similar to: Finding the true src in CDR

Displaying 20 results from an estimated 6000 matches similar to: "Finding the true src in CDR"

2003 Dec 09
1
Outbound iax dialing to one #
What I am trying to do is in the 3rd option dial my cell# thru voicepulse I just can't figure how to construct the line [inevans] exten => s,1,setcallerid(${CALLERID}) exten => s,2,Dial(MGCP/aaln/1@Egraph-1,10,tr) exten => s,3,Dial(iax2/passwod@voicepulse.com/ Where do I put the # to dial 18708573287 thanks James Schenck Egraph Design Inc. Arkansas Online Internet Services (870)
2005 Jun 22
1
missing cdr records
I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>
2005 Oct 05
1
Help! Extensions
Hello How do I fix this.... [IPComms-in] exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN01},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN02},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN03},2,SetCallerID(${CALLERID}) exten =>
2004 Nov 22
3
Zap - 256 format frames
Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --
2005 Mar 10
3
SetCallerID({$NEWCALLERID})
I am trying to SetCallerID to a variable I have defined. This obviously is wrong. It actually sets the caller ID to $NEWCALLERID. I have search through the examples on wiki but wasn't able to find something similar to see what I was doing wrong. Could someone tell me the correct way to SetCallerID to a defined variable? exten => 2125551212,5,SetCallerID({$NEWCALLERID}) exten =>
2003 Aug 20
2
PRI CallerID problem
Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can
2005 Oct 13
1
SetCallerID Problem
My number is not submitted. I updated my asterisk but this error still occurs coz of the "" in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack -- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack
2004 Sep 20
2
1 extension entry for multiple purposes?
Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar
2004 Oct 06
1
how does agent logoff if you supply extension?
Per the wiki: Logging off 1. call the extension for AgentCallbackLogin 2. enter your password followed by # 3. when asked for the extension number just press # But if your exten=> is this: exten => 2010,1,AgentCallbackLogin(3333|3044@mycontext) How do they logoff per the wiki's directions? If you use ACBL as above, it never asks you for the extension number because you have
2004 Dec 13
1
incoming call from pstn to fxo not working with Asterisk
When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling
2005 Mar 17
1
Include/Macro not working right...
Hey guys. Thanks for the help on the Pattern matching, I got that working pretty nicely. the next problem I have is that I'm using an include file, but its not really working... In my extensions.conf: [incoming] exten => _NXXNXXXXXX,1,SetCallerID("Unknown Called Number") #include "numbers.conf" exten => _NXXNXXXXXX,3,Macro(Number,1000,${EXTEN}) [macro-Brand]
2005 Mar 08
2
GotoIf with Authenticate
Quick question...Im authenticate all exten except this one(2006). If I call from ext 2006 I still have to authenticate. If I call form any other ext I have to authenticate. Any suggestions? Thanks extex => s,1,GotoIf($[${EXTEN} = "2006"]?3) exten => s,2,Authenticate(731) exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) exten =>
2006 Jan 16
1
modify a cdr values..
Is it possible to modify CDR variables before insert into MySQL or CVS? how? thanks all; Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: alexm.vcf Type: text/x-vcard Size: 334 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/5a095cdc/alexm.vcf
2003 Apr 24
1
CallerID hosed
This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is
2004 Aug 17
6
dialplan woes
I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales
2004 Nov 29
2
Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?
I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten
2004 Jun 29
5
Outgoing CallerID on PRI problems
For outgoing calls made on our PRI circuit we are setting the Caller ID using the format Exten => _9XXXXXXX,1,SetCallerID(1601XXXXXXX) The monitor shows that the CallerID is being set to the specified number, but yet when the call is received on the user end the ID is always the base number of our DID. For example we have 8600-8650 as DID's but the callerid is always 8600 regardless of
2004 Jun 07
2
IAX Won't Pass Caller ID
Hi, We have to servers set up in two different networks. We are able to connect calls via IAX and they work perfectly. We do not see caller ID from clients on either side. Our Grandstream phones say Eri and our XTen phones say Asterisk. We did a debug and I am pasting the output from both servers below. We tried setCallerId in several different ways. We see the value get passed to the
2005 Aug 19
4
Overriding Caller ID
Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible
2005 Jun 19
2
outgoing call routing
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls