Jon Morgan wrote:
Hi All,
We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:
ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone
System
The company that looks after our internal phone system can no longer dial in
using their data
for the second time i'm asking in this forum,
somebody help me
my asterisk box have a problem with memory leak.
I'm scheduling to rstart the box to fix this problem
but any cleverer suggest to fix this? coz this issue
causing another problem to my AGI application...
thankyou before
Hi everyones,
I have a production server using asterisk 1.6.0.6
using php as an IVR and mssql server (on other machine)
My server attached a Sangoma A104 card (4xT1 card)
i have a problem with memory leak on that server
and causing a delay on IVR prompt. (Thats my assumption, memory leak
am looking for a good source of Monterrey DIDs
I cordially point you to asterisk-biz.
--
Sent from mobile device
On Apr 28, 2009, at 9:39 PM, Robert Augustyn robert.augus...@linqone.com
wrote:
Any pointers will be appreciated...
Sincerely,
Robert Augustyn
Any pointers will be appreciated...
Sincerely,
Robert Augustyn
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to themselves.
The logs also show asterisk bridging the incoming SIP connection with
the Phone, but when the agent hangs up, it shows the agent as hanging
up, but it's not closing the bridge.
Any ideas? Where should I begin in troubleshooting this? Thanks!
--
Regards,
Robert Broyles
Team Lead
Hi
Had an issue today where all channels connected to the telco when dialed
returned
WARNING[15366] chan_zap.c: Call specified, but not found?
in the logs,
when I removed the isdn cable and reinserted everything was fine
any ideas?
software Versions
asterisk-1.4.21.2
zaptel-1.4.12.1
So I'm guessing, I would disable any wrapup on the queue, and then in my
'h' extension pause the agent for a set period of time, with another
extension to unpause the agent if entered?
Or is there a better way to set the pause after the call is over?
Thanks!
--
Regards,
Robert Broyles
that?
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Robert Broyles
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All these questions are valid, though I want first to see that the DID does not
work then I will go and try to resolve it.
I do not have a specific issue at this moment.
Sincerely,
Robert Augustyn
p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
-Original Message-
From
Thanks
Sincerely,
Robert Augustyn
From: da...@debsinc.com [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Danny Nicholas
Sent: Friday, March 06, 2009 9:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to verify
and
not the usual 105.
Can I adjust this down anywhere?
Sincerely,
Robert Augustyn
p:519.997.3106 ext:802
m:519.817.2503
www.linqone.com
-Original Message-
From: supp...@ocg.ca [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of OCG Technical Support
Sent: Saturday, March
Great backports! :-)
This should really be merged into 1.4.
--
Regards,
Robert Broyles
Atis Lezdins wrote:
Well, i can share mine backports of queue_log into mysql for 1.4.
Basically you need two backports (that's why there are numerous
files). Realtime store/destroy allows Asterisk
Problem is, without going to 1.6, I can't get the queue log or events
posted to MySQL in realtime.
There used to be a patch out there for queue_log, but it doesn't work
with versions 1.4.21 or higher.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
I saw some
The patch I was referring to is:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging
It doesn't work for the current SVN 1.4
--
Regards,
Robert Broyles
Anthony Francis wrote:
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a
patch way back
Is there a built-in way of detecting fax tones, or a switch to T.38 on a
SIP channel? I need to periodically check some efax servers for
availability and figured the best way to ensure they are operational is
to check for tones. I've looked into Nvdetect but the company seems to
have gone out
appreciated.
Robert
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as I can - hoping the bug would 'resolve itself' -
but now I'm putting a bounty out on it.
http://bugs.digium.com/view.php?id=13691
--
Regards,
Robert Broyles
DISCLAIMER : This email and any files transmitted with it are property of
Poornam Info Vision Pvt. Ltd. This email contains
Yes, I've already posted notes on the bug.
I applied the patch, and when attempting to recompile, it fails.
--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145
Jason Parker wrote:
Robert Broyles wrote:
I saw some
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145
Jason
Actually, that's alcohol abuse. :-)
Regards,
Robert Broyles
Christian Victor wrote:
2009/3/4 Atis Lezdins a...@iq-labs.net mailto:a...@iq-labs.net
Bottle of Riga Black Balsam (45%), just have to figure out a way
to send it :)
Balsam??? By mail? Doesn't that count as liquid
Yea, that patch was tried, and doesn't resolve the issue either.
I will hold out on the bounty a little longer... maybe it will be
resolved soon. It's pretty important for us.
--
Regards,
Robert Broyles
Jason Parker wrote:
Tilghman Lesher wrote:
On Wednesday 04 March 2009 10:24:16
. In the meantime, I need to be able to
capture unanswered calls in the previously semi working method.
