ts packets, SIP INFO, ...)
Thanks in advance.
Cesc
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re
or install new apps.
Thanks a bunch!
Cesc
-- Forwarded message --
From: Cesc Santa <[EMAIL PROTECTED]>
Date: Fri, Nov 28, 2008 at 3:26 PM
Subject: Asterisk RTP pager
To: [EMAIL PROTECTED]
Hi,
I came across your "RTPpage" application and just made me very happy.
If
Hi,
I am trying to use an Avaya 4602 phone, which I just updated from a
very old SIP software to the latest I could find on avaya's site
(032207). The upgrade went fine and it gets registered on the Asterisk
server.
Now, a couple of glitches, though.
- The phone's web server is not working ... so
On 8/10/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Fri, Aug 10, 2007 at 11:35:32AM +0200, Cesc Santa wrote:
> > Hi,
> >
> > I have asterisk 1.2.18.
> > I just took a peak at the command: > show translation
> > and I saw that I can only con
inline
On 8/10/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
> On Fri, 10 Aug 2007, Cesc Santa wrote:
>
> >
> > I can use them ... but cannot do translation ... only "pass-through" ...
> it
> > complains that
> > there is no function to
inline ...
On 8/10/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
> On Fri, 10 Aug 2007, Cesc Santa wrote:
>
> > Hi,
> >
> > I have asterisk 1.2.18.
>
> Installed from binary or compiled by yourself?
I compiled it myself ...
> I just took a peak at
hem?
or for each?
How do I get them to work? not just pass-through ... I need conversion.
Thanks a lot!
Cesc
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h
Hi,
I have asterisk 1.2.18.
I am trying to get asterisk to react to an (out of dialog) REFER ...
see below. I get a 603 (no dialog) ... and in the code (sip.conf:3277)
a comment being able but not supporting it??
Any pointers would be great ... is it a configuration option?
REFER sip:[EMAIL PROT
... not even GSM.
Would the Digium G729 license be good both for SIP and H323?
Cesc
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Hi,
I will add the report ... though I find the system a bit cumbersome
for sporadic users like me.
Oh, and you are right ... without chan_h323 asterisk shuts down just fine.
Regards,
Cesc
On 5/8/07, Joshua Colp <[EMAIL PROTECTED]> wrote:
Cesc wrote:
> Hi,
>
> I hope this ge
Hi guys,
I had the same problem ... and then remembered that my asterisk
1.2.9.1 compiled just fine ...
So, i tried that Makefile ... and voila! :)
See attached patch ...
Cesc
On 5/8/07, nik600 <[EMAIL PROTECTED]> wrote:
On 5/7/07, nik600 <[EMAIL PROTECTED]> wrote:
> i am exper
.1.17.3
#16 0xb7e0d4f0 in exit () from /lib/tls/libc.so.6
#17 0x080bddd7 in quit_handler (num=135324439, nice=3, safeshutdown=1,
restart=0) at asterisk.c:945
#18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xb830)
at asterisk.c:1104
#19 0x0809811b in ast_cli_command (fd=1, s=0x8151900 &
time (1:0/0/1)
-- Executing Wait("H323/ip$192.168.1.100:1940/4096", "1") in new stack
-- Executing Playback("H323/ip$192.168.1.100:1940/4096",
"/etc/asterisk/sounds/pbx-invalid") in new stack
-- Playing '/etc/asterisk/sounds/pbx-invalid' (la
I connect the 2 phones
in that one ISDN? and, can i call one from the other in this fashion?
Cesc
On 10/9/06, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
On Mon, Oct 09, 2006 at 10:55:51AM -0500, Ejay Hire wrote:
> Hi. A "cross-over" cable won't work, the isdn network p
s to avoid
the need for an NT connection (via a PBX).
If the above is not possible ... where can I buy a cheap, small,
simple ISDN PBX with at least two NT ports, so that i can connect my
two phones and call each other?
Tks!
Cesc
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inline ...
On 9/28/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Thu, Sep 28, 2006 at 03:24:27PM +0200, Cesc wrote:
> Hello people!
>
> I have an inquiry (not a doubt ;D ). Actually, two.
>
> I am trying to run asterisk on an embedded Power PC platform on which
> we ha
space is taken by the
modules. My question is, can asterisk work with just the chan_sip.so,
chan_h323.so and the codec_*.so? is there any other module needed? I
need only be able to bridge sip to h323, no extra fancy stuff needed
(parking, echo, blah, blah, ... )
Tks a lot!
pre-loaded SIP phones, what part number is the correct? and what about the service contract ... when is it needed?
Thank you very much ...Cesc
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Hi,
Quickie ... is the alcatel ip touch 4068 (or any other in that series)
sip enabled?
If not, does alcatel have a sip-enabled phone?
Cesc
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eds to be rebooted.
huh? i didn't do extensive testing ... but on my 1.2.9.1 it seemed to
work in a stable way ... i would reboot every now and then, but it
would easily resist more than a few calls ...
Cesc
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gnugk.ini and h323.conf file I can
look at to get this all running?
gnugk has no clue of sip.
gnugk is a gatekeeper for h323 ... you can do stuff with the dialled
numbers ... forwarding the call to various [asterisk] h323 gateways
and the like ... but the conversion of h323 to sip is done in
ast
;H323 call).
- in asterisk, h323.conf ... the codec configuration ... i commented
all lines related to it ...
;disallow=all
;allow=all
;allow=gsm
;disallow=g723.1
(just in case)
(again, trial-error)
Cesc
On 6/21/06, Pawel <[EMAIL PROTECTED]> wrote:
Hallo group members
Could You tell me a
... just to make sure i get the best. Any experiences??
Cesc
On 6/20/06, Alberto Sagredo <[EMAIL PROTECTED]> wrote:
Im using several Asterisk Box with chanh323 from asterisk, and it works
fine.
Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability.
A fail (crash) last mont
implementations. And in my particular
knowledge, i know asterisk was not the choice.
Just my 2 cents.
Cesc
On 6/20/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> -Original Message-
> From: Patrick [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, June 20, 2006 12:05 PM
> To:
pped with the latest sources tarball) Or that it
is possible to configure via some obscure setup file?
Thanks!
Cesc
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
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? i have a flat test network, and i would like
the RTP streams to be sent directly end to end (sip phone to h323
phone). It should be possible ... but is it possible with asterisk?
Thanks!
Cesc
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that if it works for you it does for me ...
what version of asterisk you are running? :D
Cesc
On 6/16/06, Tigran Kocharyan <[EMAIL PROTECTED]> wrote:
It should do the job!
In my setup, I call from an IAX phone to an h323 Gateway, and all is
fine. The opposite direction also works fine.
a reduced amount of users (so name mapping
could even be manual ... no problem).
Cesc
On 6/15/06, Gary Richardson <[EMAIL PROTECTED]> wrote:
I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP
setup. It works. There are issues, but that has more to do with Unity
vo
minisip (http://www.minisip.org) - it is LGPL, GPL ... windows support
is in testing status
On 6/15/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
None of those are open source that I recall.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
tell me.
I don't know if it makes a difference, but most of the calls would go
from the H323 side to the SIP side ... but i don't really want to
restrict SIP->H323.
Thanks a lot!
Cesc
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