Hello,
I'm working on an Asterisk configuration for a call center, and they
bill based on the time spent talking to an agent, but not for any time
spent waiting in a queue. The CDR information contains the entire
duration of the call as billable seconds, including time spent waiting
in the
Le Tue, 14 Apr 2009 10:28:42 +1000, David Klaverstyn
d...@klaverstyn.com.au a écrit :
Hi All,
I'm in the process of writing an install script and I would like to
change some settings for the install process but I don't want the
user to go into menuselect and make the changes manually.
Is
I wouldn't approach this by trying to rework the CDRs at all; CDRs are
fundamentally low-level call records. They correspond to calls.
If you need logic to support a billing model for some specific
application (i.e. time after connect to agent), I would approach that
from a higher layer of
Thanks to everyone who replied. I'll let you know if I have any issues
with Fax for Asterisk...
Regards
Ian
Hello
I am going to try the new Digium Fax for Asterisk product. I'm planning
to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs.
I'm looking at Grandstream
Hello all-
As a general rule, the INPUT FW rules for SIP should be 5060 UDP and
1-2 UDP right?
Is TCP used in any part of the SIP structure?
Michael
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Hi michael,
you should open both tcp,udp 5060,5061 too and as you mentioned between
1-2.
On Tue, 2009-04-14 at 20:37 +1200, Michael wrote:
Hello all-
As a general rule, the INPUT FW rules for SIP should be 5060 UDP and
1-2 UDP right?
Is TCP used in any part of the SIP
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I' try to implement an automatic callback mechanism, just for local SIP calls..
Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten =
OK, thanks. If I could convince them to use info, would that be
better as far as the duration is concerned?
on Monday 04/13/2009 Brent Davidson(br...@texascountrytitle.com) wrote
John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to
On Tue, 14 Apr 2009 20:47:29 you wrote:
Hi michael,
you should open both tcp,udp 5060,5061 too and as you mentioned between
1-2.
AFAIK 5061 TCP is for TLS SIP which isn't used much yet?
Is TCP the default for 5060, with UDP as fallback, or is this provider
dependent?
Michael
Can you do a 'sip show peer bt201' and show us what the output is? I don't
see anywhere that indicates that your phones are even registered or trying to
talk to Asterisk.
Also do a 'sip show peers' and show us what that gives you.
From: jonas.kell...@telenet.be
To:
On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote:
Can somone spot the problem? Is someone using t38modem with asterisk
successfully?
Cheers,
Florian
The best advice I can offer is to give up now and use Callweaver otherwise you
can spend hours, or days, with no working result.
Hello,
Substitution is a classic PBX feature.
How would it best be defined ?
For me, it's the ability to configure a telephone so that it would behave
almost exactly like the one you would regularly use.
My questions are :
- how should behave the phone you regularly used ?
- how should incoming
Hi!
I'm trying to get t38modem working and started with the loopback mode. I
installed everything according to
http://www.voip-info.org/wiki/view/T38modem+configuration+with+Asterisk
and used a stock asterisk 1.4.20.1 compiled from source. Openh323,
ptlib_unix and t38modem are compiled from
Dears
How to disallow asterisk to send the keep alive 200 ok message to the peers
and trunks.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express
Hey there,
I'm trying to convert some call recordings from asterisk we have in .gsm
format to something I can pipe through ffmpeg - wav would be good, mp3
would be amazing!
I've been trying playing with sox but I don't seem to be getting too far
with
1239101491.30.gsm -ql -r 64000 -t wav
please look at
http://www.voip-info.org/wiki/view/Asterisk+SRTP
and try compilerun clients with srtp (linksys,grandstream,aastra,
qutecom, eyebeam, ...)
digium need feedback for srtp inclusion to 1.6.3.0
http://bugs.digium.com/view.php?id=5413
if you need additional info, i'm on jabber -
On Tue, Apr 14, 2009 at 7:39 AM, Tim Dobson li...@tdobson.net wrote:
I'm trying to convert some call recordings from asterisk we have in .gsm
Why not use sox for this purpose?
sox mygsm.gsm -r 8000 -c 1 mywave.wav resample -ql
Once it's a wav you can mp3 it with lame or your preferred encoder,
Another satisfied CentOS customer here.
CentOS 5.2 running 1.6.0.1 on a Dell 2950 with a Sangoma A104D, call center
environment, 20 employees doing outbound dialing and answering queue calls
(migrating the other 380 seats next month)
ZERO issues, not a single dropped call reported, no crashes,
Hey,
I record the message in ULAW
exten = s,1,Record(${A_record}:ulaw,0,60)
After that I call sox with this command:
/usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1
$wav_fl
Regards,
Arjan Kroon
Mobillion BV
-Oorspronkelijk bericht-
Van:
My suggestion is to use a tool made specifically for this - we happen to
sell one, but there are many options with different prices and licencing
model. Don't reinvent the wheel and concentrate on added value.
l.
2009/4/14 Scott Gifford sgiff...@suspectclass.com
Hello,
I'm working on an
On Tue, Apr 14, 2009 at 7:11 AM, Michael mich...@networkstuff.co.nz wrote:
On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote:
Can somone spot the problem? Is someone using t38modem with asterisk
successfully?
The best advice I can offer is to give up now and use Callweaver otherwise you
On Wed, 15 Apr 2009 00:30:43 David Backeberg wrote:
Once it's a wav you can mp3 it with lame or your preferred encoder,
but encoding and playing mp3s takes more cpu than just playing it in
gsm, or stopping after sox and playing as a wav.
Has anyone got any suggestions based on previous
On Wed, 15 Apr 2009 00:43:45 you wrote:
Now for the part I do know something about. Native asterisk fax
support and native asterisk sip support improved in 1.6. With 1.6
there is a built-in app_fax module which works quite well for sending
fax over SIP with T.38. I found the configuration and
- Scott Gifford sgiff...@suspectclass.com wrote:
The CDR information contains the entire
duration of the call as billable seconds, including time spent
waiting
in the queue. I would like the billable seconds to only include the
time spent actually talking to an agent.
You're absolutely
Turn off the 'qualify' setting on the peer(s) in sip.conf to stop it
sending OPTIONS pings.
As for it responding with a 200 OK to such, there is no way to turn
that off.
--
Sent from mobile device
On Apr 14, 2009, at 8:14 AM, Khaled W. Chehab kche...@xplorium.com
wrote:
Dears
How to
Michael wrote:
On Tue, 14 Apr 2009 20:47:29 you wrote:
Hi michael,
you should open both tcp,udp 5060,5061 too and as you mentioned between
1-2.
AFAIK 5061 TCP is for TLS SIP which isn't used much yet?
Is TCP the default for 5060, with UDP as fallback, or is this provider
On Tue, Apr 14, 2009 at 4:15 PM, Jared Smith jsm...@digium.com wrote:
- Scott Gifford sgiff...@suspectclass.com wrote:
The CDR information contains the entire
duration of the call as billable seconds, including time spent
waiting
in the queue. I would like the billable seconds to only
On Tuesday 14 April 2009, Michael wrote:
asterisk-1.6 with app_fax built-in
Try 1.6. You'll be glad you did.
While I have not tried Asterisk 1.6 because I settled on Callweaver
at the time (which has native T38 support), I *strongly* recommend
going with software that has native T38
On Tue, 14 Apr 2009, Arjan Kroon | Mobillion wrote:
I record the message in ULAW
After that I call sox with this command:
/usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1
$wav_fl
What are -1 and -2 for? Both sox 12.17.5 and 12.18.1 say they are
invalid.
Thanks in advance,
On Tue, Apr 14, 2009 at 9:52 AM, Florian Hackenberger
f.hackenber...@chello.at wrote:
With asterisk 1.6, is it possible to use hylafax, or would asterisk
terminate the fax calls itself?
With app_fax integrated into asterisk-1.6, you have an 'infinite'
modem pool that you control through the
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Even I can see in the CLI debugging the status is ringing
-my idea is to add music on hold stop when asterisk detect --
SIP/OPNS-096456c0 is ringing line
In
Scott Gifford escribió:
Hello,
I'm working on an Asterisk configuration for a call center, and they
bill based on the time spent talking to an agent, but not for any time
spent waiting in a queue. The CDR information contains the entire
duration of the call as billable seconds, including
David Backeberg wrote:
It may be possible to use hylafax, but
I don't know how or why you would.
The reason *why* is generally due to support issues.
For one, HylaFAX probably has a better T.30 implementation in its Class
1 driver than does app_fax. At least that historically has been true.
Khaled W. Chehab wrote:
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Remove the 'm' out of your dial command:
m([class]) - Provide hold music to the calling party until a requested
channel
You are closing in. what does users.conf look like?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 3:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
In sox 14.0.1 the -1 is a one byte sample, -2 is a two byte sample,
therefore the command is sampling one-in, two-out.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, April 14, 2009
This is what I derive from your emails so far:
1. Asterisk is up and running
2. you have two SIP Peers that are running but not registered (sip show
peers shows not monitored, phones don't go off-hook, dialtone)
3. I've experienced similar problem with a X-Lite softphone. When the
Put register=yes in the BT201 and GXP1200 contexts of sip.conf
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To: asterisk-users@lists.digium.com
Subject:
You are doing 210 (enter), 210(dial) or 210 (#)? You have to engage the
dialer in some fashion.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 12:20 PM
To:
To the best of my knowledge, the only way for you to control the
duration sent to the PSTN lines is for you to be directly connected to
the lines so you can set the tone duration in zapata.conf / dahdi.conf
or to use inband signalling.