--
Regards,
Robert Broyles
Steve Murphy wrote:
On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where
FYI to everyone...
It was an issue on Vitelity's end on the gateway I was assigned to. They
switched me, and it's working fine now.
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
I turned on DTMF debugging. It looks like the extra digits coming in
are less than
in once.
When testing the dialplan internally, it accepts only the digits that I
key in.
Anyone else experienced this?
--
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Robert Broyles
DISCLAIMER : This email and any files transmitted with it are property of
Poornam Info Vision Pvt. Ltd. This email contains confidential
Btw, I'm using Asterisk SVN-branch-1.4-r178640
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has had this problem.
When calling from an outside line, and entering the digits during the
read() part of my
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has
Not at all.
In fact, I found that relaxdtmf=yes is now available for sip.conf as of
1.4 as well.
However, that didn't resolve the problem.
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking
Yea, I tried that too. I have it: dtmfmode=rfc2833
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert
.
Can someone else check this on their system, and see if this is a problem?
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria
Glad I could help!! :-D
Leonja Cerebro wrote:
To Robert Broyles,
Thank you very much, it is very helpful information.
Regards,
Leonid
2009/2/18 Robert Broyles rob...@poornam.com mailto:rob...@poornam.com
Hi,
You might want to check out this tutorial:
http://hostseries.com
Hi,
You might want to check out this tutorial:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
It's a good place to start.
--
Regards,
Robert Broyles
Leonja Cerebro wrote:
Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk
of Asterisk B (registered
Hi,
When I transfer a call to an extension, the person I call does not have any
idea when that transfer happened so it is a guessing game.
Is there a way to send a beep to the caller just before transferring the call?
Preferably by setting something in FreePbx?
Sincerely,
robert
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
Regards,
Robert Broyles
oumar ndiaye wrote:
Hi,
My queue used to work fine until I upgraded to 1.6. I am getting the
message:
No application 'AgentCallBackLogin' for extension (default, 31001, 1)
After
Why don't you use followme if you want to do that?
In fact, you can have followme, plus the local agents as mentioned in
the previous alternative that I mentioned.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Look at example #2, and adapt it for your needs.
--
Regards,
Robert Broyles
Philipp Kempgen
You guys... grr...
I'm still on 1.4 svn ... not ready to even think about 1.6 OR 1.8(when
it's released) for production right now. :-)
--
Regards,
Robert Broyles
Rob Hillis wrote:
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i safely ignore it?
Thanks
Robb
On 04/02/2009 00:24, Mark Michelson wrote:
Robert Boardman wrote:
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i
Hi,
Is that reliable? Any known issues? or recommended setups?
I am planning on adding the spa2002 devices and attaching the terminal to it.
Will this work well?
Sincerely,
Robert Augustyn
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I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG suggestion to peruse the FAQ first.
I agree with this 100%
I'm still pretty new to the mailing
to read,
they ask stupid questions because they're too lazy do to the footwork.
Robert Broyles wrote:
I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG
Jared Smith wrote:
On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote:
I'm still pretty new to the mailing lists myself. I don't consider
myself a novice Asterisk user, but one of my biggest 'complaints' is
the lack of a well documented FAQ or Manual for Asterisk
Hi,
I have several Aastra 57i phones connected to 1.4.22 version of asterisk and
when I call the queue these phones are part of I get few of these phones
ringing with a delay ... as much as 18 secs or not at all ...
What could be the problem?
Thanks
[Jan 22 03:06:19] VERBOSE[13842] logger.c:
Anyone know how soon this will be patched?
Or are we waiting on the new CDR structure/method?
Steve Murphy wrote:
On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
. 942's do, for an extra $20.
Regards,
Robert Broyles
Julian Lyndon-Smith wrote:
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
Julian
Are you planning on connecting your two Asterisk servers with SIP or IAX?
Check out this tutorial if using SIP:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
You should be able to adapt it to your needs. Good luck!
Paul wrote:
Can anyone tell me how I can completely move an
Hi,
I'm curious if anyone knows of any possibility to use video VOIP client
(like Ekiga or Linphone or...) under Linux that could be operated by
touchscreen friendly GUI (bigger buttons, large keypad, etc...) ?
I like Ekiga, but GUI is small and cannot be operated via touchscreen... But
maybe
Hi,
We have zabbix running and would love to be able to monitor our asterisk box
with it.
I believe that some sort of SNMP is build in 1.4+ correct?
Where do I find more info or a how to on what is supported and how to use it?
Thank you.
___
--
The Blackberry community has been begging for a SIP client for awhile.
Apparently there are some restrictions within the Blackberry OS. But
with the newer Blackberry models including wifi abilities, we should be
seeing something released soon... I hope! **Fingers Crossed**
Eric Moniz wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
waiting on the Queue.
