Philipp Kempgen wrote:
> Mojo with Horan & Company, LLC schrieb:
>> Eric Wieling wrote:
>>> The word "Dialing..." and "Calling..."
>>>
>>> As in "Dialing 911, please wait..."
>>>
>>> and as in "Calling 911, please wait..."
>>>
>> oooh boy wouldn't I be frustrated if I heard that instead of a r
wrote:
> Jay,
>
> What error?
>
>
> Jay Moore wrote:
>> How do I include a file (not a context) in AEL? #include "filename"
>> returns an error.
>>
>> Thanks,
>> Jay
>>
>> ___
>>
How do I include a file (not a context) in AEL? #include "filename"
returns an error.
Thanks,
Jay
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Steve Totaro wrote:
> Jay Moore wrote:
>> Greetings, List.
>>
>> I'm having a problem where my recorded calls are skipping every 4-5
>> seconds are so. I can hear the caller (or callee) just fine and then a
>> second or so of silence followed by the pe
Doug Lytle wrote:
> Jay Moore wrote:
>> Hi list.
>>
>> I'm new to IVRs and trying to set up one that toggles an auto-forward
>> flag on or off for specific accounts.
>>
>>
>
> Why don't you post what you've currently written and we&#
Hi list.
I'm new to IVRs and trying to set up one that toggles an auto-forward
flag on or off for specific accounts.
I'd like to have my users dial an extension and then be prompted to
enter the account number. (done)
Next I'd like it to jump to the appropriate line in the dial plan that
cor
Once again, my initial message goes ignored or unreceived. Let's try
this one more time, shall we? :)
Jay Moore wrote:
> Greetings, List.
>
> I'm having a problem where my recorded calls are skipping every 4-5
> seconds are so. I can hear the caller (or callee) just fin
Greetings, List.
I'm having a problem where my recorded calls are skipping every 4-5
seconds are so. I can hear the caller (or callee) just fine and then a
second or so of silence followed by the person talking again. I'm
saving my calls as .gsm files and it's worked fine for the past 11
mon
rom some custom AGI programming, although
> it won't be very complicated. But I don't know that there's a business
> rules engine that does exactly what you're looking for right out of the
> box.
>
> On Thu, 6 Dec 2007, Jay Moore wrote:
>
>> Not sure if my
Not sure if my original message made it through. Going to try this
again. :)
---
Greetings, List.
I would like to implement a procedure in my call center but am not sure
the best way to implement it. I'm hoping I can describe it here and
that I'll receive some feedback and/or suggestions
Greetings, List.
I would like to implement a procedure in my call center but am not sure
the best way to implement it. I'm hoping I can describe it here and
that I'll receive some feedback and/or suggestions on how to proceed.
Here's my situation:
My call center fields calls regarding interne
Steve Totaro wrote:
> Jay Moore wrote:
>>> This is a FREE SERVICE provided by Bochter Services and it is not going
>>> away any time soon.
>>>
>> Except now, right, pal?
>>
>> Your site is down, you see. A shame, that.
>>
>>
>
> This is a FREE SERVICE provided by Bochter Services and it is not going
> away any time soon.
Except now, right, pal?
Your site is down, you see. A shame, that.
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Stephen Bosch wrote:
> I think it is trying to play mp3 files.
Yes, this appears to be the case, I presume(see below). I am not sure
why as I've always had .gsm files for playback.
mode => quietmp3
>
> Is this mode appropriate when you're using gsm audio files?
I could not find any info
;t work. I have to actually do a "stop now"
> and then "asterisk" to get it to work again. * restarts and MOH works
> fine. No clue why, but I have seen it on multiple versions of *.
>
> Jay Moore wrote:
>> Folks, I have somewhat of a serious issue here. M
Folks, I have somewhat of a serious issue here. My music on hold
mysteriously stopped working. I have made no changes to my Asterisk box
in the past month and up until an hour ago, MoH was working fine (as far
as I know).
CLI:
-- Started music on hold, class 'default', on channel 'IAX2/lobby-
Greetings, List.
With my current setup, I record all incoming calls to my queues. My
problem is that once a call is transferred out of a queue, recording
stops. How can I make it so recording continues even after a call is
transferred?
If you need me to post any dialplan or conf logic, pleas
> Hopefully that helps clarify things!
