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

[Matt Riddell]

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'>);

Problem 1: The url is http but gets redirected to https in the browser
Problem 2: Although the wiki doesn’t state it, you need at least version 14 of 
Asterisk and I was using version 13.  In the process of updating now___
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[asterisk-users] [asterisk-app-dev] Asterisk 13 ARI Playback of audio via HTTP

[Matt Riddell]

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 /sounds resource 
to query for available sounds on the system. You can also use specify a media 
file which is consumed via HTTP (e.g sound:http://foo.com/sound.wav)

But I get

[2019-02-18 11:18:29] WARNING[23860][C-0002f9e3]: file.c:774 
ast_openstream_full: File http://www.nch.com.au/acm/8k16bitpcm.wav does not 
exist in any format

Cheers,

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

[Matt Riddell]

> 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?
> 
> 
I think you’re probably going to be looking for:

ari.recordings.getStoredFile(
  {recordingName: val},
  function (err, binary) {}
);

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

[Matt Riddell]

> 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 applications subscribed to the channel. In this case 
> you can discriminate the events on the basis of application name.

Yeah you’re right - I was registering 3 applications so get three events - even 
though the channel was only in one application.

Thanks,

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

[Matt Riddell]

Hiya,

When I hang up on a call to my stasis app I’m getting multiple channelDestroyed 
events for the same channel:

app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone

I’m only registering once to the event:

ari.on('ChannelDestroyed', channelDestroyed);

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

[Matt Riddell]

> 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?
>> 
>>var bridge = ari.Bridge();
>>bridge.create({
>>type: 'holding',
>>name: event.application+" bridge"
>>}, function(err, bridge) {
>>bridge.addChannel({
>>channel: incoming.id
>>}, function(err) {
>>console.log("Added to bridge")
>>console.log(bridge.channels).   ; 
>> <——— This Line
>>console.log(err);
>>});
>> 
> 
> I believe you are accessing the snapshot, essentially, of the bridge at the 
> time it was created in which case there would be no channels. You would need 
> to retrieve an up to date snapshot to get the current state.

Yeah cool that worked:

ari.bridges.get({bridgeId: bridge.id}, function (err, newBridge) {
console.log("New Bridge: "+newBridge.channels)
});
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Re: [asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

[Matt Riddell]

> 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?
>> 
>>   var bridge = ari.Bridge();
>>   bridge.create({
>>   type: 'holding',
>>   name: event.application+" bridge"
>>   }, function(err, bridge) {
>>   bridge.addChannel({
>>   channel: incoming.id
>>   }, function(err) {
>>   console.log("Added to bridge")
>>   console.log(bridge.channels).   ; 
>> <——— This Line
>>   console.log(err);
>>   });
>> 
> 
> I believe you are accessing the snapshot, essentially, of the bridge at the 
> time it was created in which case there would be no channels. You would need 
> to retrieve an up to date snapshot to get the current state.

Yeah cool that worked:

ari.bridges.get({bridgeId: bridge.id}, function (err, newBridge) {
console.log("New Bridge: "+newBridge.channels)
});
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[asterisk-users] [asterisk-app-dev] ARI Node JS Bridge.addChannel

[Matt Riddell]

Hiya,

I would have expected this to show the channels in the bridge inside the 
anonymous function - it shows the bridge is empty though?

var bridge = ari.Bridge();
bridge.create({
type: 'holding',
name: event.application+" bridge"
}, function(err, bridge) {
bridge.addChannel({
channel: incoming.id
}, function(err) {
console.log("Added to bridge")
console.log(bridge.channels).   ; <——— This 
Line
console.log(err);
});

Reason being, I’m creating a queue need to move channels between bridges 
depending on agent/customer status etc

Cheers,

Matt Riddell
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Re: [asterisk-users] shell dialplan application blocking

[Matt Riddell (lists)]

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 the shell script is 
> finished?
> 
>same => n,Set(PUSHRESULT=${SHELL(sendpush.sh)})
> 
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Re: [asterisk-users] Asterisk chan_sip registration attempts

[Matt Riddell (lists)]

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 you help me with Asterisk 11.21.2 and AsteriskNow platform.
> 
> The problem is:
> 
> My Asterisk PBX has SIP (chan_sip) trunk to provider.
> 
> Asterisk periodically loses trunk registratrion:
> 
> sip show registry:
> 
> Hostdnsmgr Username   Refresh State   
>  Reg.Time 
> X.X.X.X:5060N  105 Unregistered
>
> 
> This happens sometimes once per 4 hours, sometimes once per a week.
> 
> I don't see any patterns.
> 
> sip.conf:
> 
> registerattempts=0
> 
> registertimeout=20
> 
> peer confifuration:
> [-friend]
> disallow=all
> host=192.168.1.1
> defaultuser=
> fromuser=
> callerid=
> secret=
> type=friend
> qualify=yes
> allow=ulaw
> allow=alaw
> nat=no
> rtpkeepalive=10
> dtmfmode=rfc2833
> insecure=port,invite
> context=from-trunk-ISP1
> fromdomain=
> registration string:
> 
> register=:@/
> 
> where:
> 
>  is our ISP-provided phone number
> 
>  is our ISP-provided SIP secret
> 
>  is our ISP SIP server IP address
> 
> I don't have NAT between the ISP and my server (the ISP server IP address is 
> in Asterisk's sip.conf Localnet scope) but as I can see there is ISP's 
> routers between my Asterisk and the ISP SIP server.
> There is not any firewall between the ISP and my Asterisk. The firewall rules 
> on Asterisk host allow any traffic from host to the ISP and allow 5060/UDP 
> from ISP to Asterisk host.
> 
> 
> When I restart Asterisk I can see successful registration to the ISP. And it 
> works. I can make calls in any direction but after some time I have the 
> "Unregistered" status and calls don't work.
> 
> When I make tcpdump I can see about 40-60% failed REGISTER attempts (401 
> forbidden) and 100% failed OPTIONS attempts (is it qualify packets?) with 401 
> forbidden.
> 
> The interesting thing is: the last REGISTER packet from my Asterisk to the 
> ISP has "OK" response and there was not any REGISTER attempts after this 
> packet, only OPTIONS packets.
> The second interesting thing is: this ISP has been working for about 6 or 
> even 12 months before this problem happened.
> 
> 
> What should I check to understand and solve my problem?
> 
> 
> P.S. I'm sorry for my bad English(
> -- 
> Dmitriy Ermakov
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Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition… so many questions!

[Matt Riddell]

> On 22/09/2017, at 8:08 AM, Jonathan H <lardconce...@gmail.com> wrote:
> 
> Removing the "record in Asterisk/store as file/convert file/upload
> file  <> receive stream/save file/convert file/playback in Asterisk"
> part of the sequence would save vital seconds 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.

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

[Matt Riddell]

> On 31/05/2017, at 3:52 PM, Steve Edwards <asterisk@sedwards.com> wrote:
> 
> On Wed, 31 May 2017, Daniel Tryba wrote:
> 
>> On Wed, May 31, 2017 at 01:39:25PM -0700, Steve Edwards wrote:
>>>> What bugs you about the output format?
>>> 
>>> It's been a while, but as I recollect, it included the date/timestamp in the
>>> file name of the 'ring buffer' which meant that each time the host was
>>> rebooted, dumpcap didn't know the files from the previous run should be
>>> deleted when they 'aged out.'
>> 
>> 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 me to maintain :)


Easier just to use logrotate no?

