similar to: Routing incoming call based on caller id

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you

I have this in my config: exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/1${EXTEN}) exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/${EXTEN}) exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/trunk/1${EXTEN}) exten =>

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

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

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle -------- Original Message -------- Subject: voicemail locked up Asterisk 1.4.13 Date: Thu, 01 Nov 2007 20:57:27 -0500 From: Lyle Giese <lyle at lcrcomputer.net> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> I am running Asterisk

I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

Does anyone have the NPA NXX list for North America in comma delimited file? Only looking for: NPA, NXX, City, State Just seems it should be available free somewhere? Then comes the next step, the setting callerid name based on the NPANXX to city, state -Mark -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.3 - Release Date:

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database with City and State. Actually it's for an Asterisk routing app I'm working on. I see several vendors that want a few bucks to those that want an arm and leg. I expect this is published somewhere by some government agency, but Google hasn't got me to it yet. Jon -------------- next part --------------

I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006 +++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006 @@ -16,6

Does anyone know enough about the implementation of AstDB to know whether the data structure is a Hash function, a Balanced-Tree, a b-Tree, or a Linked List? I'm trying to estimate the lookup 'cost' of a AstDB with around 160,000 keys? Obviously I already know that it WILL WORK, but the question is whether the data structure is optimal in the Berkeley DB AS IMPLEMENTED in Asterisk.

I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called xxxxx:xxxxxxx@iax2.fwdnet.net/18007425877 -- Call accepted by 65.39.205.121 (format ULAW) -- Format for call is ULAW -- Hungup