similar to: Knowing incoming call technology and channel [SOLVED]

I've been using asterisk for 3+ years now, I love it, but it doesnt love me back. :-) It was crashing frequently and seemingly randomly prior to this latest upgrade. Not sure what version it was running prior to upgrade (it was probably an old CVS HEAD from 2+ years go.) Anyway, currently running 1.4.21.2. == Problem == Problem is that its crashing for seemingly no reason at all, no

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

Hi all, I know when doing a Dial, when the called party hangs up, we have a few different ways to redirect the calling party to other parts of the dialplan. In this case, I have someone who would like to do the opposite... When the calling party hangs up after a Dial(), redirect the called party to another location. I'm not sure how else to describe what the user wants to do, but I'm

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

Hello, I am a newbie in Asterisk IP PBX but I am very impressed by its functionalities. I have read that It can work over IP network and across the PSTN. I am not very sure how it works over the PSTN.. In case if people have not yet the Internet or SDL access, I would like just to know if it is possible to interconnect two IP LANs using the traditionnal analog Network and Asterisk PBX? In such I

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

> > Thanks Josiah Bryan, > > I do not have any dns server running on my asterisk server, we do have an > external DNS server working in the data center; the IP of this dns server is > 10.4.1.5... > > Following you will see my main configuration: > > /etc/resolv.conf: > > domain localdomain > search localdomain > nameserver 10.4.1.5 > nameserver 10.4.1.2

Hey all - Attached is quick perl script i whipped together this morning to dial from Kontact (well, KAddressBook). You'll need Net::Telnet (perl -MCPAN -e 'install Net::Telnet') for it to work. You'll also need to make sure you've got a valid user/secret in /etc/asterisk/manager.conf on your * server for this to work. Usage: /usr/bin/asteriskdial --from

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

Hello I've tried a bunch of things, but still get errors/warnings when using the IF() function: ============== TEST #1 exten => h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)} ]?${CALLTIME}.wav)}) [Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax IF(<expr>?[<true>][:<false>]) ============== TEST #2 exten =>

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

I'm trying to figure out how to use virsh (or something else) to list all the IP addresses (or MAC addresses if needed) for each virtual NIC, preferably with it domain affiliation also listed. Is this possible? Thanks, JSR/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I'm stuck with a problem using plyr to process a rather large junk of data. What I'm trying to do is applying a moving average to all the subparts of the dataframe (the example data can be found here https://dl.dropboxusercontent.com/u/2414056/testData.Rdata). require(plyr) load("testData.Rdata") applyfilter<-function(x){ return(filter(x,rep(1/5, times=5))) }

Hi there, I tried to use Voicemail from a PRI interface but it didn't work because pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY messages which are normally handled by a bri-stuffed libpri. Unfortunately a wrong if condition stops keypad messages from being passed to the channel driver. The patch attached to this mail removes the 2 lines from the code. Did I

Hi, I'm trying to use RODBC without having to set up a DSN, using hte direct connection string in odbcDriverConnect. My connection attempt looks something like: > odbcDriverConnect(connection = "SERVER=localhost;DRIVER={/usr/lib/odbc/libmyodbc.so};DATABASE=myDB;UID=reader;PASSWORD=insecure;") And this returns the message: Warning messages: 1: In odbcDriverConnect(connection =

Hello list, getting warning : *syntax error, unexpected '<token>'* dialplan : exten => pbx,n,Macro(CheckNetworkProblems,${custID}) exten => pbx,n,NoOp(status = ${STATUS}) exten => pbx,n,GoToIf($["${STATUS}"="congestion"]?backup:nocongestion) CLI : [Sep 9 12:27:07] -- Executing [pbx at cust:15] NoOp("SIP/test13-0000002a",

Dear Prof. Ripley, Thank you for reminding me to the possibility to place the dots as first argument. If this is the solution against partial argument matching, we e.g. cannot safely define a plot method which gives a demo plot in case of no object (because we have to obey the class paramter conventions). plot.someobject <- function(x=NULL, ...) # another example would be paramter pairs

I have asterisk 1.2.18 running on a new system we just installed. Although I've used AGIs many times in the past, I'm stumped on this one. It may just be a simple issue that I need another eyeset to look at. My AGI does the following: #!/usr/bin/perl #Load a few modules... use Asterisk::AGI; use DBI; $AGI = new Asterisk::AGI; #Grab input from Asterisk my %input =