Joseph wrote:
> Does anybody know if the process of upgrading firmware on "Linksys
> SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org
> http://www.voip-info.org/wiki/view/Sipura
>
>
I'm pretty sure it works - I used it to upgrade a (god help me) SPA 9000
the othe
On Wed, Aug 20, 2008 at 7:44 AM, Joseph <[EMAIL PROTECTED]> wrote:
>
> Does anybody know if the process of upgrading firmware on "Linksys
> SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org
> http://www.voip-info.org/wiki/view/Sipura
>
I just recently upgraded mine.
Ruddy,
I've used deadagi for years with perfect success.
If it's a perl agi module, you need to make absolutely sure that
you're using 'use strict' and 'use warnings' in the main agi file -as
well- as any includes. You'll need to test your agi while in console
mode, so any of the perl warn
Hi!
Ruddy Gbaguidi wrote:
> I'm using DeadAgi and has set AGISIGHUP to no because I don't want my
> script to stop if the user hangs up.
> But when it reach the end of the script, the child process should die.
> And I don't see why I only have this trouble with perl agis.
>
Can you check if yo
Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA"
in Linux is the same as on Sipura 3K as described on voip-info.org
http://www.voip-info.org/wiki/view/Sipura
--
#Joseph
GPG KeyID: ED0E1FB7
___
-- Bandwidth and Colocation
michel freiha wrote:
>> Hi all,
>>
>
>
> I'm getting the following error when trying to make a PSTN call from
> asterisk server:
>> end_sound = (null)
> [Aug 19 20:51:17] WARNING[18945]: channel.c:3025 ast_request: No channel
> type registered for ''
> [Aug 19 20:51:17] WARNING[18
> Hi all,
I'm getting the following error when trying to make a PSTN call from
asterisk server:
> end_sound = (null)
[Aug 19 20:51:17] WARNING[18945]: channel.c:3025 ast_request: No channel
type registered for ''
[Aug 19 20:51:17] WARNING[18945]: app_dial.c:1183 dial_exec_full: Unable
I'm using DeadAgi and has set AGISIGHUP to no because I don't want my
script to stop if the user hangs up.
But when it reach the end of the script, the child process should die.
And I don't see why I only have this trouble with perl agis.
Eric "ManxPower" Wieling wrote:
> Your script is not catch
Your script is not catching SIGHUP, which is what Asterisk uses to tell
the AGI the channel went away.
Ruddy Gbaguidi wrote:
> Hi all.
> I'm using asterisk 1.4.21.2 and when I run
> ps -ef |grep defunct,
> I can see a lot of my perl agi still pending there.
> The channel has been cleaned up in as
Hi all.
I'm using asterisk 1.4.21.2 and when I run
ps -ef |grep defunct,
I can see a lot of my perl agi still pending there.
The channel has been cleaned up in asterisk.
I don't have this kind of problem with python or php.
I'm using ubuntu ...
Anyone has an idea ?
I've tried export LD_ASSUME_KERN
J.M. schrieb:
> Where is this "debug log" that is mentioned? Most web sites I've read
> mention a /var/log/asterisk/full file, however, I do not have a
> /var/log/asterisk/full file on my system.
You need to enable it in logger.conf.
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de -
On Tue, 2008-08-19 at 10:59 -0500, J.M. wrote:
> Where is this "debug log" that is mentioned? Most web sites I've read
> mention a /var/log/asterisk/full file, however, I do not have
> a /var/log/asterisk/full file on my system.
The full log only gets created if you tell Asterisk to create it. Yo
> You should expect that; in fact, that's what the 'TB' in 'TBCT' stands
> for... for a time, there are two B-channels involved. TBCT is a method
> of taking two existing already connected B-channels and linking them
> together into the network, it is not a 'transfer' facility where you
> provide a
Edwin Quijada wrote:
> Hi!
> I wanna know if here somebody has installed gnudialer ?
> I installed but i dont know how to run it
> Anybody has a cluee?
>
>
You would probably have more success reading all the README's and online
help.
If that does not provide the answer you can ask on the GnuD
I have a similar error to this:
http://lists.digium.com/pipermail/asterisk-users/2005-February/080917.html
Where is this "debug log" that is mentioned? Most web sites I've read
mention a /var/log/asterisk/full file, however, I do not have a
/var/log/asterisk/full file on my system.
jm
__
Kevin,
Thanks for the hint (did´nt know that there is such list). Sorry for
disturbing this list. As soon as I have fugured out how to subscribe to the
GUI list, I will ask my questions there.
