[ansible-project] Understanding relationship between -K & become vs ansible_ssh_user & ansible_ssh_pass

I am creating a simple playbook to add the current user's public key onto the remote host. My playbook fails with error "Failed to connect to the host via ssh: Permission denied". I am running the playbook as non-root, and I can succesfully ssh to the remote host as root. I run the following

[ansible-project] Understanding relationship between -K & become vs ansible_ssh_user & ansible_ssh_pass

I am creating a simple playbook to add the current user's public key onto the remote host. My playbook fails with error "Failed to connect to the host via ssh: Permission denied". I am running the playbook as non-root, and I can successfully ssh to the remote host as root. I run the

[asterisk-users] Interpreting pjsip.conf

I am looking at the pjsip.conf file shipped with asterisk, and trying to understand it. For example, there are 3 transport-X sections as noted below. Does this mean I could uncomment all 3? Must I uncomment 1? Is the -X portion of [transport-X] arbitrary? ; Basic UDP transport ;

Re: [asterisk-biz] Need help with one-way audio

Hi Don - you are welcome to call Telium for assistance: [logo-75x75] telium T: (519) 266-4357 x270 E: mdup...@telium.ca W: www.telium.ca Confidentiality Warning: This message and any

Re: [Assp-user] assp, the end

Just another opinion - but I too have been stuck on ASSP problems, posted questions, and never got answers. I've learned to live with the problems and just have to weigh the benefits of a broken ASSP installation vs no ASSp installation. I understand the OP's concerns, and understand why he

[asterisk-users] Pass variable to voicemail script

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? --

[asterisk-users] Ast under CentOS 7 - slice messages

I'm building a CentOS 7 Asterisk and find my system log full of messages like this: Mar 5 17:07:01 pbx2 systemd: Started Session 823 of user asterisk. Mar 5 17:07:01 pbx2 systemd: Starting Session 823 of user asterisk. Mar 5 17:07:11 pbx2 systemd: Removed slice user-1001.slice. Mar 5

[asterisk-users] Ast 13 always uses slin internally?

I've ported an Asterisk 10 installation to Asterisk 13, and I've noticed that whenever Asterisk plays my audio files it uses the slin format. I have not converted ANY of my audio files, which means asterisk must be converting my wav files to slin on the fly. Is this the new standard for

Re: [asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user

I don't think you can do this natively within Asterisk, but take a look at SecAst (from http://www.telium.cahttp://www.telium.ca/ ). There is a free edition you can download right from the web site. SecAst will monitor the rate at which a user/device places calls to detect potential fraud.

Re: [asterisk-users] Branch based on call volume

Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re: [asterisk-users] Branch based on call volume On 27Jun, 2015, at 15:34, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: Is there a simple way to get call volume from a particular trunk within the dialplan

[asterisk-users] Branch based on call volume

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -- _ -- Bandwidth

Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

I think you are mixing up answers and general advice. FreePBX was intended to get you over the dialplan creation hurdle (the biggest challenge for people new to Asterisk). In regards to the LinkSys they are compatible and you do find them in enterprises, but admins are trying to get rid of

Re: [asterisk-users] asterisk google contacts

Take a look at the smartCID script available from www.telium.cahttp://www.telium.ca/? It does a web based CID lookup on incoming calls, you can at least use that as a starting point for development... From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Am I cracked?

I'm guessing this is a small/home system? I suggest you install SecAst from this site: www.telium.ca It's free for small office / home office and will deal with these types of attacks and more. It can also block users based on their Geographic location (based on the phone number it

[asterisk-users] Results of security honeypot experiment - scraping for IP's/credentials ?

The results of a security experiment were published this week, in which an Asterisk PBX was set out in the wild to see who would attack it and how: http://www.telium.ca/?honeypot1 What I find particularly interesting is that people/bots are scraping support websites looking for valid IP's of

Re: [asterisk-users] Anonymous SIP calls

Dupuis Cc: Asterisk Users List; byrn...@harte-lyne.ca Subject: RE: [asterisk-users] Anonymous SIP calls On Thu, March 26, 2015 22:29, Michelle Dupuis wrote: You have to consider whether you really want anonymous calls, or you just want to enable SIP calls from trusted companies/partners

Re: [asterisk-users] Anonymous SIP calls

You have to consider whether you really want anonymous calls, or you just want to enable SIP calls from trusted companies/partners. The latter means setting up routes to these companies and (ideally) registration between peers. If you really want anonymous calls, then you will have to setup

Re: [asterisk-users] Asterisk API

As you've probably discovered, most of the API toolkits are half baked and poorly maintained. The Java interface is not great for performance and is suffering from the above too. From our experience (including customer specific and commercial apps) using the AMI directly is the best way to

[asterisk-users] When are /proc/dahdi files created

Can someone tell me when the /proc/dahdi files are created for spans? Are they created when asterisk starts (or the asterisk init script) - if not what script creates them? -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Investigating international calls fraud

Do you have DISA setup? We're seeing lots of attackers running scripts that send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it wasn't a stupid employee forwarding an inbound call to a 9xxx number etc). Have a look at SecAst (www.generationd.com) - it detects

[asterisk-users] Best way to get dahdi status

I'm creating an app that needs to read the status of all dahdi spans and channels, etc. (whatever is needed to tell a user the state of their DAHDI connections). What is the best way to do that? I see dahdi-tools available from the command line, asterisk CLI commands, and AMI commands.

Re: [asterisk-users] SEMI OFF-TOPIC - Fail2ban

I'd suggest taking a look at the free edition of SecAst (www.generationd.com). It handles these messages perfectly (and can also use AMI security events) - so you don't need to constantly be updating fail2ban rules. It's a drop in replacement for fail2ban. -M- P.S. My opinions are my own

[asterisk-users] Reset calls processed counter

When I issue the CLI command 'core show calls' I see how many calls have been processed by Asterisk since it started; eg: 0 active calls 198 calls processed Is there a way to reset the calls processed counter without having to shutdown and restart asterisk? --

Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?

There are lots of ways to solve this, and NOT to solve this. Don't start adding lots of rules to iptables (or deep per packet inspection requirements) as this will hurt capacity...and it doesn't really solve the problem Take a look at http://www.voip-info.org/wiki/view/Asterisk+security If

Re: [asterisk-users] Asterisk secure fine tune - stop attack

You can also take a look at SecAst (www.generationd.com).The free version is a drop-in replacement for fail2ban but also add a lot more intelligence (and no need to update regex's etc). There's also geographic IP fencing so you can block attacks by country / region / city etc., only allow

Re: [asterisk-users] incoming calls fall into echo test mode

You might get a better response on the FreePBX forum. (FreePBX adds pre-built dialplan elements onto standard asterisk. This forum is more for Asterisk) But some suggestions: SSH to your PBX enter the Asterisk CLI set verbose to 10 Call into the problematic number ...and watch where the call

Re: [asterisk-users] Attack on Sip server.

If you have a small Asterisk installation install the free version of SecAst: http://www.voip-info.org/wiki/view/SecAst+(Asterisk+Intrusion+Detection+and+Prevention) For general Asterisk security info check this out: http://www.voip-info.org/wiki/view/Asterisk+security -=Michelle=- All

[asterisk-users] SSL/TLS weakness impact on Asterisk authentication

After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Thanks! --

[asterisk-users] Login by AMI ok, by AJAM fails

I have setup an Ast 11.6 host and I want to login via AJAM. I setup manager.conf, http.conf described in the docs. When I login via the AMI it works fine (see below), but when I login via AJAM the same credentials fail (see further down) Can someone tell me how to fix this? ---

Re: [asterisk-users] Login by AMI ok, by AJAM fails

: Friday, May 16, 2014 3:25 PM To: Asterisk Users List Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails - Original Message - From: Michelle Dupuis mdup...@ocg.ca To: Asterisk Users List asterisk-users@lists.digium.com Sent: Friday, May 16, 2014 2:43:30 PM Subject: [asterisk

Re: [asterisk-users] Login by AMI ok, by AJAM fails

! From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis mdup...@ocg.ca Sent: Friday, May 16, 2014 3:39 PM To: Asterisk Users List Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails You're

Re: [asterisk-users] Login by AMI ok, by AJAM fails

PM To: Asterisk Users List Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails - Original Message - From: Michelle Dupuis mdup...@ocg.ca To: Asterisk Users List asterisk-users@lists.digium.com Sent: Friday, May 16, 2014 3:39:35 PM Subject: Re: [asterisk-users] Login by AMI ok

Re: [asterisk-users] Login by AMI ok, by AJAM fails

actually rawman and manager are very different, and you don't need cookies just to test login. However, I found the problem: I forgot quotes around the curl command. Thanks! -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Asterisk 1.8.22

Another alternative is SecAst (Asterisk intrusion detection system). Grab the free version from www.generationd.comhttp://www.generationd.com/? It does everything fail2ban does, plus you have the option of blocking IP's based on geograhic origin, detecting suspicious call patterns, etc.

[Assp-test] crash on max files limit (when using DB)

I'm running ASSP 2.4.1 (14097), and once a week I find my assp crashed due to too many files open. I have the system limit set to 30 files. At the tail of the assp log I see the following errors: rker_2] Error: Worker_2 accept to client failed IO::Socket::INET=GLOB(0x7f762ec3d060)

[Assp-test] Despite DB: setting still writing files to spam folder

I'm running ASSP version 2.4.1(14097)? and have DB: set for all files/caches. I see the assp database is created, as well as other tables. However, while ASSP is running I still see new message files being created in the spam folder. Why? ASSP has been restarted many times and I startup I

Re: [asterisk-users] Asterisk Call Redirection

These are at completely different levels of the ISO stack...question is making sense to me. (What does it mean to divert a call to a serial port). Do you mean route a call over a link that is ppp/dialup and connected to another endpoint on the other side of that link? If so you would have to

Re: [asterisk-users] Asterisk 1.6

Take a look a SecAst from www.generationd.comhttp://www.generationd.com/ It does everything fail2ban does and more, including blocking users by geography (we exclude all of Asia and Africa), detection of break-in patterns (even if someone guessed your un/pw), detect changes in dial rates, etc.

Re: [asterisk-users] Asterisk 1.6

To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk 1.6 On Friday 04 Apr 2014, Michelle Dupuis wrote: Take a look a SecAst from www.generationd.comhttp://www.generationd.com/ It does everything fail2ban does and more, including blocking users by geography (we exclude all of Asia and Africa

Re: [asterisk-users] Asterisk 1.6

If you know your users are all from with your country, or state, or even city, you could restrict geographic access in your secast.conf file like this: ruledefault=deny ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA The above would: - By default deny all source IP's anywhere

[asterisk-users] Commercial vs Users list (was Asterisk 1.6)

IMHO: If you're announcing a product, selling a product, etc. it belongs on the commercial list. If you're asking/answering questions about Asterisk and the ecosystem I think you can mention commercial products too. (We don't want to pretend they don't exist, and then steer users to only

[asterisk-users] Best zwave controller for MH

I (canadian) store has a deal on for the vera lite controller: http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107sku=VEP-STARTER1 but this looks different than the vera lite green white:

Re: [asterisk-users] Best zwave controller for MH

?oops...wrong list :) From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis mdup...@ocg.ca Sent: Friday, March 28, 2014 5:43 PM To: Asterisk Users List Subject: [asterisk-users] Best zwave controller

Re: [asterisk-users] Security log format / content

: Michelle Dupuis mdup...@ocg.ca To: Asterisk Users List asterisk-users@lists.digium.com Sent: Thursday, March 27, 2014 12:55:21 AM Subject: [asterisk-users] Security log format / content I've noticed that the Asterisk (v11) security log captures attempts do dial without first authenticating

[asterisk-users] Numbers hackers call

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... --

Re: [asterisk-users] Numbers hackers call

To: Asterisk Users List Subject: Re: [asterisk-users] Numbers hackers call On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone

[asterisk-users] Security log format / content

I've noticed that the Asterisk (v11) security log captures attempts do dial without first authenticating, and places the number dialed into the accountid field. I'm trying to distinguish between failed attempts to register and attempts to dial without registering, but the security log treats

Re: [asterisk-users] Replying to Posts

After each line of text, please also dip the corner of your keyboard into your ink well to ensure your writing can been seen. Calling something natural because it used to be that way isn't always correct. -MD- P.S. Notice how little we see PS in posts...now that we can also edit our own

Re: [asterisk-users] High Availability with Asterisk

Some food for thought: If you use DRBD, then you will mirror corruption from one system to another. You also cannot selectively pick files in a folder to mirror (you will mirror a lot!) As well, DRBD struggles as peers are set further apart (latency) or number of changes increases. A lot of

[asterisk-users] Asterisk intrusion detection/prevention, georgaphic IP banning, etc. (new software)

I'm looking for some beta testers to provide feedback on an Asterisk intrusion detection prevention program we're releasing soon. As a quick overview, the program provides: - banning based on geographic location of source IP (Continent, country, region, city, etc) - detection and banning based

Re: [asterisk-users] Telco with multipe SIP servers

Markus, We are developing an Asterisk intrusion detection prevention tool which will allow you to limit connections by geographic region (continent/country/region/city), and include/exclude IP subnets, etc. If you are interested let me know off-list (we're looking for beta testers!).

