Re: [asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

Answering the below for search engine’s sake. > On Feb 18, 2019, at 11:23, Matt Riddell wrote: > > Hey, trying to use ARI with NodeJS - this doesn't work: > > play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav' > <http://www.nch.com.au/acm/8k16bitpcm.wav'&g

[asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

Hey, trying to use ARI with NodeJS - this doesn't work: play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav'); should it? https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation says: A sound file located on the Asterisk system. You can use the

Re: [asterisk-users] [asterisk-app-dev] ARI-client Node.js objects

> On Jan 11, 2019, at 10:46, Gilles VERRIEZ (SERENEO) > wrote: > > Hi, > > I would like to get the audio resource from a record in order to send it > threw AJAX request with my ARI-client Node JS source. I thought > Playback.media_uri could help me but it's value is undefined. Any ideas? >

Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

> On Jan 11, 2019, at 11:14, Jean Aunis wrote: > > Le 11/01/2019 à 16:47, Matt Riddell a écrit : >> Hiya, >> >> When I hang up on a call to my stasis app I’m getting multiple >> channelDestroyed events for the same channel: > > It may happen if several

[asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

); Is this normal? I’m writing like a CDR on channel destroyed so don’t want to write it multiple times. Should I keep an array of channels and only write if I haven’t seen the event for that channel before? Cheers, Matt Riddell ___ asterisk-app-dev

Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

> On Jan 7, 2019, at 12:25, Joshua C. Colp wrote: > > On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote: >> Hiya, >> >> I would have expected this to show the channels in the bridge inside >> the anonymous function - it shows the bridge is empty though?

Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

> On Jan 7, 2019, at 12:25, Joshua C. Colp wrote: > > On Mon, Jan 7, 2019, at 1:23 PM, Matt Riddell wrote: >> Hiya, >> >> I would have expected this to show the channels in the bridge inside >> the anonymous function - it shows the bridge is empty though?

[asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

ges depending on agent/customer status etc Cheers, Matt Riddell ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/li

Re: [asterisk-users] shell dialplan application blocking

Use AGI Kind regards, Matt > On Jun 4, 2018, at 02:28, Benjamin Marty wrote: > > I'm calling a script which needs to wait a certain time and also hold the > call for this time. But the script dialplan application seems to work non > blocking. Is there a way to hold the call/dialplan till

Re: [asterisk-users] Asterisk chan_sip registration attempts

Maybe the provider has added an extra gateway and it is not processing accounts correctly. If they had one before and now two then 40-60% registration fails would show that. Kind regards, Matt > On Oct 10, 2017, at 06:27, Dmitriy Ermakov wrote: > > Hello! > > Could

Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition… so many questions!

nds of silence and caller > annoyance. At least in older versions you can use EAGI to get a handle to the audio stream. You can then pipe that stream to something like bluemix using Node.js and have a handle to the incoming recognition in realtime too. -- Cheers, Matt Riddell _

Re: [asterisk-users] OT: Want to capture all SIP messages

; >> Solvable by by writing a cleanup script that deletes files over a >> specific age, just a basic find in the daily crontab: >> find /path/to/captures -type f -name 'pattern*' -mtime +X -exec rm {} \; > > Been there, done that. Just 1 more thing for

Re: [asterisk-users] softphone instead of desktop phones

I use Bria on all of the above. Kind regards, Matt > On Apr 29, 2017, at 10:35 AM, Thomas wrote: > > Hello, > Iam lookong for an Softphone for iPhor oder Android smartphone using togehter > with an headset. > I tried Zoiper and CSipSimple but quality was bad compared

Re: [asterisk-users] Call List Campaign to an IVR

one of them goes to voicemail the other will stop ringing. Typing calls this happens in a few ms (after post dial delay). Because they are both going out at the same time with the same provider this is super quick. -- Cheers, Matt Riddell ___ ht

Re: [asterisk-users] Call List Campaign to an IVR

. If your phone is ringing and another comes in it will go to voicemail. Call waiting allows you to get notified of a call if you're already on one, not multiple incoming at the same time. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/ne

