Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-08 Thread Lenz Emilitri 
2013/1/5 joachim 

>
> You are pretty much limited to measuring the delay and the jitter.
> The delay you can somewhat estimate prior to the call (with qualify for
> example).
> The jitter / packetloss you can only figure out when the call is already
> up for a while. (e.g. you might have no issues the first minute, but maybe
> packet loss will come in bursts after a minute).
>

A few years ago I spoke to a Finnish company that had a commercial solution
for automated MOS estimation. So something exists though I have not tested
it first-hand.
l.

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Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-08 Thread Leandro Dardini 
2013/1/8 Lenz Emilitri 

>
> 2013/1/5 joachim 
>
>>
>> You are pretty much limited to measuring the delay and the jitter.
>> The delay you can somewhat estimate prior to the call (with qualify for
>> example).
>> The jitter / packetloss you can only figure out when the call is already
>> up for a while. (e.g. you might have no issues the first minute, but maybe
>> packet loss will come in bursts after a minute).
>>
>
> A few years ago I spoke to a Finnish company that had a commercial
> solution for automated MOS estimation. So something exists though I have
> not tested it first-hand.
> l.
>
>
For MOS calculation I use voipmonitor, but it computer it at the end of the
call. The voipmonitor guy is very handsome, maybe you can sponsor a patch
to have the MOS calculation in real time. An external software can get it
and halt the call if needed.

Leandro
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Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-08 Thread joachim 


A few years ago I spoke to a Finnish company that had a commercial 
solution for automated MOS estimation. So something exists though I 
have not tested it first-hand.

l.

--
You need a lot of data to calculate a MOS score, you will need the 
actual call.
The only solution i can think of is that the phones start a fake call as 
soon as they are in focus and the server calculates some scores based on 
the fake call. When the client calls, the fake call is terminated and 
replaced with a real call.


About the qualify, i don't know how to get the timing results from 
within the dialplan, i'm not even sure it's possible without patching.


Z.

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[asterisk-users] Monitor extensions status.

2013-01-08 Thread Luis H. Forchesatto 
Greetings.

I got two extensions on my asterisk that autenticates from outside our
network, via internet. Is there a way to monitor, in certain time periods,
if they are available (online) and send some sort of notification if they
don't?

There are two extensions to monitor, they belong to the same queue. Both
must be available to receive calls at the same time and if one or both are
offline I must be notified. They stand behind NAT so making Nagios monitor
will either report wrong extension status (monitoring the NATing
server/router) or simply useless (unless there's a plugin to monitor
asterisk extensions).

But anyway...I'll be open to opinions.

My environment:

- Asterisk 1.6.2.13
- Server running Elastix 2.0.0
- DAHDI v. 2.3.0.1

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Re: [asterisk-users] Monitor extensions status.

2013-01-08 Thread Leandro Dardini 
2013/1/8 Luis H. Forchesatto 

> Greetings.
>
> I got two extensions on my asterisk that autenticates from outside our
> network, via internet. Is there a way to monitor, in certain time periods,
> if they are available (online) and send some sort of notification if they
> don't?
>
> There are two extensions to monitor, they belong to the same queue. Both
> must be available to receive calls at the same time and if one or both are
> offline I must be notified. They stand behind NAT so making Nagios monitor
> will either report wrong extension status (monitoring the NATing
> server/router) or simply useless (unless there's a plugin to monitor
> asterisk extensions).
>
> But anyway...I'll be open to opinions.
>
> My environment:
>
> - Asterisk 1.6.2.13
> - Server running Elastix 2.0.0
> - DAHDI v. 2.3.0.1
>
>
Doing a nagios probe to check for extension status is a matter of just few
lines... I think you can have it done by a developer for less than $30

Leandro
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Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-08 Thread Dmitry 


When i worked in an internet provider with asterisk telephony solution - we 
used Aqua (http://www.sevana.fi) to measure voice quality. several nettops were 
spread across our network. The nettop called to our asterisk, the asterisk 
saved this voice file to the disk, then this file was sent to a server with 
Aqua software which compared this file to its original. then the quality 
(measured in percents) were sent to Zabbix monitoring. actually this data was 
used for analisys and it compares two files (not realtime). 

BR,
Dmitry Pavlenko




 From: Lenz Emilitri 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
 
Sent: Tuesday, January 8, 2013 2:25 PM
Subject: Re: [asterisk-users] Detect Low Quality Calls - Realtime
 



2013/1/5 joachim 

 
You are pretty much limited to measuring the delay and the jitter.
>The delay you can somewhat estimate prior to the call (with qualify for 
>example).
>The jitter / packetloss you can only figure out when the call is already up 
>for a while. (e.g. you might have no issues the first minute, but maybe packet 
>loss will come in bursts after a minute).

A few years ago I spoke to a Finnish company that had a commercial solution for 
automated MOS estimation. So something exists though I have not tested it 
first-hand.

l.

-- 

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Test-drive WombatDialer beta @ http://wombatdialer.com 
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Re: [asterisk-users] Monitor extensions status.

