similar to: call from ISDN

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten

Hello everybody. I am trying to configure an Asterisk on Debian with the Billion ISDN card. I am using mISDN. But when I call on the CLI apears this: -- Executing Dial("SIP/101-081805b8", "mISDN/1/943833473|45|tTwW") in new stack -- Called 1/943833473 P[ 1] empty_chan_in_stack: cannot empty channel 255 P[ 1] --> we have already send Release_complete == Everyone is

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hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz

Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time.

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten =>

On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote: > On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote: > > > > I can add it onto the end of the variable in the Dial() command: > > > > Dial(${FRED};transport=tcp,${timeout},TtWw); [ the part you trimmed ]: and it has the desired effect, but of course specifying at the point where the Dial command is sets

On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote: > > You specify the transport in the SIP URI. For example: > > sip:test at example.com;transport=tcp Hrm. This is probably going to be pretty basic, but some googling didn't seem to come up with anything. How do you do this when you are assigning to a variable in extensions.conf such as: FRED=PJSIP/outgoing/sip:fred at

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten =>

In the following macro, a call is dialed and control branches according to DIALSTATUS, much like the default std-exten macro. What I'm trying to figure out is how to regain control when the call is answered. ; Standard extension logic [macro-stdexten] ; ${ARG1}=Extension ${ARG2}=Device(s) to ring exten => s,1,NoOp(stdexten ${EXTEN}) exten =>

Hi all, Why subscripting a one column matrix drops one dimension? > x<- matrix(rnorm(100), ncol=1) > str(x) num [1:100, 1] -0.413 -0.845 -1.625 -1.393 0.507 ... > str(x[20:30,]) num [1:11] -0.315 -0.693 -0.771 0.448 0.204 ... > str(x[20:30]) num [1:11] -0.315 -0.693 -0.771 0.448 0.204 ... This breaks: > cov(x) [,1] [1,] 0.9600812 >

Anyone know why chanspy was not included in asterisk distribution as of October. ? I tried patching my current 1.0 but seems the patches are for an older version. I posted a bounty of $250 to get this to work with the newest stable. Needs be able to monitor bridged sip calls with or without a monitoring beep. Thanks John Bittner Simlab.net

Hello, here is my problem, After looking at the mail archives, I found a description of the error I get when I use this package. At first I even tought that they were showing how to solve it. But the thing is that by saying "the programmer forgot drop=FALSE" doesn't show me how I should get rid of the problem I have looked inside the package very quickly and I found three

Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten => 105,1,Answer exten => 105,2,Background(/home/user/suport) exten => 1,1,Dial(SIP/101,30,Ttm) exten => 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option "1" it doesn't redirect to

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hello everybody, i'm from Mexico, at the time i?m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI and