similar to: Asterisk brute force watcher (was FYI)

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

Top of the morning all... So I reworked the pseudo IDS/Brute Force Asterisk script for those who want to either use it, or use it as a baseline to build a better one... The script now does a few things... It logs those with password issues, and blocks them as well. This was done to ensure that a remote user who was blocked can be found in the log. E.g., Sally the homemaker keeps fiddling

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone

Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061122/054eb688/attachment.htm

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

So I was bored yesterday and tried solving a few problems with one stone: 1) Notify me of potential brute forcers (multiple attempts to register multiple numbers from one address) 2) Notify me of (l)users who are having password issues So I whipped up a simple script to run in cron and notify me that UserX from X_IP_Space had X amout of password issues. I'm currently running this from cron

Dear list, following problem, i have some users, who are supressing their callerid. This setting is adjusted at the sip phone. So if these guys are calling internal persons nobody sees the callerid. I am looking for the following resolution: User has set his phone to anonymous, user calls somebody internal, Asterisk initials a lookup on the channel and generates a new callerid for the

Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also,

problem - occasional garbled calls, mostly remote users. T1 connected to PSTN, SIP over local LAN and internet to "remote users". NAT at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most are Polycom phones, two Aastra's. Start with QoS on LAN switches? No 2x4's please, start with 1x4's. joe a.

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

Anyone running caller id spoofing applications in the USA running asterisk? Then it's time to move them to Canada or similar. http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-t o-be-outlawed.html Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial).

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

So I have my extensions.conf (http://www.infiltrated.net/exten.stupidity.conf) shortened in case someone wants to look. Has someone encountered the following? I've racked my brain on this for too long... I have two contexts, day and night... Caller (Daytime) --> Dials an extension --> Caller hears extension ring on receiver --> Call goes through Caller (Night) --> Dials an

Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails -- #Joseph

Has anyone gotten Asterisk to compile on Solaris 10? I have tried both 1.2and 1.4 and I get errors about editline. Actually it seems that 1.4 goes through more of the process, but thats not good enough.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061116/82464f01/attachment.htm

I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant