
Hi Henrik,
I would like to do the same thing you are doing here. I want to implement an
Tadaaah,
thanks.
immediate=yes fixed it.
Loic
On Thu, 2009-04-02 at 15:11 +0300, Tzafrir Cohen wrote:
On Thu, Apr 02, 2009 at 01:06:04PM +0200, Loic Didelot wrote:
Hi,
I am trying to connect a doorbell to a Xorcom device. And the setup is
quite simple. But when I push the doorbell all I
Hi,
I want my telco to redirect all the incoming calls to my Asterisk
towards another number (connected to my old Panasonic PBX) so I can
stop Asterisk and repair my office. I tried to send the code *#21# (
Dial(mISDN/1/*#21#) ) but I get a busy channel while it is working with
an ISDN
What about *#72#?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Thursday, April 02, 2009 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] activate telco
My only comment is that I am having moral issues with assisting anyone
that is planning to call one million phone numbers to play a message and
hang up. Doesn't sound like an opt-in kind of campaign to me. When
such a thing happens to me on my home phone I get extremely angry.
j
On Wed,
Yes, we have enough car warranty calls now, just recently joined by the
reduce your credit card interest rate calls.
:-(
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday,
Dears
How can I send or force sending 180 Ringing instead of 183 back to the caller
?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
Hi All,
Does anyone have a config example for setting up SIP trunking to a
CIsco IAD2400 and are willing to share?
I've done SIP trunking to Cisco 2600's with PRI's but not to the POTS
lines on the IAD's, I'm wondering if that is possible and how to
specify the DID on the POTS line config for
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Hi Danny,
it is the code to ask the telco your status about the redirection
service...when you dial that number, you hear a voice from telco telling
you if the redirection has been activated or not.
Giorgio
Danny Nicholas wrote:
What about *#72#?
-Original Message-
From:
Cary Fitch escribió:
Yes, we have enough car warranty calls now, just recently joined by the
reduce your credit card interest rate calls.
:-(
Cary
It's unbelievable how people use all this marketing strategies that
annoy people far away the limit. Fortunately, nobody here in Colombia is
Is there a way to program an FXO device to totally ignore incoming calls?
I want to put an FXO on a Fax line so that 911 calls can be sent via that
line, but all other activity on the line is between the Fax machine and the
phone company.
Perhaps munge the ring tone detect if nothing else?
Cary
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From:
You could use ex-girlfriend logic to hang up the call without answering.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Thursday, April 02, 2009 10:38 AM
To: 'Asterisk Users Mailing List -
Hi All;
I got to know (reading on the wiki) that fxotune was have a bug, and it has
been fixed. But I do not know if my current asterisk version contain the fixed
one or not? How can I know?
My current asterisk version is 1.4.22
Any advise?
Regards
Bilal
Is it possible to have asterisk override the connection information embedded
in a SIP 200 packet with the registration information? I have multihomed
machines with softphones and they register just fine and sip works fine, but
the RTP packets get sent to the ip from the SIP connection information
On Thu, Apr 2, 2009 at 11:37 AM, Cary Fitch ca...@usawide.net wrote:
Is there a way to program an FXO device to totally ignore incoming calls?
put the port in that context :
[incoming-noanswer]
exten = s,1,Hangup()
hth
___
-- Bandwidth and Colocation
Hey !
this can drive to heart attacks
randulo a écrit :
Nice one, Olle ! :)
On Wed, Apr 1, 2009 at 9:18 AM, Olle E. Johansson o...@edvina.net wrote:
* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND
VIDEO TO MICROBLOGGING!
- Cary Fitch ca...@usawide.net wrote:
Is there a way to program an FXO device to totally ignore incoming
calls?
I want to put an FXO on a Fax line so that 911 calls can be sent via
that
line, but all other activity on the line is between the Fax machine
and the
phone company.
This is a new installation.
Here are the specs of my system:
Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11 CDT 2007 i686
Intel(R) Xeon(R) CPU E5420 @ 2.50GHz GenuineIntel GNU/Linux
08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11)
(ethernet?? first time
Hi,
In our office, we're migrating from a Cisco set up to Asterisk. We'd
like to do it gradually, so I've added an asterisk server as an H.323
gateway to the call manager so out going calls are going through
asterisk. So far so good.