Re: [Assp-test] fixes in assp 2.3.4 build 14025

Where can we download the new version? The latest version on the website is: ASSP_2.3.3_13335_install.zip (Which is crashing multiple times per day and driving me crazy) From: Thomas Eckardt [thomas.ecka...@thockar.com] Sent: Saturday, January 25,

[Assp-test] Constant crashes

About a month ago I moved to the latest ASSP 2 code, and also switched to MySQL databases for all records. About a week ago ASSP started crashing once a day, now it's a few times a day. I've included the perl dump below in case that helps...but can someone help me resolve this? I have no

Re: [asterisk-users] AMI eventmask question

Of Daniel Jenkins [dan.jenkin...@gmail.com] Sent: Thursday, January 23, 2014 9:03 AM To: Asterisk Users List Subject: Re: [asterisk-users] AMI eventmask question On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: Hi I'm creating an AMI client and I

Re: [asterisk-users] AMI eventmask question

: [asterisk-users] AMI eventmask question On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: That's an interesting link - I didn't know you could set a per user eventfilter in the conf file However, I'm hoping to do this in the AMI connection for more

Re: [asterisk-users] core show channels truncates channel names?

Of Richard Mudgett [rmudg...@digium.com] Sent: Tuesday, January 21, 2014 6:12 PM To: Asterisk Users List Subject: Re: [asterisk-users] core show channels truncates channel names? On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: When I issue a 'core show

[asterisk-users] type=peer vs type=user (depricated?)

I'm looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only use type=peer Is that still correct? Will

[asterisk-users] AMI eventmask question

I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? For anyone else looking...is there a table somewhere online that maps events to their eventmask

[Assp-test] 2.3.3 keeps crashing

I saw yesterday that someone else had the same problem. I managed to capture a traceback (below) - hopefully that helps diagnose the problems. # *** stack smashing detected ***: /usr/bin/perl /usr/local/assp/assp.pl MainLoop - next: Tue Jan 21 18:23:15 2014 terminated === Backtrace:

[asterisk-users] AMI version to Asterisk version mapping

Is there a mapping of AMI versions to Asterisk versions? eg: AMI 1.0 = Ast 1.4 AMI 1.1 = Ast 1.6 etc... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] core show channels truncates channel names?

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)

[Assp-test] stack smashing detected

I recently upgraded to the latest version of assp, and upgraded my perl as well (due to assp shutting down at night). Now I seem to get a more sever crash - coinciding with running the rebuildspamdb. Can someone help with how to fix? I downloaded only the latest assp.pl (do I need to

[Assp-test] Does ASSP generate a 550 response?

Does ASSP ever generate 550 responses to foreign mail systems sending in mail? Or only the mail host behind ASSP? (I have exchange 2007 sitting behind ASSP in case it matters) The reason I'm asking is that I see the occasional 550's in my ASSP log as shown below. I've disguised the domain

Re: [Assp-test] Does ASSP generate a 550 response?

My paste chopped off a few chars, the error in the log is mailbox is unavailable. (in case that helps) From: Michelle Dupuis Sent: Monday, December 23, 2013 5:05 PM To: assp-test@lists.sourceforge.net Subject: Does ASSP generate a 550 response? Does ASSP ever

Re: [asterisk-users] IAX2 bridge failing

: [asterisk-users] IAX2 bridge failing Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment. On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: meant to say restart didn't help either

Re: [asterisk-users] IAX2 bridge failing

Ok just restart -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried

Re: [asterisk-users] IAX2 bridge failing

meant to say restart didn't help either.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.ca] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re

