similar to: Free Alarms sound

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110516/c4d3be03/attachment.htm>

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

the schedule of my server this configured with -6:00, and this correct one with the normal hour of my country, I made the change but I don't work me [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include

Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in

Dear ALl Can we use Asterisk for only for transcoding?. if yes how many concurent call we can transcode with help of Astetrisk? For this we only need to config SIP.conf or any other file too. Thanks Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100704/f6159f70/attachment.htm

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Amit Shah <amit.shah at redhat.com> writes: > On (Mon) 29 Jul 2013 [14:26:09], Rusty Russell wrote: >> Amit Shah <amit.shah at redhat.com> writes: >> > Remove the debugfs path before freeing port->name, to prevent a possible >> > use-after-free. >> > >> > Reported-by: Jason Wang <jasowang at redhat.com> >> > Signed-off-by:

Amit Shah <amit.shah at redhat.com> writes: > On (Mon) 29 Jul 2013 [14:26:09], Rusty Russell wrote: >> Amit Shah <amit.shah at redhat.com> writes: >> > Remove the debugfs path before freeing port->name, to prevent a possible >> > use-after-free. >> > >> > Reported-by: Jason Wang <jasowang at redhat.com> >> > Signed-off-by:

Amit Shah <amit.shah at redhat.com> writes: > Remove the debugfs path before freeing port->name, to prevent a possible > use-after-free. > > Reported-by: Jason Wang <jasowang at redhat.com> > Signed-off-by: Amit Shah <amit.shah at redhat.com> Applied, Rusty. > --- > drivers/char/virtio_console.c | 3 +-- > 1 file changed, 1 insertion(+), 2

Amit Shah <amit.shah at redhat.com> writes: > Remove the debugfs path before freeing port->name, to prevent a possible > use-after-free. > > Reported-by: Jason Wang <jasowang at redhat.com> > Signed-off-by: Amit Shah <amit.shah at redhat.com> Applied, Rusty. > --- > drivers/char/virtio_console.c | 3 +-- > 1 file changed, 1 insertion(+), 2

On (Mon) 29 Jul 2013 [14:26:09], Rusty Russell wrote: > Amit Shah <amit.shah at redhat.com> writes: > > Remove the debugfs path before freeing port->name, to prevent a possible > > use-after-free. > > > > Reported-by: Jason Wang <jasowang at redhat.com> > > Signed-off-by: Amit Shah <amit.shah at redhat.com> > > Applied, > Rusty. Hey

Hey Rowland, I wanted to ask a. What is mutex.tdb, what is its purpose? b. why "mutex grab failed" started appearing in messages, what can be the probable reasons? Is pre-authentication failed related to locking failure of mutex.tdb? Dec 15 12:49:52 abc winbindd[30467]: [2018/12/15 12:49:52.764924, 0] ../source3/libsmb/cliconnect.c:1921(cli_session_setup_spnego_send) Dec 15

In the case where a virtio-console port is in use (opened by a program) and a virtio-console device is removed, the port is kept around but all the virtio-related state is assumed to be gone. When the port is finally released (close() called), we call device_destroy() on the port's device. This results in the parent device's structures to be freed as well. This includes the PCI regions

In the case where a virtio-console port is in use (opened by a program) and a virtio-console device is removed, the port is kept around but all the virtio-related state is assumed to be gone. When the port is finally released (close() called), we call device_destroy() on the port's device. This results in the parent device's structures to be freed as well. This includes the PCI regions

Remove the debugfs path before freeing port->name, to prevent a possible use-after-free. Reported-by: Jason Wang <jasowang at redhat.com> Signed-off-by: Amit Shah <amit.shah at redhat.com> --- drivers/char/virtio_console.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/char/virtio_console.c b/drivers/char/virtio_console.c index 4e380c1..e910bec

Save the free page info into a page bitmap, will be used in virtio balloon device driver. Signed-off-by: Liang Li <liang.z.li at intel.com> Cc: Andrew Morton <akpm at linux-foundation.org> Cc: Vlastimil Babka <vbabka at suse.cz> Cc: Mel Gorman <mgorman at techsingularity.net> Cc: Michael S. Tsirkin <mst at redhat.com> Cc: Paolo Bonzini <pbonzini at redhat.com>