similar to: Guest registration in SIP

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

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

When Asterisk dials an IAX destination with no registration, it very quickly comes to the conclusion that it can't make the call -- Executing [500@default:2] Dial("Zap/1-1", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default [Jan 29 21:43:15] NOTICE[1957]: chan_iax2.c:2686 __auto_congest: Auto-congesting call due to

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

Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register extension 100 but with wrong password, so registration failed; 3. A then showed it was unregistered! Failed registration of device B shouldn't kick A off, I expect A stay

Call for Participation http://www.hpdc.org/2012/ The organizing committee is delighted to invite you to *HPDC'12*, the /21st International ACM Symposium on High-Performance Parallel and Distributed Computing/, to be held in *Delft, the Netherlands*, which is a historic, picturesque city that is less than one hour away from Amsterdam-Schiphol airport. HPDC <http://www.hpdc.org> is

Call for Participation http://www.hpdc.org/2012/ The organizing committee is delighted to invite you to *HPDC'12*, the /21st International ACM Symposium on High-Performance Parallel and Distributed Computing/, to be held in *Delft, the Netherlands*, which is a historic, picturesque city that is less than one hour away from Amsterdam-Schiphol airport. HPDC <http://www.hpdc.org> is

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

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu