[asterisk-users] Asterisk MSN ?

2006-09-22 Thread Yoann Aubineau
Hi list,

Does anyone knows whether Asterisk is able to talk to MSN peers or not,
and if yes to what extend? text-only, audio, video?

Thanks
Yoann

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Re: [asterisk-users] Asterisk AGI question

2006-09-20 Thread Yoann Aubineau
Le mardi 19 septembre 2006 à 15:30 -0500, David R. a écrit :
 Can AGI be used to have a web application talk back and forth between
 Asterisk and itself?  What if the web application is on a separate
 box? 

As Stefan Reuter previously stated, there's no problem running your AGI
application remotely. It's extremely fast (hence the name FastAGI) and
it also frees the Asterisk box from any load your application could
produce (think of calculation or database query processing)

However, even though AGI is a lot like CGI for Asterisk, it doesn't mean
you can use web applications as AGI applications. Ok, you could tell
Asterisk what to do. But how would you get responses back from
Asterisk? 

Or maybe you've got a genius idea I couldn't think of. In that case, let
us know!

Yoann

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Re: [asterisk-users] Asteisk plays music on hold starting from random point

2006-09-20 Thread Yoann Aubineau
Le mercredi 20 septembre 2006 à 18:18 +0200, Giorgio Incantalupo a
écrit :
 Hi,
 I'm using mpg123 to play music on hold but it seems that Asterisk does 
 play the music from a random point: is there a way to make my music on 
 hold always starting from beginning?

Use native format audio (ulaw, alaw, gsm and the likes) not mp3.

[default]
mode=files
directory=/home/asterisk/wdeal-plateform/var/lib/asterisk/moh-native

With MP3 music-on-hold, Asterisk spawns only ONE mpg123 process (or
whatever you mp3player is). Thus, you have only ONE audio stream and all
your users hear the same music at the same time. It's recommended if you
have a huge number of users on hold at the same time.

With native format music-on-hold, Asterisk reads and streams the audio
as if it were a Playback. The music start at the beginning for EACH
user. So they DON'T hear the same sound at the same time. This method is
know to produce better quality sound than with mp3.


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[asterisk-users] Customize host in INVITE's Contact header?

2006-09-13 Thread Yoann Aubineau
Hi list,

I've got to be honnest and admit that I don't know the SIP-related RFC's
by heart. So my question may seem completely heretic to some of you.

What I want to know is whether it's possible to set, somewhere in
Asterisk's configuration files, the host part of the Contact header for
INVITE messages sent by Asterisk.

Currently (I use Asterisk version 1.2.8), I get:
|
|   Contact: sip:[EMAIL PROTECTED]

but instead I'd like Asterisk to send:
|
|   Contact: sip:[EMAIL PROTECTED]

Is this:
1) possible with Asterisk 1.2.8?
2) possible with more recent Asterisk releases?
3) planed for future development?
4) RFC compliant?

In case some of you are interested, I've already got a quirk-and-dirty
patch of my own, against chan_sip.c, to implement that feature. But I
suspect there is something wrong with it: either it could have been
achivied otherwise without having to patch anything, or this patch may
break compatibility stuff that I'm not aware of... See attached file for
more details.

Yoann Aubineau
Index: channels/chan_sip.c
===
--- channels/chan_sip.c (revision 540)
+++ channels/chan_sip.c (working copy)
@@ -343,6 +343,7 @@
 static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;

 static char default_fromdomain[AST_MAX_EXTENSION] = ;
+static char default_contactdomain[AST_MAX_EXTENSION] = ;

 #define DEFAULT_NOTIFYMIME application/simple-message-summary
 static char default_notifymime[AST_MAX_EXTENSION] = DEFAULT_NOTIFYMIME;
@@ -4774,12 +4775,15 @@
static void build_contact(struct sip_pvt *p)
{
char iabuf[INET_ADDRSTRLEN];
+   char *contactdomain = NULL;

+   contactdomain = (default_contactdomain) ? default_contactdomain : ast_inet_ntoa(iabuf, sizeof(iabuf), p-ourip);
+
/* Construct Contact: header */
if (ourport != 5060)/* Needs to be 5060, according to the RFC */
-   snprintf(p-our_contact, sizeof(p-our_contact), sip:%s%s%s:%d, p-exten, ast_strlen_zero(p-exten) ?  : @, ast_inet_ntoa(iabuf, sizeof(iabuf), p-ourip), ourport);
+   snprintf(p-our_contact, sizeof(p-our_contact), sip:%s%s%s:%d, p-exten, ast_strlen_zero(p-exten) ?  : @, contactdomain, ourport);
else
-   snprintf(p-our_contact, sizeof(p-our_contact), sip:%s%s%s, p-exten, ast_strlen_zero(p-exten) ?  : @, ast_inet_ntoa(iabuf, sizeof(iabuf), p-ourip));
+   snprintf(p-our_contact, sizeof(p-our_contact), sip:%s%s%s, p-exten, ast_strlen_zero(p-exten) ?  : @, contactdomain);
}

/*! \brief  build_rpid: Build the Remote Party-ID  From using callingpres options ---*/
@@ -12626,6 +12630,7 @@
default_subscribecontext[0] = '\0';
default_language[0] = '\0';
default_fromdomain[0] = '\0';
+   default_contactdomain[0] = '\0';
default_qualify = 0;
allow_external_domains = 1; /* Allow external invites */
externhost[0] = '\0';
@@ -12749,6 +12754,9 @@
ast_copy_string(default_callerid, v-value, sizeof(default_callerid));
} else if (!strcasecmp(v-name, fromdomain)) {
ast_copy_string(default_fromdomain, v-value, sizeof(default_fromdomain));
+   } else if (!strcasecmp(v-name, contactdomain)) {
+   ast_copy_string(default_contactdomain, v-value, sizeof(default_contactdomain));
+   ast_log(LOG_DEBUG, Value for contactdomain is '%s'\n, v-value);
} else if (!strcasecmp(v-name, outboundproxy)) {
if (ast_get_ip_or_srv(outboundproxyip, v-value, _sip._udp)  0)
ast_log(LOG_WARNING, Unable to locate host '%s'\n, v-value);
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