hi all,
I have a little problem. I'm trying to configure a2billing
(asterisk2billing) with asterisk. Everything done successfully but when I
try to call following error occur
WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application
'DeadAGI,a2billing.php' for extension (a2billing, 456,3)
When passing arguments to applications you must use parentheses.
Try:
exten = _X.,3,DeadAGI(a2billing.php)
You can omit parentheses when calling applications with no arguments,
e.g.
exten = s,1,Answer
... but not when there are parameters.
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On Jan 1, 2010,
An great an happy new year 2010 for all of you ...
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Myles Wakeham a écrit :
[...] Are there tools or
add-ons available for this that will email me when a SIP registration
goes offline?
Any suggestions for this would be greatly appreciated.
Hi Myles,
first, best wishes to the list for this new 2010 year.
To answer your question, you
Is there any way to listen SIP on multiple ports on asterisk. Is is possible
to define in sip.conf in the following way.
sip.conf
[general]
port = 5060
port = 5090
Regards,
Shariq Khan
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Also, shouldn't the .php script be located in /var/lib/asterisk/agi-bin?
On Fri, Jan 1, 2010 at 7:31 AM, Alex Balashov abalas...@evaristesys.comwrote:
When passing arguments to applications you must use parentheses.
Try:
exten = _X.,3,DeadAGI(a2billing.php)
You can omit parentheses when
On Fri, Jan 1, 2010 at 10:34 AM, Shariq Khan shariqrazak...@gmail.comwrote:
Is there any way to listen SIP on multiple ports on asterisk. Is is
possible to define in sip.conf in the following way.
sip.conf
[general]
port = 5060
port = 5090
Depending on the version of asterisk you are
Fact.
On 01/01/2010 01:06 PM, Warren Selby wrote:
Also, shouldn't the .php script be located in /var/lib/asterisk/agi-bin?
On Fri, Jan 1, 2010 at 7:31 AM, Alex Balashov abalas...@evaristesys.com
mailto:abalas...@evaristesys.com wrote:
When passing arguments to applications you must use
Friday, January 1, 2010, 7:12:54 PM, Alex wrote:
On 01/01/2010 01:06 PM, Warren Selby wrote:
Also, shouldn't the .php script be located in /var/lib/asterisk/agi-bin?
Fact.
And on a live channel must use AGI instead of DeadAGI.
And man should not topposting on a maillist...
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Best
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
like
bindport = 5060,5061 OR
bindport = 5060
bindport = 5090
I want, asterisk to listen SIP on multiple ports. so that users where SIP
port 5060 blocked, can easily register to asterisk by using an alternate
port.
Shariq Khan
Better to post your problem on Asterisk2Billing Forum
http://forum.asterisk2billing.org/
http://forum.asterisk2billing.org/Warren Selby mentioned right that your
agi is missing.
And dont change DeadAGI and ignore this error, as the script is still not
updated to use AGI for local channels.
I have AudioCodes 2xFXO / 2xFXS but can not make the FXO port to work
correctly; I can dial out on one FXO port or the other FXO, but not on both.
It depends on Sorting in : Hunt Group Setting (Ascending, Descending)
If setting is set to a Cyclic Ascending I can dial out on FXO port every
second
Un-top-posting...
On 01/01/2010 01:06 PM, Warren Selby wrote:
Also, shouldn't the .php script be located in
/var/lib/asterisk/agi-bin?
On Fri, 1 Jan 2010, Alex Balashov wrote:
Fact.
To be more specific, unless the full path (starts with a slash) is
specified, the specified path is
At 18:36 12/31/2009, Qurba Joog wrote:
Thanks for the reply Doug. Do you mean I should add nat=yes on my
nat'd extensions?
Affirmative.
On Wed, Dec 30, 2009 at 10:57 PM, Doug
mailto:d...@natel.netd...@natel.net wrote:
At 18:22 12/30/2009, Qurba Joog wrote:
You are correct.. I had the
On Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming kpflem...@digium.comwrote:
hadi motamedi wrote:
Can you please let me know if we can have different codec schemes for
audio codec in audio codec out ? I mean , in one application , we
can have our audio codec input set to G.711 a-law
On Thu, Dec 31, 2009 at 6:40 AM, hadi motamedi motamed...@gmail.com wrote:
Dear All
Please be informed that my Asterisk has sip connection to an external
sip server but the sip outgoing call will be disconnected for some
unknown reasons . Please find attached the debug log . Can you please
Hi,
I have two Digium Cards
http://www.digium.com/en/products/digital/te121.php, on connected to PRI
the other to EPABX.
We have felt problem similar to what is mentioned at,
https://issues.asterisk.org/view.php?id=15498
I have removed the VPMADT032, from the card connected to EPABX. I would
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