similar to: Priority + 101 exit

After testing some AGI's, I noticed several extra Asterisk processes. They are not zombies, but can't be killed by safe_asterisk. Nor will they die when CLI issues stop now. Then I read that each AGI spawns a separate Asterisk process. But all my AGI calls have apparently completed successfully. So there should be no reason for them to hang there. Several questions: 1) Under

I mean how do I set failure condition in AGI? My script exits with code 0 upon success, and non-zero when problems occur - the standard *nix way. But Asterisk always report "AGI Script completed, returning 0", and AGISTATUS is always SUCCESS. Yuan Liu

I don't have a particular reason to upgrade, but I'm installing a new box, so I have the opportunity to go 1.4. On the other hand, I'm not familiar with 1.4, and relatively new to Asterisk. So instead of trying to keep up with two different versions, I want to tie my handful of boxes to one, before any of them grow too complex. Is there a document about the main motivations to

I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a

The SM56 MODEM manual says it does. But when used with zaptel 1.2.12, nothing shows up. Yuan Liu

Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. Yuan Liu

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of cracks in my echo, right? I'm not hearing that. Besides, a telephony card is not involved.

My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing FXO calls. If I make a call (from an FXS channel) to a PSTN destination, and the other side answers, Asterisk will show continued ringback on the FXS channel, while the PSTN side hears silence. No error message appears. If a call from PSTN terminates on the same FXO, Asterisk can still ring the FXS channel, and when

Home someone can explain this: a Goto() command can walk within a macro, but if a digit is dialed from within a macro, the call flows back to the context that called the macro. Is there some way to "contain" the flow? Thanks. Yuan Liu

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to fxotune -d. But when trying to take action, it fxotune simply says it can't. ./fxotune -i3 -vvvv Running with parameters: doset=0 docalibrate=1 dodump=0 startdev=1 stopdev=252 calibtype=2 waveformtype=-1 delaytosilence=0 silencegoodfor=18

Is there anything that allows a logical line to extend to the next physical line? Printed files are so hard to read with blind line wraps - and my printer doesn't even automatically wrap. Yuan Liu

How to compile smsq in 1.2? It is compile in 1.4 by default. It is included in 1.2.13, but not compiled. Any rule or method to make it? Yuan Liu

Hi,All, I wang to use AGI in asterisk1.4. AGI file / myperl.agi #!/usr/bin/perl use strict; ...... print STDERR "7. Testing GET VARIABLE..."; print "GET VARIABLE EXTEN \"\"\n"; my $result = <STDIN>; &checkresult($result); ...... when the agi execute; asterisk conosle show that : AGI Rx << GET VARIABLE EXTEN "" AGI Tx >>

Hey, Is there any way to share?MySQL?connection between different agi's.Actually when call comes to asterisk box it executes various agi scripts sequentially. Each script checks various values by making a new?MySQL?connection and then execute query and then disconnects.? So,?Ideally?there should be one connection, and it should be reused between each agi and when a call is over it should be

This is rather bizarre: My TDM11 (one FXS) rings a $10 passive phone with REN of 1.0B, a cheap speaker phone of 0.3B, and a cordless phone with marked REN of 0.0B. But it couldn't properly ring this 27935GE3-B (FCC ID G9H2-7930) cordless phone rated at merely 0.1B. Rarely, the phone will crack out an occasional weak and abrupt beap, but never a normal ring. Otherwise Asterisk and TDM400P

From: Yuan Liu <liuyuan at google.com> Fix a warning thrown from virtio_mmio_remove(): Device 'virtio0' does not have a release() function The fix is according to virtio_pci_probe() of drivers/virtio/virtio_pci_common.c Signed-off-by: Yuan Liu <liuyuan at google.com> --- drivers/virtio/virtio_mmio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git

From: Yuan Liu <liuyuan at google.com> Fix a warning thrown from virtio_mmio_remove(): Device 'virtio0' does not have a release() function The fix is according to virtio_pci_probe() of drivers/virtio/virtio_pci_common.c Signed-off-by: Yuan Liu <liuyuan at google.com> --- drivers/virtio/virtio_mmio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git

I think not. In virtio_pci_common, vp_dev is allocated by kzalloc so a kfree is needed. Here vm_dev is allocated by devm_kmalloc which is "automatically freed on driver detach" from the comment (drivers/base/devres.c:769). On Thu, Nov 24, 2016 at 6:37 PM, Jason Wang <jasowang at redhat.com> wrote: > > > On 2016?11?24? 08:31, Yuan Liu wrote: > >> From: Yuan Liu

I think not. In virtio_pci_common, vp_dev is allocated by kzalloc so a kfree is needed. Here vm_dev is allocated by devm_kmalloc which is "automatically freed on driver detach" from the comment (drivers/base/devres.c:769). On Thu, Nov 24, 2016 at 6:37 PM, Jason Wang <jasowang at redhat.com> wrote: > > > On 2016?11?24? 08:31, Yuan Liu wrote: > >> From: Yuan Liu

I'm writing an AGI script and want it to dial a number on a channel connected to the PSTN. It would look something like this (pseudo-code follows): if ($a){ dial("8005551212"); }else{ dial("8665550000"); } The part I can't seem to get right is the "dial" function. I tried to mimic the dial plan like so sub dial($number){ print