Am now faced with the challenge relaying incoming calls from
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
criptos wrote:
This is a new installation.
08:08.0 Ethernet controller: Digium, Inc. Unknown device 8000 (rev 11)
(ethernet?? first time with a card like that for me)
dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.12.1
Zaptel Echo Canceller: MG2
proc dev
bilal ghayyad wrote:
Hi All;
I got to know (reading on the wiki) that fxotune was have a bug, and it has
been fixed. But I do not know if my current asterisk version contain the
fixed one or not? How can I know?
My current asterisk version is 1.4.22
Current version of fxotune (in
Holy crap!
This list is magical :D it started working... maybe yesterday was too late
when I was configuring this thing
On Thursday 02 April 2009 09:54:18 criptos wrote:
This is a new installation.
Here are the specs of my system:
Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11
Hi All;
If I need to set the opermode to King Saudi Arabia, what the name I have to
use? For example, to set it for kuwait then I use opermode=KUWAIT. So what will
be for Saudi Arabia?
Any advise?
Regards
Bilal
___
-- Bandwidth and
Did you reboot? Zaptel does not work until you reboot or do a manual
modprobe.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of criptos
Sent: Thursday, April 02, 2009 11:18 AM
To: Asterisk Users Mailing List -
Dear Mathew;
Kindly find the link of the batch tha fixed the bug:
http://bugs.digium.com/view.php?id=7136
It is written that last update was in 2008-06-07 11:36, so for that I do not
know if my asterisk and zaptel versions include this fix or not? Because I
installed them before this date.
bilal ghayyad wrote:
Dear Mathew;
Kindly find the link of the batch tha fixed the bug:
http://bugs.digium.com/view.php?id=7136
It is written that last update was in 2008-06-07 11:36, so for that I do not
know if my asterisk and zaptel versions include this fix or not? Because I
On Thu, 2009-04-02 at 10:40 -0500, Danny Nicholas wrote:
Sipaddheader(180 Ringing) might do the trick.
Danny, I appreciate your enthusiasm for helping people on the mailing
list, but unfortunately this is not the correct method of doing what the
original poster is asking about. It's not enough
Thanks
Actually, I tried everything before... I even do a genzapconf -d.
I really don't know what happened.
After I posted to the list, I made another genzapconf -d and then genzapconf -
l and the card just appeared.
Don't know the exact reason, but... it's working (the card, I still
Is there anyway a T1/PRI can ignore the ANSWERED signal and just go
straight from a dial command
to the call was answered?
I have a PBX that when calling a certain analog trunk it is not giving
me signaling that the call was
answered however I hear the PA system come off hook and give dial
We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to
Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of
meetme. Meetme seems to not be able to find a zap channel for conferencing.
We use voice introductions in our conference bridge and it seems to break
that
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
Thanks for not being too critical and for providing a good clarification.
I've been doing Asterisk for about 7 months now and realize that my answer
might or might not be correct.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To:
Hello All,
Is there anyone out there that is able to integrate a custom visual basic 6
application to Fonality’s Trixbox HUD Pro?
Thanks,
Greg
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
Humm... should the list would be magic again?
I have just intsalled, using the register, benchmark and downloared the
correct codec to my asterisk installation, but I don't have the
g729 command at my CLI...
Any advice... Do I reboot? ;D
___
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk
You should not have a G729 command on the CLI. Codecs are addressed in
sip.conf, dahdi.conf, etc. restarting Asterisk might do the trick. You
only need to reboot for a driver level change.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Danny Nicholas wrote:
You should not have a G729 command on the CLI. Codecs are addressed in
sip.conf, dahdi.conf, etc. restarting Asterisk might do the trick. You
only need to reboot for a driver level change.
This is incorrect. Digium's codec_g729a.so module does in fact add a
'g729 show'
Try replacing answer() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Documenation show that at the asterisk cli you can use the g729 show to
show the codec usage/license availability...
This is what is missing, So, I'm not sure that my licenses are being loaded.