2013-01-08 Thread Leandro Dardini 
Top and bottom post in the same email... don't open again the thread :-)

#!/bin/bash
res=`sudo /usr/sbin/asterisk -rx 'sip show peer $1' | grep Status | cut
-d\: -f 2 | cut -d\  -f 2`
if [ "$res" == "OK" ]
then
echo "OK is registered"
exit 0
else
echo "WARNING peer not registered"
exit 1


2013/1/8 Luis H. Forchesatto 

> Hmmmlooks good, but I'm looking for something that I could do.
>
> I'm not much of outsorcing.
>
> 2013/1/8 Leandro Dardini 
>
>>
>>
>> 2013/1/8 Luis H. Forchesatto 
>>
>> Greetings.
>>>
>>> I got two extensions on my asterisk that autenticates from outside our
>>> network, via internet. Is there a way to monitor, in certain time periods,
>>> if they are available (online) and send some sort of notification if they
>>> don't?
>>>
>>> There are two extensions to monitor, they belong to the same queue. Both
>>> must be available to receive calls at the same time and if one or both are
>>> offline I must be notified. They stand behind NAT so making Nagios monitor
>>> will either report wrong extension status (monitoring the NATing
>>> server/router) or simply useless (unless there's a plugin to monitor
>>> asterisk extensions).
>>>
>>> But anyway...I'll be open to opinions.
>>>
>>> My environment:
>>>
>>> - Asterisk 1.6.2.13
>>> - Server running Elastix 2.0.0
>>> - DAHDI v. 2.3.0.1
>>>
>>>
>> Doing a nagios probe to check for extension status is a matter of just
>> few lines... I think you can have it done by a developer for less than $30
>>
>> Leandro
>>
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>>
>
>
>
> --
> Att.*
> ***
> Luis H. Forchesatto
> Mail: luis_forchesa...@hotmail.com
>
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Re: [asterisk-users] IAX2 support of video

2013-01-08 Thread Danny Nicholas 
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 support of video

 

 
 
According to this:
https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
yes.
 
 

 

I have a local server with two video phones - running SIP to each phone.
Works.
Then I have an IAX2 connection from that local machine to another machine.
then a SIP connection from that machine to another machine where the same
model
video phone is in use. A call to that phone does not show video only audio.

All machines have in sip.conf:videosupport=yes

Is there something else to get SIP/IAX2/SIP video call to work?

Thanks

Jerry

 

Make sure you have the H.26X codec enabled at all points.

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[asterisk-users] Streaming/Recording audio

2013-01-08 Thread Grant Bagdasarian 
Hello Users,

I've been searching for a couple of hours now but I can't find the answers to 
my questions, so here they go:

1)  Is it possible to stream audio files from a webserver during a call by 
configuring this in the dialplan? Something like 
Playback(http://myserver.companynetwork/welcome.alaw)?

2)  After recording is finished using the Record application, is the 
recorded file(the audio stream) accessible to send off to an http handler using 
HTTP POST?

I hope someone could help me out.

Thanks,

Grant

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Re: [asterisk-users] Streaming/Recording audio

2013-01-08 Thread Danny Nicholas 
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant
Bagdasarian
Sent: Tuesday, January 08, 2013 9:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Streaming/Recording audio

 

Hello Users,

 

I've been searching for a couple of hours now but I can't find the answers
to my questions, so here they go:

1)  Is it possible to stream audio files from a webserver during a call
by configuring this in the dialplan? Something like
Playback(http://myserver.companynetwork/welcome.alaw)?

2)  After recording is finished using the Record application, is the
recorded file(the audio stream) accessible to send off to an http handler
using HTTP POST?

 

I hope someone could help me out.

 

Thanks,


Grant

 

1 . AFAIK, playback is just for local files, not web files.   There are
options like mpg123 to play streaming audio files - see
http://nerdvittles.com/?p=92

2.  As long as you save the audio stream in a normal format, this shouldn't
be a problem.

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Re: [asterisk-users] echo from channel bank

2013-01-08 Thread Justin Killen 
Valer,

Thank you for the advice - I have support tickets open with Adtran and Digium 
and we are tracking down the issue.  Hopefully it doesn't come down to adding 
more hardware, but I'll keep that in mind.

-Justin Killen

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Valer Nur
Sent: Monday, January 07, 2013 11:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] echo from channel bank

It sounds to me like you should first discuss it with adtran. The standard echo 
cancellation for Asterisk have a hard time cancelling echo generated at the far 
end, especially if the echo tail/delay is not minimal.

If adtran can not solve the problem at their end, you can use a server-side 
echo cancellation that can handle long echo tail. One option is PBXMate.