That's what it's designed for.

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

[Matt Riddell (lists)]

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 to an desktop SIP 
> phone.
> 
> Is there an better softphone?
> 
> Or are there softphone solutions for PC desktop MAC or Android with an 
> headset?
> I want to save cost for desktop phones.
> 
> thanks Thomas
> 
> 
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Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell]

> On 6/02/2017, at 1:28 PM, Tech Support <aster...@voipbusiness.us> wrote:
> 
> I remember doing the testing and two calls going out at the same time 
> don’t actually have to go out at the *exact* same time. The remote end will 
> pick up one of the two calls, but there is no guarantee which one it will be. 
> Also, if you let the first call ring too long, yes, the second call will go 
> to voicemail,  but the first call will start ringing, which is something we 
> wanted to avoid.
> John
> 

That's the benefit of doing the & thing.

The instant 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.

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

[Matt Riddell]

> On 6/02/2017, at 12:30 PM, Chris Bagnall <aster...@lists.minotaur.cc> wrote:
> 
> On 6/2/17 5:24 pm, Tech Support wrote:
>> Basically,
>> two calls are made.
> ...
>> When the first call is made for
>> such a short period, the remote end still goes off hook, but the call will
>> end before it starts to ring. Then, halfway through the first call, a second
>> call is made. Since the remote end is off hook from the first call, the
>> second call will get sent to voicemail and the message is played there.
> 
> Am I right in thinking call waiting isn't a thing on US mobile networks then? 
> In the UK, call waiting is pretty standard, and almost universally enabled by 
> default on mobile networks. AIUI the same is true for much of Europe.

Heh nah, call waiting still only allows one ringing event to happen at a time.

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.

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

[Matt Riddell (lists)]

Not really, doing the way below you don't even have to worry about it. They 
both go out at the same instant and as soon as it hits voicemail it disconnects 
the other leg. 

If you wanted you could leave it ringing for twenty minutes and it would still 
have the same effect. 

Kind regards,

Matt

> On Feb 6, 2017, at 12:29 PM, Tech Support <aster...@voipbusiness.us> wrote:
> 
> That's the basics, but you have to nail the timing just right. The timing is
> really important to do it the right way.
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Monday, February 06, 2017 12:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call List Campaign to an IVR
> 
> 
>> On Mon, 6 Feb 2017, Tech Support wrote:
>> 
>>  We were able to develop a feature to send the call to voicemail
> about 90% of the time. That way, an end user could (1) not be bothered by
> having to answer the call, (2)
>>  delete the message without listening to it, 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 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?
> 
> So how/why does this work?
> 
> I see 2 calls going out to my cell. Does the first 'busy out' my number at
> my cell provider so the second goes straight to VM? What part does the
> '0111' play?
> 
> --
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> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Call List Campaign to an IVR

[Matt Riddell]

> On 6/02/2017, at 12:24 PM, Steve Edwards <asterisk@sedwards.com> wrote:
> 
> 
>> On Mon, 6 Feb 2017, Tech Support wrote:
>> 
>>  We were able to develop a feature to send the call to voicemail about 
>> 90% of the time. That way, an end user could (1) not be bothered by having 
>> to answer the call, (2)
>>  delete the message without listening to it, 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 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?
> 
> So how/why does this work?
> 
> I see 2 calls going out to my cell. Does the first 'busy out' my number at my 
> cell provider so the second goes straight to VM? What part does the '0111' 
> play?


Yep, if you have two calls going to the same number at the same time the second 
will be busy.

Meaning it will go to voicemail.

Meaning it will be answered.

Meaning the other leg will be dropped.

The 0111 and the 1 is just so that Asterisk doesn't think that both calls are 
to the same destination and then only make one call.

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

[Matt Riddell]

> 
> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk@sedwards.com> wrote:
> 
> On Mon, 6 Feb 2017, Tech Support wrote:
> 
>> We were able to develop a feature to send the call to voicemail about 90% of 
>> the time. That way, an end user could (1) not be bothered by having to 
>> answer the call, (2) delete the message without listening to it, 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.
> 
> Love the idea. How?


exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider/1${EXTEN}@myprovider,3)

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

[Matt Riddell]

> 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 for channel statuses or
>> sending the channels to the Stasis app and recreating the logic of the
>> Queue application.
> 
> You can subscribe to events for everything, but ARI events are a subset
> of what are available over AMI so you may not get what you want. Events
> raised by applications for example would not be present. You can
> subscribe to all channels by subscribing to "channel:" using the
> /applications//subscription resource, and "bridge:" for all
> bridges.


 is the stasis app name or the Asterisk application name?

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

[Matt Riddell]

Hi,

I'm writing a node.js backend to pass events via a websocket to a CRM.

Basically what I want to do is notice when things happen (i.e. new channel, new 
bridge etc) without sending the channels to the Stasis app.

The channels I'm interested in are agents who are in a queue only because they 
are in a realtime MySQL database for the queue_member_table.

There doesn't appear to be a way to monitor general Asterisk events like you 
can in the Asterisk manager without polling for channel statuses or sending the 
channels to the Stasis app and recreating the logic of the Queue application.

Is this a correct assumption?

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Re: [asterisk-users] Streaming for ASR

[Matt Riddell]

> On 19/10/2016, at 2:32 AM, Luca Pradovera <luca.pradov...@gmail.com> wrote:
> 
> Would UnicastRTP be able to output u-law frames directly? If so, I think that 
> is all I need.

Joshua Colp did a great writeup that may work for your situation:

http://www.joshua-colp.com/broadcasting-asterisk-conferences/ 
<http://www.joshua-colp.com/broadcasting-asterisk-conferences/>

I'm still working on mine :-)

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Re: [asterisk-users] Streaming for ASR

[Matt Riddell]

> 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:
>>> 
>>> I have been working on designs for two different projects, where both
>>> of them would need to use the IBM Watson streaming ASR service.
>>> 
>>> Would it be possible to write out the audio frames as they get
>>> recorded? Watson supports 16 bit signed little endian audio, IIRC, but
>>> there are a few other raw formats available.
>> 
>> That's the main problem I was discussing at AstriDevCon. There's no way
>> currently to stream the frames to a service.
>> 
>> At the moment I'm saving to a file and then sending that file once
>> recording has finished.
> 
> The UnicastRTP channel driver allows you to send RTP to a specific target 
> address with media. Combined with Chanspy (or Snoop channels in ARI) you can 
> duplicate audio from a channel and send it off to where you want.

So originate a new channel, make one leg a UnicastRTP and the other a chanspy 
to spy on the channel you're interested in transcribing?

Cheers,

Matt Riddell
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Re: [asterisk-users] Streaming for ASR

[Matt Riddell]

> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradov...@gmail.com> wrote:
> 
> I have been working on designs for two different projects, where both of them 
> would need to use the IBM Watson streaming ASR service.
> 
> Would it be possible to write out the audio frames as they get recorded? 
> Watson supports 16 bit signed little endian audio, IIRC, but there are a few 
> other raw formats available.

That's the main problem I was discussing at AstriDevCon.  There's no way 
currently to stream the frames to a service.

