17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf)
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote:
You propably have a type=friend where the user part matches before you even
hit the peer part, where the insecure configuration parameter matches. There
is a confusion here on the From: username and the authentication
Hi!
I'd like Asterisk to set up direct media connections for calls between
clients who're both on the internet, and for calls between clients
who're both on the private network, but not set up direct media
connections for calls between clients on the internet and clients on
the private
Anyone know if my example of combining extensions.conf and realtime
extensions is doable ??
Kind regards,
Jonas.
On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote:
How about something like :
[mycontext]
exten = 100,1,NoOp(calling 100)
exten = 100,n,NoOp(going realtime)
switch =
- Chandrakant Solanki solanki.chandrak...@gmail.com wrote:
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to
- Ken D'Ambrosio k...@jots.org wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As
with
many phones, provisioning it was a bit of a PITA. The biggest
problem, as
far as I could tell, was that their firmware just doesn't seem that
stable, and is sometimes hard to
19 feb 2010 kl. 10.22 skrev Randy R:
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote:
You propably have a type=friend where the user part matches before you even
hit the peer part, where the insecure configuration parameter matches. There
is a confusion here on the
- --[ UxBoD ]-- ux...@splatnix.net wrote:
- Ken D'Ambrosio k...@jots.org wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As
with
many phones, provisioning it was a bit of a PITA. The biggest
problem, as
far as I could tell, was that their firmware just
Ken D'Ambrosio wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with
many phones, provisioning it was a bit of a PITA. The biggest problem, as
far as I could tell, was that their firmware just doesn't seem that
stable, and is sometimes hard to get to.
- I managed to
- Ken D'Ambrosio k...@jots.org wrote:
Hey, all. Got an SNOM 820 in the other day to kick the tires. As
with
many phones, provisioning it was a bit of a PITA. The biggest
problem, as
far as I could tell, was that their firmware just doesn't seem that
stable, and is sometimes hard to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday, Feb 18, 2010 at 05:36:41 -0300, Administrator TOOTAI wrote:
Hi
Hi, Daniel.
Daniel Bareiro a écrit :
[...]
Hours ago the IP changed and the domain was updated satisfactorily,
but in spite of this I was obtaining the registering
Hello,
Anyone know how I can intesify volume of an application playback()?
Thank you very much.
ye
Veja quais são os assuntos do momento no Yahoo! +Buscados
http://br.maisbuscados.yahoo.com--
AFAIK, playback/background has no gain adjustability. Two possible
work-arounds would be to adjust gain on the line/extension or to use sox to
create a louder version of the file you want to playback sox v +2
vm-goodbye.gsm vm-goodbye2.gsm
_
From:
Hi all,
we are looking for a solution which transfers device status changes
(events) like busy, picked up, detected answering machine/fax and
no valid number to another server which takes action according to
these events.
I already found the patch mentioned in the subject of this message.
Which
I think you are correct, thank you for pointing it out.
I just switch entries in sip.Cong put [pstn-9998] first' and [pstn-]
second
and the second entry was selected :-( (so you are right on).
Audiocodes gateway, has two FXO ports, I was convinced that entry is selected
based on
19 feb 2010 kl. 11.47 skrev --[ UxBoD ]--:
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
UxBoD - you really have to read the security advisory before sending out such
examples on the mailing list. Please go to http://www.asterisk.org now.
Have a nice weekend!
Thanks,
/O
--
Hi,
Many thanks Steve and Philipp for your input. I am compiling Asterisk
1.6 svn version right now and will try to integrate the T.38 Gateway
patches mentioned at https://issues.asterisk.org/view.php?id=13405 (I
have some Linux coding experience, but unfortunately my
telecommunication knowledge
I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I
installed my TDM400P into the PC, it's really slow to boot now, when it
finally does I gets stuck in a loop of reporting isac xdu no tx_busy.
Anyone able to assist?
Thanks in advance!
