Displaying 7 results from an estimated 7 matches for "doli".
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2004 May 20
8
I have put iLBC at the top
I want use iLBC and have following in mind, please help me is it possible ?
ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM
SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM
1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC
because a lack of codec).
2. SIP incoming codec should be iLBC (snom is ALAW).
3. SIP outgoing codec should be iLBC /snom
2004 Jun 28
5
Modems behind Asterisk - how?
The configuration I'm building replaces an existing PBX with Asterisk. There
are 8 existing modems that people use to call in from the outside to connect
to PC(s) on the inside to transfer data, etc. Callers access these modems by
calling the main number and then dialing an extension for the modem they
want to talk to.
What are my options for supporting these modems with Asterisk? Here are
2004 May 21
3
Asterisk and OH323
Hello,
i want to use asterisk as a gateway for H323-Phones.
But i cant get it work.
I'm using a gatekeeper on another computer. My IP-phone is registered there.
Does anybody can sent me an oh323.conf and extension.conf as examples?
Thanks in advance
Erik Bastian
--
NEU : GMX Internet.FreeDSL
Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl
2004 Jun 11
3
Simplified Voicemail app / keeping peace with cohabitants
Hello,
I have modified the VoiceMailMain application to satisfy the request of
the "local users", i.e., my wife. She lost patience with too many
options (we have one mailbox, so we don't need to forward messages, or
reply to messages, or file them in 6 different folders...) So the
modified app says "Message 1", reads the message, "Message 2", reads...
2004 Jun 20
10
One way audio
Perhaps I was a little too hasty in my conclusions of dysfunctional fax
on the SPA-2000. It turns out I have a one way audio problem on line
one of my SPA-2000. I have all the correct settings according to the
comments in the wiki, but the problem persists. However, if I do a hook
flash out of and back in to the call that isn't transmitting audio, it
works fine. My sip.conf entry for the
2004 Apr 21
1
rsync-2.6.1pre-1 hang
...ver process
can only be killed by SIGKILL; no timeout occurs on server side.
ssh is not being used for sync - only pure rsync :)
I do not have necessary skills to debug the problem, so am asking you
for help.
Below please find attached backtrace on server side of rsync:
Attaching to program: /home/doli/progz/rsync-2.6.1pre-1/rsync, process 30131
0x2a344742 in ?? ()
(gdb) bt
#0 0x2a344742 in ?? ()
#1 0x2a3a2d20 in ?? ()
#2 0x00000020 in ?? ()
#3 0x598062b0 in ?? ()
#4 0x0805a150 in make_file (fname=0x0, flist=0x0, exclude_level=69) at flist.c:879
#5 0x0805a3bc in send_file_name (f=3, flist=0...
2004 Jun 04
9
MYSQL asterisk configuration
Hello all.
I am a little (allot) lost on my next hurdle in getting an asterisk
system built.
I would like to get my asterisk servers configured exclusively from
database. I have read through the wiki on this subject but once again I
find that there is a certain level of knowledge that is assumed. As of
now I know nothing about databases in general and specifically MYSQL. I
do not know