At the moment I'm saving to a file and then sending that file once recording 
has finished.

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

[Matt Riddell]

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>

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

[Matt Riddell]

> On 17/10/2016, at 9:51 AM, Jonathan H <lardconce...@gmail.com> wrote:
> 
> Ah, no, you misunderstand. Asterisk wouldn't care one little bit what
> is on the page - Chromevox would do all that.
> A screenreader usually tabs or arrows their way about, selecting
> headings to read content.
> 
> Thus, Asterisk ONLY needs to be able to hear content FROM the browser
> and pipe it to the channel, and pass keypresses back TO the browser.
> 
> The human is the parser, if that makes sense?


Right, so you're using a prebuilt browser to do the parsing.

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

[Matt Riddell (lists)]

This is a really interesting project but I think it's going to be seriously 
hard. You're going to need to parse meaning from a site, and that's not an easy 
thing to do. 

If you're focused on a few of the bigger sites then it might be easier. 

You almost want a middle layer that can parse meaning from a site into xml or 
something. 

Then I'd work on creating objects out of each kind of tag. The problem is that 
navigation may not be the same as you'd see when visiting a site. You're not 
really going to be moving left and right. It would be more like tab works. Next 
item kinda thing. And items wouldn't necessarily be in the same order as you 
see. Pull right/left classes for bootstrap etc would make layout different. 

I would maybe check if there are any libraries that can parse HTML into objects 
first and if not then start building on yourself. The telephony side will be 
easy. 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:
> 
> Has anyone attempted making the web phone accessible? I can only find one 
> company which operated between 1996 and 2000. 
> 
> I was thinking, install Chrome with Chromevox, headless, on a server, and use 
> something like an AGI to send basic keyboard commands to navigate a page, as 
> a screenreader user would, and pipe the audio back to a channel, to be 
> streamed by Asterisk. 
> 
> (Bear with me here - it's a project for blind people involving a telephone 
> and some lateral thinking!)
> 
> And yes, I mean more than just CURL a page, tts it and then read. I'm talking 
> about using the keypad to navigate the headers and landmarks. There are just 
> enough keys to make it viable.
> 
> Of particular interest is the very high quality of the Chromevox screenreader 
> voice from Google.
> 
> Does such a framework exist? I'm aware of a project called Chromium Headless, 
> but some of the links are broken and it doesn't seem to have the 
> audio/extension part
> 
> Failing headless, what about running it on a vps in an x-window environment? 
> Not sure how I'd pass the keyboard presses to it, without using a keyboard...
> 
> Any ideas, or is the whole idea complete madness? Thanks!
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Re: [asterisk-users] Installing Asterisk on MAC native

[Matt Riddell (lists)]

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 the time I just use virtualization and spin up Debian in 
parallels. 

Kind regards,

Matt

> On Sep 20, 2016, at 4:07 PM, Glenn Geller (VDOPh)  wrote:
> 
> If you're looking for installing on a MAC, best to start searching for MAC 
> OSX install
> 
> See here: 
> http://www.voip-info.org/wiki/view/Asterisk+Getting+Started+on+MacOSX
> 
> I don't know how old this is, or if it directly applies to your task at hand, 
> but it may be a start.
> 
> Also, if you're just looking for a simple PBX for light usage, there may be 
> other options out there for MAC OS as well.
> 
> Good hunting!
> 
> Glenn Geller
> 
>> On Tue, Sep 20, 2016 at 1:58 PM, Saint Michael  wrote:
>> ​I need to install Asterisk on a MAC, native, no virtualization.
>> Has anybody done this? Are there documents on the Internet?
>> I googled it and all web sites that claimed to help installing Asterisk on
>> a MAC have disappeared. Is it possible at all?
>> Digium should actually have a MAC app in the Apple store with a PBX. It 
>> should be a paid app. I would buy it right away.
>> 
>> ​
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Re: [asterisk-users] 1000 analogue lines with asterisk

[Matt Riddell (lists)]

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 Harry.
> I will check and revert. I hope it work perfectly with asterisk.
> Regards
> 
>> On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor  
>> wrote:
>> Hi,
>> 
>> For analog, I really like telco grade channel banks.
>> 
>> I would recommend the adit 600, there is a good market on Ebay, and you can 
>> do 48 channels per adit 600, with 2 T1 interfaces.  Having onsite spares 
>> would not be an issue (cost is low).  You can put two next to each other in 
>> a rack, taking up about 2U of space per 2 channel banks.
>> 
>> You could service this with six eight port T1 cards, or with eleven/twelve 
>> quad T1 cards.  I would distribute across two, three, or even four servers 
>> for redundancy/resiliency and load balancing.
>> 
>> -Harry
>> 
>> 
>>> On 02/17/2016 12:16 AM, Goke Aruna wrote:
>>> 
 On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani  wrote:
 Sangoma 50 port FXS
>>> 
>>> 
>>> Thanks.
>>> Will I now stack 20 boxes in order to achieve the 1000 FXS lines?
>>> Regards
>>> 
>>> 
>> 
>> 
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Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Matt Riddell (lists)]

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 customer where we are putting an Asterisk PBX in 
 front of a legacy (non-VoIP) PBX. One of the requirements it that the 
 Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the 
 carrier) with the ability to go to pass through should the Asterisk PBX 
 (software or hardware level) fail.
 
 I did not see this feature in the Digium, Sangoma, Allo, or OpenVox cards.
 
 Does anyone know of a card that will do this? I know that Digium has an 
 external box (the r850) that does something similar for 2 PBXs making them 
 high availability, but in this case I only have the 1 Asterisk box acting as 
 a gateway and passing some calls out over SIP and IAX2. 
 
 Any suggestions would be appreciated.
 
 Thanks
 Eric
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Re: [asterisk-users] Looking for PRI Card with automatic fail over

[Matt Riddell]

Yep it was red fone

http://red-fone.com

7 years ago :-)

http://www.venturevoip.com/detail.php?news_id=1927

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

[Matt Riddell]

On 3Jul, 2015, at 06:17, Jerry Geis ge...@pagestation.com wrote:
 
 alsa_card_init^[[0m: snd_pcm_open failed: Connection refused
 soundcard_init^[[0m: Problem opening alsa capture device
 

If the problem is a permission thing I’d confirm that.

Run Asterisk as root and see if the problem goes away.

If so then you’ll have to work your way through devices and drivers and check 
who owns them and is able access them.

I’d also try temporarily disabling SELinux to see if that helps.

As with the first option this would be temporary - go back to running Asterisk 
as myuser or whatever and reenable SELinux regardless of the outcome.

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

[Matt Riddell]

 On 3Jul, 2015, at 13:46, Alonso Genis abge...@gmail.com wrote:
 
 Hello,
 
 I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with 
 success.
 
 I have an application that sends an action Originate to AMI for
 calling, it's working well, but when i see to Asterisk's CLI, i see 2
 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.

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

[Matt Riddell]

 I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse 
 audio running as a local user.
 
 Has anyone done that?
 What is the trick?
 
 I changed directories /var/run/asterisk, /var/spool/asterisk, 
 /var/log/asterisk,
 /usr/lib/asterisk, /etc/dahdi and all that stuff.
 
 The console channel gets errors on opening.

What errors are you getting?

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

[Matt Riddell]

 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?

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

[Matt Riddell]

 On 28Jun, 2015, at 10:19, Michelle Dupuis mdup...@ocg.ca wrote:
 
 ​I meant how many calls are in progress on a particular trunk.  (Sorry - I 
 didn't even think of the other interpretation).


Then yeah group and group count functions:

https://wiki.asterisk.org/wiki/display/AST/Function_GROUP 
https://wiki.asterisk.org/wiki/display/AST/Function_GROUP

https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT 
https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT

Just in case you’re wondering, the function to change the loudness of a channel 
is volume :-)

https://wiki.asterisk.org/wiki/display/AST/Function_VOLUME 
https://wiki.asterisk.org/wiki/display/AST/Function_VOLUME

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

[Matt Riddell]

I will post this to the Asterisk news for you.

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Re: [asterisk-users] setting outbound caller ID

[Matt Riddell]

 On 18Jun, 2015, at 13:44, Greg Woods g...@gregandeva.net wrote:
 
 I am certain that the old number that is showing up as my caller ID is not 
 present in any of my config files (that includes sip.conf and iax.conf, 
 everything in the /etc/asterisk directory has been checked). 

Did you buy the number from your carrier?  Maybe it’s set on their side for the 
trunk.

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Re: [asterisk-users] setting outbound caller ID

[Matt Riddell]

 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

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

[Matt Riddell (lists)]

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 to check whether the remote server is
 available before I issue the dial in my dial plan.
 
 Is there a better way to do it in asterisk without using unix commands?
 
 
 Many Thanks,
 Ashwin
 
 On 5/30/15, 2:06 AM, Carlos Chavez cur...@telecomabmex.com wrote:
 
 On 5/29/15 1:16 PM, Ashwin Surendran wrote:
 Hi,
 I have multiple Asterisk servers in various parts of the world all
 connected using dedicated VPN¹s.
 
 Each of these servers have iax and dahdi TRUNK configured on them.
 
 Occasionally the VPN¹s fail.
 
 What I want to be able to do is on my dial plan, use IAX if the asterisk
 server can reach the remote server using the internet OR, use TRUNK only
 if it can¹t use IAX.
 
 Any ideas on how this can be implemented on the dial plan?
Check the DIALSTATUS variable to see if the IAX trunk failed and
 then dial via DAHDI.
 
 https://wiki.asterisk.org/wiki/display/AST/Dial+Channel+Variables
 
 --
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez
 +52 (55)9116-91161
 
 
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Re: [asterisk-users] Problem with realtime mysql I can't seem to resolve

[Matt Riddell]

 On 22May, 2015, at 03:51, Jonas Kellens jonas.kell...@telenet.be wrote:
 
 Realtime seems to be loaded :
 
 *CLI realtime mysql status 
 general configured for asterisk on socket file /var/lib/mysql/mysql.sock with 
 username asterisk.
 MyAsteriskDB connected to MyAsteriskDB@127.0.0.1, port 3306 with username 
 astadmin for 12 minutes.
 [May 22 10:32:02] ERROR[11269]: res_config_mysql.c:1599 mysql_reconnect: 
 MySQL RealTime: Failed to connect database server asterisk on 
 /var/lib/mysql/mysql.sock (err 1045). Check debug for more info.

Does MySQL work ok from the console with those 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 
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Re: [asterisk-users] adding area code

[Matt Riddell]

 On 27Apr, 2015, at 16:39, Motty Cruz motty.c...@gmail.com wrote:
 
 forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. 
 
 Thanks, 
 
 
 On 04/27/2015 02:38 PM, Motty Cruz wrote:
 here is what I have: 
 exten = _9XXX,1,Set(l_HomeAreaCode=381)
 
 exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${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?

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Re: [asterisk-users] Transcoding issues with siren14

[Matt Riddell]

On 28/02/2013, at 6:08 PM, Richard Kenner ken...@gnat.com wrote:
 Sorry for a possible retransmit: the first was sent from an incorrect
 email address.
 
 I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
 
 But the transcoding from siren14 to slin32 is via slin.  First, it
 seems odd that there's no transcoder directly to slin32 since anything
 else will lower fidelity.  But, more importantly, there is transcoding
 from siren14 to slin16 and slin16 to slin32.  So why is slin used
 as the intermediate instead of slin16?


Do you have transcode_via_sln set in asterisk.conf?

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

[Matt Riddell]

On 22/01/2013, at 5:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Dears;
 
 Can someone advise me where to find a technology (open source) that let us 
 able to integrate with social media like whatsapp and facebook? And use this 
 in call center (queuing the messages and routing it for agent)?
 
 Anyone give me a light to start?


In the past I've sent calls to an agent in the queue with music on hold that 
contained a beep every 20 seconds (to remind them they're on a call) and then 
used the same code I do for screen popping to send them alternative records.  
I.E. web page, email, fax etc.  It's stored in the database that that's what 
they were working on and then when they finished working on it they just hang 
up or press * to disconnect the call.

That way you can use the standard Asterisk queues and they don't get bothered 
by anything else while they're working on it.

Facebook might be a little harder as you wouldn't necessarily know when an 
incoming request came.

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

[Matt Riddell]

On 24/01/2013, at 10:24 AM, bilal ghayyad bilmar...@yahoo.com wrote:
 They advised me to check jabber.org.
 Yes, jabber.org has a client that can send/receive and integrate with other 
 social media (facebook, msn, twitter, ... etc).
 
 But, as an Agent who can login/logout and take a calls, how can I make it to 
 be single login for voice and messages. So, if the agent is not available, he 
 will not get a calls and will not get a messages.
 
 Those who used jabber.org or who used other than jabber.org for such 
 requirement, what do you suggest? 


Ah, so yeah same as I said but when you 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.

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

[Matt Riddell]

On 3/12/2012, at 3:55 AM, Deepak Hegde deepak...@yahoo.com wrote:
 Hello,
 
 I am trying to stream an IP Camera output (h264) into a conference. The IP 
 Camera supports RTSP. 
 Searching around the web, I believe the RTSP support (was) available through 
 app_rtsp (external to Asterisk distribution).
 This, I believe, has problems and has issues compiling in Asterisk 11 (I 
 tried compiling it in Asterisk 11 and it failed).
 I may not be able to use DiaStar or i6net's VXI etc as the hardware is a low 
 end appliance with limited resources.
 
 
 Would anybody be able to suggest how to go about it?


You're probably better off asking in the Asterisk-Video mailing list.

Probably best to ask Sergio :-)

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

[Matt Riddell (lists)]

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
 
 [Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on 
 /var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
 [Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1] 
 System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3] 
 System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or 
 extension must be specified, along with tech and dest in file 
 /var/spool/asterisk/outgoing/FAX_44731.call
 [Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in 
 /var/spool/asterisk/outgoing/FAX_44731.call, deleting
 
 As you see there are two thread dealing with my call file. Now let's inspect 
 the thread 18852.
 
