Re: [asterisk-users] DTMF forwarding and Page [SOLVED] [PATCH 1/1]

2012-02-13 Thread Matteo Fortini

Nevermind,
I checked the code, and A* is not using the "F" option in MeetMe for 
Page(), so it's not working by default.

Attached is a patch which fixes the problem for me, if anyone needs it.

Matteo

Il 11/02/2012 13:53, Matteo Fortini ha scritto:

Noone knows that? Where/whom could I ask?

Thanks

Il 10/02/2012 12:30, Matteo Fortini ha scritto:

Hi,
I'd like to implement some way of controlling remote SIP clients 
while in a call, to execute remote commands.


The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is 
connected to the previous one


I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().


TIA,
Matteo
diff -Nurd asterisk-1.6.2.20.orig/apps/app_page.c asterisk-1.6.2.20/apps/app_page.c
--- asterisk-1.6.2.20.orig/apps/app_page.c	2009-01-25 14:35:48.0 +0100
+++ asterisk-1.6.2.20/apps/app_page.c	2012-02-13 13:47:53.509396266 +0100
@@ -164,7 +164,7 @@
 		timeout = atoi(args.timeout);
 	}
 
-	snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
+	snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxFdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
 		(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 
 	/* Count number of extensions in list by number of ampersands + 1 */
@@ -247,7 +247,7 @@
 	}
 
 	if (!res) {
-		snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
+		snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqFxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
 			(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
 		pbx_exec(chan, app, meetmeopts);
 	}
--
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Re: [asterisk-users] DTMF forwarding and Page

2012-02-11 Thread Matteo Fortini

Noone knows that? Where/whom could I ask?

Thanks

Il 10/02/2012 12:30, Matteo Fortini ha scritto:

Hi,
I'd like to implement some way of controlling remote SIP clients while 
in a call, to execute remote commands.


The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is 
connected to the previous one


I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().


TIA,
Matteo


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[asterisk-users] DTMF forwarding and Page

2012-02-10 Thread Matteo Fortini

Hi,
I'd like to implement some way of controlling remote SIP clients while 
in a call, to execute remote commands.


The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected 
to the previous one


I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().


TIA,
Matteo

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