there's just one factor - customer, i.e. extension in terms of Asterisk.
On 9/15/07, Joseph Bajin [EMAIL PROTECTED] wrote:
What are the factors in deciding which interface the traffic needs to
go out of?
Is it based on IP address, is it based on the terminating carrier?
--Joe
On 9/14/07,
Hi,
I would recommend instead of Using IPs in your Billing, your Prefixes.
Most of the billing softwares can to billing based on Prefix, for example
when Bill Clinton from Extension 100 is calling, add 22 or 22# in front of
the calling number 22#12345678, then your billing can do the rest based
no.
all packets come to the same h323 proxy.
and actually asterisk acts as sip -- h323 convertor.
so, for instance, Bill Clinton calls asterisk as SIP, asterisk sees it's a
Bill Clinton and sends h323 packets to the same h323 proxy as usual, but put
certain outgoing IP address
On 9/15/07,
In article [EMAIL PROTECTED],
Jonas Arndt [EMAIL PROTECTED] wrote:
Call File
=== Call File ==
channel: Local/[EMAIL PROTECTED]
maxretries: 3
retrytime: 60
waittime: 60
callerid: Test *66
application: AGI
data: test.agi|670507
=
Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
Thanks Anselm. This does clears a few things for me.
Tho, I couldnt find the patterns you mentioned in the docs(do point me
to the location if you know of it).
I started on
Am Dienstag, den 11.09.2007, 19:09 +0500 schrieb Rizwan Hisham:
The whole point of doing this is because if the user gives away his
username/password to his friends or relative and allows them to use
his account, that way we r gona have a lot more traffic in our
asterisk server.
Also we
On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote:
I'm seeing the problem on both etch and lenny releases.
Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux
I have a similar problem and am also using Debian (lenny). I'm using an
SMP kernel. Could that
On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote:
I'm seeing the problem on both etch and lenny releases.
Linux ads04 2.6.18 #2 SMP Wed Sep 12 15:45:10 EDT 2007 i686 GNU/Linux
I have a similar problem
On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote:
I'm seeing the problem on both etch and lenny releases.
Linux ads04 2.6.18 #2
Sir,
I have an audiocode MP-118 8 port external FXO gateway and i have connect pstn
line to FXO gateway now i want to dial outside call using FXO gateway and
receive all outside call. but i donot know what i have add in sip.conf and
extension.conf to make it possible.
I have also attach
So take the challenge and go with ip routing.
I've mentioned everything that you needed in previous reply.
On 9/15/07, Kate Kretz [EMAIL PROTECTED] wrote:
no.
all packets come to the same h323 proxy.
and actually asterisk acts as sip -- h323 convertor.
so, for instance, Bill Clinton
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jonas Arndt [EMAIL PROTECTED] wrote:
Call File
=== Call File ==
channel: Local/[EMAIL PROTECTED]
maxretries: 3
retrytime: 60
waittime: 60
callerid: Test *66
application: AGI
data:
Thank you for the example Isaac. I did as you mentioned and now it seems to
be working perfectly.
Saludos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Isaac Xiao
Sent: jueves, 13 de septiembre de 2007 10:33 p.m.
To: asterisk-users@lists.digium.com
hi folks,
I've built a simple PHP-script utilizing the AGI-interface. in
extensions.conf I trigger the script and pass a single value as first
argument:
exten = h,1,DeadAGI(process.php|${Enter})
On the Asterisk-console, I can actually see that the script is called
correctly (something like
In article [EMAIL PROTECTED],
Jonas Arndt [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jonas Arndt [EMAIL PROTECTED] wrote:
Call File
=== Call File ==
channel: Local/[EMAIL PROTECTED]
maxretries: 3
Your best bet is to get your VoIP service through whoever your ISP is.
If Global Crossing offered cheap VoIP (in comparision to some of their
TDM offerings), I would consider it. It's all IP in the core now
anyways, no real reason to use TDM for the last mile.
Maybe it has something to do
On Sat, 2007-09-15 at 19:25 +0530, Sanspareils Greenlans wrote:
Sir,
I have an audiocode MP-118 8 port external FXO gateway and i have connect
pstn
line to FXO gateway now i want to dial outside call using FXO gateway and
receive all outside call. but i donot know what i have add in
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
It's is possible to receive the callerid from the lines on the astribank
unit? This is my config:
[channels]
language=es
context=from-zaptel
The is the real meat of the article is that the Microsoft tsunami is
coming, no it wont be like the iPhone, it will be real and hard hitting
and unlike a tsunami, it will continue to get stronger.
Digium’s strategy is fairly straightforward. Write more code, package
Asterisk better, educate
Tony Mountifield wrote:
a separate context [foo], with a wildcard extension _X., which will match
any extension of two or more digits. I then put the extension number into
the parameter list for the AGI.
So instead of generating data: test.agi|12345 in the call file you
generate extension:
Hi Michael,
Actually parameter passed to AGI script are not Channel Variables and
they passed to PHP/AGI directly so you cannot access them using STDIN.
to access passed parameters simply use global variable argv like.
global $argv;
//Getting input data (Parameter Passed to Script)
Hi,
uncomment immediate=no
Regards
Nasir Iqbal
ICT Innovations
http://ictinnovations.com
On Sat, 2007-09-15 at 13:18 -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always
Michael Kamleitner wrote:
I've built a simple PHP-script utilizing the AGI-interface. in
extensions.conf I trigger the script and pass a single value as first
argument:
exten = h,1,DeadAGI(process.php|${Enter})
On the Asterisk-console, I can actually see that the script is called
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
It's is possible to receive the callerid from the
Guys,
The problem was that the IAX2 channel accepted both ulaw and gsm codec.
Once I dropped the gsm alternative in the iax.conf the interaction with
the AGI scripts (DTMF) worked great. So it seems that the ulaw was
picked if the IAX call was initiated by the IAX phone and hit the AGI
script
Hi Tzafrir:
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always
Sorry but your not going to drag me into another one of these you
ungrateful bastard type arguments. If you want to take the pepsi challenge
as to who contributes how much in what way then email me offline with your
list and I'll send you mine!
-Original Message-
From: Brian Capouch
In article [EMAIL PROTECTED],
Jonas Arndt [EMAIL PROTECTED] wrote:
The problem was that the IAX2 channel accepted both ulaw and gsm codec.
Once I dropped the gsm alternative in the iax.conf the interaction with
the AGI scripts (DTMF) worked great. So it seems that the ulaw was
picked if the
You never really specified what you want this for. If it is for enterprise
type installations then Aastra has a very robust SIP DECT solution
specifically designed for multiple roaming extensions. When you go through
their webinar training they provide all the calculations in terms of square
On Sat, Sep 15, 2007 at 01:30:26PM +, Chris Nestrud wrote:
On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote:
I'm seeing the
Actually Cbeyond does that and their quality of voice is much better analog
lines.
On 9/15/07, Steve Totaro [EMAIL PROTECTED] wrote:
Your best bet is to get your VoIP service through whoever your ISP is.
If Global Crossing offered cheap VoIP (in comparision to some of their
TDM offerings), I
Bogen http://WWW.BOGEN.COM
On 9/13/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI Digium
card.
I am looking for a paging system to an external speaker. I can page to
internal Polycom 501 VoIP.
But, what hardware or system do I
On Sun, 16 Sep 2007 01:29:11 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 01:30:26PM +, Chris Nestrud wrote:
On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
On Fri, 14 Sep
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