Its telling you the sound file Goodbye
does not exist in the directory it looks for sounds. If you indeed have a sound
file called Goodbye then you need to either move it to the default sounds
directory or add the path line to the command. If you dont have the
sound file youll need to
I have tested Redfones boxes. Tried
two of them and was able to re-create some issues. I did not have PRI lines but
a 24 channel em wink line so not sure if PRI is affected as well. I found
that over time we had issues with hanging zap channels. Asterisk reported
everything was just fine
Can
anyone tell me where this is coming from? I cant seem to find any information
on it anywhere. I dont believe Im using special tones
anywhere. Any ideas?
Aug 23
14:30:34 WARNING[15443]: chan_zap.c:5492 ss_thread: Unable to start special
tone on 15
_
Kevin
then Festival.
Thanks
for your input.
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1 is a service mark of Novo 1, Inc
All I can find for Flite is for AAH, does it work as well with plain
Asterisk? Is the setup the same?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, August 21, 2006 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial
You need to change:
exten = 777,4,goto(trunkretry,1,1)
to
exten = 777,4,goto(trunkretry,777,1)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Silverman
Sent: Friday, August 11, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial
This is an issue I'm having as well. Here is what I've discovered.
Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based on
the extensions.conf setup. User of phone 4000 has set a forward in the phone
to an external number, 1-555-555-. There is nothing telling Asterisk to
box? If
it is, have you established why? CPU being killed, memory starvation,
something else?
It is only happening on forwarded calls, though. I'll have to try your
workaround.
Thanks,
Mark
On 10/08/06, Kevin Savoy [EMAIL PROTECTED] wrote:
This is an issue I'm having as well. Here is what I've
the PSTN. Instead the caller gets
the message Press 1 to accept the recording. Pressing 0 again deletes the
message. How do I get this to work for outside callers calling in??
Thanks
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701
) in extensions.conf with
no luck:
exten =
o,1,DIAL(SIP/100,100)
Like Kevin, it works fine for our internal users, just
doesnt work for callers coming from the PSTN.
Thanks,
-AntD
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy
Sent: Monday, July 24, 2006 7:37
AM
: [asterisk-users] Operator in Voicemail
Are you sure this is saying exten = 0 with a ZERO and not an Oh?
Looks like a lowercase Oh to me below.
Kevin Savoy wrote:
This doesn't solve the problem. Still the same. Any other ideas
Discussion
Subject: Re: [asterisk-users] Email notification of voicemail
Aha - get rid of the leading comma for each entry..
= ,Front Desk
= ..
A.
On Jul 13, 2006, at 1:00 PM, Kevin Savoy wrote:
I've X'd out the extensions and passwords but this is all I have
Would this work?
exten = 3299,1,VoicemailMain(${EXTEN})
This way it would check the voicemail of the extension doing the dialing?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Thursday, July 13, 2006 9:51 AM
To: Asterisk
Should be span 1 for the for T1 and span 2 for the second T1 in your config.
They are both span 1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Thursday, July 13, 2006 12:47 PM
To: asterisk-users@lists.digium.com
Subject:
in your voicemail.conf:
1234 = 1234,The Marquis de Sade
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote:
I have attach=no in my voicemail.conf so
=,,Justin Hall
=,,Jason Smestad
=,,Cumi Everson
=,,Glenda Cusker
=,,Laura Sanford
=,,Gary Sundet
=,,Kevin Penner
=,,Kevin Savoy
=,,Jeff Garaas
=,,Natalie Thompson
=,,Jolene Ross
=,,Ralf Patterson
=,,Mike
The world is full of smart alecks. Thank the lords because what a boring
world this would be without us :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer
Sent: Thursday, July 13, 2006 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial
on linux gets full of these rejection notices. I cant seem to
find anywhere to tell Asterisk to stop notifying people they have voicemails.
Im
using 1.2.9.1 of Asterisk. Thanks
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
: Wednesday, July 12, 2006 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Email notification of voicemail
Kevin Savoy wrote:
Asterisk is trying to send an email to users when they receive a
voicemail. Can this be shut off? I have not entered any email
Are you sure they are sending you all 10
digits and not just the last four? Our provider just sends the last four digits
on DID. If this is the case you would have this:
exten = 4567,1,Answer()
exten = 4567,1,DIAL(SIP/user,20)
Hope this helps.
From:
[EMAIL PROTECTED]
I'm in Williston, North Dakota and we have an office in Billings, MT. He's
right. We are 500 miles form civilization! :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, July 05, 2006 10:00 PM
To: Asterisk Users Mailing
to Asterisk and then
hangs up since I have the s,1,Hangup() and Asterisk at this point doesnt
know where the call should go. I believe at this point the call is hungup in
Asterisk but NOT on the telco side.
Any
ideas how to get around this???
Thanks
_
Kevin Savoy
-Commercial Discussion
Subject: Re: [asterisk-users] Zap Channel not hanging up on Telco side
Kevin Savoy wrote:
I'm having an issue where Asterisk hangs up a call (either party hangs
up) but the telco side of the T1, both the local company and ATT,
does not receive the hangup signal from
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
I'm having an issue where Asterisk hangs up a call (either party hangs up)
but the telco side
of the T1, both the local company and ATT, does not receive the hangup
signal from
Asterisk. Therefore Asterisk thinks the channel is available but it's
this is a bug? Should I submit a
bug report?
Ive
heard to submit a bug to send it to mantis. What is this and where or how do I
do this?
Thanks
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http
I can also add that this happens on em_w lines as well. I've had issues
where callers start getting dead air when dialing out. Talking with the
phone company the lines were in an off-hook state even though Asterisk hung
up the call. I done exactly as below where I hang up before the other party
://bugs.digium.com/view.php?id=4101
On 6/28/06, Kevin
Savoy [EMAIL PROTECTED]
wrote:
Sorry
if this has been posted before but I'm having an issue where I get the
following on my CLI.
ast_read:
Dropping incompatible voice frame on Local/XX of format ulaw since our native form has
systems. Not good.
Any
help would be greatly appreciated. If I had hair left Id be pulling it
out about now.
Thanks
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1
Speaking as one of those call centers we are looking at doing a turn over to
Asterisk from our Nortel systems and are doing it ourselves. We've looked at
a lot of packages from Fonality, Signate, Aheeva and others and none fit our
needs. Each has good aspects but none have all of what we need.
Steve ,
I was actually looking forward for the same thing , do y ou have something like
this , as an example?
regards
Junaid Uppal
On 5/9/06, Steve
Totaro [EMAIL PROTECTED]
wrote:
Use an activex screenpop.
Thanks,
Steve Totaro
-Original Message-
From: Kevin Savoy [mailto:[EMAIL
something like this , as an example?
regards
Junaid Uppal
On 5/9/06, * Steve Totaro* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Use an activex screenpop.
Thanks,
Steve Totaro
-Original Message-
From: Kevin Savoy [mailto:[EMAIL PROTECTED
: Kevin Savoy [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Fri, 5 May 2006 15:31:41 -0500
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer
The problem with what is in wiki is that these calls are being sent
, 2006 10:15
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer
Does the phone ring, just not auto-answer?
Thanks,
Steve Totaro
From: Kevin Savoy
[mailto:[EMAIL PROTECTED]
Sent: Monday, May 08, 2006
comes in?
Thanks,
Steve Totaro
From: Kevin Savoy
[mailto:[EMAIL PROTECTED]
Sent: Monday, May 08, 2006 11:27
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer
Correct. We have to hit the
answer
From: Kevin Savoy
[mailto:[EMAIL PROTECTED]
Sent: Monday, May 08, 2006 12:12 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer
Because we will have many of these phones
in remote locations and we dont want
This may be the way to go but not the best. Our agents frankly aren't the
brightest people and I can see them forgetting it as soon as it is said to
them, or they are not paying attention and missing the announcement but it
is something to look into. Thanks
-Original Message-
From:
database. I dont get any
error messages anywhere telling me why it stops. As far as tail and perl are concerned
everything is fine.
We
will be using this for a call center and need more reliability. Anyone got one
working?
Thanks
_
Kevin Savoy
Business Unit Telecom
Analyst
of these support it.
Thanks
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1 is a service mark of Novo 1, Inc
___
--Bandwidth
with Auto Answer
Kevin Savoy wrote:
Can anyone recommend a phone to use in an inbound call center
environment that has an auto answer feature? We don't want the agents
having to acknowledge the call. The call should just activate on the
headphones. We have tried Grandstream 2000, Polycom
: Re: [Asterisk-Users] Call Center Phone with Auto Answer
Kevin Savoy wrote:
Can anyone recommend a phone to use in an inbound call center
environment that has an auto answer feature? We don't want the agents
having to acknowledge the call. The call should just activate on the
headphones. We
agents phone for anymore then 5 seconds when there are other agents out
there waiting to take that call. Any ideas?
Thanks
_
Kevin Savoy
Business Unit Telecom Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1 is a service mark of Novo 1, Inc
___
--Bandwidth and Colocation provided
994 (B1706AVA) Haedo
Buenos Aires, Argentina
Tel: 54 11 4650 1775
Fax: 54 11 4650 4295
www.infodax.com.ar
-Mensaje original-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy
Enviado el: Martes, 18 de Abril de
2006 04:03 p.m.
Para: 'Asterisk Users Mailing List
hiccupping.
Can
anyone shed any light on where to look? Any help would be desperately
appreciated.
Please
help.
_
Kevin Savoy
Business Unit Telecom
Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1
-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy
Enviado el: Martes, 18 de Abril de
2006 03:30 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] PRI
blocking on incoming calls
Ok here is our setup. We are using Asterisk 1.2.6 and
Zaptel 1.2.5. We
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