similar to: CDR and CallerID

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

hello i am using ast-rad-acc.pl from portaone connected with asterisk manager. my (%cdr) = @_; $cdr{'CALLERID'}, $cdr{'DNID'}, these are empty why these two variables are not working on hangup any comments thanks Kamran Ahamd __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

Hi, Using asterisk 1.4.21.2. For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? -- http://www.lesculturelles.net

I'm looking for some information on how the CDR gets the data for the source and destination on the records. My current system sets the callerid to private, via SetCallerID(Private<>) in the dialplan. Unfortunately, this means there's records in my CDR that have no source on them, and as such I'm unable to bill... Any ideas? Sherwood McGowan -------------- next part

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

Hi, all Sorry for null subject last mail. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database

I've recently noticed some oddities in my CDR records. In some cases the original CallerID that I've set in the .conf file for the extension showed-up in the CDR as the originating extension (on Zap/ devices on the channel bank), and in other places it was the one that I set using Set(CALLERID(num)=<something>) (SIP/ devices). Digging around a little in the source (and doing some

Hi Gang, Running Asterisk 1.4SVN using Polycom 501 phones. Just enabled CallerID and for the most part it works as good as you'd expect anything to from the phone company to. Except: on about 1 out of 10 transfers, instead of getting a callerid of "joe cool <100>" or "abc company <205-555-1212>" I just get "asterisk". Here is what

Hi I have an ISDN connection with 100 DIDs assigned to it... What I'm trying to achieve is set the proper outgoing callerID while showing the local caller's extension in the CDR. There is a behaviour that I just can't explain. the callerid field in sip.conf is set as : callerid="Jean-Yves/E" <300> the callerid in iax.conf is set a: callerid="Jean-Yves/E"

I have been shaking down a dialplan for SIP fax to efax. The basic senario is an ATA on the same subnet as the Asterisk 1.2 box (avoid RTP packet lose and thus fax crash), calling a 'fax extension' and envoking rxfax then email. I leverage off of context: from-internal-additional-custom, so as not to have it overwritten by FreePBX. In extension_custom.conf I have:

Hello, I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1. I've dedicated a context to sipp in my dialplan. Everything works OK expect that calls from sipp comes in with a CallerID set to sipp and this sipp value is stored in CDR. 1. I can change the value of the CallerID but how can I have the calls from sipp traced in CDR with a customized src field value ?

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Both of our production asterisk servers are dumping core when making writes to our cdr tables. Here is a backtrace of the problems we are having: #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at util.c:347 347 if (tds_ctx && tds_ctx->err_handler) { (gdb) bt #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at

When receiving inbound calls from a Vonage Softphone extension, I'm unable to view/maniupulate calledid data. but it shows up in the CDR records and on called handsets.. any ideas? exten => asda,n,NoOp(callerID is ${CALLERID}) exten => asda,n,NoOp(CallerID is ${CALLERIDNAME}) exten => asda,n,NoOp(CallerID is ${CALLERIDNUM}) -- Executing [asd at pstn-in:2]

Let's say i have a network of * boxes connected via IAX, one of them is a "switch", one or more are the "gateways". - An IAX[2] "customer" register himself on the "switch" (and gets an accountcode for te purpose of cdr) - The customer places a call to the "switch", the switch does some magic and decides which "gateway" the call

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

Hi, Asterisk 11 doc says CDR(src) value is read-only (see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). For various reasons, I would appreciate to change its value so that it my own presentation rules instead of telco rules. Very often, I'm connected to telcos through DAHDI (and ISDN). For instance, telco presents calls with 123456789 while I would prefer a normalized

All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 7961G or Polycomm Soundpoint SIP 430 IP phones. The problem is with CallerID showing up in some