similar to: Very basic questions

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channel (Context

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hello Everyone, I'm using Mandrake 10 and trying to learng Samba 3.0.2a. I compile the source and install it alright on my Mandrake linux (./configure, make, and make install). Here what I'd done after the installation: Add to /etc/services file swat 901/tcp Add to /etc/inetd.conf file swat stream tcp nowait.400 root /usr/local/samba/sbin swat Since swat binary in the

So I've been testing asterisk under LXC for a few days now and am very happy with the results. My test server is a 1.8GHz Celeron with 256KB cache and 512MB RAM and I have 20 containers each running asterisk (and apache/php,sendmail and a few other minor things) More for fun than anything else, I've tried daisy-chaining instances together - so 20 asterisks running on the same host, 0

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

Hi, I compiled and am running the latest CVS but strange things are now happening.. it looks like asterisk is randomly declaring my calls to be fax calls, complaining and then sending the calls into a black hole... if I hangup the calls below (soft hangup) asterisk locks up and I have to kill the process. NOTICE[21526]: File chan_zap.c, Line 3520 (zt_read): Fax detected, but no fax

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

Hello Sir I am getting problem in plotting in CCA . Could you please help me? I wrote the below command but I don't know why it is taking only first 5 env data rather than all 9. > strain.data <- read.xlsx("Dee rhiz.xlsx", sheetName="strain", header = T, row.names = 1) > env.data <- read.xlsx("Dee rhiz.xlsx", sheetName="env", header = T,

Hi I'm looking for a pinout for the above. Note this has what i'd call RJ45 sockets (or someone smart can correct me). I need to plug into BT (rj13?). And, yes I've googled (glad I'm not chinese) and have tried the suggested, just plug in a 6 connector rj11 and i didnt work atall. On a side note, I cannot beleive that Digium dont have this information on there site, it

I'm new to Asterisk and have been attempting various configurations. I'm having problems with the basics of SIP to SIP phone communications within my own network. I've configured two phones ( Xten X-Lite) and whenever I dial either one I get errors as follows: *auto-congestion SIP/Phone 1 *SIP/Phone 1 is circuit-busy *Everyone is busy at this time I usually don't post such

> > For the record, I just workarounded it in pocl by borrowing the > BreakConstantGEPs code from SAFECode. But for SPIR specs, IMHO, this should > be reconsidered. Yes, I agree. On 24 September 2012 15:08, Pekka Jääskeläinen <pekka.jaaskelainen at tut.fi>wrote: > Well, > > To be honest I'm not very comfortable with the whole constant GEP > idea. It's a

I don't know if the toy program in chapter 4 of the tutorial implementing Kaleidoscope in llvm with C++ is part of your regression suite, but with the version of llvm I installed last weekend, it does not compile: hendrik at lovesong:~/dv/llvm/tut$ g++ -g toy.cpp `llvm-config --cppflags --ldflags --libs core jit native` -O3 -o toy toy.cpp: In member function ‘virtual llvm::Value*

The patch prevents KEIL MDK compile warnings, like:   warning:  #1035-d: single-precision operand implicitly converted to double-precision Actually ARM Cortex-M4F has only a *single precision* (float) FPU. It's suit for all platforms. See the comment at the begin of patch file. Sincerely Forrest Zhang -------------- next part -------------- Specify the floating point constant with single

Micah, Boaz, Do you guys have any ideas about how to fix this issue? Cheers, James On 24 September 2012 16:04, James Molloy <james at jamesmolloy.co.uk> wrote: > For the record, I just workarounded it in pocl by borrowing the >> BreakConstantGEPs code from SAFECode. But for SPIR specs, IMHO, this >> should >> be reconsidered. > > > Yes, I agree. > >

Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

When I type: asterisk -rx "core show channels" I usually get Channel Location State Application(Data) SIP/pstn-4444-000003 7807574622 at internal: Up Dial(SIP/77807574622 at pstn-9998 SIP/pstn-9998-000003 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMITEL123456 1296197222 at entryhome<mailto:1296197222 at entryhome> Ringing