On Thursday 02 April 2009 12:35:18 Danny Nicholas wrote:
You should not have a G729 command on
The Asterisk.org development team has released Asterisk versions 1.2.32,
1.4.24.1, and 1.6.0.8. These releases are available for immediate download from
http://downloads.digium.com/.
This update for Asterisk includes a security fix for chan_sip. Please see the
associated security advisory for
Asterisk Project Security Advisory - AST-2009-003
++
| Product | Asterisk |
On Thu, Apr 02, 2009 at 09:54:18AM -0600, criptos wrote:
This is a new installation.
Here are the specs of my system:
Linux asterisk 2.6.21.5-smp #2 SMP Tue Jun 19 14:58:11 CDT 2007 i686
Intel(R) Xeon(R) CPU E5420 @ 2.50GHz GenuineIntel GNU/Linux
08:08.0 Ethernet controller:
criptos escribió:
Humm... should the list would be magic again?
I have just intsalled, using the register, benchmark and downloared the
correct codec to my asterisk installation, but I don't have the
g729 command at my CLI...
Any advice... Do I reboot? ;D
Did you load your brand
codec is on place:
r...@asterisk:/usr/lib/asterisk/modules# ls chan_*
chan_agent.so* chan_iax2.so* chan_mgcp.so* chan_phone.so* chan_skinny.so*
chan_zap.so*
chan_alsa.so* chan_local.so* chan_oss.so* chan_sip.so*
chan_unicall.so*
r...@asterisk:~# asterisk -v
On Thu, Apr 02, 2009 at 09:22:52AM -0700, bilal ghayyad wrote:
Hi All;
If I need to set the opermode to King Saudi Arabia, what the name I
have to use? For example, to set it for kuwait then I use
opermode=KUWAIT. So what will be for Saudi Arabia?
$ grep -i saudi
asterisk*CLI module load codec_g729a.so
asterisk*CLI [Apr 2 14:06:10] WARNING[3732]: loader.c:605 inspect_module:
Module 'codec_g729a.so' does not provide a description.
[Apr 2 14:06:10] WARNING[3732]: loader.c:662 load_resource: Module
'codec_g729a.so' could not be loaded.
[Apr 2 14:06:10]
Hi Jerry,
If you are calling that number from Asterisk via T1 and you do not get
ANSWER/CONNECT message
from that particular number/line then a workaround might not work.
It's simply because there's no connectivity. You might only have
early audio one way from that PA line to you but not the
check if you loaded the module
show modules like codec_g729
or simply try to unload/load codec_g729.so
Martin
On Thu, Apr 2, 2009 at 1:25 PM, criptos crip...@aullox.com wrote:
Humm... should the list would be magic again?
I have just intsalled, using the register, benchmark and downloared
criptos wrote:
asterisk*CLI module load codec_g729a.so
asterisk*CLI [Apr 2 14:06:10] WARNING[3732]: loader.c:605 inspect_module:
Module 'codec_g729a.so' does not provide a description.
[Apr 2 14:06:10] WARNING[3732]: loader.c:662 load_resource: Module
'codec_g729a.so' could not be
I'd rather put
Wait(3600) than Hangup(). Furthermore hangup would probably not work
since the line was not taken offhook.
Asterisk would do cleanup on the logical zap channel but then the next
ring would create another zap channel and so on till the line stops
ringing.
Martin
On Thu, Apr 2,
Do you know how to play a musiconhold or ... but when its ringing the user
will hear the Ring Back Tone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 9:44 PM
To:
Kindly its too important to me
If any one can help me on a command can force asterisk to send 180 and 183
sip message in the same time
Regards
Do you know how to play a musiconhold or ... but when its ringing the user
will hear the Ring Back Tone
-Original Message-
From:
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
i want to configure a x100p card an use it with asterisk, so i download,
compile and install:
asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9
i try almost everything i found on the net but without success:
if i run
Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic
Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
descague compile e instale lo siguiente:
asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9
Sin embargo no logro configurar la tarjeta con
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back and another come
sin and then both customers can hear each other { which i think is VERY
dangerous [image: Wink] }
Any Solutions ?
Hi -
Does anybody know if an FXS generates line voltage when Dahdi/Zaptel
is disabled?
