similar to: "Reserved" extensions?

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten=> 501,1,Congestion() exten=> 501,2,Hangup() exten =>h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =>h,2,Hangup() With the above, the call comes into the trigger number, then the call

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

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

Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are

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

On my wild learning curve, I encountered numerous occasions when a channel remained in "Congestion" state after a Congestion() step without going to the next step, which is Hangup(). I couldn't find a definite pattern but it seems to happen when a channel is hung up by the other party or by some other action. Any recommendation about preventing such? 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

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

Greetings everyone, today I spent the last part of my day trying to find a parse error inside this snip: http://pastebin.ca/740081 If there's anyone who can shed some light on why my GosubIf condition is throwing a parse error, I'd greatly appreciate your insight. This was really frustrating and is probably a stupid mistake. Regards, -Michael

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