hallo,
i experienced codec translation problems between my sip adapter and
different sip providers
i use a grandstream ht286 sip adapter and an asterisk server, the
asterisk server is registered with two sip providers, sipgate and
voipbuster, i would like to use ilbc o. g726 because of bandwith,
hallo,
maybe somebody could help me, i try to bring my asterisk server to
native bridge two iax2 channels, on my old asterisk server (Asterisk
CVS-v1-0-03/23/05-10:07:13) it is working,
since i have installed the latest cvs code, asterisk stayes always in
the middle of my iax clients? i used
hallo,
i just experienced that all meetme rooms share the same voice data, if i
connect to 499, it could be heard in all other rooms (498,500, 501)
could sombody help me, why does asterisk send the voice out of all rooms if
i only connect to one?
thanks for help,
alex
meetme.conf
conf =>
hallo all,
could sombody please help me,
i dont know why nativ bridging is not working when i choose the ilbc codec,
with speex it is working,??
iaxcomm (ilbc) --->asterisk --> ( asterisk2 --> sip grandstream (alaw) )
\-native bridge--/
1. if i use
hallo,
i tried to setup my extentions,conf like this but it never jumps to the
busy part (102)
asterisk always plays the unavail msg, also when i am connected to another
iax channel (conferece room) and no more channel on my client is available.
could sombody give me a hint what could be wrong?
hallo,
i trying to dial with a python script via the manager interface, it works
ok but i would like to send a soud file name as a variable to the dialplan,
so that i can call a number and send it a different soundfile i choose in
my pyton script.
the problem is, that the * dials correct and se
hallo,
does sombody know how to get the callerid from iax.conf ( callerid="name
"<1234>) via the manager interface?
Action: IAXpeers
gives only the Name/Username but not the call number?
any ideas how to do this?
thanks,
alex
___
Asterisk-Users mailing
hallo all,
i have a quicknet LineJACK card and it seems to work ok, the only problem
is, that when i use this in extentions.conf,
exten => _[1-9]., 1, Dial(IAX2/krath:[EMAIL PROTECTED]/${EXTEN},50,Ttr)
exten => _[1-9]., 2, Congestion
it dials only 2 digits, e.g when i dial 1234 it dials only 12,
re is what my asterisk server shows during connection (unable to transfer):
tahnks,
alex
Connected to Asterisk CVS-v1-0-12/02/04-14:33:02 currently running on snd
(pid = 3792)
Verbosity is atleast 5
-- Registered 'atuc' (AUTHENTICATED) at 82.82.238.221:30512
== Parsing '/etc/a
re is what my asterisk server shows during connection (unable to transfer):
tahnks,
alex
Connected to Asterisk CVS-v1-0-12/02/04-14:33:02 currently running on snd
(pid = 3792)
Verbosity is atleast 5
-- Registered 'atuc' (AUTHENTICATED) at 82.82.238.221:30512
== Parsing '/etc/a
hallo,
could somebody help me, i would like to select ilbc as preferred codec but
dont disable gsm totally, i can only make a call with ilbc if i disable the
gsm codec in iax.conf,
if i enable gsm and make call to the same enpoint, always the gsm codec is
choosen as audio codec.
any idea whats
At 20:22 02.11.2004, you wrote:
what's the package to fetch? debian?
f.
on testing:
ap-get install libspeex1 speex libspeex-dev
on stable you have to compile from source,
alex
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/ma
At 19:17 02.11.2004, you wrote:
At 17:45 02.11.2004, you wrote:
Atuc [EMAIL PROTECTED] wrote:
> does anybody know, how to enable the the new iaxcomm client (with speex
> codec!!) to work with asterisk?
>
> i get a "Unable to find a path" error??
>
> i have enabled s
At 17:45 02.11.2004, you wrote:
Atuc [EMAIL PROTECTED] wrote:
> does anybody know, how to enable the the new iaxcomm client (with speex
> codec!!) to work with asterisk?
>
> i get a "Unable to find a path" error??
>
> i have enabled speex in iax.conf,
>
Have you
hallo,
does anybody know, how to enable the the new iaxcomm client (with speex
codec!!) to work with asterisk?
i get a "Unable to find a path" error??
i have enabled speex in iax.conf,
thanks for help,
alex
Nov 2 15:37:19 NOTICE[281616]: channel.c:1698 ast_set_write_format: Unable
to find a pa
At 19:19 26.09.2004, you wrote:
Yes We have one... Its going to be released in October... We are in middle
of Moving so once we move and get things settled we will be going online
Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has
many more abilities.
We are releasing softwar
At 10:05 21.09.2004, you wrote:
hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see
below lsmod),
does anybody has an idea what is wrong?
many thanks for help,
alex
sorry, i found teh proble
hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see
below lsmod),
does anybody has an idea what is wrong?
many thanks for help,
alex
snd:/usr/src/zaptelrtc# make
cc -c zaprtc.c -D__KERNEL__ -
At 19:43 15.09.2004, you wrote:
if you're using ppp, you can add an asterisk reload command in your
ip-up.local
how does it look like? do you meen over the manager server? any infos on
the net?
thanks,
alex
___
Asterisk-Users mailing list
[EMAIL PROT
hallo,
my asterisk server is working fine, the only problem i have is, that every
morning when i look at my sipgate page i am logged out, when i do a cli
reload, everything is working until the next morning.
so my question, how can i force my asterisk server to keep logged in at my
sip provider
hallo all,
i am new to asterisk, just started my first tests and tried to build up a
phoneserver with serveral iaxcomm clients running over iax2.
my question is:
now, it is possible to reach all phones over the asterisk server, but the
media stream is always routed through the server.
is it also
21 matches
Mail list logo