similar to: CDR-CSV Processing

Just FYI: IP address 89.248.168.176 has been trying to use the recently release SIP vulnerability in Asterisk to make outbound calls via our box. They are running a bank account callback scam. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jmann at txhmg.com

I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com

Is there a way to recognize if someone called our PRI using an 800 number? The DID is showing my 4 digit primary line, not anything obvious signifying that an 800 number is called? ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the

Can anyone help me start to diagnose why a Sangoma A200 wouldn't dial out LD? Local calls are fine, incoming is fine, just no LD. Bell tech has been on site and plugged into lines with his test set and was able to dial LD just fine, so it's not a LEC issue. No errors in asterisk console, using zaptel 1.4.11 and sangoma drivers 3.2.6, asterisk 1.4.18 ________________________________

Are the cordless phones on the 480i CT from Aastra registered independently in Asterisk? Such that if I have 5 of the cordless phones hooked up, each one is it's own "extension?" ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended

Is there a way to have a dundi host advertise extensions for another server? A-----------B-----------C I'd like A to reach C through B. A and C would handle the call, B would just be the DUNDi intermediary. Assuming A has 101-199 B has 201-299 And C has 301-399 A sample dundi/extensions/iax config for B is all I need. I can get single DUNDi queries running fine(A->B, B->C,

Is it normal for a PRI to reset the inactive B channels periodically(like once every hour). I'm seeing on my asterisk console successful restarts, just curious as this is all new to me. ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only

Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon? ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further

Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux 2.2.0-rc2 which is what you get when you get dahdi-linux-current.tar.gz Anyone have a workaround or patch? Error below ==================== Building modules, stage 2. MODPOST CC /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.mod.o LD [M] /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.ko make[1]:

Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I send from my PC(on the PC port of the phone) have the same VLAN tag? THe PC is sending untagged packets. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and

Can an asterisk box equipped with a Digium T1 card handle Integrated T1 circuits? I have a T1 with 768k data and the remaining channels voice, can the asterisk box do the Data routing + Voice processing? It's only going to support 4-5 users(the voice channels won't all be active obviously). ________________________________ This e-mail, facsimile, or letter and any files or attachments

I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span 2 sends to my existing phone system(Nortel). My Span1 gets sent to the context from-pri, detailed here: [from-pri] exten => _49XX,1,Set(CALLERID(all)=${CALLERID(all)}) exten => _49XX,2,Dial(Zap/g2/${EXTEN},,twk) exten => _49XX,3,Congestion() exten => _49XX,4,Set(CALLERID(all)="") exten =>

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts

I'm curious if anyone has implemented the following: Need to setup an on-call queue, that activates after 5PM and de-activates at 8AM, also that activates/deactivates on demand(I'm thinking a feature code here). The "agents" need to log in via cell phones, and when calls come in from outside to the asterisk system, it'll need to call the cell phone agents that are active.

I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1. Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass

Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3 Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the "lock-up", IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk. Box is debian w/ asterisk built from scratch. My setup is

I've been considering replacing a PRI with SIP or IAX trunks. The monthly cost difference is marginal, but it would save a bit on the hardware side and soft trunks would be easier to manage. I can't help but wonder what I would be giving up? I'd like to hear some "lessons learned" from those who are doing it or decided, for whatever reason, it's a bad idea.

I am trying to use the sas.get function to bring a SAS dataset into R. After calling the sas.get function I got an error message stating that 'sas' is not a recognized internal or external command, etc. I am guessing that I need to specify the internal command to invoke SAS in the sasprog option for the sas.get function, but I don't know how to determine what that command might be.