similar to: OT - log rotation [solved]

Hi, I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? If there is an A->B call going on, I'd like to know which side did the transfer - but whichever side does it, I get back to context

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all

Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone

Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that

Dear R users I have a list, as follows: > intvl.period.myrs $Devonian [1] 4.8 4.2 9.5 5.7 $Ordovician [1] 7.2 5.1 10.2 1.9 $Silurian [1] 4.7 3.0 7.8 2.0 3.3 1.6 2.6 2.7 I want to write a loop that will sum up the values in each part, and give me a vector containing the (in this case 3) summed values this is what I have so far: for (i in 1:length(names(intvl.periods.myrs)) {

Hello Everyone I have an R script (and a source file which I keep my functions) that I need to run on 70 data sets (each consisting of a pair of files). I wish to run these data sets in a computer cluster that is run by my uni (HOWEVER they cannot help me with this problem but say it is do-able) the cluster is clever enough that if i set my data up as follows: within one folder called

Hello everyone I have problem with axis labels on graphs, I have my code as below: plot(0,0,xlim=c(1,ncol(PA)),ylim=c(1,nrow(PA)),main="Stratigraphic Range",xlab="Time Bins",ylab="Taxa",cex.axis=1.5,cex.lab=2,cex.main=2.5,mgp=c(5,1.5,0),xaxt="n") text(1:(length(strat_name)), y= 0, adj=1, srt=45,labels=strat_name,xpd=TRUE, cex=1) #adds text to x

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

The help pages of R-1.2.2 include several pages on various time series functions, but when I try to use these functions they appear not to be available .... am I missing something obvious, or are these functions not yet built? Chris Rogers ----------------------------------------------------------------------- L C G Rogers, Professor of Probability tel:+44 1225 826224 Department of

Hi all R users ! I'm using gamm function from Simon Wood's mgcv package, to fit a spatial regression Generalized Additive Mixed Model, as covariates I have the geographical longitude and latitude locations of indexed data. I include a random effect for each district (dist) so the code is fit <- gamm(y~s(lon,lat,bs="tp", m=2)+offset(log(exp.)), random=list(dist=~1),

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

Can anyone tell me the trick for obtaining the smoother matrix from smooth.spline when there are non-unique values for x. I have the following code but, of course, it only works when all values of x are unique. ## get the smoother matrix (x having unique values smooth.matrix = function(x, df){ n = length(x); A = matrix(0, n, n); for(i in 1:n){ y = rep(0, n); y[i]=1; yi =

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I'm trying to debug DTMF issues I have with certain endpoint conferencing systems (external, 3rd party). On our A*k server I log DTMF, and I see that coming through in the log. What I'd like to see is what is sent onto our VoIP carrier over SIP. I can do a tcpdump of the packets, but what am I then looking for? Would it be in the RTP audio stream or within the SIP