Hello All,
I don't even know the relevancy of my question. Please answer me if my
question have some sense.
I have recently implemented an asterisk server with freepbx. I have created
100 extentions and i can make successful calls between extensions from
anywhere. But my office have three
Hi,
I'm currently testing my FreePbx Box to work with our Avaya PBX to allow
dialing outgoing international call and FreePBX extensions to avaya PBX
Extensions calling.
Unfortunately no luck to do it successfully. Any help would be much be
appreciated, here is the sample codes I already
I have tried asterisk on windows XP using Cygwin and it worked fine.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antonio Modesto
Sent: Thursday, July 28, 2011 1:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi
Any company proving radius based billing for asterisk only for
accounting ,not authenication and atherization.Please provide some links
Thanks
Nikhil
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-- Bandwidth and Colocation Provided by
Hi,
Did you created your normal Inbound and Outbound routes in freepbx? For use
with your zap channels?
You'll problably have to change your routes on your pbx too...
Regards,
Carlos M Cruz
2011/7/28 michael k mich...@inapp.com
Hello All,
I don't even know the relevancy of my question.
On 28/07/11 02:58, Mike Diehl wrote:
Any ideas?
Mike.
I'd go on site if possible and see what actually happens at 19:00. Set
up a wireshark trace capturing all traffic through their router.
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-- Bandwidth and
On 28/07/2011, at 8:41 PM, Paul Hayes p...@provu.co.uk wrote:
On 28/07/11 02:58, Mike Diehl wrote:
Any ideas?
Mike.
I'd go on site if possible and see what actually happens at 19:00. Set up a
wireshark trace capturing all traffic through their router.
--
I am picking a cleaner
On 27/07/11 19:41, Claude Hayn wrote:
The office manager freaks out each time and starts randomly rebooting
devices in no particular order including the UPS, PBX, Asterisk Gateway,
firewall and router.
Ahh that old chestnut. That's never a good thing, try to tell them not
to do this,
Dear All
I've setup in lab a model include
a *handse*t (nokia 6021 in supported
listhttp://www.voip-info.org/wiki/view/chan_mobile)
connect to an *bluetooth dongle* (Cambridge Silicon Radio, Ltd Bluetooth
Dongle (HCI mode)) attached to an *PC *(install Asterisk 1.6.2.19 and Bluez
3.7 with
On Thu, 2011-07-28 at 09:57 +0100, Paul Hayes wrote:
On 27/07/11 19:41, Claude Hayn wrote:
The office manager freaks out each time and starts randomly rebooting
devices in no particular order including the UPS, PBX, Asterisk Gateway,
firewall and router.
Ahh that old chestnut.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, July 28, 2011 1:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connect asterisk to normal telephone PBX
Hello All,
Hello everybody,
We have an asterisk 1.8.4.1 setup, connected to a PRI line.
We're currently facing an issue where asterisk does not recognise the event
when the called party declines/cuts the call. This happens specifically over
calls on a PRI line. For calls over SIP, call decline event is
Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
Hello everybody,
We have an asterisk 1.8.4.1 setup, connected to a PRI line.
We're currently facing an issue where asterisk does not recognise the
event when the called
On Thursday 28 Jul 2011, Ishwar Sridharan wrote:
Is there a reason why asterisk doesn't recognise the call decline, and
does it need any configuration changes to enable this?
What are you seeing for ${HANGUPCAUSE} when this happens ? (Put a line such
as
exten = y, n, NoOp(Hangup cause
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Thursday, July 28, 2011 9:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
Hi list,
I have Asterisk speaking with google talk, is there any way to set or at
least hide my google voice number when I call others?
thanks for your help,
AHJos
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On 07/28/2011 09:22 AM, A.H. Jos wrote:
Hi list,
I have Asterisk speaking with google talk, is there any way to set or
at least hide my google voice number when I call others?
Set a different 'callerid' on either your outgoing sip.conf peer?
--
Alex Balashov - Principal
Evariste Systems LLC
I am new to asterisk but I think that with gtalk things are different!!!
In my extensions.conf I have:
exten = 2103,2,Dial(SIP/sip.generic,20); twinkle client
exten = _33X,1,Set(CALLERID(num)=01215063743); that doesn't work
exten =
On 11-07-26 03:21 AM, Gilles wrote:
Are there just not enough interest and too many, deep, Linux-specific
assumptions in the code, that would explain why Asterisk was never
officially ported to Windows?
Cost?
For me to run Asterisk under Windows requires me to purchase a Microsoft
license
Thanks for the reply. I am using an analog phone in normal PBX. I have an
extension called 199 in asterisk and an extension 264 in analog PBX. So how
do i create an inbound or outbound routes for call between these two
extentions ?
