[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 ; ;[transpor

[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 17:07:1

[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 Aster

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.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

List Subject: Re: [asterisk-users] Branch based on call volume On 27Jun, 2015, at 15:34, Michelle Dupuis mailto:mdup...@ocg.ca>> wrote: Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? Do you mean large number of calls or h

[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 and

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 ada

Re: [asterisk-users] asterisk & google contacts

Take a look at the smartCID script available from 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 on behal

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 attempted

[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

ing groups, industry groups, etc. dedicated to VoIP security. They exist for a reason - this is a HUGE problem. It's easy to get over confident and a mistep in security can cost you your job and your company a small fortune. ____ From: James B. Byrne Sent

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 y

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 http://ww

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 cal

[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. What

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 ac

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 i

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 opin

[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! -- __

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 http://

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

users] Login by AMI ok, by AJAM fails - Original Message - > From: "Michelle Dupuis" > To: "Asterisk Users List" > Sent: Friday, May 16, 2014 3:39:35 PM > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > You're right -

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

ed destroy, doing it now! From: asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis 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 right - but I tried

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

Users List Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails - Original Message - > From: "Michelle Dupuis" > To: "Asterisk Users List" > Sent: Friday, May 16, 2014 2:43:30 PM > Subject: [asterisk-users] Login by AMI ok, by AJAM fails >

[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? --- Con

Re: [asterisk-users] Asterisk 1.8.22

Another alternative is SecAst (Asterisk intrusion detection system). Grab the free version from 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. -=

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

[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 non-c

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 i

Re: [asterisk-users] Asterisk 1.6

1.6 On Friday 04 Apr 2014, Michelle Dupuis wrote: > Take a look a SecAst from www.generationd.com<http://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 > pa

Re: [asterisk-users] Asterisk 1.6

Take a look a SecAst from 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] Security log format / content

e log) From: asterisk-users-boun...@lists.digium.com on behalf of Michael L. Young Sent: Thursday, March 27, 2014 2:42 PM To: Asterisk Users List Subject: Re: [asterisk-users] Security log format / content - Original Message - > From: "Michelle Dupuis" > To:

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

?oops...wrong list :) From: asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis Sent: Friday, March 28, 2014 5:43 PM To: Asterisk Users List Subject: [asterisk-users] Best zwave controller for MH I (canadian) store has a deal on for the vera

[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=8930107&sku=VEP-STARTER1 but this looks different than the vera lite green & white: http://www.amazon.com/Mi-Casa-Verde-VeraLite-Controller/dp/B007005364/re

[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] Numbers hackers call

call On 26 Mar 2014, at 15:05, Michelle Dupuis mailto: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 break down that dial string with an explanation? The 011 lo

[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] 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 p

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!). Miche

Re: [asterisk-users] AMI eventmask question

To: Asterisk Users List Subject: Re: [asterisk-users] AMI eventmask question On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis mailto: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

Re: [asterisk-users] AMI eventmask question

boun...@lists.digium.com] On Behalf 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 mailto:mdup...@ocg.ca>> wrote: Hi I&#x

[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 categories?

[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 typ

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

um.com] On Behalf 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 mailto:mdup...@ocg.ca>> wrote: When I issue a &#

[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)" IAX2/FONEMI

[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

Re: [asterisk-users] IAX2 bridge failing

ist Subject: Re: [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 mailto:mdup...@ocg.ca>> wrote: meant to say resta

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 Subjec

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

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

Re: [asterisk-users] IAX2 bridge failing

st Subject: Re: [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

[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 au

[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 a

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" mailto:mdup...@ocg.ca>> wrote: I need to disable/enable a peer after hours automatically, and am thinking about doing s

[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 looki

Re: [asterisk-users] Failed to authenticate user 1000; 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 '"te

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: asterisk-users-boun...@lists

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

... For redundant/failover of Asterisk checkout HAAST at 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 ac

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

Check out smartCID on 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 mys

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=- __

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

mand “core show 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 t

[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. Tha

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-boun...@lists.dig

[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. --

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

aster...@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

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

: asterisk-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:

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

From: 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 Du

[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] 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 t

[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 AT

[asterisk-users] View # active calls in a context

We have a multitenant Asterisk 1.4 installation for multiple small business, and we need to report how many calls a single business has active at one time. Is there a way to VIEW how many calls are up in a single context? (Or some other way to accomplish the same)? Thanks -- __

Re: [asterisk-users] Speech recognition in asterisk using google voice API

Wow - nice! A few quick questions: 1. How long can the recording be for translation? 2. Any limitation on how much text the return (transcribed) variable can hold? 3. Any commercial / terms of use limitations? From: asterisk-users-boun...@lists.digium.com [ast

