similar to: Local channel characteristics

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

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

I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. Yuan Liu

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

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

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.

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

For a simple call file like Channel: Zap/g1/XXXXXXX RetryTime: 60 WaitTime: 30 Context: from-file Extension: s Priority: 1 I noticed that s@from-file started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only

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

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

On 2016?11?25? 10:47, Yuan Liu wrote: > 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). I see, thanks. Reviewed-by: Jason Wang <jasowang at redhat.com> > > On Thu, Nov 24, 2016 at

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