similar to: PRI Question

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 need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abra?os Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/b8593cb1/attachment.htm

asterisk*CLI> show channels Channel Location State Application(Data) Zap/3-1 (None) Up Bridged Call(Zap/47-1) Zap/47-1 8178062105 at from-nort Up Dial(ZAP/g1/2105||TWK) Zap/25-1 (None) Up Bridged Call(Zap/1-1) Zap/1-1 4999 at from-pri:2 Up Dial(Zap/g2/4999||twk)

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call

Sorry for disturbing, but I sent some messages today and I am not seeing them on this list. Can sombody tell me, in case this message appear on the list. Thank you

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've installed Sven Slezak's Notify module. He gives the follow as an example line to put into extensions.conf exten => s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/ sunnybook) I understand what is going on with this line but I don't know where in the extensions.conf file to put it? Thanks, Chris Griffin cgriffin@33keys.com

I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from '<sip:reg-1@pbx.domain.com> I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20) exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20 ;end macro-forward1 exten =>

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

Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '45629314783bd11604363618632f07b9@201.x.x.x' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received

Hello, I created a new field named pre_dst of type varchar(80) exactly like dst field in cdr table. In the dialplan I put: exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1}) and when I call, all goes fine except that pre_dst has always NULL value in cdr. Do you know why? Is something wrong I did? I know that original fields in cdr are only readable, but in this cas pre_dst is one I created

Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews

Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. ________________________________ This e-mail,

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]:

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

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

Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 "lines" registered w/ the server? 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 ________________________________ This e-mail, facsimile, or letter and any files or

Is there a way to tell asterisk, via a sip.conf peer, what IP address to send a packet out of? I've got multiple NICs in my box, each with it's own public IP. I need the SIP messages to originate from any one of the IPs depending on which number was originally called(and therefore where the packet originally came from). My fear is that it will listen on all IPs fine, but only respond

Originally posted by: mailto: Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I