[asterisk-users] Streaming for ASR

2016-10-19 Thread Luca Pradovera 
Hello,
(sorry for not continuing the thread, I had set the list to digest).

Would UnicastRTP be able to output u-law frames directly? If so, I think
that is all I need.
Does anyone know what the EAGI output is? Raw RTP?

Best regards,

Luca
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[asterisk-users] Scenario of confbrige & calls!

2016-10-19 Thread Mandar Khire 
Hi,
I learn basic confbridge .

I am thinking to solve following scenario (Hypothetical):-

1. there are few sip entities like 6001,6002...600N & few like 7001,7002
etc. These 700X people only knows sip entities not know any conference room
numbers.

2. Every sip entity (only 6XXX) should have own conference room like
6001 have 1001, 6002 have 1002,,600N have 100N.

3. Every sip entity should be waiting in its own conference.
(that can be done by dialing 100X from 600X's softphone.)

4. Now if 7001 want to talk with 6001, he/she dial 6001 from his/her
softphone as he/she not know about conference room number.

5. 7001 should automatically enter in conference 1001 & talk with 6001
which already waiting there.

6. Same way there should be N conference run at same time. (N = as much as
possible).

Question is:-

1. How can be this happens?

2. What should I write in extensions.conf or other conf files so this can
be happens?

Need help!

Mandar P. Khire
+919769419340
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Re: [asterisk-users] Streaming for ASR

2016-10-19 Thread Matt Riddell 

> On 19/10/2016, at 2:32 AM, Luca Pradovera  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/ 


I'm still working on mine :-)

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Re: [asterisk-users] Streaming for ASR

2016-10-19 Thread Joshua Colp 

Matt Riddell wrote:



On 19/10/2016, at 2:32 AM, Luca Pradovera mailto: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/


I did indeed, although I think the options have since changed.

channel originate UnicastRTP/127.0.0.1:5001//c(g722) extension 1000@test

Would be the current.

As well running from git is currently needed as a bug was fixed which 
would cause a crash in UnicastRTP.


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[asterisk-users] CONNECTEDLINE endpoint support

2016-10-19 Thread marek cervenka 

hi,

i'm testing CONNECTEDLINE function

https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

example dialplan

same => n,set(CONNECTEDLINE(name,i)=aastra)
same => n,set(CONNECTEDLINE(name-pres,i)=allowed)
same => n,Set(CONNECTEDLINE(num,i)=5551212)
same => n,Set(CONNECTEDLINE(num-pres)=allowed)
same => n,dial(SIP/sipline501,,I)

it only works with mitel(aastra 6767i) phones

i tested - grandstream 2130, jitsi, blink, microsip, zoiper - nothing worked

what devices working for you with CONNECTEDLINE function?

thanks

Marek



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Re: [asterisk-users] CONNECTEDLINE endpoint support

2016-10-19 Thread Doug Lytle 
>>> On Oct 19, 2016, at 8:37 AM, marek cervenka cerva...@gmail.com wrote:

>>> i'm testing CONNECTEDLINE function
>>> what devices working for you with CONNECTEDLINE function?

Cisco and Polycom

Doug

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Re: [asterisk-users] Configuration management and update deployment - what do you use?

2016-10-19 Thread Duncan 


On 19/10/16 05:57, Ludovic Gasc wrote:

+10 for Ansible.

We use that on our production.



Okay, I will investigate Ansible

Thanks very much

Cheers Duncan

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2016-10-18 14:00 GMT+02:00 marek cervenka >:


ansible.com 


Dne 18/10/2016 v 11:46 Duncan napsal(a):

Hi All

We have about 15 different asterisk boxes around the place and
on my list has been automate deployment updates and keep a
revision history. They are mostly not publicly accessible, and
external SIP access is closely firewalled , so updates happen
straight away when its something like heartbleed, but take a
while to trust/test new releases.

Our boxes are Ubuntu LTS - mostly 14.04 at the moment. We use
Freebpx as the configuration front end and so that tends to be
a more manual update, although there is an API we could use to
keep things in step. We run backups from freepbx and archive
those as well as any specific asterisk settings missed. At the
moment our scale means manual is okay, but automation would
make it easier if the learning curve and new issues aren't too
high.

We compile asterisk from source as the packages aren't usually
quite what we want.

I was just curious how people deploy asterisk across multiple
platforms and keep them all up to date?

What tools are good for this sort of thing?

Thanks very much

Cheers Duncan




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[asterisk-users] tcpenable

2016-10-19 Thread Jerry Geis 
I am playing with tcpenable... on 13.11.2

so in sip.conf I have

tcpenable=yes
tcpbindaddr=192.168.1.8:5070

but when I "telnet localhost 5070" I get no connect.

 iptables -L -n -v | grep 5070
0 0 ACCEPT tcp  --  *  *   0.0.0.0/0
0.0.0.0/0state NEW tcp dpt:5070
firewall is good.

Is my syntax not correct above to run on port 5070 for SIP over TCP?

Thanks,

Jerry
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Re: [asterisk-users] tcpenable

2016-10-19 Thread Steve Edwards 

On Wed, 19 Oct 2016, Jerry Geis wrote:


I am playing with tcpenable... on 13.11.2


This may yield clues:

sudo netstat --all --numeric --program | grep asterisk

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Re: [asterisk-users] tcpenable

2016-10-19 Thread Jerry Geis 
Hi Steve,

>
>  netstat --all --numeric --program | grep asterisk
tcp0  0 192.168.1.8:50700.0.0.0:*   LISTEN
 32379/asterisk
tcp0  0 0.0.0.0:50380.0.0.0:*   LISTEN
 32379/asterisk
udp0  0 0.0.0.0:27270.0.0.0:*
32379/asterisk
udp0  0 0.0.0.0:50000.0.0.0:*
32379/asterisk
udp0  0 192.168.1.8:50600.0.0.0:*
32379/asterisk
unix  2  [ ACC ] STREAM LISTENING 78383649 32379/asterisk
/var/run/asterisk.ctl


Jerry
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Re: [asterisk-users] tcpenable

2016-10-19 Thread Joshua Colp 

Jerry Geis wrote:

I am playing with tcpenable... on 13.11.2

so in sip.conf I have

tcpenable=yes
tcpbindaddr=192.168.1.8:5070 

but when I "telnet localhost 5070" I get no connect.


You are explicitly binding to 192.168.1.8 port 5070. Trying to connect 
to localhost won't work. You will either have to bind to 0.0.0.0 or 
connect to 192.168.1.8 instead.


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Re: [asterisk-users] tcpenable

2016-10-19 Thread Steve Edwards 

Jerry Geis wrote:



I am playing with tcpenable... on 13.11.2

but when I "telnet localhost 5070" I get no connect.


On Wed, 19 Oct 2016, Joshua Colp wrote:

You are explicitly binding to 192.168.1.8 port 5070. Trying to connect 
to localhost won't work. You will either have to bind to 0.0.0.0 or 
connect to 192.168.1.8 instead.


Or bind to 127.0.0.1 - aka localhost :)

0.0.0.0 means 'bind to all addresses on the host.'

The 'bind' and the 'telnet' need to match. You probably mean to bind to 
the host's IP address (slightly more secure) so the 'telnet' needs to 
match that IP address.


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