HI,
I want to allow user to press 0 to the voicemail if the user don't
want to wait in the queue. Below is what I set but it doesn't work.
Anyone can help? ango
file: features.conf
[applicationmap]
opervm = 0,self/both,Macro,opervm
file: extensions.conf
...
exten =
- Paul Hales pdha...@optusnet.com.au wrote:
Digium PSTN cards seem to work.
PaulH
OpenVox works well.
Best Regards,
--
SplatNIX IT Services :: Innovation through collaboration
___
-- Bandwidth and Colocation Provided by
Has anyone been able to do the following:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
2. Put Record on the main screen when a call is active. This would
eliminate having to press the 'more' softkey.
Thanks,
Matt
Hi All,
please provide some help.
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones Asterisk SIP Trunk Call Manager - CME
-
On Thu, May 21, 2009 at 10:04 AM, Matt Darnell mattdarn...@gmail.com wrote:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
Not possible AFAIK.
2. Put Record on the main screen when a call is active. This would
eliminate having to press
Hi,
I am not sure if I am doing something wrong, but I can't get MeetMe to
work with GSM codec (Asterisk 1.6.1 SVN r190371).
My config files below:
sip.conf:
[general]
context=common
canreinvite=no
bindport=5060
bindaddr=78.105.1.127
disallow=all
allow=alaw
allow=gsm
rtptimeout=600
Hi List,
I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that
jitterbuffer is only effective on the receiving channels.
My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
office.
Questions:
1. To enable jitter buffer on SIP channels it seems I have to
Hello!
Thanks...I set up a Samba mount, which works ok, except that Asterisk
confuses a wave file as a wav49 file. I think it may have something do with
the way Samba supports case sensitivity. Since Windows is not very
aggressive when it comes to being case sensitive, I am thinking that Samba
What exactly are tyou trying to achieve?
l.
2009/5/20 Kurian Thayil kurianmtha...@gmail.com
Hi All,
I am trying to implement ACD using Asterisk 1.2.18 and I've chosen
AgentCallbackLogin for login purpose. One AGI is written which will actually
get executed when agent dials '1001' (say) from
Hi guys,
I'm trying to write hangup causes from asterisk into the CDR record.
Using version 1.4.24.1 at the moment, but no joy so far.
Has anyone implemented this?
___
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El miércoles 20 de mayo del 2009 a las 21:19:18 -0300,
Daniel Bareiro escribió:
I load the modules wctdm and dahdi. But when I execute in Asterisk
CLI dahdi show channels, I get the following error message:
No such command 'dahdi show channels'
On Thu, May 21, 2009 at 06:38:27AM -0300, Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
El miércoles 20 de mayo del 2009 a las 21:19:18 -0300,
Daniel Bareiro escribió:
I load the modules wctdm and dahdi. But when I execute in Asterisk
CLI dahdi show channels, I
While I have not needed to do this for myself, I believe you can create this
functionality quite easily using Polycom's 'Enhanced Feature Keys' (EFK's).
IIRC, EFK's are available in the newest firmware revision 3.1.x and newer.
-Karl
- Original Message -
From: Matt Darnell
On Wed, May 20, 2009 at 6:58 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
We have been working with the ReceiveFax application for some weeks now in
order to receive faxes in T.38 and it works fairly well, but there are some
faxes that for some reason we are not able to receive
hi,
i'm searching solution for playing media(moh,prompts,voicemail,recordings
- wav format) from adobe flash player (web browser)
flash cannot play wav directly (imho)
i must convert files to any other format on-the-fly
- i cannot use mp3 because of royalties
- next option is swf (with
On May 21, 2009, at 5:59 AM, Karl Fife wrote:
While I have not needed to do this for myself, I believe you can
create this
functionality quite easily using Polycom's 'Enhanced Feature
Keys' (EFK's).
IIRC, EFK's are available in the newest firmware revision 3.1.x and
newer.
-Karl
- Ondrej Valousek webs...@s3group.cz escreveu:
Hi List,
I have a question regarding jitterbuffer in Asterisk 1.4.24. I see
that
jitterbuffer is only effective on the receiving channels.
My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
office.
Questions:
1.
Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Wednesday, May 20, 2009 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We did an opensource Java Applet that plays GSM files _very_ simply if
that helps.
I'd accidentally removed it from our website, but it is back now -
improved
with a javascript interface supporting load, play and pause actions.
http://www.westhawk.co.uk/software/playGSM/PlayGSM.html
The
Hi,
Did anyone tried static build of asterisk 1.6 version?
Installation fails when tried with static build.
warning: Using 'initgroups' in statically linked applications requires at
runtime the shared libraries from the glibc version used for linking
asterisk.o: In function `cli_prompt':
Manoj Panicker - FOES wrote:
Hi
Which is the best interface card to connect* PSTN* line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa. The Asterisk is in LAN and is reachable from all
Hi Vinicius.
/ 1. To enable jitter buffer on SIP channels it seems I have to enable
// and
// force it, right?
/
Not sure about the forcing part (don't know exacly how it works), but I
always set jbforce=yes to be sure.
Ok, thanks!
/ 2. If I enable and force jitter buffer, Asterisk would
You should try Answer before Dial on the Monitored call. Bridging can be
very unhappy.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline
Sent: Wednesday, May 20, 2009 8:52 PM
To:
What are you getting if you do a dahdi_cfg -vv?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Bareiro
Sent: Thursday, May 21, 2009 4:38 AM
To: asterisk-users@lists.digium.com
Subject: Re:
Sangoma as well.
Also ATA's such as what used to be called the Sipura 3000
Cisco 3810's with SIP IOS will give you up to 6 analog ports, and on the
really low end, if you can still find an X100 card, at least for US PSTN
lines.
John Novack
--[ UxBoD ]-- wrote:
- Paul Hales
Neeraj Chand escribió:
Hi guys,
I'm trying to write hangup causes from asterisk into the CDR record.
Using version 1.4.24.1 at the moment, but no joy so far.
Maybe something like this could do the job:
exten = h,1,Set(CDR(userfield)=${HANGUPCAUSE})
You can use the accountcode field as
Rilawich Ango wrote:
I want to allow user to press 0 to the voicemail if the user don't
want to wait in the queue. Below is what I set but it doesn't work.
Anyone can help? ango
None of that is necessary, but reading the documentation is. app_queue
already supports the caller using a DTMF
2009/5/18 Danny Nicholas da...@debsinc.com
I'd love to see this as well. After a few days of trying 1.6.1 (from
1.4.21) I dropped back to 1.4.25-rc1 and that is going pretty well.
Which issues did you get ?
I'm about to deply a 1.6.1 system it does seem to work ok in a pure SIP
environment.
My issues are all DAHDI/POTS related. Unfortunately, our present
communication depends on the POTS lines, so Im back to 1.4.25-rc1 as stated
earlier.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent:
FWIW, asterisk processes its' voicemail in FIFO (First in First out) fashion
using msg.* to store the messages, so it sends msg, then msg0001,
etc. You could write a shell or perl or C script to do a bubble sort on
all voicemails for a user.
Here is a listing of two voicemails
ll
sasirekha jaganathan wrote:
Did anyone tried static build of asterisk 1.6 version?
Installation fails when tried with static build.
warning: Using 'initgroups' in statically linked applications requires
at runtime the shared libraries from the glibc version used for linking
asterisk.o:
Danny Nicholas wrote:
You should try Answer before Dial on the Monitored call. Bridging can be
very unhappy.
Hi Danny.
Already done earlier in the dial plan, when the call first comes in but
before it gets routed to the part that I showed. Thanks for looking
though!
Barry
What you say...Martin (asteriskl...@callthem.info):
check if your dahdi card still takes interrupts at this point
dahdi_test should return some numbers close to 99%
Martin
Thanks, I'll try running d_t next time it happens. Are you suggesting
that either 1) the card is no longer generating
To clarify:
Inbound - Answer
Outbound - Answer (again)
Dial.
If I missed that, please disregard.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline
Sent: Thursday, May 21, 2009 9:05 AM
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Robin Rodriguez wrote:
still rather frustrating getting the EFK working. If needed I could
post that portion of sip.cfg to get you started.