Re: [asterisk-users] Interesting attack tonight & fail2ban them

1. I checked the log and I don't see any registration attempt, so I *assume* they simply send an invite, and so they are in the external/outside context of my dialplan. So they are trying to reach extensions which don't exist. If they succesfully registered they would be on the internal contex

Re: [asterisk-users] Interesting attack tonight & fail2ban them

...@lists.digium.com] On Behalf Of Andrew Furey [andrew.fu...@gmail.com] Sent: Wednesday, December 28, 2011 11:37 PM To: Asterisk Users List Subject: Re: [asterisk-users] Interesting attack tonight & fail2ban them On 29 December 2011 12:07, Michelle Dupuis wrote: > I thought that it might be worth adding a

Re: [asterisk-users] Client - registers but unreachable

Here is more of a SIP debug log: As you can see Asterisk retries four times but I assume the softphone is not responding? --- Really destroying SIP dialog '637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4' Method: OPTIONS Reliably Transmitting (no NAT) to 172.31.254.53:9653: OPTIONS sip:230bb@17

Re: [asterisk-users] Client - registers but unreachable

The BB is using wifi, on the same subnet as the asterisk server so no need for NAT. There is no keep alive option on the softphone (very simplistic settings) Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] Interesting attack tonight & fail2ban them

fail2ban work fine? Regards On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mailto:mdup...@ocg.ca>> wrote: I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on f

[asterisk-users] Client - registers but unreachable

I have a softphone I'm trying on a blackberry, that registers on my Asterisk, can make outgoing calls, but can't receive calls. There is very little traffic with this phone (see debug below - as the phone registers), and sip show peers confirms it is "unreachable". Any suggestions? Is this jus

[asterisk-users] Interesting attack tonight & fail2ban them

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 2

Re: [asterisk-users] android won't play wav49: how to change format

um.com] On Behalf Of jon pounder [j...@inline.net] Sent: Friday, November 25, 2011 8:03 PM To: Asterisk Users List Subject: Re: [asterisk-users] android won't play wav49: how to change format On 11/25/2011 06:39 PM, Michelle Dupuis wrote: > There is a script on www.generationd.com designed

Re: [asterisk-users] android won't play wav49: how to change format

There is a script on www.generationd.com designed for Asterisk. It will convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc. and then email the message. It's a one line change to add to asterisk - very handy. (We use it for Android phones, nice to see call info in

Re: [asterisk-users] State of Asterisk+Virtualization+Timing

VMware is moving all server products to their ESXi engine. (The old VMware "server" and ESX products are moving to legacy status - with these you could actually do stuff on the kernel). ESXi is no longer a kernel you can mess with, can't install drivers, etc. ESXi is being treated as an appli

Re: [asterisk-users] State of Asterisk+Virtualization+Timing

Although you say "SIMPLE"...not all virtualization hosts allow software installation. On VMware the host has become an appliance you can't really mess with... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Beh

Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?

If one server is supposed to carry the full load of the other during failure, then you have to size each server to handle 100% load - so load balancing is pointless. Checkout haast at www.generationd.com and read the docs on how it does failover...certainly good fo

[asterisk-users] C wrapper for AMI?

Has anyone written a C wrapper to ease development with the AMI? I found a couple of c++ ones, but not C. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live int

[asterisk-users] Controlling max simultaneous calls for a group/.call files

We are building an app that will initiate outbound calls using .call files, and each call can be a different duration (eg: 1min to 5min). These calls will go through an Asterisk service with other calls/apps running. I need to control the MAX number of channels in use so I don't overload this

Re: [asterisk-users] Aastra phone # key in dialplan

If you check the archives you might find the original messages on this topic from a few years ago... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies [davies...@gmail.com] Sent: Wednesday,

Re: [asterisk-users] Aastra phone # key in dialplan

We ran into this a few years ago. Polycoms and Grandstreams worked fine with #xxx extensions, but Aastra's would not. Could not dial extensions beginning with # We chased Aastra tech support for 2 weeks. They acknowledge the bug, and we were told they would fix this in their next firmware re

Re: [asterisk-users] Free CNAM

Cool topic! Our company (generationD) developed some CID scripts for free use, and we would be interested in building and hosting this service. On the spec side, how do we avoid users claiming numbers belonging to others? (Could be an admin nightmare) Do we allow number ranges? Do we require c

[asterisk-users] standalone PRI-to-SIP converter

I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks! -- _ -- B

Re: [asterisk-users] receive faxes

I think the OP's point was that open source should mean: Free to modify Free to contribute code Free to use. Leaving the first two but taking away the "free to use" really takes the F out of FOSS. There have been other posts discussing Digium's license requirements, code ownership, etc. I thi

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