:-
Thanks
Tommaso Calosi
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-08270aa8 and Zap/13-1
Please see the atteched files containing details.
SETUP
SETUP ACK
ALLERTING
CONNECT
CONNECT ACK
DISCONNECT
RELEASE
RELEASE ACK
Asterisk 1.2.22
Zaptel 1.2.19
Libpri 1.2.5
Addons 1.2.7
Sounds 1.2.1
Tommaso Calosi
Jul 18 12:51:48 DEBUG[4200] chan_sip.c
Gordon Henderson wrote:
On Sun, 21 Jan 2007, Cristian Draghici wrote:
IDEfisk is based on iaxclient (iaxclient.sourceforge.net) which will
reject a call with the BUSY signal if there is no available line in
the softphone to take the call.
This means you need to configure IDEfisk to use only
I have this problem with Asterisk 1.2.4 I hear other party's voice only
when I speack or i make some noise. Otherwise i hear nothing. Futhermore
every time i receive a call , this message is displayed : -- Started
music on hold, class 'my_class', on SIP/ some random public ip address
Giorgio Incantalupo wrote:
Hi Tommaso,
have you tried to search for noise suppression? I remember some phone
has a function to automatically suppress it so the caller does not
hear anything and thinks the other party has hung up.
Giorgio Incantalupo
Tommaso Calosi wrote:
I have
Title: Messaggio
My sip phones
are connected to asterisk PBX 1.2.4. The PBX is connected to the provider
through IAX2 connection. Sometimes randomly the voice is stopped and both caller
and called don't hear the other's voice. During this silence period Asterisk is
not logging any errors.
Title: Messaggio
I have a preoblem
with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2
firmware. All have same settings in the advanced panel. On2 phones when I
press the hold or transfer key nothing happens and * does not start the
musiconhold. In the The hold
) it works. The strange thing is that on gxp2000 it works in both
ways.
Bye
and thanks
Tommaso Calosi
-Messaggio
originale-Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Franklin
WebbInviato: giovedì 27 aprile 2006 21.23A: Asterisk Users
Mailing List - Non-
Title: Messaggio
Ihave 13 Snom 320 with asterisk 1.07
bristuffed. The problem is that sometimes on random basis, when one customer is
placed on hold and another call arrives, the customers are put in conference
with each other. This look very strange to me, but I've disabled the confernce
setting.
This is the answer from snom
Dear Tommaso Calosi,
Is the initial problem still present with V5.5?
However Call join on Xfer works like this: When this feature is turned
to on, you will connect an incoming call to, for example, a colleague
you already have on hold by pressing TRANSFER. You
I agree... Snom firmware are buggy. You can use any version, you'll
always find a bug, expecially for 320's. I think instead of developing
the XML minibrowser they should make the phones to work properly.
Dovid Bender wrote:
I was transporting it in my suitcase when I flew from NY to FL. When
I have this problem on misdn 0.2.1:
in extension.conf i have such a situation;
[misdn_incoming]
exten = 06786541,1,Dial(SIP/203)
where SIP/203 is a GXP-2000.
I want to make the 203 to answer just one call at the same time, so i've
disabled the call waiting feature on the phone, but when I do
Guido Hecken wrote:
I looked long and hard at the LAN and it was basically narrowed down to
the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly into the Catalyst did *not*
In your extension.conf, in the misdn context you defined in
/etc/asterisk/misdn.conf you have to add something linke the following line
[from-pstn]
exten = 0108680550,1,Dial(SIP/201)
If you don't want to have to write a string for each called extension,
you can put something like. Obviously
I want that incoming callers to hear a welcome message while the phones
ring. I know I can use Dial with the m(class) option to make the same
with musiconhold, but the problem is that musiconhold does not start
from the beginning of my mp3 file. If I use Playback or Background, the
phones do
Subject: Re: [Asterisk-Users] Playback welcome message while phones
ring,please help
I believe if you use the new native music on hold feature it always
plays the music on hold starting from the beginning.
On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote:
I want that incoming callers to hear
the disable call waiting in asterisk as well, but I have
not been able to find any documentation for this. I have found this
http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting
about call waiting, but it's quite unusefull.
Thanks
Tommaso Calosi
problem with
my queue extensions, and the way to resolve has been to use
call-limit=1 in extensions.
i hope this helps.
Tommaso Calosi escribió:
I'm trying to disable call waiting for Linksys SPA-941, but
unfortunately as far as I have seen, there are no parameters on the
web interface regarding
you can either use the call_limit for each internal or if you wish
something centralized ( for example a maximum total of 30 concurrent
calls ) you can use the superdial macro
http://www.voip-info.org/wiki/view/Superdial+macro
Patrick wrote:
On Tue, 2006-06-20 at 09:20 +0200, bram
I have had the same problem too, I solved resetting the phone to
factory defaults
Edward de Zeeuw wrote:
I'll take a look first thing tomorrow and let you know what I find. Thanks!
Edward
Colin Anderson wrote:
In the Snom web management page under Advanced make sure Challenge response
Who knows something interesting about the new BRI digium card b410p ?
For example, will it use the misdn driver or the native zaptel? Any
interesting links will be appreciated too.
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I think you'll need to set the jumpers on the card in order to specify
the NT ports.
Jean-Louis curty wrote:
Hi everybody
I hope that somebody can help me with the following
I have
2 quadbri cards
2 - 1t0 cards
1 pabx alcatel 4200
I would like to connect my asterisk to the alcatel
Does anybody knows how to transfer calls from Sipura SPA 1001 configured
as asterisk internal ?
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