 ]# grep \[18852\] /var/log/asterisk/full.2 
 [Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on 
 DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
 [Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
 [Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME:  NUM: 90312xxx
 [Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer 
 capability: 0x00 - SPEECH
 [Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1] 
 System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:2] 
 SendFAX(DAHDI/i1/0312xxx-b08, /tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new 
 stack
 [Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel 
 'DAHDI/i1/0312xxx-b08' sending FAX:
 [Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
 /tmp/Qg90Ox5YGF5kYkJu.tif
 [Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3] 
 System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS  
 /var/spool/asterisk/outgoing/FAX_44731.call) in new stack
 [Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel 
 'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
 [Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup 
 'DAHDI/i1/0312xxx-b08'
 [Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to 
 DAHDI/g0/0312xxx
 
 It seems that the thread 18852 executes it normally but the thread 26842 
 deletes my call file. And when I inspected the asterisk log file, i saw that 
 the thread 26842 is deleting all my call files.
 
 Here is my custom_extensions.conf file:
 
 [asteriskgw_fax]
 exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}  
 /var/spool/asterisk/outgoing/FAX_${ID}.call)
 exten = s,2,SendFAX(${FAXFILE},zdfs)
 exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}  
 /var/spool/asterisk/outgoing/FAX_${ID}.call)
 
 And here is a sample of call file:
 
 Channel: DAHDI/g0/0312xxx
 MaxRetries: 0
 RetryTime: 60
 Context: asteriskgw_fax
 Extension: s
 Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
 Set: ID=44884
 Callerid: 90312xxx
 Archive: Yes
 
 
 
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Re: [asterisk-users] XMPP sendtodialplan

[Matt Riddell]

On 20/09/2012, at 3:59 PM, Noah Engelberth n...@directlinkcomputers.com wrote:
 I’ve been working on an interactive XMPP interface so users at my office can 
 interact with the timeclock and queues by XMPP (in addition to IVR menu, 
 which has been running just fine for quite a while before the XMPP 
 interface).  I’m using sendtodialplan=yes to handling the incoming 
 unsolicited messages, and typically will have at least one point of 
 interaction where Asterisk requests authentication from the user and then 
 waits with XMPP_RECEIVE for the response.  Asterisk then processes the reply 
 and finishes out the “call” as expected, no problems there.  However, I’m 
 seeing some behavior that I don’t really expect after the call finishes.
  
 After my call finishes, sendtodialplan triggers again one time for each 
 message that was sent back (and caught by XMPP_RECEIVE) during the just 
 completed call.  I’ve avoided unwanted extra processing by using XMPP_RECEIVE 
 on a short timeout to process incoming unsolicited messages, so the net 
 effect of this extra trigger is one or more “XMPP calls” that trip the 1 
 second timeout on this XMPP_RECEIVE and then fall through and clean up.  
 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 _.

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

[Matt Riddell]

On 20/09/2012, at 4:07 PM, Steve Edwards asterisk@sedwards.com wrote:
 On Thu, 20 Sep 2012, Joseph Acquisto wrote:
 
 Any ideas on how asterisk could accept an email (such as an email to SMS or 
 num...@mybox.org sort of thing) and make a phone call to a specific number 
 and make 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.

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

[Matt Riddell]

On 7/08/2012, at 7:38 PM, Arstan Jusupov arst...@gmail.com wrote:
 Correct me if I'm wrong but phono works with voxeo tropo.


Phono can also be used to make SIP calls directly to and from your Asterisk 
servers via Voxeo.

I use it in my CRM package to provide a softphone that logs into call queues 
and makes/receives calls.

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

[Matt Riddell (lists)]

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 systems.  The CLI command is not extensible without breaking
 existing systems.
 Richard,
 
 Thanks - I tried the CoreShowChannels AMI and it says:
 
 Response: Follows
 Privilege: Command
 No such command 'CoreShowChannels' (type 'core show help CoreShowChannels' 
 for other possible commands)
 --END COMMAND--
 
 In my manager.conf I have 
 read = system,call,command,agent,user,reporting
 write = system,call,command,agent,user,originate,reporting
 
 Did I miss something?
 
 Jerry
 
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Re: [asterisk-users] html/js/flash/air SIP clients?

[Matt Riddell]

On 2/08/2012, at 2:27 PM, Arstan Jusupov arst...@gmail.com wrote:
 Dear list,
 I am looking for an open source SIP client(or any SDK) that can work on a 
 browser. It may be based html5, javascript, flash, adobe air. I have done 
 some research myself and I would like to ask the community 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

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Re: [asterisk-users] extending fallback numbers

[Matt Riddell]

On 3/04/2012, at 6:42 PM, Paolo Supino wrote:
 Hi
 
 Isn't there a better way of doing it that, other than hard coding the
 extension groups?
 
 exten = _20XX,1,Dial(SIP/${EXTEN},30)
 exten = _20XX,n,Dial(SIP/2000,30)


Change to:

exten = _2XXX,1,Dial(SIP/${EXTEN},30)
exten = _2XXX,n,Dial(SIP/${EXTEN:0:2}00,30)

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Re: [asterisk-users] concurrent channels limit

[Matt Riddell]

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?

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

[Matt Riddell]

On 29/09/11 12:53 AM, Nick Khamis wrote:

Hello David,

I have this discussion also on the -dev mailing list. and suggested
that we use a database hook to trigger the originate process (pleasee
see Outbound Call Implementation). However, compiling it directly
into asterisk as a realtime moodule insted of using AMI etc...


I'm pretty sure nobody recommended you compile it directly into Asterisk :-)

Asterisk Realtime does not support polling a database.  It is for 
loading configuration from.  Basically it is a way to replace 
(supplement) the config files.


Originating a call is not a config file.

Call origination happens via the following means:

1. Asterisk Manager
2. Call files
3. VoIP/Analogue Phone/Line (includes GoogleTalk etc)

What you're wanting to do (initiate a call from a database record) is 
something that you would be better doing externally and using an 
existing 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.


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

[Matt Riddell]

On 27/09/11 2:57 AM, Nick Khamis wrote:

That would be amazing! And would allow for more possibilities. A call
is made with the simple insertion of new call data. Some directions on
this please?


Just write a program that polls the database, if it sees a record it 
runs an Originate command via 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

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

[Matt Riddell]

On 27/09/11 10:51 AM, Danny Nicholas wrote:

Matt - how dare you tell a man asking for a fish to learn how!


:-)  It would have taken way too much time to explain all the steps.

Although, having said that I am doing a tutorial at Astricon on how to 
use the Asterisk Manager, so if he's at Astricon I could teach him one 
of the steps :-)


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

[Matt Riddell]

On 27/09/11 11:04 AM, Nick Khamis wrote:

Hello Matt,

Thank you so much for your response. The first two are not a problem.
I would rather write the app in C and compile it directly into
asterisk. Are there
any direction on how to work with the dial app? Any types of threads *
may use etc...


I wouldn't compile it directly into Asterisk for a few reasons:

1. It's not likely to ever get accepted into the mainstream codebase as 
it would require a dependency on MySQL.  This means you would 
continually need to adapt your code as Asterisk changes.


2. If you were to write something that could be accepted into the 
mainstream code base, it would need to support multiple database systems 
and would likely need to be integrated with Asterisk realtime.


3. If there was a problem in the code it would bring down Asterisk, 
whereas if it was external it would just bring down the app.


4. If you ever wanted to support a cluster of Asterisk machines it would 
likely require significant changes.


If the app were written in C but external to Asterisk you would overcome 
all of these.  The only exception being that you would want to check the 
Asterisk Manager version when connecting.


You could then make the app GPL or whatever if you wanted to distribute 
it and then other people could help maintain it.


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

[Matt Riddell]

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 :-)


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Re: [asterisk-users] espeak module for asterisk

[Matt Riddell]

On 23/08/11 12:48 AM, virendra bhati wrote:


When I try this application I get these error list. As per the documents
there is some header files and other files is required before installed.
Please help me ...

I am using cenos 1.4 with centos 5.5

[root@cent69 asterisk-espeak-0.4]# vi README
[root@cent69 asterisk-espeak-0.4]# make
gcc -pipe   -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g-D_REENTRANT -D_GNU_SOURCE -O6   -fPIC -c
-o app_espeak.o app_espeak.c
app_espeak.c:61:30: error: espeak/speak_lib.h: No such file or directory


It looks like you haven't installed espeak :-)

The application is to connect Asterisk to espeak.

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

[Matt Riddell]

If the peers are SIP you could do:

akl*CLI manager show command SIPpeers
Action: SIPpeers
Synopsis: List SIP peers (text format)
Privilege: system,all
Description: Lists SIP peers in text format with details on current status.
Variables:
  ActionID: idAction ID for this transaction. 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.


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Re: [asterisk-users] Increasing volume ?

[Matt Riddell]

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?

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Re: [asterisk-users] Increasing volume ?

[Matt Riddell]

On 4/08/11 2:12 AM, Zeeshan Ali Shah wrote:

Hi, I am running asterisk with konference .  tried to increase the
conference voice but not success

i tried to add in diaplain
SetGlobalVar(Set(VOLUME(TX)=10))
SetGlobalVar(Set(VOLUME(RX)=10))


Should be:

SetGlobalVar(VOLUME(TX)=10)
SetGlobalVar(VOLUME(RX)=10)

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

[Matt Riddell]

On 16/07/11 12:29 AM, Michelle Dupuis wrote:

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 server. What is the best way to ensure I stay within an arbitrary
limit (eg: 10 simultaneous call files in process at once)?
The call files will be written to the spool directory by a bash file, so
ideally the bash file should have visibility into the number of .call
files in process.


Use the Asterisk manager instead.

Or do something like:

asterisk -rx'core show channels'|grep active channels|cut -d' ' -f1

and either store the output in a variable or pipe it to a file.

The other alternative would be to do core show channels concise|grep 
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 :-)


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

[Matt Riddell]

On 12/07/11 9:29 AM, Steve Edwards wrote:

Many times, I've made the statement that you can execute hundreds of
AGIs written in C in the time it takes to load an interpreter and parse
a script written in PHP or Perl.


It would be interesting to see the same types of tests run against 
fast-agi - personally if I write an agi that will be called 1000 times 
I'm going to leave it running and have network requests against it 
rather than starting and stopping every time.


Interesting tests nonetheless - I would have been pretty concerned if 
straight C wasn't faster :-)


Mind if I post it to the Daily Asterisk News?

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

[Matt Riddell]

On 1/06/11 11:03 PM, Örn Arnarson wrote:

Hi Matt,

Yes, passing two carriage returns. I login successfully. Here's
example output (with my comments in [])

Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.1
action: login
username: phpagi
secret: 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

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

[Matt Riddell]

On 1/06/11 5:38 AM, Örn Arnarson wrote:

The problem presents with a php script that opens a socket directly to
AMI, a telnet client from the local machine to the AMI, but not when I
telnet from the machine to a remote machine running AMI.


Are you 100% sure it's happening when you use telnet on the local machine?

Are you passing two carriage returns after logging in?

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Re: [asterisk-users] make calls from DID

[Matt Riddell]

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

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Re: [asterisk-users] make calls from DID

[Matt Riddell]

On 27/05/11 5:27 AM, virendra bhati wrote:

Hi,
Thanks for replay ...

How to get incoming call from DID to server ?


You might want to read:

http://downloads.oreilly.com/books/9780596510480.pdf

Or you can support it and future books by buying a copy:

http://oreilly.com/catalog/9780596009625

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

[Matt Riddell]

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

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

[Matt Riddell]

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 :-)

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

[Matt Riddell]

On 13/05/11 4:38 PM, Jose P. Espinal wrote:

Hello,

After installing Asterisk from source in Slackware 13.1, I get the
following error:

Error loading module 'res_config_odbc.so':
/usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
ast_odbc_clear_cache

Then a core dump.


Do a backtrace on the core dump.

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

[Matt Riddell]

On 15/05/11 12:40 PM, Steve Edwards wrote:

Adding a couple of lines to root's crontab like:

# Min hour DOM month DOW command
# --
# */5 * * * * /etc/init.d/iptables stop

make it easy to enable an 'iptables failsafe' (by un-commenting the last
line) while you're fiddling about.


What a great idea! I've never thought of doing that!

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

[Matt Riddell]

On 17/05/11 3:25 AM, vip killa wrote:

I'm using a perl daemon i wrote to connect to AMI and perform actions.
The daemon connects to asterisk via AMI at start up. Is there anyway to
check if the AMI connection is still alive, for example every 2 seconds.
if the connection is not alive, re-connect 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.


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

[Matt Riddell]

On 17/05/11 1:36 PM, Steve Edwards wrote:

On 15/05/11 12:40 PM, Steve Edwards wrote:

Adding a couple of lines to root's crontab like:

# Min hour DOM month DOW command
# --
# */5 * * * * /etc/init.d/iptables stop

make it easy to enable an 'iptables 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 hosts.


Yeah exactly - hence my excitement at the idea :-)

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

[Matt Riddell]

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 ;-)

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Re: [asterisk-users] Plotting fxotune dump

[Matt Riddell]

On 11/05/11 12:21 AM, Ross Cameron wrote:

Can anyone shed any light on how you plot the data outputted by fxotune
-d into a graph similar to the one in the link below?

http://www.voip-info.org/storage/users/65/31065/images/673/medium.jpg


That looks like a plot from oslec rather than fxotune.

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

[Matt Riddell]

On 11/05/11 11:39 AM, Sherwood McGowan wrote:

On Tue, May 10, 2011 at 5:38 PM, Nic Colledge n...@njcolledge.net
mailto:n...@njcolledge.net wrote:

+1 I agree with all of this.

Just wanted to add that there’s another good reason for not
contacting people off-list and that is when you do so you leave no
record of your conversation that others can use in future to solve
the same problem.

You can learn a lot just by reading the archives.

Very good point Nic! Oh and thanks for the +1...Usually when I get on a
rant, I piss everyone off ;-)


+1 from me too.  The other thing is that when you answer to say the 
problem has been solved this goes into the archives meaning that people 
can use Google to answer their own questions rather than having to even 
ask the list.


There have been times when I've searched for a solution to a problem, 
found like 10 answers, and nobody has said whether they work or not so 
you have to try all of them.


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

[Matt Riddell]

On 11/05/11 3:11 PM, John Wu wrote:

Hi Enrico
thanks I do what u said but meet this problem:
[May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File
fail.wav does not exist in any format
[May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to
open fail.wav (format 0x2 (gsm)): No such file or directory
[May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/IMSI460020656177633- for fail.wav


When you playback a file in Asterisk you don't provide the extension.

So you'd do Playback(fail) rather than Playback(fail.wav)

That way if you have 10 files all called fail (i.e. fail.wav, fail.gsm, 
fail.ulaw, fail.alaw etc etc) it will pick the one which matches the 
phone calls.