Re: [asterisk-users] Call List Campaign to an IVR

or (3) listen to the > message when it was most convenient for them. That way, they were in control > and things were done on >> their terms. > >> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> >> wrote: >> >> Love the i

Re: [asterisk-users] Call List Campaign to an IVR

>> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> wrote: >> Love the idea. How? > > On Mon, 6 Feb 2017, Matt Riddell wrote: > >> exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3) > > Amazing. Who knew? > &g

Re: [asterisk-users] Call List Campaign to an IVR

X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full I

Re: [asterisk-users] Subscribe to events via ARI from node.js without sending to Stasis

> On 27/11/2016, at 6:44 PM, Joshua Colp <jc...@digium.com> wrote: > > On Wed, Nov 23, 2016, at 06:41 PM, Matt Riddell wrote: >> >> There doesn't appear to be a way to monitor general Asterisk events like >> you can in the Asterisk manager without polling f

[asterisk-users] Subscribe to events via ARI from node.js without sending to Stasis

. Is this a correct assumption? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] Streaming for ASR

adcasting-asterisk-conferences/ <http://www.joshua-colp.com/broadcasting-asterisk-conferences/> I'm still working on mine :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_

Re: [asterisk-users] Streaming for ASR

> On 17/10/2016, at 4:07 PM, Joshua Colp <jc...@digium.com> wrote: > > Matt Riddell wrote: >> >>> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradov...@gmail.com >>> <mailto:luca.pradov...@gmail.com>> wrote: >>> >>>

Re: [asterisk-users] Streaming for ASR

nd then sending that file once recording has finished. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange

Re: [asterisk-users] Surfing the web via Asterisk.

I'm guessing you're going to be wanting something closer to this: https://www.npmjs.com/package/speech-rule-engine <https://www.npmjs.com/package/speech-rule-engine> -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily As

Re: [asterisk-users] Surfing the web via Asterisk.

You'd really want to see if you can get ChromeVox as a library rather than as a browser though - otherwise you're going to be limiting yourself to using one concurrent channel and hacks like jack audio to move the audio from the browser to the channel. -- Cheers, Matt Riddell __

Re: [asterisk-users] Surfing the web via Asterisk.

. You'd use agi or something to navigate the object you create and tts to describe current position. The hard part will be parsing the HTML even though most HTML is broken :-) Kind regards, Matt Riddell > On Oct 17, 2016, at 9:00 AM, Jonathan H <lardconce...@gmail.com> wrote: >

Re: [asterisk-users] Installing Asterisk on MAC native

It pretty much just works the same way as Linux. you might need to use brew to install a few prerequisites but I've got it running on my MacBook Pro without any major problems. It's good for testing things but I wouldn't use a MacBook as an office server or anything. And to be fair most of

Re: [asterisk-users] 1000 analogue lines with asterisk

There is definitely no way you should put 1000 lines on a single box. To be honest I do wonder what you want to do with 1000 lines as your description probably changes the recommendations. Kind regards, Matt > On Feb 17, 2016, at 5:09 PM, Goke Aruna wrote: > > Thanks

Re: [asterisk-users] Looking for PRI Card with automatic fail over

There was a product called something like red box or similar that I saw around 5 years ago. Probably not entirely helpful but maybe Google will help. Kind regards, Matt On Aug 3, 2015, at 9:50 AM, Eric Klein eric.kl...@greenfieldtech.net wrote: Hi all, Strange request, I have a

Re: [asterisk-users] Looking for PRI Card with automatic fail over

Yep it was red fone http://red-fone.com 7 years ago :-) http://www.venturevoip.com/detail.php?news_id=1927 -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

and reenable SELinux regardless of the outcome. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels

calls for just one originate: It looks like you’re originating with local channels. Try with /n at the end (not sure if that helps anymore) and also send us your originate line. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

. The console channel gets errors on opening. What errors are you getting? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com

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

On 27Jun, 2015, at 15:34, Michelle Dupuis 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 how loud the call is? -- Cheers, Matt Riddell

