OK, I'll redirect question to users list. Thanks.
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerris, Michael
MI
Sent: Monday, June 13, 2005 9:06 PM
To: Asterisk Developers Mailing List
Subject: RE: [Asterisk-Dev] Keeping users, extensions,voic
On Tue, 2005-06-14 at 00:09 -0400, Jared Mauch wrote:
> Index: chan_sip.c
> ===
> RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
> retrieving revision 1.759
> diff -u -r1.759 chan_sip.c
> --- chan_sip.c9 Jun 2005 22:41:1
so, I have the hitach WIP-5000 (1.5.8) and asterisk
crashes with refer from it because it doesn't include a refered-by
header. while i'm not here to argue the case of the sip-ua, i'd
rather not have asterisk crash.. so please see included diffs
that will keep it from writing a core.
> Irakli Natsvlishvili
>
> Hello,
>
> I have one question regarding *. Default configuration for
> asterisk is to keep configuration(s) in ordinary text based
> config files.
>
> My question is simple: is it possible to keep those config
> info (at least, to start from - sip.conf, extensions
Hello,
I have one question regarding *. Default configuration for asterisk is to
keep configuration(s) in ordinary text based config files.
My question is simple: is it possible to keep those config info (at least,
to start from - sip.conf, extensions.conf and voicemail.conf) on a database,
whic
We use it. Works great!
- D
On Jun 13, 2005, at 10:14 PM, Shanon Swafford wrote:
Take a look at http://www.sipsak.org/
Haven't tried it yet though.
Shanon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nicholas Wong
Sent: Monday, June 13, 2005
Take a look at http://www.sipsak.org/
Haven't tried it yet though.
Shanon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicholas Wong
Sent: Monday, June 13, 2005 10:32 AM
To: Asterisk-Dev@lists.digium.com
Subject: [Asterisk-Dev] sip tester?
I'm work
I'm working on some sip code, but I don't have any way to test it.
Is there a debug app or something where I can send a sip message with
the headers I need to test?
Thanks,
Nick Wong
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cdr.c: In function `submit_unscheduled_batch':
cdr.c:939: warning: implicit declaration of function
`use_ast_mutex_lock_instead_of_pthread_mutex_lock'
cdr.c:941: warning: implicit declaration of function
`use_ast_mutex_unlock_instead_of_pthread_mutex_unlock'
cdr.c: In function `do_cdr':
:P
Hi Franco. I have a PHP script that connects successfully and receives
the responses correctly. I think that it would help you to read the
manager.c file in the Asterisk source code. Also, its difficult to
know why you are not getting the whole response from asterisk, since
the code you show us doe
On Fri, 10 Jun 2005, Ben Kramer wrote:
> Now onto the subject title! I noticed in the ast_frame structure that
> there is a delivery timeval struct. I cant determine if this is a time
> stamp or not. Could I use this to determine if the audio frames I have
> received are old, and are safe to dro
Title: Messaggio
For a presence application, I need to know the
status (free, ringing, talking) of the phones in the sip network managed by
asterisk; to do this I tried to use Manager Api. A
little sample java program connects correctly with Asterisk server and sends the commands to the engin
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