search for: oneliners

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

Dear All I have a problem with a TDM400 card with 4 x FXS modules. The card carries extensions only and there are no incoming lines. I can make a call to the extension on this card with no problems. However, when I try and call out I just get a busy signal. I also get an error message (as shown at the bottom). Is this a problem? Configs below: [root at asterisk etc]# more zaptel.conf

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hi, I noticed that it takes around 5 sec before the phone hang up immediately if SIP response 486 "Busy Here" was received. How to change it so that it will hangup immediately. >From the asterisk CLI, I am seeing ocalhost*CLI> -- Executing Macro("SIP/6200-70bb", "oneline|SIP/6203") in new stack -- Executing

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

On Tue, Jan 7, 2014 at 8:48 PM, Sean Silva <chisophugis at gmail.com> wrote: > sean:~/pg/llvm/llvm % git log --oneline --since='1 month ago' | wc -l > 706 > sean:~/pg/llvm/llvm % git log --oneline --since='1 month ago' ./test | wc -l > 317 Wouldn't this also catch commits to code generation that added tests as well? Diego.

Has anyone had any problems with the voicemail app in the last weeks cvs? I was using a checkout from 10-31 and everything worked fine, but the last few days I get a hangup right after it calls the voicemail app. No sound, just an immediate disconnect. Nothing has changed in my extensions.conf, etc. Here is what I have in extensions.conf: [macro-oneline] exten => s,1,Dial(${ARG1},20,tr)

http://pastebin.com/raw.php?i=rykHdvBh bix.hu and www.yahoo.com are "pingable" test sites. 127.0.0.1 could not be pinged [firewall drops all icmp] i have a "oneliner" that echoes if theres "internet connection or no". $ ping -W 1 -c 2 bix.hu &gt;&amp; /dev/null &amp;&amp; ping -W 1 -c 2 www.yahoo.com &gt;&amp; /dev/null &amp;&amp;

Greetings, DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) "registers", then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on my desktop.I think), and then * goes to vmail. Here's the debug output: affinity*CLI> iax debug IAX Debugging Enabled Rx-Frame Retry[N/A]

Hi, I have two local SIP extensions (both bt100). One is on remote location behind another nat (16), but everyithing seems to be setup correctly as it can register and is listed as OK(57ms). However I can only call in one direction between those two. Extensions are defined in same context: exten => 11,1,Macro(oneline,SIP/11) exten => 16,1,Macro(oneline,SIP/16) both using same macro

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

Hi, I would like to announce a new set of LNT -O3 performance testers. In a discussion titled "Question about results reliability in LNT infrustructure" Anton suggested that one way to get statistically reliable test results from the LNT infrastructure is to use a larger sample size (5-10) as well as a more robust statistical test (Wilcoxon/Mann-Whitney). Another requirement to

On Wed, Jan 8, 2014 at 4:07 PM, Sean Silva <chisophugis at gmail.com> wrote: > > > > On Wed, Jan 8, 2014 at 7:58 AM, Diego Novillo <dnovillo at google.com> wrote: >> >> On Tue, Jan 7, 2014 at 8:48 PM, Sean Silva <chisophugis at gmail.com> wrote: >> >> > sean:~/pg/llvm/llvm % git log --oneline --since='1 month ago' | wc -l >>

I have a file in stata format, which I have read in, and I am trying to create a text file. I have exported the data using various delimiters, but I'm unable to read it back in. I originally read in the file with: library(foreign) myData <- read.dta("mydata.dta") I then exported it with write.table using comma, tab, and exclamation marks as a delimiter. When I was unable to

As first discussed in the giant "New LLVM git repository conversion prototype" thread, I've pushed a git tag that allows git-describe to be used for generating human readable commit names on master. The tag is "llvmorg-10-init", to be interpreted as "the initial commit with major version number 10". The idea is to add similar tags along with future version bumps.

I tried some of the scripts in the toolkit (opensnoop, execsnoop, dtruss), and I keep seeing errors like this: dtrace: error on enabled probe ID 3 (ID 113: syscall::exece:return): invalid kernel access in action #5 at DIF offset 0 Anyone know what could be the problem? TIA This message posted from opensolaris.org

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension