IE6, FF3 (no difference regardsless of used browser)
GUI 2.0, Revision 3677
-
I tried to add a new SIP-provider using Trunk - VOIP Trunk - + New
SIP/IAX Trunk. Unfortunately the provider uses user account information of
the following format: Bluesip/username
This leads to the following
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
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?
bkruseAll you have to do is click 'Add Service Provider'.
I tried: Trunks -
Tzafrir,
TCDo 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
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 didn't
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
External GW [G729]
|
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:
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
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
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
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
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 pedantic
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
headers
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.
You
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. 126
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 particular to
Vadim Lebedev vadim at 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,
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 bought a
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
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
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
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 GnuDialer
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
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. You
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 -
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
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
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
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
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[18945]:
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
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 your
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
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. It's
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 other
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