Please do! Just having the example could be helpful for those of us
preparing to tackle this kind of
Hello Everyone,
I am receiving following error message will making Zaptel on Cent OS 5.2.
make[1]: Entering directory `/usr/src/zaptel-1.4.12.1'
echo You do not appear to have the sources for the 2.6.18-92.el5 kernel
installed.
You do not appear to have the sources for the 2.6.18-92.el5 kernel
Farooq Hussain farooqhussain...@gmail.com wrote:
Hello Everyone,
I am receiving following error message will making Zaptel on Cent OS 5.2.
make[1]: Entering directory `/usr/src/zaptel-1.4.12.1'
echo You do not appear to have the sources for the 2.6.18-92.el5 kernel
installed.
You
On opensuse 11.0, I had to install my kernel source using zipper (that's y
not I - email self corrected).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Farooq Hussain
Sent: Thursday, May 21, 2009 9:27 AM
To:
Kevin P. Fleming wrote:
Rilawich Ango wrote:
I want to allow user to press 0 to the voicemail if the user don't
want to wait in the queue. Below is what I set but it doesn't work.
Anyone can help? ango
None of that is necessary, but reading the documentation is. app_queue
already
Lyle Giese wrote:
Manoj Panicker - FOES wrote:
Hi
Which is the best interface card to connect* PSTN* line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa. The Asterisk is in LAN and
On Thu, May 21, 2009 at 09:32:02AM -0500, Tim Nelson wrote:
yum -y install kernel-devel kernel-headers
kernel-devel is the one you'll need . Sadly you'll get one of a newer
version . If booting to a newer kernel is not an issue, I suggest you
install the newer kernel (which is recommended
Hi David,
That's very similar to a setup I made. And I was troubleshooting
similar problems. Let me ask you a question:
Are you quite confident that the inbound faxes that fail are going to
succeed on an ordinary fax machine?
At least I'm sure of a couple of calling numbers that I know are
Not that I;m exactly a big fan of NFS but... why would you choose to
implement a filesystem that was designed to emulate Windows shares for your
UNIX-type environment? You have to kind of expect odd problems like this
when you choose to use things for other than their intended purpose. Samba
I
I'd be interested in this as well... I;m coming up to an upgrade cycle and
trying to decide if I should upgrade to the latest 1.4 or 1.6.1
When others that have commented on this say they have had problems with PSTN
connections, are you referring to T1 or POTS? I;m in a T1 scenerio, so if
Mark Michelson wrote:
Not to undermine Kevin's requests to read what is documented, I can say that
what you want actually will not be presented by running core show
application
Queue in the CLI.
As file would say... 'osnap'
In my haste to respond this morning while eating breakfast I
openSuse 11
Asterisk 1.4.23.1
Asterisk GUI 2.0 Latest SVN version
I set up some page groups using the Asterisk GUI and found that when I
hang up the paging phone it causes Asterisk to restart. So far no one
has been on the phone at this time so I am unsure if it hangs them up
but it
Hi, in MFC-R2 signaling there is a value Calling party category signal
(e.g., normal subscriber, high-priority subscriber, operator, coin-operated
telephone)
How can I get that information in my Asterisk??
Is there any similar value in SIP?
Thanks
___
Danny Nicholas wrote:
To clarify:
Inbound - Answer
Outbound - Answer (again)
Dial.
Hmmm... that seems like it would be from the department of redundancy
department but I gave it a try, both before and after the Monitor()
command with the same result... it fails.
Thanks!
Barry
Hi, in MFC-R2 signaling there is a value Calling party category signal
(e.g., normal subscriber, high-priority subscriber, operator, coin-operated
telephone)
How can I get that information in my Asterisk??
That depends on which MFC-R2 solution are you using for Asterisk. The
2 most known are
Hi,
We are looking for the best outbound rate to US48 termination, in any
quality lines (for call centers resale).
If you offer volume discounts, please quote for:
- up to 1 million min/month
- over 1 million min/month
We currently use in total up to 3 million min/month and are planning on
It is already a macro, not sure about passing an array of numbers.
Alex Samad wrote:
On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote:
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
[snip]
I left the busy after dial
Couldn't he also just do a sip set debug to view the responses coming
back?
