Yes they did.
On 3/26/12, SamyGo wrote:
> Good to know, hope our replies did some help :)
>
> On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku wrote:
>
>> Hi,
>>
>> Thanks for the support. Issue solved. Somehow the routes on the fxo
>> gw were not working.
an IP
> 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I
> don't know how I can keep the PBX on this subnet, and also connect it via
> eth to the other vpn modem and give it another IP which is on a
> 192.168.200.* subnet.
> Any pointers?Thanks!
Hi,
Thanks for the support. Issue solved. Somehow the routes on the fxo
gw were not working.
On 3/21/12, James Mutuku wrote:
> Hi,
>
> I have configured a route on the fxo to send all incoming sip traffic
> to the "fxo" ports.
>
> I will try set the specific di
r dialled number.
> See your FXO gateway configuration Web-UI for outbound patterns OR verify
> that the FXO has its outbound line configured and working properly.
>
> On Wed, Mar 21, 2012 at 5:20 PM, James Mutuku wrote:
>
>> I am setting up asterisk->fxo gw.
>>
>&
www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
www.zetu.co.ke
Has your o
ax-Forwards: 70
From: "pbxserver" ;tag=as66c75bd7
To:
;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06
Contact:
Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
--
Best Regards,
James Mutuku Ndeti
Ag
rier and then try to send the sip
>> traces in human readable format. From those traces it'd be more clear what
>> is the issue from the carrier end rejecting the calls...Maybe your credit
>> expired !!
>>
>> Regards,
>> Sammy.
>>
>> On Wed, Mar
ts-busy-now&pls-try-call-later, noanswer[0m") in
new stack
[0K-- Playing 'all-circuits-busy-now.gsm'
(language 'en')
[0K == Spawn extension (macro-outisbusy, s, 4) exited non-zero on
'SIP/3000-0014' in macro 'outisbusy'
== Spawn extension (fro
ing to them about their phones
> has been so-so. Historically we've had awful experiences with other
> Chinese phone vendors and have stopped considering products from Chinese
> companies. We did not actually try Yeastar products.
>
>
> On Wed, Nov 30, 2011 at 3:39 PM, Ja
Hi,
I am looking into advising a client on the pro's and cons of using
Installing asterisk on a server vs appliance(e.g digium mypbx). the
appliance seems cheaper initially.
>From experience, what would be pro and cons for either option?
--
Best Regards,
James Mutuku Ndeti
Agile
http://www.google.co.ke/search?q=asterisk+for+call+centers&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization im
the handset
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve
XX.,4,Hungup()
Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error
SIP/Y.Y.Y.Y-35dc is circuit-busy
Are there any settings I am leaving out?
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your
I have tried all that. I just can't trace the value. this maybe the wrong
list. I just thought someone might know
On Thu, Sep 17, 2009 at 5:39 PM, wrote:
> Well, why not disable it from the GUI and see what changes -- this is
> sort of the wrong list, but maybe someone knows more fully.
>
__
that deals with
enabling/disabling followme. Please help
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find ou
I got it running a few days ago. I am using php4. Itshould run on php5
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asterisk-users mailin
I got my answer
dialparties.agi is not generated. It get's copied from the core module when
you 'Apply Configuration Settings' which is what allows it to be updated
with new core modules. It is also copied when you reload asterisk
James
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Hellos,
I have asterisk 1.2 and freepbx 2.3. I have edited the agi
script(dialparties.agi). Everytime I restart asterisk, the file gets
overwritten. How do I make sure my changes are not overwritten? What
generates dialparties.agi?
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems
increasing
bandwidth?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve
Hellos,
I need to send personal directory from asterisk to the ersonal directory of
the linksys spa 942. Is this possible?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship
sable followme. Is this value stored in the database?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CR
I have included that but my scripts goes silent at
AGI Rx << EXEC Flite "Hello 1215, you have dialed 1220."
AGI Tx >> 200 result=0
Below is my script
#!/usr/bin/php -q
new_AsteriskManager();
$agi->answer();
$callext = $agi->get_variable("DNID");
$callext=$callext['data'];
show hints");
but I am getting the output below on the cli debug
AGI Rx << EXEC show hints
AGI Tx >> 200 result=-2
AGI Rx << VERBOSE "EXEC show hints returned -2" 1
AGI Tx >> 200 result=1
>From My understanding -2 means "failure to find application"
I am new to AGI. I have written my first php agi script that gets the
extension dialed and says it back the caller using flite. I am stuck on how
to pass the comand asterisk –rx “core show hints to asterisk and get the
data back.
This isn’t the recommended way, but it does work: Let’s say exten
i_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> AGI Rx << EXEC Flite "Hello, ."
-- AGI Script Executing Application: (Flite) Options: (Hello, .)
-- Playing '/tmp/flite_buf_VTgzTg' (language 'en')
As you can see,
Hello,
I am looking for a follow me script, where users can toggle follow me from
their extensions and add follow me numbers from their extensions.
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization
It's long gone.
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http:
I did am not the one who started the project. the client has been running
1.2 for years and they needed additional features set up
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The project I am working on is really big. Unless I upgrade during
christmas(by then the project will be several months overdue). Just googled
further and saw some patches. I will try them and see.
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Hello,
>From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says
that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based
channels (i.e. chan_sip).
I am using 1.2 and Ind there is no reason to upgrade. Are there any
developments on this?
--
Best Regards,
Ja
. Extension A sees the second light blinking and hears the beeps (currently
working).
4. Extension B is notified that extension A is on another call.
Where do I start...I have looked in the following call
variables--$dialstatus and $devstatus but I can't get them working
--
Best Regards,
...the above post lacked some details
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't
figure how to use it. My dial plan gives "1215 has state 0" on all
scenarios.