On Thu, Jul 28, 2011 at 1:39 PM, Carlos M Cruz
On Thu, Jul 28, 2011 at 4:46 AM, Duncan Turnbull dun...@e-simple.co.nzwrote:
On 28/07/2011, at 8:41 PM, Paul Hayes p...@provu.co.uk wrote:
On 28/07/11 02:58, Mike Diehl wrote:
Any ideas?
Mike.
I'd go on site if possible and see what actually happens at 19:00. Set
up a wireshark
2011/7/27 Vladimir Mikhelson v...@mikhelson.com:
Do you have any network devices or VPN tunnels in between the Asterisk
and Avaya?
Hi , the server does not have connections vpn I have and it in the
same LAN that avaya
The reason I am asking it looks like a potential networking issue.
ok,
Google Voice will show your number no matter what, there was a problem with
abuse when they let you send the CID in the early days. Pretty sure there is
nothing you can do about it.
From: A.H. Jos
Sent: Thu 7/28/2011 9:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] hide
Hi,
I've been trying to get MoH files to sound decent. I've got a hold of
Royalty-free Classical music (a safe choice for most of my customers) and
I`ve been trying to convert them to the normal telephony/Asterisk format
using sox. Unfortunately, it sounds really bad. I don't expect concert
On 07/28/2011 10:53 AM, Mike wrote:
Hi,
I’ve been trying to get MoH files to sound decent. I’ve got a hold of
Royalty-free Classical music (a safe choice for most of my customers)
and I`ve been trying to convert them to the normal telephony/Asterisk
format using sox. Unfortunately, it sounds
On Thu, 28 Jul 2011, Mike wrote:
I’ve got a hold of Royalty-free Classical music (a safe choice for most
of my customers) and I`ve been trying to convert them to the normal
telephony/Asterisk format using sox. Unfortunately, it sounds really
bad.
I convert files using:
sox
Hmmm, if alwaysauthreject is already breaking RFC rules then why not break
another rule for the greater good? It would only add another layer of
security.
Maybe: *alwaysregreject=yes*
*
*
*To drop SIP packets for both unauthorized registers and anonymous calls.
Keep it off by default and then
Personally I like to just hook up an old ghetto blaster / boombox to the
line in port on my sound card :)
Kidding aside - I think audio quality for MoH is not always going to sound
as nice as you might want.
I mostly stream online radio over my MoH and the quality is not the
greatest.
Maybe
On 07/28/2011 10:53 AM, Mike wrote:
Hi,
I've been trying to get MoH files to sound decent. I've got a hold of
Royalty-free Classical music (a safe choice for most of my customers)
and I`ve been trying to convert them to the normal telephony/Asterisk
format using sox. Unfortunately, it
Do you mean that was possible to set the CID in the early days of GVoice?
On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell te...@brummell.net wrote:
Google Voice will show your number no matter what, there was a problem
with abuse when they let you send the CID in the early days. Pretty sure
Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure other
VTSP's) do.
From: A.H. Jos
Sent: Thu 7/28/2011 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hide google voice number
Do you mean that was possible to set the CID
On 07/28/2011 11:57 AM, Mike wrote:
I should have said I am trying this both from a landline using ulaw, and
from a Polycom phone using g729 codec. G729 is noticeablty worst, as you`d
expect, maybe this is what is reported by my customers.
Is there any way to have a decent g729 file, or
I should have said I am trying this both from a landline using ulaw,
and from a Polycom phone using g729 codec. G729 is noticeablty worst,
as you`d expect, maybe this is what is reported by my customers.
Is there any way to have a decent g729 file, or should I just give
up and
On 07/28/2011 01:29 PM, Mike wrote:
I should have said I am trying this both from a landline using ulaw,
and from a Polycom phone using g729 codec. G729 is noticeablty worst,
as you`d expect, maybe this is what is reported by my customers.
Is there any way to have a decent g729 file, or
On Thu, 28 Jul 2011 12:04:38 +0500, Faisal Hanif fai...@vopium.com
wrote:
I have tried asterisk on windows XP using Cygwin and it worked fine.
Would you mind explaining how to do this?
Thank you.
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-- Bandwidth and
Interrupting - you have to not use DAHDI (SIP Only) and make sure you have
the necessary libs downloaded in your Cygwin install.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, July 28,
On Thu, 28 Jul 2011 12:46:03 -0500, Danny Nicholas
da...@debsinc.com wrote:
Interrupting - you have to not use DAHDI (SIP Only) and make sure you have
the necessary libs downloaded in your Cygwin install.
It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has
someone written an
If they have, it would probably be on www.nerdvittles.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Thursday, July 28, 2011 1:02 PM
To: asterisk-users@lists.digium.com
Subject: Re:
- Original Message -
On Thu, 28 Jul 2011 12:46:03 -0500, Danny Nicholas
da...@debsinc.com wrote:
Interrupting - you have to not use DAHDI (SIP Only) and make sure you
have
the necessary libs downloaded in your Cygwin install.