Re: [asterisk-users] IAX2 bridge failing

: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: Some more details...I noticed that the call is bridged, and audio goes one way. However, the dial command still times out after 35 seconds (approx), and exists non-zero. While the channels are up, I did an core show channel xxx

Re: [asterisk-users] IAX2 bridge failing

? Or something I can fix through config? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.ca] Sent: Thursday, December 12, 2013 5:08 PM To: Asterisk Users List Subject: [asterisk-users] IAX2

[asterisk-users] IAX2 bridge failing

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong.One rare occasion it works fine, but usually there is no

[asterisk-users] AMI version vs. AST version

Is there a mapping of AMI versions to Asterisk versions somewhere? For example, Asterisk 1.4 includes AMI version 1.0 (at least that's what I see when I connect to Ast 1.4 via telnet to the AMI port) Also, doe the AMI version changes reflect changes to the AMI commands? If so, is there also

Re: [asterisk-users] Disable peer from AMI

someone tries to use it during the 'off' time. no need for anything as brutal as disabling it in sip.conf. On 2013-10-23 12:37 AM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI

[asterisk-users] Disable peer from AMI

I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... --

[asterisk-users] What linux distro most popular for Asterisk

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm

Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

Gareth: Did you check if your message (or security) log recorded anything during these attempts? If so, can you post the content of the logs during this attack? M From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf

[asterisk-users] Registration failure event from AMI

Is it possible to detect the failure of an agent to register with Asterisk via the AMI ? When I try to register with Asterisk 1.4 using an invalid password I don't see any event in the AMI, but see this in the messages log: [2013-10-05 22:05:03] NOTICE[24598] chan_sip.c: Registration from

Re: [asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example

Be careful with DRDB singe failing drive/corruption on one peers takes down the other too... Check out haast as well (at www.generationd.com) for a commercial asterisk clustering solution. Michelle (GenerationD Systems) From:

Re: [asterisk-users] I need a second opinion on a new phone system deployment

... For redundant/failover of Asterisk checkout HAAST at www.generationd.comhttp://www.generationd.com The HAAST product sits between Linux and Asterisk, monitors for failures etc, and then fails over to another Asterisk box. It effectively creates a low-cost cluster, moving IP's etc to

[Assp-test] ASSP mistakenly returning 5.1.1 user unknown ?

I'm running ASSP version 2.2.1(13020) and when my mail server (box behind assp) is rebooting, ASSP is supposedly responding with a 5.1.1. user unknown (according to the MTA upstream). I checked the assp maillog for the time in question and I see this line: May-21-13 20:20:53 m1-82053-139071

[Assp-test] Why is spam prob score 0

My ASSP uses test mode. causing the subject to be prefixed with [SPAM] which I catch downstream. (in case that matters). The problem I'm having is that mail with a faked from address (different from MAIL FROM) is getting through unmarked. Below is an analysis of such a message. As you can

Re: [asterisk-users] block one number in incoming calls

Check out smartCID on www.generationd.comhttp://www.generationd.com This script allows lookup of incomming calls based on number and either Block (no ring), endless ring (ignore), or pass through to asterisk. It allows allows rewriting of CID name based on number. All numbers stored in a

Re: [asterisk-users] monitoring asteriks

take a look at AsteriskControl script at www.generationd.com This is a free script that monitors, responds to IP address changes, etc. and restarts asterisk. You can also use HAAST (commercial) at same site - it can check for missing registrations etc and restart asterisk too. -=M=-

[Assp-test] Becoming too complicated!

Based on my own experience and that of others (looking at the postings including the current MSGID discussion), I can help but wonder if ASSP configuration has gotten too complex. I'm not suggesting we drop features, but maybe the who way it is configured needs to be rethought!? Instead of

[asterisk-users] Counting calls in progress from AMI

I want to track the number of calls up at any given time, through the AMI. I found the Link and Unlink commands as the most likely candidates - is that the right way? Also, a comment on the wiki suggests that Link may be called several times for a single bridge if transcoding is required.