From: Justin Killen 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Tuesday, January 8, 2013 12:17 AM
Subject: [asterisk-users] echo from channel bank

I have several adtran 624 with 24 FXS ports hooked up to analog phones.  The 
adtran is connected to asterisk via a channelized T1 into a digium TE820.  I 
have hardware echo canceling enabled on all channels/spans, but there is still 
echo on the lines for both calls out of the trunk, as well as 
station-to-station calls.  I've checked the output of 'dahdi show channel x' 
and see the echo turned on:

dozer2*CLI> dahdi show channel 149
Channel: 149
Description:
File Descriptor: 158
Span: 7
Extension: 98300326
Dialing: no
Context: from-zaptel
Caller ID:
Calling TON: 0
Caller ID subaddress:
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: DAHDI/149-1
Real: DAHDI/149-1
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently ON
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Onhook


I have echocanceller set to HWEC in dahdi/system.conf for all spans, and I have 
echocancel=yes in chan_dahdi.cfg

I've tried reading some echo cancelation articles, but they seem to be focused 
on trunks and not on stations.  Also, I'm not 100% sure if this is an issue 
that I should focus on with asterisk, or if it's something I should first take 
up with adtran?

I'm using dahdi 2.6.1, asterisk 10.10.0.

Thanks in advance,
-Justin

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Re: [asterisk-users] Dialplan - working out when users answer

2013-01-08 Thread Andrew White 
Hey Satish,

I've worked this out. I'm sorry, you were completely right and the context is 
fine. I was testing without answering the call, so the Dial was never 
connected! Doh!

Thanks heaps for your help, it's all working perfectly.

Cheers,

Andrew

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Tuesday, 8 January 2013 12:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer

HI Andrew,
Show your queuecontrol context. You should have  extension s with priority 1 in 
this context.
--Satish Barot

On Mon, Jan 7, 2013 at 12:08 PM, Andrew White 
mailto:and...@computersforall.com.au>> wrote:
Hi Satish,

Thanks for your response - sorry on the slow reply.

So I've tried the following in the dialplan:

exten => 
direct,n,Dial(${QUEUEEXTS},${RINGTIME},U(queueControl,direct^CONNECTED))

This has a very strange behavior - the NoOp that is in 
queueControl,direct,n(CONNECTED) does not show up, however I get the following:

[2013-01-07 17:31:39] ERROR[19135]: app_stack.c:420 gosub_exec: Attempt to 
reach a non-existent destination for gosub: (Context:queueControl, Extension:s, 
Priority:1)

I've also tried with a macro:

exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME},M(inboundconnected))
[macro-inboundconnected]
exten => s,1,NoOp(Inbound connected!)

It definitely seems like it's being called, but again no NoOp:

-- Executing [direct@queueControl:11] Dial("SIP/1000-47f1", 
"SIP/1000,20,M(inboundconnected)") in new stack

I would expect some kind of error if I was doing this wrong - have I missed 
something?

Thanks for your or anyone elses help in advance!

Andrew


To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer

On Wed, Dec 19, 2012 at 12:44 PM, Andrew White 
mailto:and...@computersforall.com.au>> wrote:
Hi Satish/list,

Looks like I spoke to soon.

I have the following in my dialplan:

Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))

And after confirming with a "dialplan show" it was definitely in there, I 
continued to get this:

ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a non-existent 
destination for gosub: (Context:queueControl, Extension:s, Priority:1)

I can't quite work out why it would be trying to s/1 instead of 
direct/CONNECTED =/.

Any ideas?

Thanks!
In your case, direct and CONNECTED have to be arguments and not the extension 
and priority value respectively. Calling Subroutine from dial will always start 
execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using ^ 
as a delimiter.

--Satish Barot



To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dialplan - working out when users answer

Thanks Satish, fantastic advice. I didn't even think to look into the dial 
options - doh!

Thanks very much,

Andrew


On Wed, Dec 19, 2012 at 10:53 AM, Andrew White 
mailto:and...@computersforall.com.au>> wrote:
Hey guys,

I've got a part of my dialplan that dials multiple people:

exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME})

Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. 
SIP/100&SIP/101&SIP/105 etc

This works great, however I want to see if I can find a way to work out (and 
run an AGI script) when the call is picked up by someone.

Thanks all!

Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.

--Satish Barot

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[asterisk-users] .call file retry issue in Asterisk-10.11.1

2013-01-08 Thread pankaj pandey 
Hi,

I am working on Asterisk-10.11.1,I tried to generating outbound call through 
.call file and facing a issue that call retry was happening after call 
Answered.Is it bug in that Version or i missed some thing.
Here is my call file is-

Channel: DAHDI/G1/09990212758
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: menu
Extension: 1234
Priority: 4


Please suggest.


 
Thanks & Regards,
Pankaj Pandey
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Re: [asterisk-users] Streaming/Recording audio

2013-01-08 Thread Grant Bagdasarian 
Hello,

For some reason I did not receive any replies related to my question by mail, 
but I found the topic back on the online mailing archives. I hope by supplying 
the same subject this email will be logged in my previously created topic 
instead of a new one. If it does not, I apologize.

Regarding my second question, is the recorded stream also available in the 
dialplan after the recording has finished?

Regards,

Grant
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