Jeff LaCoursiere wrote:
On Wed, 20 May 2009, John Regal wrote:
Thanks for the reply and apologize for the double post. My original post
landed in another thread and thought it may have been missed...
I
Hi Lenz,
Here is my objective. Planning to implement queue using Asterisk 1.2.18.
So created a queue named testqueue in queues.conf and then created
agents for this. Now, our actual requirement is to collect the callerid
from the inbound call and search in the DB (customer list) and display
the
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.25. Asterisk 1.4.25 is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves several crash issues, DTMF related issues, and CDR related
issues.
For a summary
The Asterisk Development Team is pleased to announce the release of
Asterisk-Addons 1.6.0.2. Asterisk-Addons 1.6.0.2 is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves a potential crash issue in the ooh323 channel driver, and
I have one asterisk server where most of the calls that go through AMD get
stuck in it, even if the analysis time of 3 seconds has already ended.
It doesn't move to the next priority (which is checking the AMD_STATUS).
Executing 'show channels' shows that the calls are stuck in the AMD app.
I
Gavin Henry wrote:
Is there any document on the reasons for the 1.6.0 and 1.6.1 branches?
I remember reading something but can't find it again.
Was it stability versus new features?
I'm currently playing with 1.6.1
The difference is in regards to new features.
Instead of waiting 1-2+
Hi guys,
I'm running Asterisk 1.2.13 on a Debian Linux system (that was just the
version that was packaged for it). I've been using monitor() to record
calls, with fairly satisfactory results - at least until the last few
months.
I've been recording VoIP calls, and using monitor() with no
On 21 May 2009, at 22:02, Nikhil Nair wrote:
I'm pretty stumped here; I can only imagine that, for some reason,
not all
silence is being recorded in the sound files.
Silence suppression might be enabled somewhere? Asterisk doesn't like
that generally, so might screw recordings too..
Steve
Here's a little story on all the cheap guys trying to get the best rate on
any route out there ( lcr and others).
Anyone have 0.01 to Mexico billed 1/1 ?
When customers call us to ask if we sell Cuba termination for 50c/min, I
sometimes joke and tell them sure, I'll
On 05/21/2009 09:11 AM, Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Robin Rodriguez wrote:
still rather frustrating getting the EFK working. If needed I could
post that portion of sip.cfg to get you started.
Please do! Just having the example could be helpful for
Hi Gang,
I've got 1.4.25-rc1 up and running pretty good now. The only
difficulties I have left to conquer are:
1. FFA won't receive a fax from a DELL A990 (failed on 6 out of 7
attempts with wrong protocol or timeout).
2. DAHDI dial makes a clicking/static sound on line
(monitor legs are out of sync)
On Thu, 21 May 2009, Nikhil Nair wrote:
I'm running Asterisk 1.2.13...
A more modern version wouldn't hurt.
I've been using monitor() to record calls, with fairly satisfactory
results - at least until the last few months.
If you don't need the legs separate,
Hi Nikhil,
Several of these out of sync issues have been resolves in many recent
versions
of Asterisk. I'm not sure if many of the out of sync issues were reported
against 1.2 when it was receiving bug updates, so you may need to move to
Asterisk 1.4 in order to get these updates.
Y,
Because the scheduler usually uses the dahdi timer to run ... and if
the timer has stopped
then the frames/events will not go out and finally you get the scheduler full
Martin
On Thu, May 21, 2009 at 9:14 AM, Hose hose+aster...@bluemaggottowel.com wrote:
What you say...Martin
it should work just fine; do you have the GSM codec compiled/loaded
core show modules like codec_gsm ... ?
OR that particular version has a BUG...
Martin
On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote:
Hi,
I am not sure if I am doing something wrong, but I
Thanks all. I figure out to exit the queue by setting context in queue.conf.
On Thu, May 21, 2009 at 11:20 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Mark Michelson wrote:
Not to undermine Kevin's requests to read what is documented, I can say that
what you want actually will not be
hello, i made a experimental patch for libpri to have NT/PTMP mode,
answers please on asterisk-dev at:
http://lists.digium.com/pipermail/asterisk-dev/2009-May/038455.html
Kristijan
2009/5/14 Kristijan Vrban vrban.l...@googlemail.com
good news, i just made my isdn device ring! ok, after it
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