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

://wiki.asterisk.org/wiki/display/AST/Function_VOLUME https://wiki.asterisk.org/wiki/display/AST/Function_VOLUME -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk

Re: [asterisk-users] Product CDR/Queue/Meetme

I will post this to the Asterisk news for you. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP

Re: [asterisk-users] setting outbound caller ID

the number from your carrier? Maybe it’s set on their side for the trunk. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com

Re: [asterisk-users] setting outbound caller ID

At any rate, if I can figure out the right way to set the caller ID explicitly, and assuming they honor it if I do, then none of this will matter. Ok, so just do exten = s,n,Set(CALLERID(all)=“Greg” 5551234) https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID -- Cheers, Matt

Re: [asterisk-users] How to use TRUNK only if IAX fails?

The command he gave you was in Asterisk. Why do you not want to call it to try it? Then you can fail over to the other trunk if the IAX link is down. Kind regards, Matt On May 30, 2015, at 2:03 AM, Ashwin Surendran ashwin.surend...@now-health.com wrote: Many Thanks Carlos, I was hoping

Re: [asterisk-users] Problem with realtime mysql I can't seem to resolve

parameters? I.E. mysql -u astadmin -p (then type mysecret) The debug logs and the normal logs don’t match in time - do you have some that do? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com

Re: [asterisk-users] adding area code

=${l_HomeAreaCode}${EXTEN:-1}) exten = _9XXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; Got SIP reponse 503 Service Unavailable” Can you send us the console output when you make the call? -- Cheers, Matt Riddell ___ http

Re: [asterisk-users] Transcoding issues with siren14

? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call

Re: [asterisk-users] Integration with Social Media, Email and Web call center

on it. Facebook might be a little harder as you wouldn't necessarily know when an incoming request came. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http

Re: [asterisk-users] Integration with Social Media, Email and Web call center

get a jabber message you just send a call to the queue with one leg pointing to a particular music file and screen pop the relevant data. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com

Re: [asterisk-users] Support for IP Camera streaming (RTSP) channel to a conference

about it? You're probably better off asking in the Asterisk-Video mailing list. Probably best to ask Sergio :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX

Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files

There's no priority in your call file. Sent from my iPhone On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote: Hello, I noticed that when i move a call file to outgoing directory, two asterisk threads are dealing with it. ]# grep FAX_44731.call /var/log/asterisk/full.2

Re: [asterisk-users] XMPP sendtodialplan

. However, when I initially started setting this up, my expectation was that sendtodialplan would only trigger on messages that weren’t solicited. What does your dialplan look like? Are you using _. -- Cheers, Matt Riddell ___ http

Re: [asterisk-users] accept email and make phone call?

an announcement? I imagine the first part is the big question. procmail could be a useful tool to trigger a script to create an Asterisk call file. You can also use /etc/aliases to run a script when mail is received in an account. -- Cheers, Matt Riddell

Re: [asterisk-users] html/js/flash/air SIP clients?

queues and makes/receives calls. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] quick questions on version 10

I thought this was discussed and it was going to be left in? Sent from my iPhone On 23/08/2012, at 2:30 PM, Jerry Geis ge...@pagestation.com wrote: The AMI action CoreShowChannels deprecated the CLI concise command because the output of the AMI action is extensible without breaking existing

Re: [asterisk-users] html/js/flash/air SIP clients?

if they have any further hints for me. Real life experience would be awesome. You might want to have a look at http://phono.com -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php

Re: [asterisk-users] extending fallback numbers

/${EXTEN:0:2}00,30) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com

Re: [asterisk-users] concurrent channels limit

On 31/03/2012, at 3:28 AM, Syco wrote: But if I change the dialplan, remove background and wait functions, add play with a g729 audio file instead, I could do again just 80 concurrent call. How many g729 licenses do you have? You sure you're not transcoding? -- Cheers, Matt Riddell

Re: [asterisk-users] Asterisk Realtime Time Dial App

trigger for the actual call (i.e. one of the above). It's like saying you want to write an application to be built into Asterisk to initiate a call and play a message. You'd instead use one of the above, then the Playback or Background application to play the message. -- Cheers, Matt Riddell