It's OK, I don't mind using a VoIP gateway instead of a
On Thu, 28 Jul 2011 13:08:33 -0500, Danny Nicholas
da...@debsinc.com wrote:
If they have, it would probably be on www.nerdvittles.com
It looks like The Incredible PBX runs on CentOS
www.nerdvittles.com/index.php?p=740
--
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--
you have to select musical compositions that are 'less incompatible'
with G.729's compression methods. When we chose the current MOH selections
included with Asterisk, our initial list was much larger, but we had to
remove some because they sound terrible when compressed with G.729.
Thanks
The dialplan is very simple. When the call comes in, we hand the call over
to adhearsion.
This is how the dialplan looks:
;group 0 will be used for incoming calls
EXOIN = DAHDI/g0
;group 11 for outgoing
EXOOUT = DAHDI/G11
;This will be used by adhearsion
EXOCID=
[general]
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Hash: SHA1
In
http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=markup,
line 180 states:
Voicemail now runs the externnotify script when pollmailboxes is
activated and notices a change.
My voicemail.conf configuration for my LDAP vm storage is
Hi AJS,
Our dialplan doesn't have a Dial() statement as that's taken care of by
adhearsion.
However, I added exten = y, n, NoOp(Hangup cause was ${HANGUPCAUSE})
at the end of our context, restarted asterisk.
The log doesn't have anything new.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at
Hi Eric,
There weren't any lines with PRI channel = in the chan_dahdi.conf
However, I added the lines you'd mentioned, near the top of the file. Still,
no difference in either the behaviour or the asterisk output.
Please note that as soon as the call lands on asterisk, we pass the control
over
From: Mike l...@net-wall.com
Sent: Thursday, July 28, 2011 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MoH - conversion command
I should have said I am trying
1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk
DAHDI configs.
2) Can't troubleshoot when everything important is masked by an AGI script.
Reproduce the problem using standard dialplan stuff.
-Original Message-
From:
From: Mike l...@net-wall.com
Sent: Thursday, July 28, 2011 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MoH - conversion command
I should have said I am
Hi,
I'm looking to disable rejecting calls from my call center employees. They
are using Polycom phones. Is there a way to either disable the reject/DND
features on the Polycom phones (don`t think so) or have the Asterisk PBX
ignore Got SIP response 486 Busy Here back from 12.23.34.45
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On 07/28/2011 02:42 PM, Barry L. Kline wrote:
Is there some other parameter required to get this to fire or am I
reading more into that sentence from the CHANGES document than is
actually there?
Sorry for replying to my own post, but I've done
In order to get the proper encoding for Asterisk, you must provide the
correct values for each of these characteristics. In your case, they
are as follows:
rate = 8000
data size = 8-bit (byte)
data encoding = gsm
channels = 1 (mono)
Therefore, the command you would use to
On 07/28/2011 03:47 PM, Mike wrote:
I’m looking to disable rejecting calls from my call center employees.
They are using Polycom phones. Is there a way to either disable the
reject/DND features on the Polycom phones (don`t think so) or have the
Asterisk PBX ignore “Got SIP response 486 Busy
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, July 28, 2011 3:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Disabling Polycom
I'm looking to disable rejecting calls from my call center employees.
They are using Polycom phones. Is there a way to either disable the
reject/DND features on the Polycom phones (don`t think so) or have the
Asterisk PBX ignore Got SIP response 486 Busy Here back from
12.23.34.45
All;
In a linked server environment, running Asterisk 1.6 I am noticing that when
a call is placed from server A to server B (via 4 digit extension) and
server B ext has a FMFM to call their mobile, the mobile phone shows the
default caller ID setting on server B instead of the actual caller id
On 7/28/2011 11:31 AM, Bruce B wrote:
Hmmm, if alwaysauthreject is already breaking RFC rules then why not
break another rule for the greater good? It would only add another layer
of security.
Maybe: *alwaysregreject=yes*
*
*
*To drop SIP packets for both unauthorized registers and anonymous
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Hash: SHA1
I found what I believe to be a bug and have submitted it:
https://issues.asterisk.org/jira/browse/ASTERISK-18207
Please correct me if I'm wrong.
Barry
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Hi team,
Can any one help me to implement dialplan in which conversation between a-party
and b-party (call patch) needs to be recorded.
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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-- Bandwidth and Colocation
Hi Vinod,
You Need to look in MIxmonitor application on asterisk.
http://www.voip-info.org/wiki/view/MixMonitor
http://www.the-asterisk-book.com/unstable/applikationen-mixmonitor.html
Where you can find easy dialplan
On Fri, Jul 29, 2011 at 4:35 AM, Vinod Dharashive
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