Edit: Found the page that shows all available lists:
http://lists.digium.com/mailman/listinfo
Best reg
Dear friends,
As a community service, FailSafeVoip is providing a free US Based Echo
Test. The service is running on a high performance asterisk box and is
connected via a fully TDM T1-PRI. The test server is based in Michigan.
The test extension is written simply as:
s,1,Answer
s,2,Echo
s,3,Ha
On Aug 19, 2008, at 1:44 AM, [EMAIL PROTECTED]
wrote:
> i read a few articles online about the possibility to setup a
> "buzzer" door system to PBX using asterisk!
I took a somewhat unique approach, based on reading recent postings.
We already had intercoms without door relays. So, I bough
Vadim Lebedev mbdsys.com> writes:
I've found the root of the problem and fixed it:
http://bugs.digium.com/view.php?id=13341
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
I have tested pedantic=yes just in a peer configuration and it don't
work, is a global setting. (tested with asterisk 1.2.21)
slds.
Philipp Kempgen wrote:
> Alex Balashov schrieb:
>
> [pedantic]
>
>> is peer-specific in this context, so it's just like any other
>> option that is also particul
Alex Balashov schrieb:
[pedantic]
> is peer-specific in this context, so it's just like any other
> option that is also particular to certain peers.
Really? I don't think so.
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 12
On Tue, Aug 19, 2008 at 07:35:03AM -0500, Kevin P. Fleming wrote:
> Jay R. Ashworth wrote:
> > I'll assume you've watched it on a PRI, so I'll defer, but I wouldn't
> > expect that myself; I would expect that when you tell the switch to
> > transfer it, you go immediately from one B channel to 0.
>
Nestor A. Diaz schrieb:
> Philipp Kempgen wrote:
>> Did you enable pedantic=yes in sip.conf?
>>
> thank you very much for your help, it fix the problem.
>
> Is there any other issue that i have to take in mind for placing calls ?
I don't know the Huawai softswitch. Just noticed the multiline
On Tue, August 19, 2008 8:48 am, Nestor A. Diaz wrote:
> Philipp Kempgen wrote:
>> Did you enable pedantic=yes in sip.conf?
>>
> thank you very much for your help, it fix the problem.
>
> Is there any other issue that i have to take in mind for placing calls ?
> is there any option for set up ped
Philipp Kempgen wrote:
> Did you enable pedantic=yes in sip.conf?
>
thank you very much for your help, it fix the problem.
Is there any other issue that i have to take in mind for placing calls ?
is there any option for set up pedantic for selected peers ? i use
broadvoice too and it requires
Jay R. Ashworth wrote:
> I'll assume you've watched it on a PRI, so I'll defer, but I wouldn't
> expect that myself; I would expect that when you tell the switch to
> transfer it, you go immediately from one B channel to 0.
You should expect that; in fact, that's what the 'TB' in 'TBCT' stands
fo
Nestor A. Diaz schrieb:
> I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i
> can succesfully connect other softphones like Zoiper, but when it comes
> to Asterisk SIP Client, the system doesn't authenticate, i have the
> following configuration:
[...]
> i attache the logs
Klaus Ruebsam wrote:
> IE6, FF3 (no difference regardsless of used browser)
> GUI 2.0, Revision 3677
> -
It will be much more effective for you to post messages about
Asterisk-GUI to the asterisk-gui mailing list.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The
Hello Asterisk People,
I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i
can succesfully connect other softphones like Zoiper, but when it comes
to Asterisk SIP Client, the system doesn't authenticate, i have the
following configuration:
peer: 10.220.0.2
username: 48577
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] ---> [Asterisk ]
> External GW [G729]
|
|---
> Why is it bad? In all Asterisk config files, the '>' after the '=' is
> superfluous for defining extensions, variables, etc. Try it. Having
> exten=123,1,... is perfectly valid and does not affect how Asterisk works
> in any appreciable way.
>
Ok Tighlman,
Thank you for the information,
i d
Tzafrir,
TC>Do you have such (working) stanzas for some providers?
as mentioned adding providers to providers.conf seems not to work by means
of the GUI. Yes I may of course add them manually to providers.conf, but
they won´t show up in the selection list.
However providers via Trunks -> VOIP Tr
On Tue, Aug 19, 2008 at 07:44:42AM +0200, Klaus Ruebsam wrote:
> GUI used: current 2.0 branch out of SVN
> browser used: IE6 as well as FF 3
>
> >> What is the proper way to add additional providers to the dropdown list?
> bkruse>All you have to do is click 'Add Service Provider'.
>
> I tried: Tr
Dear All,
I have the following scenario:
the customer is registerd on an openser server and is trying to make a PSTN
call...I configured the Openser to send PSTN calls to my asterisk server who
should send the call to the PSTN gateway...
I would like to ask please how should I configure the asteri
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