--
On 18 February 2010 00:14, Michael Graves mgra...@mstvp.com wrote:
On Wed, 17 Feb 2010 17:12:01 +, Steve Davies wrote:
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[7.590966] Zaptel Version: 1.4.12.1
[7.590966] Zaptel Echo
- Katerina Borin katerin.bo...@gmail.com escreveu:
Hello,
I have transcoding card TC400P installed in server running Debian with
Asterisk 1.4.23. Everything seams to be fine and after I boot up
server I see in dmesg:
7.590966] Zapata Telephony Interface Registered on major 196
[
Is it possible to split sip.conf into two files (sip1.conf sip2.conf)?
I have an Audiocodes gateway with two FXO ports, and (according to info I
received, and it appears to be correct) Asterisk find the peers based on their
IP
and not on their IP+PORT. Thus, Audiocodes with two FXO ports
I think sip.conf will allow the inclusion of a second (or greater) sip2.conf
file. This might only apply to extensions.conf, but I'm betting all .conf
files are processed with the same parser.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
like a USB port or
I'm suspecting you might be correct; so it will not make much difference.
--
Joseph
On 02/19/10 10:29, Danny Nicholas wrote:
I think sip.conf will allow the inclusion of a second (or greater) sip2.conf
file. This might only apply to extensions.conf, but I'm betting all .conf
files are
--[ UxBoD ]-- wrote:
Would be nice if the VPN support could be back ported to the 360s.
Never going to happen, there isn't enough flash memory to store the
code. The Snom370 has had OpenVPN support for quite a while though.
cheers,
Paul.
--
On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti
mleone...@evolutionce.com wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work
David Backeberg wrote:
You could always use ConfBridge(), starting in 1.6.2.*, which does not
require DAHDI/Zaptel, and therefore doesn't require a timer.
It *does* require a timer (all conferencing requires a timer), but it
does not require a DAHDI/Zaptel timer, there are other options
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need to set it to a known value or insert 0's at the beginning until it
is 10 digits in length.
My
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
like a USB port or
Jerry Geis escribió:
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need to set it to a known value or insert 0's at the beginning until it
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
CLI: show function LEN
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hi Leif,
Thanks for the information. I checked the /tmp/ folder and there was core
files and I tried to back trace it but it was not showing the cause of
that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from
past few days its going on fine. I have also researched and
On Fri, Feb 19, 2010 at 09:21:46AM -0700, Joseph wrote:
Is it possible to split sip.conf into two files (sip1.conf sip2.conf)?
I have an Audiocodes gateway with two FXO ports, and (according to info I
received, and it appears to be correct) Asterisk find the peers based on
their IP
and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI
-- Executing [...@from-internal:1]
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd l...@media.mit.edu wrote:
Hello David,
Thanks so much for your message!
Please check my comments inline below...
David Backeberg wrote:
On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd l...@media.mit.edu wrote:
Hello there,
I'm trying to figure out how
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd l...@media.mit.edu wrote:
How much control do the ssh processes have over the call, if any?
It occurred to me that I might be answering this backwards.
So from the perspective of server A, trying to talk to a remote system
B running asterisk, server A
Joseph wrote:
Is it possible to split sip.conf into two files (sip1.conf sip2.conf)?
I have an Audiocodes gateway with two FXO ports, and (according to info I
received, and it appears to be correct) Asterisk find the peers based on
their IP
and not on their IP+PORT. Thus, Audiocodes with
Hi,
I have a case where SIP channels will not be destroyed, resulting in
further calls to ChanIsAvail() to fail.
The process (I believe) to replicate this is the following:
- Make a call to another SIP phone that is an intercom call (Auto-Answer)
- For whatever reason, the phone happens to go
On 02/19/10 18:38, Edwin Lam wrote:
So I can only use one context for incoming calls. If I split the sip.conf
into two files will it make any difference.
there might be an include directive in sip.conf (i can't confirm)
however Asterisk will see it as one big sip.conf so it will do
absolutely
40 matches
Mail list logo