Re: [asterisk-users] Asterisk Realtime Time Dial App

the Asterisk Manager. Things you would need to learn: 1. The language you use for the program (i.e. C/PHP/Java etc) 2. How to create/use MySQL databases 3. How the Asterisk Manager works -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php

Re: [asterisk-users] Asterisk Realtime Time Dial App

at Astricon I could teach him one of the steps :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution

Re: [asterisk-users] Asterisk Realtime Time Dial App

. You could then make the app GPL or whatever if you wanted to distribute it and then other people could help maintain it. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php

Re: [asterisk-users] Asterisk Realtime Time Dial App

On 27/09/11 11:03 AM, Danny Nicholas wrote: Will you be recording this presentation for those of us who can't get to Astricon? Dunno whether they'll be recording - I haven't done an Astricon presentation for a few years now :-) -- Cheers, Matt Riddell

Re: [asterisk-users] espeak module for asterisk

haven't installed espeak :-) The application is to connect Asterisk to espeak. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com

Re: [asterisk-users] How to get presence using AMI

. Will be returned. Basically you should connect to Asterisk, go into the console (asterisk -r) and then type manager show commands. Read through them, learn what they all do and you'll pretty quickly get a feel for what you can do and how to do it. -- Cheers, Matt Riddell

Re: [asterisk-users] Increasing volume ?

On 4/08/11 9:16 PM, Zeeshan Ali Shah wrote: Tried below, but it still no improvement Zeeshan SetGlobalVar(VOLUME(TX)=10) SetGlobalVar(VOLUME(RX)=10) Have you tried just doing Set(VOLUME(TX)=10) and then 5 etc to make sure you are actually changing the volume? -- Cheers, Matt Riddell

Re: [asterisk-users] Increasing volume ?

(VOLUME(RX)=10) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

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

accountcode (where accountcode is what you originated the call with) and then do a wc -l on the output to count the lines. That's the cue for someone to explain some cleaner way to do it with a single bash command :-) -- Cheers, Matt Riddell ___ http

Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl

it to the Daily Asterisk News? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] AMI buffering event output?

: supersecretpassword events: on Response: Success Message: Authentication accepted It seems somewhat impossible that you would be getting different results from different hosts. Are you using the same login? What if you use the external IP rather than 127.0.0.1 -- Cheers, Matt Riddell

Re: [asterisk-users] AMI buffering event output?

on the local machine? Are you passing two carriage returns after logging in? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com

Re: [asterisk-users] make calls from DID

On 27/05/11 1:08 PM, Cobra 2 wrote: I was trying really hard to not say RTFM. Some people might not be aware that TFM exists :D -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com

Re: [asterisk-users] make calls from DID

-- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] Manager logged on/off messages

On 20/05/11 4:05 AM, Ishfaq Malik wrote: Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? displayconnects = no inside /etc/asterisk/manager.conf -- Cheers, Matt Riddell

Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- Cheers, Matt Riddell

Re: [asterisk-users] 1.8.4 Core Dump after installing from source

a backtrace on the core dump. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] iptables for Asterisk - Any good guides out there?

) while you're fiddling about. What a great idea! I've never thought of doing that! -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] AMI check if connection is alive

to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? Send an Action: Ping\r\n\r\n command. You should receive a response. Run a timer on it and if you don't get a response reconnect. -- Cheers, Matt Riddell

Re: [asterisk-users] iptables for Asterisk - Any good guides out there?

failsafe' (by un-commenting the last line) while you're fiddling about. On Tue, 17 May 2011, Matt Riddell wrote: What a great idea! I've never thought of doing that! I wish I could take credit for it :) I had a similar 'gee, how obvious' epiphany after having locked myself out of way too many

Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

On 12/05/11 9:31 PM, Steve Totaro wrote: PS 42 is the answer, now what is the question. :) Heh, that might be one example where top posting would make sense ;-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News

Re: [asterisk-users] Plotting fxotune dump

. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not

, and nobody has said whether they work or not so you have to try all of them. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com

