Pete Kay wrote:
Hi all,
I want to allow different users to have their own unique MOH. Is
there anyway to do it? Asterisk does not have a realtime MOH feature
but I am wondering if there is anyway to get around it?
Store each user's MOH settings in the database, check those settings
Hi all,
I am hoping someone can help me out on this. I want to be able to interrupt
MOH every X seconds after the DIAL command is executed. The interrupt
greeting is something like please wait while we transfer your call. How
can I do that? Within the DIAL options, I can't see any announce
On 31/03/2008, Greg Oliver [EMAIL PROTECTED] wrote:
For a 7965, you might try loadinformation to be 335.. I have had to
match up CCM tk.prod values to match on newer phones in the past to be
what cisco uses in their internal database before I could get them to
work. Although, leaving those
Hi all,
I want to allow different users to have their own unique MOH. Is there
anyway to do it? Asterisk does not have a realtime MOH feature but I am
wondering if there is anyway to get around it?
Thank you for your suggestion.
Thanks,
Pete
___
--
On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
I am hoping someone can help me out on this. I want to be able to
interrupt MOH every X seconds after the DIAL command is executed. The
interrupt greeting is something like please wait while we transfer your
call. How can I do that? Within
Hi
I want to know how we can use Explicity call transfer by using of Asterisk
with extensions.conf
Regds
Santosh
asterisk-users@lists.digium.com
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Tilghman Lesher wrote:
On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
I am hoping someone can help me out on this. I want to be able to
interrupt MOH every X seconds after the DIAL command is executed. The
interrupt greeting is something like please wait while we transfer your
call. How
Can I assume after exten=2,1,Playback(thanksfortakingthecall) there's more
logic, or does asterisk handle the connection between both parties at that
point?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, March 31, 2008 9:34
Hello !
I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8.
I did m-a a-i zaptel to upgrade
what i still get is:
cat /sys/module/zaptel/version
1.4.7.1
It doesn't seem that it wants to load the new driver, any ideas ?
Kind regards,
Jan
--
Jan Prunk janprunk AT
hi all
I am using asterisk 1.4.15 I have a problem in conference .The conference
room is not getting hangup after disconnecting tha call also.It shows
disconnection on the x lite phone but when i run show channels on asterisk
cli it showr meetme room is reserved.
thanks
Rahul
On Tue, Apr 1, 2008 at 10:09 AM, Jan Prunk [EMAIL PROTECTED] wrote:
Hello !
I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8.
I did m-a a-i zaptel to upgrade
what i still get is:
cat /sys/module/zaptel/version
1.4.7.1
It doesn't seem that it wants to
On Tue, Apr 01, 2008 at 04:09:48PM +0200, Jan Prunk wrote:
Hello !
I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8.
I did m-a a-i zaptel to upgrade
[ which is the Debian rough equivalent of 'make; make modules-install' ]
what i still get is:
cat
Hello,
I am setting-up a system to place outgoing calls for a certain
number of minutes (as allowed per the customer's account). I would
like to break into the long distance channel to announce 1 minute
left, etc. What asterisk command can I use to do this?
Thank you in advance for
Jan Prunk,, wrote:
Hello !
I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8.
I did m-a a-i zaptel to upgrade
what i still get is:
cat /sys/module/zaptel/version
1.4.7.1
It doesn't seem that it wants to load the new driver, any ideas ?
Kind regards,
Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
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Rizwan Hisham
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On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote:
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n)
The 'n' flag tells chan_local not to optimize itself out of the call
path. Without the 'n' flag, chan_local will
Rizwan Hisham wrote:
Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
Yes, and you will too when you read localchannel.txt in your Asterisk
source code docs directory.
--
Consulting for
Rule of thumb: you first try without the /n; if the new behaviour is
different from expected, add the /n
:)
Just my $0.02
l.
On Tue, 01 Apr 2008 17:33:05 +0200, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote:
Does anyone know the purpose of /n
On Tue, Apr 1, 2008 at 7:29 AM, Chaya Zipora Rosenberg
[EMAIL PROTECTED] wrote:
Hello,
I am setting-up a system to place outgoing calls for a certain
number of minutes (as allowed per the customer's account). I would
like to break into the long distance channel to announce 1 minute
On Tuesday 01 April 2008 09:29:21 Chaya Zipora Rosenberg wrote:
I am setting-up a system to place outgoing calls for a certain
number of minutes (as allowed per the customer's account). I would
like to break into the long distance channel to announce 1 minute
left, etc. What asterisk
Are you asking if both sides of the conversation will hear 1 minute
left - then no, I'd rather if just my customer hears the message.
Thanks
Henry Cobb wrote:
On Tue, Apr 1, 2008 at 7:29 AM, Chaya Zipora Rosenberg
[EMAIL PROTECTED] wrote:
Hello,
I am setting-up a system to place
That might not be where your voicemail files live, but if that IS, maybe
asterisk currently goes 'the person at extension XYZ is [on the
phone,unavailable] rather than playing greetings out of there. Do you
have an Old folder in there? an INBOX folder? Then it's probably the
right spot. I'd
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
I call into the dialplan and try to play demo-congrats and I hear nothing.
Firewall is disabled.
Everything is on the 192.168.1.X network for this simple configuration.
The tftp server is giving the polycom phone the config files.
Any
I am using asterisk 1.4.18 with a polycom phone.
sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
I call into the dialplan and try to play
On Wed, Mar 12, 2008 at 1:57 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 16:27, Wed 12 Mar 08, Steve Totaro wrote:
Try Callweaver.
Thanks,
Steve Totaro
or app_conference for asterisk.
That does the trick for me on OpenBSD where you dont have
ztdummy.
Is app_conference
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
/ I call into the dialplan and try to play demo-congrats and I hear nothing.
//
// Firewall is disabled.
// Everything is on the 192.168.1.X network for this simple configuration.
// The tftp server is giving the polycom phone the
At 01:13 PM 4/1/2008, you wrote:
Is app_conference stable now?
I've never made it through a thousand calls without a crash. (With a
busy call center this doesn't take all that long.)
-HJC
I have deployed a MEETME conference bridge based on a FARM of
asterisks with 6,000 conference ports
http://svn.digium.com/view/zaptel?view=revrevision=4121
Pure VoIP is the wave of the future!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
Disclaimer: We all know what day it is today... :-)
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My own fear from hell is having to call tech support!
My personal experience is that I get better support, faster, from
mailing lists than from a paid for tech support. If it's a common
failure, Google will get you there faster than you can dial 1-800. If
it's more unusual, the paid for
can you give an example?
Thanks
Tilghman Lesher wrote:
On Tuesday 01 April 2008 09:29:21 Chaya Zipora Rosenberg wrote:
I am setting-up a system to place outgoing calls for a certain
number of minutes (as allowed per the customer's account). I would
like to break into the long distance
Olivier wrote:
And what about SIP support ?
Should it be removed in 1.6 or 1.8 ?
SIP might be important to retain for legacy/historical reasons, but I do
suggest ditching that long, long-deprecated INVITE method and dialog.
Nothing but sheer madness issues forth from it. I have never seen
Hi,
Is it possible to both use Digium B410P and bristuff'ed 1.4 Asterisk, now ?
I've heard BRI support in Asterisk is about to change with 1.6 but I'm not
sure I understood what the plan is.
If someone has a clue, l would delighted to learn about it.
Cheers
And what about SIP support ?
Should it be removed in 1.6 or 1.8 ?
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Olivier wrote:
And what about SIP support ?
Should it be removed in 1.6 or 1.8 ?
Where have you been? SIP's been deprecated since 1.2.
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To
Tim Nelson wrote:
http://svn.digium.com/view/zaptel?view=revrevision=4121
Pure VoIP is the wave of the future!
Yeah, April fools...
-Ron
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To
You may be able to achieve the desired result using queues rather than
Dial statements.
Overkill perhaps, but it's the only way I can think to implement it at
the moment.
John Millican wrote:
Tilghman Lesher wrote:
On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
I am hoping
Alejandro Cabrera Obed wrote:
Can Asterisk control the RTP open ports the voip clients use ??? Or the
RTP open ports depend on the voip clients ???
It depends on the VoIP clients.
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Steve Davies wrote:
Could you point me at some reference material for how this differs
from KS, and what compatibility issues this might cause with other
equipment? Has anyone tried this in the UK? Would BT even understand
the request for ground-start signalling?
KS (Kewl Start) simply
On Tue, 1 Apr 2008, Mike Trest - Personal wrote:
At 01:13 PM 4/1/2008, you wrote:
Is app_conference stable now?
I've never made it through a thousand calls without a crash. (With a
busy call center this doesn't take all that long.)
I have deployed a MEETME conference bridge based on a
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian
Etch system. On the recorded voice mail messages, the volume is really low
when retrieving them with my cell phone. I tried with multiple cell phones
with the volume level high and still, the same
Do you see the same volume issues dialing in with a 'normal' POTS phone?
Are these the standard recordings that come with Asterisk or some custom
recordings?
...brig
Brig C. McCoy
ThyssenKrupp Access Corp
Network Administrator
Grandview, MO 64030
816-767-5577
Hi,
Olivier a écrit :
Hi,
Is it possible to both use Digium B410P and bristuff'ed 1.4 Asterisk, now ?
I've heard BRI support in Asterisk is about to change with 1.6 but I'm
not sure I understood what the plan is.
If someone has a clue, l would delighted to learn about it.
You should
John Meksavan wrote:
level high and still, the same problem. I tried to increase the rxgain
to 12.2 in the zapata.conf file and it had no affect
You'd want to fiddle with the txgain(Transmit)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
When people leave me messages, both on the cellphone and POTS phones, on the
recorded Asterisk voicemail message volume is really low. I could barely hear
my voicemail messages, when retrieving them, either cellphone or POTS line.
The voice mail prompts and sound recordings are fine, but the
How does the g729 encoder/decoder count in regards to the total number
of licenses and how does it count an encoder/decoder? I looked on the
wiki and don't really see anything that explains it. In other words,
how do the calls below count (assume no reinvite)?
g729 phone calls into voicemail
Hello! I'm having a bit of an issue with one of my installations that I cannot
figure out. For some reason, when two people are in a call (both local to the *
box, same subnet, pure SIP), the call will randomly be placed on hold and
provide MOH to the other party. We're using Polycom IP430
Here's a nice discussion of the issue:
http://www.voipuser.org/forum_topic_3857.html
...brig
Brig C. McCoy
ThyssenKrupp Access Corp
Network Administrator
Grandview, MO 64030
816-767-5577
From: [EMAIL PROTECTED]
[mailto:[EMAIL
When g729 phone calls another g729 phone and you are not recording
calls or doing meetme with them then license is not required ... g729
phone calling g711 will require a license to transcode the g729 side (
no license for g711 side of call ) . In short anytime u need to
convert g729 into some
Hello
Newbie question here: I have a box running Ubuntu Linux 7.10 gutsy
gibbon, and have a single Digium TDM410E card, with 1 FXO module
fitted and connected to my landline. I have it answering the landline,
directing to SIP phones, diverting to voicemail etc - and it works
great. What
Steve Edwards wrote:
4) How do YOU find an Iaxy on your network?
I was most easily able to find them by watching my DHCP server logs.
You're right about the -b switch to ping, that's required.
Moj
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Enable dtmf on the logger.conf and see if you get some # or ** or whatever
key you have configured at features.conf for transfer, maybe you could see
something into the logs. I get something similar with some Linksys PAP2.
Adria Vidal
On Tue, Apr 1, 2008 at 10:15 PM, Tim Nelson [EMAIL PROTECTED]
Hello everybody, i'm from Mexico, at the time i´m working on a production
server with asterisk 1.2.25 + spandsp-0.0.4 +
libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I
installed this version of astunicall that i downloaded from
http://www.moythreads.com/astunicall/
Hi folks,
I'm trying to install asterisk with radius cdr support.