Below is my script. I am dialing 1215,then I get a prompt on the state of
extension 1215. I have set extension
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't
figure how to use it.
Below is my script. I am dialing 1215,then I get a prompt on the state of
extension 1215. I have set extension 1215 to busy(by making several calls to
other extensions). I still get the prompt "1215 has a
thanks danny for the reply. I am looking into using flite to read out the
prompts. if i ma ask...are there other voices other than the mechanical
robotic male voice available for flite.? I have searched over the internet
and I can't seem to find any
___
-
Hellos,
I am looking for the sounds used in this ivr example
http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with
6900.
Any assistance is welcome.
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your
Hellos,
I have using asterisk 1.2 and freepbx 2.3. I need users to disable and
enable followme from there phones. I can't see any support for it. Is this
possible/available.? I have googled and I can't get information on it
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+25
Hellos,
I have astersist 1.2 working with freepbx. I want to tie pin codes to
extensions(users). How do I do this?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
the extension is busy. I want to notify the second,
> > third and fourth callet that they are on extension is on another call.
> Which
> > variable do I use on my dial plan
> >
> > --
> > Best Regards,
> > James Mutuku Ndeti
> > Agile Systems Limited
>
sion is on another call. Which
variable do I use on my dial plan
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find o
37;40m", " [1;35;40m [0;37;40m") in new stack
-- Executing [1;36;40mGotoIf [0;37;40m(" [1;35;40mSIP/1215-fc5b
[0;37;40m", " [1;35;40m1?skiprg [0;37;40m") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [1;36;40mGotoIf [0;37;40m(" [1;3
oIf[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",
"[1;35;40m1?skipblkvm[0;37;40m") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing
[1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",
"[1;35;40m1?theend[0;37;40m") in new st
Hello,
I have asterisk and linksys ip phones setup. If an ext. is busy on phone,
the ext. user is notified by a beep. I want to configure asterisk to
notify(voice) the caller that the extension is busy on a call.
Is this possible?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
Hellos,
I want to setup Asterisk+a2billing for over 10,000 extensions for voip
resale. Has anyone done this before. What are the hardware requirements and
challenges?
James
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asteri
Hello(s),
I know this might be test book question or one best suited for google but I
will take the risk of asking. Here I go. What common
routine maintenance tasks do you run on your asterisk box?
Thanks
James.
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ind transfer and making macro-stdexten (or your equivalent)
dial that variable in the case of the dial status being treated as BUSY.
To get a 'busy' will involve single line phones, or disabling call
waiting on the phone receiving the call.
regards,
PaulH
James Mutuku wrote:
Hellos,
I
Hellos,
I want to configure asterisk so that if exten A transfers a call to
exten B, and B is either busy or the call is not answered, the call
returns back to A. Is this possible?
Please help
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
ad
Hellos,
I want to configure asterisk so that if exten A transfers a call to
exten B, and B is either busy or the call is not answered, the call
returns back to A. Is this possible?
Please help
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:
Hello,
I am searched the net for tutorials on how I can Integrate vicidial with
trixbox. I can't find any. Anyone who knows where I can get one?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;
allow you to have
redundant servers.
External gateways would also be easier to scale when you need more lines.
[/quote]
2008/8/1 James Mutuku <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>
Hi list,
I need advice on which solution to implement, asterisk
appliance
Hi list,
I need advice on which solution to implement, asterisk appliance
A50 or just install linux on a pc and get tdm cards. Any comments?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
emai
user is dialing something other than 3000 and that extension
is not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
Hi list,
Have installed trixbox and I am working with
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten => 3000,1,dial(s
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten => 3000,1,dial(s
Hi,
has anyone worked with nxtvox(www.nxtvox.com) fxo cards? What is their
quality?
James
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Hi,
has anyone worked with nxtvox fxo cards? What is their quality?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+25
e "call disconnection" from the line(POT).
Steve Totaro wrote:
Some customers are locked into two year contracts.
That was the answer I got when adding four POTS lines to a system with
four BRIs...
Thanks,
Steve Totaro
On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku <[EMAIL PROTE
to have 2 T-1s dropped into the
> location?
>
> Michael
>
> On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote:
>
>
>> Adit 600 48 FXO.
>>
>> On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku <[EMAIL PROTECTED]> wrote:
>>
>>>
or Adit, I think Rhino has a pretty low priced one but I
have never used so cannot comment. I can tell you that the Adtran or
Adit is rock solid.
Thanks,
Steve Totaro
On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku <[EMAIL PROTECTED]> wrote:
Please advice on "channel bank"
Ste
d comments
Brian J. Murrell wrote:
> On Sun, 2008-06-15 at 11:03 -0400, Steve Totaro wrote:
>
>> On Sun, Jun 15, 2008 at 10:53 AM, Brian J. Murrell
>> <[EMAIL PROTECTED]> wrote:
>>
>>> On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote
Please advice on "channel bank"
Steve Totaro wrote:
> I would suggest a channel bank populated with FXO cards muxing to a T1.
>
> Thanks,
> Steve T
>
> On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku <[EMAIL PROTECTED]> wrote:
>
>> Hi,
>> I n
Hi,
I need to get an fxo gateway/card for a high traffic asterisk
installation. Please advice on which gateway/ fxo cards
Thanks
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Hi,
I need to know if the following features are available on asterisk
and their quality
-SMS
-Call control, budgeting and monitoring
-Video conferencing
-support for 500 extensions
-fax
-audio and video conferencing
and
1. Call accounting showing calls made
2. Call
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