RE: [Asterisk-Dev] Keeping users, extensions, voicemail and so on in DB

2005-06-13 Thread Irakli Natsvlishvili
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

Re: [Asterisk-Dev] chan_sip crash w/ Refer [patch]

2005-06-13 Thread Jared Smith
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

[Asterisk-Dev] chan_sip crash w/ Refer [patch]

2005-06-13 Thread Jared Mauch
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.

RE: [Asterisk-Dev] Keeping users, extensions, voicemail and so on in DB

2005-06-13 Thread Jerris, Michael MI
> 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

[Asterisk-Dev] Keeping users, extensions, voicemail and so on in DB

2005-06-13 Thread 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.conf and voicemail.conf) on a database, whic

Re: [Asterisk-Dev] sip tester?

2005-06-13 Thread Darren Sessions
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

RE: [Asterisk-Dev] sip tester?

2005-06-13 Thread Shanon Swafford
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

[Asterisk-Dev] sip tester?

2005-06-13 Thread Nicholas Wong
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 ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://li

[Asterisk-Dev] asterisk asterisk.c, 1.155, 1.156 cdr.c, 1.38, 1.39 channel.c, 1.196, 1.197 loader.c, 1.40, 1.41 pbx.c, 1.246, 1.247

2005-06-13 Thread Brian West
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

Re: [Asterisk-Dev] manager api and response data

2005-06-13 Thread Moises Silva
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

Re: [Asterisk-Dev] ast_frame

2005-06-13 Thread steve
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

[Asterisk-Dev] manager api and response data

2005-06-13 Thread francohit
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