It does immensely. Thanks a ton!
Jay
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Jared Smith wrote:
> On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote:
>> My boss would like some statistics on how many calls are answered out of
>> specific queues during a given time period, and I'm not sure how exactly
>> to obtain those stats.
>
> It
Greetings, list!
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats. Here's how our queue system works.
1) Call comes in and enters our 'ring' queue where the phones ring for
1
Greetings, List.
I have my Asterisk box setup with 8 Centrex lines that were "left over"
from our old PBX system. My boss is asking me to set up Asterisk so
that he can flash hook and make an outgoing call on the same line to
have a 3 way call.
This is what he wants to do:
1) Incoming call o
dialplan.
>>
>> Are you sure there is no leak in your dialplan, because asterisk cant
>> transfer your caller to an extension it cant find. There must be leak,
>> check if you are using any wrong extension patterns like _XXX. or
>> something like that.
>>
>&g
The way I have my dialplan set up, the callers shouldn't be able to make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel => 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include cal
Greetings, folks.
I'm having a problem with blind transfers. It seems that, despite not
having the T flag set, callers are able to use the blind transfer option.
Scenario is this:
- Asterisk 1.2.14
- Caller calls into our call center on one of our many phone numbers.
- Call gets placed into qu
Thanks,
Jay
Doug Lytle wrote:
Jay Moore wrote:
Hi folks!
I did notice that sox has a -v flag for adjusting volume, but danged
if I can find documentation online that'll tell me what parameter to
pass.
Doing a 'man sox' does wonders:
-v volume Change amplitude (float
Hi folks!
I'm having a problem where my music on hold is just blaring to my
callers. I've tried several different formats (converting using mpg123
and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail.
Every file plays way too loud.
I did notice that sox has a -v flag for
Ok, I'll bite. This is the 4th message like this I've gotten today. I
don't speak French but it looks like an autoresponder. If so, why is it
replying back to the list, why not on every message sent, and why is it
incrementing the issue number?
Or am I missing something?
Jay
[EMAIL PROTEC
Hi folks. I'm having a problem with a SIP-enabled device that doesn't
seem to want to register after it reboots. If I program the device
manually via its interface, it registers just fine. However, once I
reboot it, it fails to register with Asterisk, despite all the proper
information being
I do it by calling my own extension. If it's me calling me, it passes
me direct to VoicemailMain. If it's someone else calling me, it rings
my phone as normal:
exten => 202,1,GotoIf($["${CALLERIDNUM}" = "202"] ? 5 : 2)
exten => 202,2,Dial(SIP/jay,10,tT)
exten => 202,3,VoiceMail([EMAIL PROTECT
You'll have to check the horse-wiki and pray it never goes down.
Alternatively, you could get a Cisco horse. While it may cost more, at
least you'll have a number you can call for tech support should your
horse throw a shoe.
The downside being, of course, if you want to modify your horse (e.
vn the you can edit the features.c file and add the lines
mentioned in the notes back to the file, then make and make install.
On 1/25/07, Jay Moore <[EMAIL PROTECTED]> wrote:
Bruce,
I'm running 1.2.14. I am not willing to switch to 1.4 yet due to the
stability issue. From what I
Jim,
I have 2 TDM400s in my * box (as well as a T1 card). I use all 8 ports,
and aside from some minor echoing during peak periods, it's running
smooth as ice.
Jay
Jim Freeze wrote:
Hello
I have a working * server with a TDM card and 4 FXO ports.
We have 4 lines now and need to add 2 more
s?
Thanks,
Jay
Bruce Reeves wrote:
Jay,
there is a bug in Mantis regarding this, a change was made to allow native
bridging of parked calls. The change has been reverted in a more recent SVN
version of 1.2. See http://bugs.digium.com/view.php?id=8804
On 1/25/07, Jay Moore <[EMAIL PROTECTED]>
Here's how it's currently working:
1) Call comes in
2) Operator parks call (700)
3) Operator picks up call on another phone (701)
4) Operator tries to transfer to a different phone (we use #0) but the
transfer doesn't work.
We can transfer initial callers all we want and it works fine. Once a
IMO, the 480i, by a LONG shot.