For example in the above example you were making a call in the GSM 
format but it couldn't find a file called fail.wav.gsm so it looked for 
fail.wav.* and couldn't find anything.


Basically just drop the extension.

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

[Matt Riddell]

On 10/05/11 2:32 AM, Naomi Rosenberg wrote:

Hi,

Apologies if this is a duplicate - been having mail server issues and I don't 
think I managed to send it when I tried this morning.

It seems there is no .conf syntax highlighting script available for gedit. I'm 
thinking of putting one together myself, but don't want to reinvent the wheel.

So I'm just enquiring if anyone knows of one that already exists that i've 
missed.


There have been some editor highlighting stuff submitted to the list in 
the past but as far as I'm aware, nothing for gedit.  If you do put 
something together post it here and I'll post it to the Daily Asterisk News.


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

[Matt Riddell]

On 5/05/11 10:46 PM, Shawn L wrote:


Yes, I'm talking about mid-call.

I do have rtptimeout and qualify set, both to 30 seconds, which should
be plenty of time.
I set them both because if a phone moves out of range, and never comes
back, asterisk was keeping the channel open way to long.


Yeah, but 30 seconds doesn't start from when it disappears.  It's every 
30 seconds.  So that could be like 1 second etc.


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

[Matt Riddell]

On 6/05/11 8:35 AM, Ira wrote:

At 01:07 PM 5/5/2011, you wrote:

I am not saying using production servers to test, rather reproducing
your production setups in a test environment. You would then create
test plans or test cases of the features you use in Asterisk. Once
documented, for each and every RC of Asterisk you go through the steps
outlined in your test plan / case, confirming this work as expected
and then documenting the results.


Yes, but in my world there is one Atom powerd Linux box running
Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has
to be on my production box and I'm more than happy to run beta software
on that box. My comment is just that the protocol for me helping you is
not clear to me. I have been beta testing since 1985 when I was able to
crash Brief on the Novell network I used at work.


Were you beta testing using your production servers then?

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

[Matt Riddell]

On 6/05/11 3:14 PM, Ira wrote:

At 03:00 PM 5/5/2011, you wrote:


Yes, but in my world there is one Atom powerd Linux box running
Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has
to be on my production box and I'm more than happy to run beta software
on that box. My comment is just that the protocol for me helping you is
not clear to me. I have been beta testing since 1985 when I was able to
crash Brief on the Novell network I used at work.


Were you beta testing using your production servers then?



Yes, I use my one and only server for testing. Brave and foolish soul
that I am!


:-)

Fair enough then!

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

[Matt Riddell]

On 3/05/11 4:01 AM, Hans Witvliet wrote:

Just a thought
If Digium / the community realy want an objective way of deciding
whether can/should migrate to any other version, you realy need a
feature-matrix (pethaps starting from version 1.2.*)

And for every and each version a statement 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!

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

[Matt Riddell]

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?


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Re: [asterisk-users] receive faxes

[Matt Riddell]

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?

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

[Matt Riddell]

On 5/05/11 10:21 AM, Shawn L wrote:

  I have a situation where we have an asterisk box that is extending
several Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310).   Everything works great,
except when the cordless
phone walks out of range of one access point and into range 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.

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

[Matt Riddell]

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?


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Re: [asterisk-users] Fading voice problem

[Matt Riddell]

On 3/05/11 10:16 PM, Eduardo Leones wrote:

Guys,

I'm having problems in the fading voice calls, receptive and active,
that in SIP accounts. While few people using the system, calls are
perfect, but it beats the normal use of connections (average 30
concurrent), the voice begins to fade from people.

Soon I figured some network problem, I did a tcpdump and analyzed by
wireshark ...the strange thing is this ...

all packets that arrive on the server asterisk are normal or jitter,
latency ... But whenAsterisk sends packets to the network or the ISP ...
maggoty packages are ... jitter of150ms on average ... latency of more
than 1000 ms ...

That is, by the way is not the network itself, but the network on the
machine ...

Dropped iptables to make sure no influence ... I changed the network
card and cables... did nothing more ...

Anyone have any ideas to help me and chase to find 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?

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

[Matt Riddell]

On 28/04/11 8:00 PM, virendra bhati wrote:

Hi,

Asterisk server is running on this machine then I tested and I got this
message after run the script.


What user are you running the script as?

It looks like you're running it as a web server when Asterisk is running 
as root?


Try running the script from the commandline while logged in as root to 
confirm:


php script.php

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

[Matt Riddell]

On 28/04/11 10:53 PM, virendra bhati wrote:

Hi,

I am running this script from wabsite and want to make it just like
FreePBX show status information of  Asterisk, mysql etc.

*Asterisk is running as root* at my end.

When I start script from command prompt then I am getting error message..

[root@cent68 mtnl]# php temp.php
PHP Parse error:  syntax error, unexpected ',' in
/var/www/html/mtnl/temp.php on line 3

Parse error: syntax error, unexpected ',' in /var/www/html/mtnl/temp.php
on line 3


You have an error on line 3 of your code - probably somewhere around the 
comma.


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

[Matt Riddell]

On 29/04/11 10:10 AM, Alec Davis wrote:

Thanks Matt.

There seems to be an unresolved deadlock since the birth of 1.8.
Using the most basic feature of a PBX, try to pickup some elses ringing
extension - DEADLOCK.

But I'm on to it, https://issues.asterisk.org/view.php?id=18654 and it's
more uptodate review https://reviewboard.asterisk.org/r/1185/


Yeah, not sure why that one's not affecting me.

I'm using the Set(_PICKUPMARK=1) thingy with a little bit of logic from 
DB functions and customers don't seem to be hitting it.


I'm not using the *8 thing though.

Yeah, just checked one system 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.

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Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

[Matt Riddell]

On 29/04/11 11:19 AM, Jan Bakuwel wrote:

Hi,

I'm about to deliver a production system based on Debian Squeeze and
Asterisk 1.6.2.9-2+squeeze1 from the Debian repositories. Asterisk 1.8
packages for Debian  Ubuntu are available from packages.asterisk.org.
Observing some recent discussions on this list, it seems that 1.8 might
not yet be ready for production use. Would whoever kindly makes the
Asterisk 1.8 packages available also consider doing that for 1.6
releases? If the build environment has been set up for 1.8, I'd imagine
it would be easy to set up something similar for 1.6 releases?


You shouldn't put *any* system into production unless you have a clear 
list of what features you will be providing, and have a way of testing 
that those features work :-)


If you do this then every update can be tested to work with those 
features, and a customer's system shouldn't crash, no matter what 
version you're using.  As I've said 1.8 is working under these 
circumstances for me in production.


One thing I'll note though is that as time goes on and you get better at 
these types of things you'll come up with some pretty crazy tests - and 
still customers will do things you couldn't possibly have thought to test.


So, long story short I recommend:

1. Make a list of the applications and modules you'll be using and a 
list of ways they'll be used.


2. Disable everything else

3. Test these apps/functions in the most intense way you can think of

4. Move the system 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.


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

[Matt Riddell]

On 29/04/11 11:51 AM, Ernie Dunbar wrote:

On 29/04/11 5:06 AM, Ira wrote:

At 05:56 AM 4/28/2011, you wrote:

If I can install 1.8 and
know that I can turn off things to get to 1.4 solidness, then I
don't
have a problem with this kettle of fish. BTW, where does 1.10 fit into
this
conversation?