Sample CDR data BEFORE the upgrade:
2008-10-30 12:46:47;\John\
If you don't want to use the AEL, but want an easy way to have agents
login and out, check out this quick tutorial:
http://hostseries.com/agentcallbacklogin-alternative/
Ariel Dorfman wrote:
i have done some research, but there says that i can use a function called
AgentCallbackLogin, but
Murf,
Thanks for the update. I look forward to seeing this one resolved. This
is just the issue that I'm facing. Looks like there's a patch already
posted on the bug. I'll wait for the bug to be closed or pushed to
release. Thanks again.
Robert
Steve Murphy wrote:
On Mon, 2009-01-05 at 12
With regards to storing queue_log data in mysql, it depends on the
Asterisk service your running.
1.6.x check out the following:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
1.2.x OR 1.4.x check out the following patch/solution:
this.
Thanks for your time.
--
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Robert Broyles
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Would this set the periodic-announce filename just for this call?
Thanks!
--
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Robert Broyles
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Hmm,
exten = s,1,Playback(/home/Sounds/greeting)
exten = s,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue2)
exten = s,n,Queue(CSR)
It's not working. It just plays the default announcement.
Same goes for:
exten = s,n,Set(GLOBAL(PERIODIC_ANNOUNCE)=/home/Sounds/queue2)
Btw, I'm using v1.4.22
That just plays back my announcement file before the caller enters the
queue.
It's still playing the default file once in the queue.
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Thanks. The thread points to an issue with the periodic-announce not
playing if the queue is set to ring, instead of musiconhold.
I have musiconhold with my queue.
My sample queue for testing purposes:
[CSR]
musiconhold = classic
retry = 1
strategy = ringall
joinempty = yes
Okay thank you.
This is something that I'm trying to avoid. I want to have one single
Queue, but based on the incoming DID, have a different periodic-announce
file played.
It would be awesome to be able to set all of the queue settings from the
dialplan, if so wished:
examples of what I mean:
It is not the source port being changed, it looks like the destination port is
being changed.
robert
-Original message-
From: pe...@networkoblivion.com
Sent: Mon 29-12-2008 09:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com;
Subject
it?
Sincerely,
Robert Augustyn
519-997-3106 ext:802
www.linqone.com
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Hi all
Been messing about with the single port cards for a number of years, but
never got good results, I was thinking of giving them another go over
Christmas and was wondering if anyone would share there recent
experience, as to which driver works best MISDN BRISTUFF etc with the
latest
Hi All
I Have an ISDN 30 circuit passing through an asterisk box to a legacy
pbx, all is working well but I have had a problem that modems do not
work, I thought of turning off echo cancelation but I cann t seem to
find the ial switch do do it, could someone point me in the right
direction to
On Wed, Dec 03, 2008 at 11:33:08AM -0500, Mike wrote:
Hi,
Been playing with Call parking, and I can`t help but wonder if I am doing
something incorrectly. The way I understand it (using default config in
features.conf), is I would transfer a call to extension 700, which would
park the
On Wed, Dec 03, 2008 at 10:56:48AM -0600, Danny Nicholas wrote:
The way I made this work was to set up 200 as my parker and I do transfer,
200, transfer.
exten = 200,1,Answer
exten = 200,n,Park(701)
That will work but only for one call park slot. If that's what you
want then great.
If you
On Tue, Dec 02, 2008 at 05:04:25PM +0530, Max Alex wrote:
Hi All,
I need to stop the transfer feature on particular sip user.
I am using linksys phone and it has set the forwarding enable to another
user.
I have three users 2101, 2102, 2103.
2102 is registered in linksys phone with
On Wed, Dec 03, 2008 at 11:13:49AM -0600, Danny Nicholas wrote:
This actually works for multiple slots. When 701 is occupied, * finds next
defined slow.
Does it announce what that slot is before doing it?
Rob
--
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL
On Wed, Dec 03, 2008 at 06:23:32PM +0100, BERGANZ François wrote:
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk.
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outbridge.png
But I have that
On Wed, Dec 03, 2008 at 03:27:28PM -0200, Sebastian wrote:
The thing is I have to wait checking a database value to change the state,
that duration is not long, but on any case I don't know when will be ready
to go on.
If I use MusicOnHold app the dialplan get stuck there and there's no
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible?
Thanks
Robb
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zap channel on one card to zap channel on another
Robb
Alex Balashov wrote:
You mean a zap-to-zap call hairpinned into the same adaptor, or another one?
Robert Boardman wrote:
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even
thanks
Found that but sometimes I need to detect dtmf ie when playing back a
recording
Robb
Philipp Kempgen wrote:
Robert Boardman schrieb:
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible
Hi,
I need to be able to unable and disable iax2 trunks from the cli?
Is there a way to do it if so how?