The 480i is easier to use, looks nicer, has better audio quality, easier
to read, and has a great speakerphone. The web-interface is also
leagues better than the tripe the Polycom phones have.
The only issue I have with the 480i, is that it's a little unintuiti
Yeah. 1.2.14.
I heard bad things about 1.4 not being all that stable. I'm hesitant to
move to it.
Jay
Julian Lyndon-Smith wrote:
1.2 series ?
I think that 1.4 has that fixed. At least, that's what my team leaders
are telling me ;)
Julian.
Jay Moore wrote:
I have a proble
I have a problem where my recorded queue calls stop recording once the
call is transferred to a different extension. Is there some additional
parameter I need to set so recording continues? Is it even possible to
do this?
Thanks,
Jay
___
--Bandwidt
Having a problem here that I can't seem to find a fix for.
PSTN call comes in, operator answers, transfers call to a phone behind
an IAXy.
Caller hears no sound after being transferred.
IAXy can hear caller, but not vice versa.
Client reads:
NOTICE[11342]: channel.c:1950 ast_read: Dropping i
I have 8 Zap channels, 25-32, all of which have their own line.
My zapata.conf file looks similar to:
group=1
context=context_1
signalling=fxs_ks
channel => 25
group=2
context=context_2
signalling=fxs_ks
channel => 26
and so forth
Greetings,
I am using MixMonitor to record my outgoing calls. It seems that
MixMonitor will not write to a directory if it doesn't exist (ie - it
doesn't create a new directory if needed).
I have checked to ensure permissions are properly set, and if I manually
create the directory, MixMoni
;
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no "."
- Upgrade sox to the latest version which fixes the funny behaviour
Cheers,
--
Ex Vito
On 12/28/06, Jay Moore <[EMAIL PROTECTED]&
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the call
Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing Li
Greetings folks.
I seem to be having a problem where calls made from an IAX device (three
single-line phones attached to IAXys) do not play the ring tone when
calling out. There's a dial tone when I pick up the phone, and the call
goes through just fine, it just doesn't ring. All my SIP phon
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones for
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (whic
v 15 12:54
swift_f87b365372c500c76e497087ac7e470a
On 11/15/06, Jay Moore <[EMAIL PROTECTED]> wrote:
Are you including the file extension?
Jay
Tom Vile wrote:
> I am trying to get the example input.php working from PHPAGI but it
will
> not
> playback the letters that I put
Are you including the file extension?
Jay
Tom Vile wrote:
I am trying to get the example input.php working from PHPAGI but it will
not
playback the letters that I put in because of this error:
Nov 15 14:25:22 WARNING[18678] file.c: File
/tmp/swift_f87b365372c500c76e497087ac7e470a does not exi
Actually, while I was waiting for an answer, I figured out my problem.
If I have any further questions, however, I'll be sure to post. Thanks!
Jay
Dovid B wrote:
Post away.
- Original Message - From: "Jay Moore" <[EMAIL PROTECTED]>
To:
Sent: Thursday, Nove
Before I make any serious gaffes, is this an acceptable place to post
PHPAGI questions as well? I can't seem to find a dedicated mailing list
for it. If not, any suggestions?
Thanks,
Jay
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as
Peter Bowyer wrote:
On 04/09/06, Jay Moore <[EMAIL PROTECTED]> wrote:
Marco: Ah I see. There's a [general] context. I'm pretty new to this
Asterisk stuff and I didn't realize there was a general context that you
could do things like global includes. Thanks, I'
o be an ass about it, pal. Not all of us are as adept
at this as you are.
Jay
Marco Mouta wrote:
So the #include could be made just after the [general] section o
extensions.conf? outside of any specific context, i think this was the
question.
On 9/4/06, Peter Bowyer <[EMAIL PROTECTED
Tzafrir Cohen wrote:
On Thu, Aug 31, 2006 at 03:52:00PM -0500, Jay Moore wrote:
I have a question on how I can better organize my .conf files.
I have 3 different groups of people who use my VoIP service. Let's call
them 'Office', 'Factory' and 'Public'
I have a question on how I can better organize my .conf files.
I have 3 different groups of people who use my VoIP service. Let's call
them 'Office', 'Factory' and 'Public'. In my Asterisk directory, I have
created three folders: 'office', 'factory' and 'public', inside each of
which has a sip
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