Re: [asterisk-users] how to play music when dial fail or time out

for fail.wav.* and couldn't find anything. Basically just drop the extension. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com

Re: [asterisk-users] asterisk syntax highlighting for gedit

it here and I'll post it to the Daily Asterisk News. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

, but 30 seconds doesn't start from when it disappears. It's every 30 seconds. So that could be like 1 second etc. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

servers then? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

soul that I am! :-) Fair enough then! -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

if it is: - discontinued - tested - test finalized, result indicating it is fully and identically functional - test finalized, result indicating that this feature is changed in either behaviour of configuration - not yet tested. +1 From me - this would be fantastic! -- Cheers, Matt Riddell

Re: [asterisk-users] Res: Fading voice problem

Are you trunking the calls via IAX2 or something? Are you using a jitter buffer? Are you sure about the direction? Do you get the same problem if you use something like sipp to create 30 LAN calls and one Internet call? -- Cheers, Matt Riddell

Re: [asterisk-users] receive faxes

On 5/05/11 3:02 AM, vip killa wrote: Honestly Digium's Asterisk is not a quality project. Though it has lead the way in innovative open-source VoIP, it's a flawed and chaotic project. Hence, I refuse to pay Digium. So why do you use it? -- Cheers, Matt Riddell

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

of another (cisco 1100 series APs). What actually happens? It shouldn't be disconnecting the call. Do you have rtptimeout or something? Qualify=x? You could try disabling both these options. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

On 5/05/11 11:40 AM, Sherwood McGowan wrote: ChanIsAvail + dialplan routing to call parking lot Problem is, I think he's talking about mid call - so ChanIsAvail will have returned success - oh unless you can run it in the h exten? -- Cheers, Matt Riddell

Re: [asterisk-users] Fading voice problem

the problem? PS: The server is a CentOS 5.5 - 32 bit ... I've tested the 64bit tb but with the sameerror ... What's your CPU usage like? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com

Re: [asterisk-users] how to know status of asterisk from php

the script from the commandline while logged in as root to confirm: php script.php -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] how to know status of asterisk from php

, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

and they're definitely using it: /var/log/asterisk/cdr-custom# grep Pickup Master.csv |wc -L 196 196 times since I upgrade them on the 11th of February. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http

Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

to production. The thing here is that if you're able to provide the same system to multiple customers then it doesn't end up being such a crazy list of things to check. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

and it resolved the problem. This was not a crash and was caused by the fact that we were doing something that most people aren't (using chan_lcr in Asterisk 1.8). If everyone's calls did this when they saw a 302 redirect it certainly would have shown up on the issue tracker. -- Cheers, Matt Riddell

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

to connect to irc.freenode.net or even a plugin in Firefox. Once you're connected to IRC you can join chat rooms. There are some like #asterisk for discussion about Asterisk and #asterisk-bugs for discussion about Asterisk bugs. Post back here if you have any problems connecting. -- Cheers, Matt

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

to create the problem - (gaps in the message sequence): Ah, which explains why I'm not seeing that too - we do attach=yes, delete=yes -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com

Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

-21 1.8.X, LTS 2010-10-21 2014-10-21 2015-10-21 Where STD is Standard and LTS is Long Term Support. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP

Re: [asterisk-users] AST-2011-006: Asterisk Manager User Shell Access

everything and added them back in as required. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions

Re: [asterisk-users] PAP2T auto answer?

You could try: exten = *701,1,Set(__SIPADDHEADER=Call-Info:sip:192.168.101.1\; answer-after=1) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http

Re: [asterisk-users] Softphone IAX

the legality of g729 patents etc - we've been there a million times. Eduardo that last sentence wasn't aimed at you :-) If you're looking for a low bandwidth codec in a sofphone you might consider Speex? -- Cheers, Matt Riddell ___ http

Re: [asterisk-users] R: No Internet, no asterisk

do - the problem (AFAIK) is that Asterisk is unable to contact the DNS server, not that it doesn't return a result it likes. Therefore a caching nameserver (bind9 etc) should solve it. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php

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