Re: [asterisk-users] Counting calls in progress from AMI

channels verbose” From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, October 18, 2012 9:58 AM To: Asterisk Users List Subject: [asterisk-users] Counting calls in progress from AMI I want to track the number

[Assp-test] ASSP 2 installation question: moduleLoadErrors.txt

I'm installing ASSP 2 on a fresh new Centos 6 x64 system and am getting pretty far along. However, there are 4 errors (in the moduleLoadErrors.txt file) I can resolve - see below. Here's what i've done: 1. I install perl-Net-SNMP using yum, so why is error #1 there? 2. This module is not

[Assp-test] ASSP not trapping obvious forged signatures

A month ago I upgraded ASSP 2.2.1(12137) (and also moved to a new partition). Since then, my volume of spam getting through ASSP has increased considerably. Attached below is an example - clearly forged sender. Why isn't ASSP trapping this? I attached the analysis of the header below.

[Assp-test] Missing DB_File wrong

I see this message during rebuild: Aug-14-12 06:11:33 warning: 'useDB4Rebuild' is set to on, but 'BerkeleyDB' nor 'DB_File' are available - the rebuild spamdb process uses the internal 'orderedtie' and will possibly require more time and a large amount of memory - check

Re: [Assp-test] Antwort: Missing DB_File wrong

it. Thomas Von:Michelle Dupuis mdup...@ocg.ca An: assp-test@lists.sourceforge.net assp-test@lists.sourceforge.net, Datum: 14.08.2012 15:32 Betreff:[Assp-test] Missing DB_File wrong I see this message during rebuild: Aug-14-12 06:11:33 warning: 'useDB4Rebuild' is set

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

That's how we do it - write to a memory based (ramdisk) disk then write to HDD upon call completion. We haven't tried a SSD but that may be necessary depending on your call volumes. From: asterisk-users-boun...@lists.digium.com

[asterisk-users] Gigaset in the USA

Does anyone know if Gigaset is for sale in the USA? Based on my assessment of phones and features, i would like to try the N300IP base along with C610H phones. I can only find the handsets on ebay, no retailers in USA. And I suspect they are using European frequencies. --

[asterisk-users] Intro to DECT vs IP

We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push

Re: [asterisk-users] Intro to DECT vs IP

: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez [car...@televolve.com] Sent: Friday, June 29, 2012 4:58 PM To: Asterisk Users List Subject: Re: [asterisk-users] Intro to DECT vs IP On Fri, Jun 29, 2012 at 1:22 PM, Michelle Dupuis mdup

Re: [asterisk-users] Intro to DECT vs IP

-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall [aster...@lists.minotaur.cc] Sent: Friday, June 29, 2012 6:27 PM To: Asterisk Users List Subject: Re: [asterisk-users] Intro to DECT vs IP On 29/6/12 11:16 pm, Michelle Dupuis wrote: Can you

Re: [asterisk-users] Intro to DECT vs IP

...@lists.minotaur.cc] Sent: Friday, June 29, 2012 8:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Intro to DECT vs IP On 30/6/12 12:12 am, Michelle Dupuis wrote: I like the look of the C610H. Is there a matching DECT base station by Gigaset? I use the N300IP. Supports 3 active SIP calls I

Re: [asterisk-users] OOh323 log fills with : In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)

you, Vladimir On 6/5/2012 8:58 AM, Michelle Dupuis wrote: We have an Ast 1.6 installation which is connected to an Avaya using ooh323. Something is causing the log to fill with In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log

[asterisk-users] OOh323 log fills with : In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)

We have an Ast 1.6 installation which is connected to an Avaya using ooh323. Something is causing the log to fill with In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log to grow to 300MB in just 5 minutes, which eventually overloads the

[asterisk-users] IAX ATA can't register

I have an ATCOM ATA that is trying to connect to an asterisk server using IAX. The ATA and Asterisk are on the same subnet, not firewall/nat etc. Below is a a log excerpt, showing the REGREQ received, and then Asterisk goes on to send lots of REGAUTH...and this continues for a while, but the

[Assp-test] Still not catching falsified sender domain

I'm still trying to get settings right (and I think I'm close), but ASSP is failing to catch really obviously faked domains! I put the header below, and you can see that 168-226-66-116.speedy.com.ar is pretending to be usps.com. I run my mail through netdorm (and have setup netdorm correctly

[Assp-test] Assp bug spftestmode (was: Still not catching falsified sender domain)

you a clue as to why it is accepting it? On Fri, 30 Mar 2012 09:30:53 -0400, Michelle Dupuis wrote: I'm still trying to get settings right (and I think I'm close), but ASSP is failing to catch really obviously faked domains! I put the header below, and you can see that 168-226-66-116

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