Sincerely,
Robert Augustyn
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I have found that the messages are not played as the hangup cause clears
down the channel and passed hangup to the other end
should I have progress() before the dial command?
Robb
Martin Smith wrote:
Hi Robert,
I'd recommend the following options for Dial() so that you corroborate
operator
Thanks for the reply
Could you be a little more specific?
Thanks
Robb
Martin Smith wrote:
Hi Robert,
I'd suggest tweaking the Dial() arguments so that you (1) allow early
audio, (2) don't force it play ringing to the calling party, and (3)
modify any other options to be as relaxed
Hi all,
Is it possible to have * playing an mp3 file in the way old tape system
worked?
Sincerely,
Robert Augustyn
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Hi All
Just been looking at stats for one of my sites, and I'm conserned about
the number of error cause codes being returned from the telco
for example
12000 calls processed
131 are cause code 31* normal. unspecified.*
139 are cause code 28 * invalid number format (address incomplete).*
-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Boardman
Sent: Thursday, November 20, 2008 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes
asterisk on a single
port.
eg all traffic to port 5070, re-route it to 5060
Robert
On Mon, Nov 17, 2008 at 3:54 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
We are planning to shift our sip users from one platform to another.
(basically from one asterisk server to another). the problem
Sriram wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via
span2)- legacy pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which
is in
the Asterisk and avaya talking to each other.
Thanks
Krishna
On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Krishna Sumanth Chava wrote:
Hi * Users,
I ran into a problem when I was trying to communicate an avaya
Krishna Sumanth Chava wrote:
Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP
Office talk to asterisk with SIP Trunking. I had successful calls from
asterisk to Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or
? It would have to include easy way for billing.
Thank you.
robert
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Is there anything like that?
Any experiences?
Sincerely,
Robert Augustyn
www.linqone.com http://www.linqone.com/
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Thank you,
How do I embed it into the web site though?
robert
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fred Posner
Sent: Monday, November 03, 2008 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Looking for a web
On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
Hi,
I have a strange problem with our Asterisk installation. Outgoing calls
are handled by the following lines:
exten = _0[2-9]X.,1,Set(CALLERID(num)=0403${CALLERID(num)})
exten = _0[2-9]X.,2,SET(CALLERID(num)=${IF($[
locking issue where multiple
things are trying to write to the same file at once?
Or perhaps you have a blank voicemail.conf.new that it can't erase, sitting
about somewhere?
Maybe try running asterisk under strace to see what happens when you try to
change a password.
Rob
--
Robert Lister
On Fri, Oct 31, 2008 at 11:39:31PM +, Robert Lister wrote:
On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
Hi,
I have a strange problem with our Asterisk installation. Outgoing calls
are handled by the following lines:
exten = _0[2-9]X.,1,Set(CALLERID(num
have run two software video phones and I had marginal results with it when
displayed on large LCDs, delay and blockines ware the problems I have run
into ...
Sincerely,
Robert Augustyn
http://www.linqone.com
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-users] Is anyone using * for 2 way
video conferencing?
On Wed, 29 Oct 2008, Robert Augustyn wrote:
Hi,
One of my clients, wants to use * box to run weekly
meetings between
remote locations over the internet.
What would be the best configuration for this? We are talking about
two
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP
it is mentioned MWI is now working.
In my testings with lastest 02123 firmware, MWI is blinking when
missed calls but
Olivier wrote:
2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults
in multiport sipura/Linksys you cannot access individual ports you have to
address them by the group
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort
Sent: Sunday, October 26, 2008 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
Sent: Thursday, October 23, 2008 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk video
Gordon Henderson wrote:
On Thu, 23
Hi
I'm trying to get the status of an extension that has DND set using the
service code, or trying to disable the service codes altogether so that
I can do them in the dialplan if needed
any advice wout be appriciated
Thanks
Robb
___
-- Bandwidth and
Thank you very much.
-Original Message-
From: [EMAIL PROTECTED]
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Julien Claassen
Sent: Thursday, October 09, 2008 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ringtones for the console
Julien,
I would love to see this solution so please upload the code.
Thank you very much.
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julien Claassen
Sent: Tuesday, October 07, 2008 4:06 AM
To: Asterisk Users Mailing List - Non
through a sound card for overhead paging
Robert Augustyn wrote:
Ok then how do you make that an night_bell as your extension?
We have an after hours IVR, press 1 if you know the party
that you're trying to reach, press 2 for Dial By Directory
and press 3 for the night bell
Julien,
Thank you, I need a file which when played sounds like a phone ringing ...
:)
robert
-Original Message-
From: [EMAIL PROTECTED]
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Julien Claassen
Sent: Tuesday, October 07, 2008 3:51 PM
To: Asterisk Users Mailing List - Non-Commercial
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