similar to: Extensions in macro

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

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

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

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

In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in the s extension. Goto() is used in examples. Is the prefix "s-" mandatory? Is it related to the original extension "s"? (Apparently Goto(${DIALSTATUS}) won't work for me.) Yuan Liu

Hi I have been asked to ind out if there is a way to use asterisk to answere a voice fax modem so it can provide an answering service and record messages ? -- Gregory Machin gregory.machin@gmail.com www.linuxpro.co.za

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk throws out congestion and hangs up. I set verbose to 6 and debug to 6, but all Asterisk cares to display in console is -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' Are these combinations forbidden? Yuan Liu

I'm still learning some of the basics. Can someone explain in layman's terms what's the difficulty for Asterisk to support SIP/TCP (and even RTP/TCP)?

The variable ${CONTEXT} stores the value of the current context. However if we are in a macro that will be the name of the macro. How do I access the name of the local channel's context. For example: [macro-test] exten => s,n,NoOp(Context ${CONTEXT}) CLI shows: -- Executing NoOp("Local/2592@1100006-2000-e802,2", "Context macro-test") in new stack I want to get

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