Personally, 1.8 has never lasted more than 12 hours on my box without
dying and once I figured out how it dies, every beta and every release
will fail within moments if I followed the same very short test script.
I did put up a bug report on the problem once and was told within
moments it wasn't a bug, but I'm not smart enough to understand what I'm
supposed to do to troubleshoot and the same configuration has always run
on 1.2, 1.6 and 1.10 so from my perspective, it's a bug.


What's the URL to the bug you submitted?

I'm running 1.8 here 24/7 with no problems other than the ones that Alec
Davis fixed.  I've got it running in I think 4 installations and we're
not getting any core dumping or anything - obviously I'm only using a
subset of the full functionality and most modules are not included.


What features do you have disabled? It would be helpful to know this for
future 1.8 implementation, although right now we can't quite use it yet.


The opposite of what we're using :-)

We've been reworking our GUI software to work on embedded systems as 
well as larger so we use:


AGI for all outbound calling logic - our licensing code sets up routes 
for the customers (i.e. which providers they're using etc) and then they 
chose order (i.e. VoIP followed by Analogue etc).  If a destination 
can't be matched via an outbound route then the call is passed back to 
the dialplan.


Applications/Functions we use:

Macro, Dial, VoiceMail, VoiceMailMain, Goto, GotoIf, GotoIfTime, Hangup, 
UserEvent, Answer, Playback, Record


Some others:

* Pickup application with PICKUPMARK
* DB Functions
* We don't use Asterisk Realtime for these systems
* Call transfers etc are all done by the phones themselves
* DAHDI for timing - even if it's just DAHDI_dummy
* MeetMe (we haven't started using confbridge yet)
* Set application for variables
* hints
* SIPAddHeader or Set(__SIPADDHEADER=
* Outbound calling via IAX2, DAHDI and SIP - depending on the customer
* RFC2833 or Inband DTMF (depending on issues)

And that's it.

We don't use any of the imap voicemail stuff, don't usually use Google 
Talk or anything.  Don't usually use Jabber. Try to stay away from Local 
channels wherever possible.  Restart Asterisk in the middle of the night 
in case there are any memory leaks.


If we ever have any problems we try to track it down to the exact 
revision that caused the problem, read the commit and try and submit a 
bug entry with as much detail as possible.  It's pretty unusual for you 
to be the only person experiencing a bug so normally if you come across 
something you'll see other people with the same problem.  If you don't 
it's because you're doing something different to the majority of users 
or it's a very new bug.  So you first look at what you're doing that's 
different (we use chan_lcr occasionally as BRI isn't working for us with 
DAHDI - LCR has caused some issues).  If it is caused by doing something 
in a way that is different then see if you can do it how most people 
would.  If it still causes an issue, either fix it or submit a ticket. 
You can usually work around most things.


For example we had a problem last week where an incoming call to a DDI 
had a 302 redirect from the phone to another number - i.e. the person 
was out of the office so they redirected to their cell.  When the call 
went back to Asterisk it used the local channel and made an outbound 
call to the cellphone.  After 2 seconds of ringing it would hangup and 
head back to the desk phone - that would redirect it back to the 
cellphone etc etc.


It turned out that for whatever reason the LCR channel wasn't happy with 
the redirection - when tested with the incoming call coming from IAX 
instead of LCR it worked fine.  We then thought that maybe it was 
because the LCR channel hadn't been answered.  We added an Answer() 
before sending the call to the phone 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.


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

[Matt Riddell]

On 29/04/11 1:16 PM, Ira wrote:

Well, I've no idea how to do that. I can duplicate the problem every


IRC is an online chat system like MSN or Skype except that it's more 
like a mailing list - you can talk to lots of people at the same time.


On Windows you can use a program like mIRC 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.

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

[Matt Riddell]

On 29/04/11 2:15 PM, Alec Davis wrote:

Let me try to better describe the test senario that I found, and have been
commited to 1.4svn, 1.6.2svn 1.8svn and trunk.
All aspects need to be thrased out though.

Leave Phone-A 2 new messages, and for this example we only have 2 new
messages.

Now 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


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Re: [asterisk-users] Asterisk 1.6.2.18 Now Available

[Matt Riddell]

On 29/04/11 12:01 PM, Jan Bakuwel wrote:

Rather than testing and finding issues that have already been resolved,
I'd prefer to have an efficient way to upgrade Asterisk to released
versions. A package system provides an efficient way to do this. The
fact that something like packages.asterisk.org exists seems to prove my
point. Upgrading the system obviously doesn't mean you won't have to do
any testing but it should make the testing more efficient - at least for
stable releases.


Each to their own - I find it easier to patch particular issues rather 
than potentially introduce new issues but hey :-)


Asterisk 1.6.2 won't be receiving any bug fixes though as it has gone to 
security only, so I wouldn't personally put it in production.


We're in a kinda interesting scenario - 1.4 is the most stable by far.

So if you're happy with the features in 1.4 that's what you should use 
for production.


If you're willing to do a little extra work right now then you should be 
going with 1.8 as it will be supported for quite some time (at least 
till October 2014).


See:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Series, TypeRelease DateSecurity Fix Only   EOL
1.2.X,  STD 2005-11-21  2007-08-07  2010-11-21
1.4.X., LTS 2006-12-23  2011-04-21  2012-04-21
1.6.0.X,STD 2008-10-01  2010-05-01  2010-10-01
1.6.1.X,STD 2009-04-27  2010-05-01  2011-04-27
1.6.2.X,STD 2009-12-18  2011-04-21  2012-04-21
1.8.X,  LTS 2010-10-21  2014-10-21  2015-10-21

Where STD is Standard and LTS is Long Term Support.

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

[Matt Riddell]

On 23/04/11 8:45 AM, Tzafrir Cohen wrote:

So here's a mini poll:

Do you have a manager interface user that does not have all the read and
write permissions? If so: how have you managed to do so?

* Reading documentation / source
* An existing sample
* Trial and Error


Trial and Error - removed everything and added them back in as required.

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Re: [asterisk-users] PAP2T auto answer?

[Matt Riddell]

You could try:

exten = *701,1,Set(__SIPADDHEADER=Call-Info:sip:192.168.101.1\; 
answer-after=1)


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Re: [asterisk-users] Softphone IAX

[Matt Riddell]

On 19/04/11 1:19 AM, Eduardo Leones wrote:

Anyone know a good IAX2 softphone for Windows that has g729 and it is free?


That's not going to happen.  g729 is not free so how can a softphone 
that uses it be free unless it isn't honouring the licenses.


Don't use this as an excuse to discuss 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?


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

[Matt Riddell]

On 20/04/11 1:58 AM, Mark Deneen wrote:

2011/4/19 Niccolò Bellidarkbas...@gmail.com:

Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:

Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and 
internet is offline.


srvlookup = no didn't help.

What about putting my provider's name in /etc/hosts?
Should it solve the problem?

A caching nameserver is not a viable solution because I want it working
even after a month without internet access.


Wouldn't a caching nameserver just return NXDOMAIN if it couldn't
contact the authoritative server for that domain?


It should 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.

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