search for: video_mod

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single vid...

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show profile bridge default_bridge" I see: Video Mode: no video I can change it to follow_talker, last_marked or first_marked and it does change, it is just the sfu option that does not seem to be valid. I am using Asterisk 15.1.2 fo...

...havez wrote: >> I am trying to get the "Mega Phone" demo working on my office PBX >> but there seems to be a problem when trying to set the default bridge to >> sfu mode. I have the following configuration in confbridge.conf in the >> default_bridge section: video_mode = sfu but when I do a "confbridge >> show profile bridge default_bridge" I see: >> >> Video Mode: no video >> >> I can change it to follow_talker, last_marked or first_marked and >> it does change, it is just the sfu option that does no...

...Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >> I followed the blog post and I can get video from the conference if >> I configure the bridge as follow_talker so I know everything is working >> on the pjsip side. The only problem is that video_mode = sfu is >> apparently not valid in either confbridge.conf or via the dialplan and I >> get no video with that option. > The option, when set, will show up as "no video" if you do the > "confbridge show" as you mentioned. That's a bug which is why I > m...

...On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>>> I configure the bridge as follow_talker so I know everything is working >>>> on the pjsip side. The only problem is that video_mode = sfu is >>>> apparently not valid in either confbridge.conf or via the dialplan and I >>>> get no video with that option. >>> The option, when set, will show up as "no video" if you do the >>> "confbridge show" as you mentioned. That...

...;syslinux/loadfile.h> #include <syslinux/linux.h> #include "cpuid.h" int main(int argc, char *argv[]) { const char *kernel, *initrd; void *kernel_data, *initrd_data; struct initramfs *rd; size_t kernel_len, initrd_len; char cmdline[1024]; uint32_t mem_limit = 0; uint16_t video_mode = 0; int i; unsigned p; char _b[64]; #define pause() (fgets(_b, sizeof _b, stdin)) s_cpu cpu; openconsole(&dev_stdcon_r, &dev_stdcon_w); for (i = 0; i < argc; i++) { printf("argument %d: '%s'\n", i, argv[i]); } pause(); if (argc < 5) { fprintf(stde...

...token_index[1]; -unsigned long symbols_markers[1]; +const unsigned int symbols_markers[1]; --- 2009-07-10.orig/xen/include/asm-x86/config.h 2009-07-13 13:56:07.000000000 +0200 +++ 2009-07-10/xen/include/asm-x86/config.h 2009-07-13 14:01:35.000000000 +0200 @@ -122,10 +122,12 @@ extern unsigned int video_mode, video_fl #define PML4_ADDR(_slot) \ ((((_slot ## UL) >> 8) * 0xffff000000000000UL) | \ (_slot ## UL << PML4_ENTRY_BITS)) +#define GB(_gb) (_gb ## UL << 30) #else #define PML4_ENTRY_BYTES (1 << PML4_ENTRY_BITS) #define PML4_ADDR(_s...

...the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found...

...the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found...

...pointer is valid but meaningless. > */ > static const char *find_argument(const char *cmdline, const char *argument) > { > @@ -166,6 +169,7 @@ int bios_boot_linux(void *kernel_buf, si > struct syslinux_memmap *amap = NULL; > uint32_t memlimit = 0; > uint16_t video_mode = 0; > + uint8_t bootflags = 0; > const char *arg; > > cmdline_size = strlen(cmdline) + 1; > @@ -200,6 +204,14 @@ int bios_boot_linux(void *kernel_buf, si > } > } > > + if (syslinux_filesystem() == SYSLINUX_FS_PXELINUX && > +...

...For boolean arguments, the returned pointer is valid but meaningless. */ static const char *find_argument(const char *cmdline, const char *argument) { @@ -166,6 +169,7 @@ int bios_boot_linux(void *kernel_buf, si struct syslinux_memmap *amap = NULL; uint32_t memlimit = 0; uint16_t video_mode = 0; + uint8_t bootflags = 0; const char *arg; cmdline_size = strlen(cmdline) + 1; @@ -200,6 +204,14 @@ int bios_boot_linux(void *kernel_buf, si } } + if (syslinux_filesystem() == SYSLINUX_FS_PXELINUX && + find_argument(cmdline, "keeppxe")) { + e...

...all $0xc000, $3 + movw %cs, %ax # In case messed by BIOS + movw %ax, %ds + movw %ax, %ss # Need this? How to ret if clobbered? + +1: + # for acpi_sleep=s3_mode + testl $2, wakesym(video_flags) + jz 1f + movl wakesym(video_mode), %eax + call mode_setw + +1: + # Show some progress if VGA is resumed + movw $0xb800, %ax + movw %ax, %fs + movw $0x0e00 + ''L'', %fs:(0x10) + + # boot trampoline is under 1M, and shift its start into + # %fs to referenc...

Hello, This series has been split into two patches, one for arm and one for x86. I figured that this was easier than doing it as a single combined patch, especially as the changes are functionally independent. x86 has been boot tested, but arm has not even been compile tested as I lack a suitable cross compiler. However, the changes are just text replacement, so I dont expect any issues. The

...simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Re...

...when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTERISK-28995 - res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) * ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) * ASTERISK-28978 - acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) * ASTERISK-28975 - res_http_websocket: Text payload data doesn't necessary include trailing zero...

...when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTERISK-28995 - res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) * ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) * ASTERISK-28978 - acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) * ASTERISK-28975 - res_http_websocket: Text payload data doesn't necessary include trailing zero...

...on a 150dpi display and because I've not modified the background loading code). Would it be interesting to develop it further or would it be considered overbloat ? If yes, I have the problem of choosing the "right" video mode: should I provide a kind of "switch (video_mode)" option ? Something along the line: MENU VIDEO 1920 1200 [anything here executed if width>= 1920 and height >= 1200] MENU VIDEO 1280 1024 [anything here executed if width>= 1280 and height >= 1024 (but either width < 1920 or height < 1200)] MENU VIDEO x y [anything here...

...ssaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found...

...ing support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be f...

...ssaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found...