One thing you might try is researching the SipDtmfMode
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From:
On Tue, Apr 14, 2009 at 10:46 AM, Lee Howard fax...@howardsilvan.com wrote:
David Backeberg wrote:
It may be possible to use hylafax, but
I don't know how or why you would.
The reason *why* is generally due to support issues.
What I was specifically getting at in the context of that response
just for a test, run service iptables stop as root on the asterisk
server and then reboot your phones. after that, try again and see if
the phones are making communications with asterisk.
you can turn the firewall back on with service iptables start
jonas kellens wrote:
Hi there,
this is
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Steve Edwards had (IMO) the best answer to this. Here is an example of
how-to in a regular dialplan:
[sat]
exten = s,1(start),Noop(in test Section)
exten = s,n,AGI(satver.agi|)
exten = s,n,Set(Play1=record/silence)
exten = s,n,Set(Play2=record/silence)
exten = s,n,Set(Press1=record/silence)
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Lee Howard wrote:
David Backeberg wrote:
It may be possible to use hylafax, but
I don't know how or why you would.
The reason *why* is generally due to support issues.
For one, HylaFAX probably has a better T.30 implementation in its Class
1 driver than does app_fax. At least
Tim Dobson wrote:
1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
as ffmpeg borks at it:
Gah I meant
sox 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
--
Thanks everyone! Really appreciate it - all the documentation via google
is to convert *to*
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
On Tue, 14 Apr 2009, Danny Nicholas wrote:
Put register=yes in the BT201 and GXP1200 contexts of sip.conf
I admit to being a 1.2 Luddite, but isn't register asking your Asterisk
server to register with another endpoint? As in:
register = username:password:[authid]@sip client/peer
Hi masters!
I've this Asterisk 1.4.15 running. yesterday I had to change the
firewall schema that I had before.
I use to have a FW that would be my network FW/Proxy and do the NATs
for Asterisk. This FW was receiving too many requests from my LAN and
it was making the Asterisk 'cut' the calls or
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html
It means that a SIP device is re-using an old authentication challenge.
If it still registers and can place calls, there's no problem to worry
about. It's just a warning.
-Original Message-
From:
Since we don't know if he's having his phone's register, why not have
asterisk try to register them? If not, just another dumb suggestion from
me...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
awerf...@hotmail.com wrote:
Hi Friend,
Oh boy, this is fun.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and Colocation
On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
on Tuesday 04/14/2009 Kristian Kielhofner(kristian.kielhof...@gmail.com) wrote
On Mon, Apr 13, 2009 at 5:32 PM, John covici cov...@ccs.covici.com wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
Steve Underwood wrote:
Lee Howard wrote:
David Backeberg wrote:
It may be possible to use hylafax, but
I don't know how or why you would.
The reason *why* is generally due to support issues.
For one, HylaFAX probably has a better T.30 implementation in its Class
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Let's just simplify this a LOT:
Your phones have no dialtone. This means they are not registering
with asterisk. I see in your sip.conf, for both you phones, you have:
host=X.X.X.X
If you specify an address here, your phones will not register.
Instead, to make your phones register, set it to:
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Sorry - this is a bit off topic, but there is almost certainly someone
here who will know the answer... Perhaps even a snom employee :)
In recent snom firmware releases, the following sequence always causes
a call to be sent from line 'n'
Receive call on Line 'n' (where n 1)
Press Hold
Dear Ben,
I tried a lot ,Kindly can you give me an example on how to do that using a
macro.
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 5:53 PM
To:
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must
David Backeberg wrote:
What I was specifically getting at in the context of that response was
a comparison of dynamic modem pool versus fixed-size modem pool. When
faced with the choice between a fixed-size modem pool or one that
would grow or shrink dynamically with demand, I think the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial, but
I wanna do it during the dial, when user A is waiting for user B, answering the
phone. This should be possible, right?
I hope anyone knows if this is possible.
Is there a way in the dialplan to figure out which agent in a ring all queue
answered a line? I'd like to take specific action based on the agent upon
hangup.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando,
I have a same problem i add to the question
Francisco
Ryan M. Colbert wrote:
Is there a way in the dialplan to figure out which agent in a ring all
queue answered a line? I’d like to take specific action based on the
agent upon hangup.
Ryan M. Colbert
Director of Information
Try to parser queue_log file in real time and catch the event CONNECT
Regards,
Luis Morales
On Tue, Apr 14, 2009 at 12:44 PM, Ryan M. Colbert
ryan.colb...@rissman.com wrote:
Is there a way in the dialplan to figure out which agent in a ring all queue
answered a line? I’d like to take specific
Khaled W. Chehab wrote:
Dear Ben,
I tried a lot ,Kindly can you give me an example on how to do that using a
macro.
Remove the 'm' out of your dial command:
Doug
I'm not Ben, but I'll answer.
Shows us what your macro looks like and we'll chime in with some pointers.
Doug
--
I will summarize everything again and try to answer all the questions
asked while I was away.
First I stop Asterisk :
[r...@asterisk asterisk]# /usr/sbin/asterisk -r
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes
There is something wrong with my IPtables !!!
When i do :
service iptables stop
I see my phones register on the CLI !!
I can place a call and the phone rings !! I see a whole lot of
SIP-requests on the CLI with SDP-message in body !! That's good news...
What is wrong with my IPtables-rule
At least in version 1.6.0.x you can specify a macro to be executed when the
agent answers the queued call. This is an argument to the queue application.
Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,g
osub][,rule])
The optional macro parameter will run a macro on
CLI core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
On Tue, Apr 14, 2009 at 9:09 PM, Jim Dickenson dicken...@cfmc.com wrote:
At least in version 1.6.0.x you can specify a macro to be executed when the
agent answers the queued call. This is an argument to the queue application.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So
_
From: Cary Fitch [mailto:ca...@usawide.net]
Sent: Tuesday, April 14, 2009 1:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk-beginner : cannot make phone calls
using Asterisk
May I suggest divide and conquer?
I
Sorry for the brief interruption. The user spamming the list with
vacation replies has been removed.
JT
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct:
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Oh,
Thanks for answering Doug
I am using exten = _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
kindly can you wrote down a macro to stop the MOH RTP in order to let the
GW inband early media rtp heard by the caller
Regards
-Original Message-
From:
Anyone have a asterisk pager script that dials a list of pager numbers,
enters the phone number, and enters pound to send the page?
I am looking for a perl or php solution. Ideally I would like to set the
pager area-code and prefix on the fly as the company I work for owns a few
blocks of pagers
On Tue, Apr 14, 2009 at 10:28:42AM +1000, David Klaverstyn wrote:
Hi All,
I'm in the process of writing an install script and I would like to change
some settings for the install process but I don't want the user to go into
menuselect and make the changes manually.
Is there a way to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
No problem : ) I tried it again, here is the log output:
-- Executing [...@from-pbx:1] Set(Zap/31-1, EXITCONTEXT=callback) in
new stack
-- Executing [...@from-pbx:2] Dial(Zap/31-1, SIP/236||d) in new stack
-- Called 236
--
I have a function *1 that starts and stops recording in a call. I use a
function so I can use MixMonitor.
It works well, however I would like to make it a little more integrated for
my users. We have GXP 200 hardphones. So far I've been able to configure a
softkey using the speeddial option to
Your problem is that you put your line after the REJECT line. Your line is
never reached. Move it up one line, before the REJECT, and it will work as
expected.
// T
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas
On Tuesday 14 April 2009 13:04:02 jonas kellens wrote:
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
For one, these two rules are reversed. For two, you've failed to create holes
in your firewall for
This worked:
Dial(SIP/5198881...@telasip-gw4,,D(12345678))
however, the problem now exists in the disconnection. Asterisk tries to
bridge the call, play dtmf but never disconnects. What is there a specific
syntax to the D command that specifies a disconnect period.
I am thinking a better
Khaled W. Chehab wrote:
Thanks for answering Doug
I am using exten = _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
Change this to:
_X.,n,Dial(SIP/OPNS/${EXTEN}|300)
The m was causing the music on hold.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
John Todd wrote:
Sorry for the brief interruption. The user spamming the list with
vacation replies has been removed.
Bless you!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Man :)
I want the MOH play until Asterisk receives 180 ringing or 183 from the
termination GW.
Here I want to stop the MOH and let the user hear the early media RBT
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Khaled W. Chehab wrote:
Man :)
I want the MOH play until Asterisk receives 180 ringing or 183 from the
termination GW.
I don't think you'll be able to mix and match via the dial application.
You may have to try using AGI for this. That, I can't help you with.
Doug
--
Ben Franklin
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
To the best of my knowledge, the only way for you to control the
duration sent to the PSTN lines is for you to be directly
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