I got freeradius up and running, so following radius instructions
inside asterisk source package, I've installed radiusclient-ng and
relative headers.
But when I start configure(asterisk 1.4.18.1) I got:
checking for rc_read_config
Paul Whitby wrote:
Hello
Newbie question here: I have a box running Ubuntu Linux 7.10 gutsy
gibbon, and have a single Digium TDM410E card, with 1 FXO module
fitted and connected to my landline. I have it answering the landline,
directing to SIP phones, diverting to voicemail etc - and
We currently have an application used by the trucking industry to find
freight to move. Now, the trucker does a search around Boston (for example)
and gets 100 loads returned. They start at the first and call the company
who has the freight, the company may say, sorry, someone just booked that
I am trying to use call files that dial and play a wave file
on 3 asterisk boxes console dsp.
This is working.
The 3 boxes are noticeably out of sync. From using 3 different call files
(time to process) I'm sure is the time delay.
Is there a way to get these audios more in sync?
Jerry
I don't entirely remember - I was writing this code from memory.
Have you done any testing?
PaulH
On Tue, 2008-04-01 at 08:47 -0500, Jeremy Mann wrote:
Can I assume after exten=2,1,Playback(thanksfortakingthecall) there's more
logic, or does asterisk handle the connection between both
Doug Lytle wrote:
John Meksavan wrote:
level high and still, the same problem. I tried to increase the rxgain
to 12.2 in the zapata.conf file and it had no affect
You'd want to fiddle with the txgain(Transmit)
Doug
He might actually want to deal with rxgain, because it
Sync the clocks on your asterisk boxen using NTP or whatever, and then
'touch' the call files into the future so each asterisk waits before
processing it...? Might get them closer.
Another option is get all three boxes into the same meetme room, waiting
a few seconds for them to be ready if
TODAY I have managed to hack the iPhone and install Asterisk on it.
Detailed instructions to follow.
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Hi,
I have just gotten my first Asterisk box up and running, and it is running
great. I am working on this project with the plans of possibly implementing
it in a business environment. The problem I am coming up against is that the
business I am planning on implementing this setup in is using
I think that's still a better idea than using a dump the caller into
meetme hack and is actually what I was going to suggest.
If you want something simpler than a queue then inject the sounds into
the moh already.
On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote:
You may be
That makes sense. A call from 729 to 711 would require one encoder and
one decoder, right?
So if you have 10 licenses, is it 10 total encoders+decoders, or 10
calls (some may require encode, or decode, or both)? Because I had 10
licenses, but my encoders+decoders was more than 10 and calls
Hi
I've a Sangoma A200D with 2FXO and 2FXS. When using it with only
the FXO module, it's all good. But when I put in the FXS module and
connect the power, logs tells me not enough power.
Mar 31 14:11:54 phone kernel: [ 4761.246931] wanpipe1: Module 1:
Failed to powerup within 600 ms
Short answer: No.
However you can use ATA devices like PAP-2's to connect to your existing
modem bank and as long as your latency is constant, get decent results.
I myself have gotten 33.6k on a regular basis with such a setup and have
called the world using cheap SIP/IAX providers with
Hi friends,
Is there anyway to have Asterisk to wait for 1 second before sending a DTMF
using the D() option?
Thanks for your suggestion.
Pete
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If you are asking about dial command on analog lines, here is what i do :
exten = _NXX,1,Dial(ZAP/g1/ww${EXTEN})
that should give you 2 seconds before actually start dialing, its good way
to wait for analog lines to stabilize first before dialing.
On Tue, Apr 1, 2008 at 9:49 PM, Pete Kay
Hi
Thank you for your reply. I am looking for sending the exension number such
as 100 immediately after the called party picks up.
I am hoping to send the digits after the call is picked up by the called
party instead of before. Is this something that can be done?
I can't see anything in the
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