Thanks,
Noah
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote:
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
i want to configure a x100p card an use it with asterisk, so i download,
compile and install:
asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9
i
I've been trying to build h323plus (both the release and svn) for
chan_h323 on Fedora 10. No joy. I posted on the h323plus ml, but no
response.
Anybody here actually built it on Fedora? Wanna share your secrets, or
even better a specfile?
sean
___
--
nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y
/etc/asterisk/chan/dahdi.conf
2009/4/2 Manolet Gmail mano...@gmail.com
Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic
Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
descague
Hello,
I have a fairly standard call center. I'm running 1.4.23.1. I am
trying to get a mechanism where :
1- Employee A can have the phone at his desk ring when a call comes in the queue
2- When already on a call, employee A does not hear a beep in his
phone because another call is trying to come
On Thu, 2 Apr 2009, Haim Dimer wrote:
The issue is the that the agent needs to wait on the phone for a call to
come in. I read
http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin but
it will be deprecated and the doc/queues-with-callback-members.txt means
that I would have
On Thu, Apr 2, 2009 at 4:38 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote:
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
i want to configure a x100p card an use it with asterisk, so i download,
compile and install:
system.conf:
# Global data
loadzone= us
defaultzone = us
el archivo /etc/asterisk/chan/dahdi.conf no existe, estoy usando
asterisk 1.2 de todas maneras como dije ztcfg -v tampoco muestra la
tarjeta, asi que de ninguna forma la va a ver asterisk, si hago en
asterisk dahdi show
Yes, I have the patch around here. I think it's the one you said at
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
Due to the res_agi patch excedes the size limit for this mailing list,
(40Kb) I wasn't able to attach it on this post, so you can find it at
How do I manually set up a ring group?
All the info I've Googled tells me how to do this using Trixbox or FreePBX.
I am using standard Asterisk 1.4 configuring at the CLI.
Michael
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Follow instructions from the following line to configure Asterisk 1.2 with
zaptel drivers for the X100P. If you are using dahdi-linux drivers instead
of zaptel, then instead of zaptel.conf you need to have a properly
configured /etc/dahdi/system.conf and instead of /etc/asterisk/zapata.conf
use
A group of phones that ring all at once?
Like:
exten =
5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/522600140
5,20)
Take out the line breaks.
Or were you looking for something else?
CF
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Fri, 03 Apr 2009 12:32:03 you wrote:
A group of phones that ring all at once?
Like:
exten =
5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
0 5,20)
Take out the line breaks.
Or were you looking for something else?
CF
That is what I am currently doing -
Hello!
I am trying to configure my digium TE220 dual-span pci express card
with Dahdi. I seemed to have managed to set up the card with the
Dahdi kernel, as demonstrated by executing dahdi_scan:
[1]
active=yes
alarms=RED
description=T2XXP (PCI) Card 0 Span 1
name=TE2/0/1
manufacturer=Digium
Is that a Bug in asterisk and meetme file ?
On Fri, Apr 3, 2009 at 2:27 AM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back and another come
sin and then
On Thu, Apr 02, 2009 at 05:55:04PM -0500, Manolet Gmail wrote:
On Thu, Apr 2, 2009 at 4:38 PM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote:
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
i want to configure a x100p
On Thu, Apr 2, 2009 at 4:36 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello!
I am trying to configure my digium TE220 dual-span pci express card
with Dahdi. I seemed to have managed to set up the card with the
Dahdi kernel, as demonstrated by executing dahdi_scan:
[1]
active=yes
Then you need to edit /etc/dahdi/system.conf
manually and add
fxsks=1
then dahdi_cfg -vv
then check if wcfxo module takes interrupts
dahdi_test
Martin
On Thu, Apr 2, 2009 at 5:55 PM, Manolet Gmail mano...@gmail.com wrote:
What's the output of:
lsmod | grep ^dahdi
r...@lhserver:~#
maybe you have to call the dahdi_scan with some argument to
autogenerate config files ...
for any dahdi T1/E1 card you have to work properly you have to have
/etc/dahdi/system.conf
configured with span= and bchan= and dchan= keyword
and /etc/asterisk/dahdi*.conf with channel = keyword.
check it
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