similar to: CallerID disappear from CDR on transfer

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] exten

Has any one used this? U(x[^arg[^...]]): x - Name of the subroutine to execute via Gosub arg - Arguments for the Gosub routine Execute via Gosub the routine <x> for the *called* channel before connecting to the calling channel. Arguments can be specified to the Gosub using '^' as a delimiter. The Gosub routine can set the variable ${GO

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hello all, I need that Asterisk Always use Reason in a CANCEL. How to do? thank you *Fran?ois * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/f3a91f36/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: francois.vcf Type: text/x-vcard Size: 400

Hi All; Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4? Appreciate ur kindly help. Regards Bilal

Dear All. I want to become a wholesale VoIP traffic Provider , and i don't have a experience about the software used this career . I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk telephony server gave me all i need to start my business . thanks -- Best Regards Mohamed Daif -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

Hi everyone, Please forgive me if my question is simple and my code terrible, I'm new to R. I am not looking for a ready-made answer, but I would really appreciate it if someone could share conceptual hints for programming, or point me toward an R function/package that could speed up my processing time. Thanks a lot for your help! ## My dataframe includes the variables 'year',

Suppose I have the following dataframe called test: test<-data.frame(year=rep(1990:2003,5),id=gl(5,length(1990:2003)),eif=as.vector(sapply(1:5,function(z){a<-rep(0,length(1990:2003));a[sample(1:length(1990:2003),sample(1:2,1))]<-1;a}))) year id eif 1990 1 0 1991 1 0 1992 1 0 2000 1 1 1994 1 0 1995 1 0 2001 1 0 1997 1 1 .... I want to create a new variable in

I would lilke to shape upload ftp bandwidth in a dual ISP setup [shorewall show connections] tcp 6 431215 ESTABLISHED src=192.168.2.89 dst=83.xxx.xxx.23 sport=1487 dport=21 src=83.xxx.xxx.23 dst=10.0.11.2 sport=21 dport=1487 [ASSURED] use=2 mark=1 [tcdevices] #INTERFACE IN-BANDWITH OUT-BANDWIDTH $EIF 970kbit 245kbit $LIF 970kbit 245kbit

Hi, I'm trying to create a function, jim(p) which varies depending on whether the value of p is odd or even. I was trying to use th eIf function, but i cant work out a formula to work out if p is odd or even. Thanks, Dave __________________________________________________ Yahoo! Plus For a better Internet experience http://www.yahoo.co.uk/btoffer

Peter Jakobi wrote: > Hi Jiri, > > you already tested an ancient userspace-only rp-pppoe? > I''ve still one active, and trying this you could validate > your the rest of your setup, before looking at possible > kernel changes anew (strace, ltrace, logs, whatever)... > Hi Peter. No I did not. Do I have to compile it? Can you give me more info what to do?

At 2013-08-08 22:28:33,"Tobias Grosser" <tobias at grosser.es> wrote: >On 08/08/2013 01:29 AM, Star Tan wrote: >> Hi all, >> >> >> I have summarized the top 10 compiler passes for Polly when compiling LLVM test-ssuite. Results can be viewed on: >>    

On 08/08/2013 06:27 PM, Star Tan wrote: > At 2013-08-08 22:28:33,"Tobias Grosser" <tobias at grosser.es> wrote: >> On 08/08/2013 01:29 AM, Star Tan wrote: >>> Hi all, >>> >>> >>> I have summarized the top 10 compiler passes for Polly when compiling LLVM test-ssuite. Results can be viewed on: >>>

I've been trying to get an AGI "transfer" to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use "dial" in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer caller to remote extension [Description]

Hello Bug people, I have an unexpected behavior and am unsure whether the problem is in my thinking, my implementation or the program R. Basically I get two different answers depending on how I call a function which takes other functions as arguments as indicated below. To me it should make no difference if f is a function that returns the function g then z(f(x)) whould give the same as y<-

I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's "Direct Bridge" variable. I have

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream