How to enable VOIP?

[Francis]

I am experiencing some difficulties when trying to get the VOIP
working.
This is where I am so far:
- I installed BBB using the apt-get package. I also installed the
desktop sharing module. This all works fine.
- Then I went to the "Install from source" guide to find out how to
install asterisk and zoiper. Which I did, but that part was a bit
tricky because I needed to filter out the stuff that was already taken
care of by the package install.

After having installed asterisk and zoiper and running all the little
config scripts I seem to be stuck. Can someone help me by answering
the following questions?

- Do I need to add the 'echotest' user as a SIP account to zoiper?
- If yes, then what should I specify as domain, username, password and
callerID? (Is 'echotest' the username or the callerid?)
- Why do you need to dial echotest/600? Is this for testing purposes,
or essential to enable VOIP?
- Is zoiper an application that should run on the server only, or also
on the clients?
- If yes, then which username should they use?
- The BBB UI has a headset icon in its top bar. How is this related to
asterisk/zoiper?

I'd greatly appreciate your help on this.

[Maciej Sawicki]

Hi,

On Tue, Nov 3, 2009 at 09:40, Francis <francis....@gmail.com> wrote:
> - Is zoiper an application that should run on the server only, or also
> on the clients?

Zoiper is sip phone. I think it should be run on clients only.

> - The BBB UI has a headset icon in its top bar. How is this related to
> asterisk/zoiper?

I tested this icon in it works like flash camera. It need flash
permissions for recording. It can work without asterisk.

regards,
Maciek Sawicki

[Francis Rammeloo]

When I press the the headset icon nothing happens. I tested this on Linux, Mac and Windows.

[Maciej Sawicki]

On Tue, Nov 3, 2009 at 10:13, Francis Rammeloo
<francis....@gmail.com> wrote:
> When I press the the headset icon nothing happens. I tested this on Linux,
> Mac and Windows.

i just tested it on bbb demo site on win xp. When i clicked this icon
flash asked me about permissions. After enabling recording i was moved
to voice group. Few days ago i tested it on VM with the same effect.

regards,
Maciek

[Francis]

I just tested it on the demo server and it indeed works.
When I try with my own server it doesn't work (the webcam does
however). So something must be wrong with my asterisk/zoiper setup
server-side. Some answers to my earlier questions would be very
helpful.

Grts,
Francis

On 3 nov, 10:22, Maciej Sawicki <viroos...@gmail.com> wrote:
> On Tue, Nov 3, 2009 at 10:13, Francis Rammeloo
>

[Maciej Sawicki]

On Tue, Nov 3, 2009 at 10:40, Francis <francis....@gmail.com> wrote:
>
> I just tested it on the demo server and it indeed works.
> When I try with my own server it doesn't work (the webcam does
> however). So something must be wrong with my asterisk/zoiper setup
> server-side. Some answers to my earlier questions would be very
> helpful.

Unfortunately I can help you more. Even worse i will hijack this
thread and ask one more question:

How can I configure asterisk for using with external voip account. For
example those are instructions form my provider:
https://www.freeconet.pl/forum/viewtopic.php?t=2905. But I'm not sure
about conferencing calls. Will they work out of the box?

TIA for help

regards,
Maciek Sawicki

[Richard Alam]

Hi Francis,

Please see answers below.

Please check logs at /usr/share/red5/log (sip.log, access.log) to see
what may be wrong. The voip app needs more work which we plan on for
the 0.63 release. But first we have to release first 0.62 which we are
wrapping (documenting) now.

Hope this helps.

Richard

On Tue, Nov 3, 2009 at 4:40 AM, Francis <francis....@gmail.com> wrote:
>
> I am experiencing some difficulties when trying to get the VOIP
> working.
> This is where I am so far:
> - I installed BBB using the apt-get package. I also installed the
> desktop sharing module. This all works fine.
> - Then I went to the "Install from source" guide to find out how to
> install asterisk and zoiper. Which I did, but that part was a bit
> tricky because I needed to filter out the stuff that was already taken
> care of by the package install.
>
> After having installed asterisk and zoiper and running all the little
> config scripts I seem to be stuck. Can someone help me by answering
> the following questions?
>
> - Do I need to add the 'echotest' user as a SIP account to zoiper?
> - If yes, then what should I specify as domain, username, password and
> callerID? (Is 'echotest' the username or the callerid?)
> - Why do you need to dial echotest/600? Is this for testing purposes,
> or essential to enable VOIP?

The echotest is just a test account. This way you can verify that the
asterisk extensions are properly setup.
The echotest account was create using the "cat asterisk-sip.conf >>
/etc/asterisk/sip.conf" command from
the install instructions. Please see asterisk-sip.conf for the
echotest account information.

> - Is zoiper an application that should run on the server only, or also
> on the clients?

The zoiper is an application on the user's computer. It can be used to
join a conference within BBB if the
user chooses not to use the built-in voip in BBB client.

> - If yes, then which username should they use?

You can create then user accounts. There's a few (user1, user2, user3)
defined in asterisk-sip.conf.

> - The BBB UI has a headset icon in its top bar. How is this related to
> asterisk/zoiper?

This is the built-in voip client. It's another way for a user to join
the voice conference without the need to install
another app like zoiper.

>
> I'd greatly appreciate your help on this.
>
> >
>

--
---
BigBlueButton
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton

[Francis Rammeloo]

Thanks for your answer Richard. I'd be happy to use the Flash based voip client instead of zoiper. If I understand correctly, even without using zoiper I still need Asterisk right?

2009/11/3 Richard Alam <ritz...@gmail.com>

[Richard Alam]

Hi,

Yes, you still need asterisk. Asterisk is used as the voice conference
server while zoiper is just a client.

Richard

[Hugo Flambó]

And what about connect from zoiper to the voice conference?
On installation howto it says to dial extension 1500 ... but I don't
find this extension anywhere in asterisk conf files.... so of course
this can not be done from zoiper...

Do you know how to configure asterisk so we can join a voice
conference with zoiper?

Thanks,

BR

Hugo

[Richard Alam]

Hi Hugo,

Use one to the test user accounts in /etc/asterisk/sip.conf.

For example,
;
; BigBlueButton: Setup sample user to connect over VoIP
[user1]
type=friend
username=user1
insecure=very
secret=secret
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=bigbluebutton
allow=all

Setup your Zoiper to be able to register with your asterisk server
using the above info. You'll be able to see the progress in the
Asterisk console to see if it registers.

On the command line, type:
> asterisk -vvvvvvvr ----> those are lots of letter v for verbose

Then from zoiper, just dial your conference number (e.g. 85115). Watch
the console to see if it's joining the conference.

If you run into issues, post your console log here.

You may have to add "debug" in you /etc/asterisk/logger.conf
i.e.
console => notice,warning,error,debug


Richard

[Hugo Flambó]

Hello Richard... Thanks for your reply...

I can not get it using zoiper... some trace logs



1 -Asterisk console result after the commenction to the CLI
asterisk -vvvvvvvr
Asterisk 1.4.25, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <mark...@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.25 currently running on inmove-debian (pid = 3751)
Verbosity is at least 7


2 - First I enter the application thought the web/flex interface as moderator...

3- I click on the voice conference icon..accept the flash warning for
using microphone...

4 - got this log/traces on asterisk CLI

-- Remote UNIX connection
-- Registered SIP '3001' at 127.0.0.1 port 5071
-- Saved useragent "mjsip stack 1.6" for peer 3001
[Nov 4 18:08:56] NOTICE[3913]: chan_sip.c:12985
handle_response_peerpoke: Peer '3001' is now Reachable. (7ms / 2000ms)
-- Executing [85115@bbb-voip:1] MeetMe("SIP/3001-08d60880",
"85115|cdMsT") in new stack
-- Created MeetMe conference 1023 for conference '85115'
-- <SIP/3001-08d60880> Playing 'conf-onlyperson' (language 'en')
-- Started music on hold, class 'default', on SIP/3001-08d60880


5 - Try to connect using zoiper .... created a profile there
domain=> localhost , user => user1, password => secret
And Register the user with Asterisk ....

6 - Got on asterisk CLI the traces/logs from the user registration before:

-- Registered SIP 'user1' at 127.0.0.1 port 5061
-- Saved useragent "Zoiper rev.5415" for peer user1
[Nov 4 18:14:38] NOTICE[3913]: chan_sip.c:12985
handle_response_peerpoke: Peer 'user1' is now Reachable. (1ms /
2000ms)


7 - using this user registered and from zoip I dial: 85115

8 - Got current traces/ logs -----> EXTENSION NOT FOUND --- rejected...

[Nov 4 18:16:54] NOTICE[3913]: chan_sip.c:14703
handle_request_invite: Call from 'user1' to extension '85115' rejected
because extension not found.


Any idea on this...?

BR

Hugo

[Richard Alam]

Can you post your entry for user1 on /etc/asterisk/sip.conf and your
/etc/asterisk/extensions.conf?

You can post it on pastebin.com and send the link.

Richard

[Hugo Flambó]

Ok

here is the link...


http://pastebin.com/m5b6a4eb4


Thanks

[Richard Alam]

I see. The context for user1 is bigbluebutton which I suspect is the
same for the other users (user2, user3).

In sip.conf, change this (context=bigbluebutton) to (context=bbb-conference).

Try it out.

If that does not work, please post your asterisk console output.

Richard

[Hugo Flambó]

Thanks ... it was that!!:D I've created those users long time ago...I
suppose that I was not updated

Here is the new log.... Just for the prosperity ;) Now I see the
zoiper user on the flex web interface and when i've connected the
music on hold stopped because that zoiper user joined the conference
and the web user I not anymore home alone!! :)

Thank you very much!!

Just a probably silly question: To have VOIP users I have to add
always them manually to the sip.conf file??
I guess that I can use asterisk-gui web interface to do that ...add
new users to the file... and reload asterisk... but this is the way to
do it?
BigBlueButton doesn't have any interface using asterisk-java for this
purpose? (like the conferences and sessions admin)... just to know...
probably I'm completely wrong with all this questions...

Thanks again.... BBB rocks!!

== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found

Connected to Asterisk 1.4.25 currently running on inmove-debian (pid = 16448)
Verbosity was 0 and is now 7
-- Remote UNIX connection
-- Unregistered SIP '3001'
-- Registered SIP '3002' at 127.0.0.1 port 5072
-- Saved useragent "mjsip stack 1.6" for peer 3002
[Nov 4 18:47:27] NOTICE[16473]: chan_sip.c:12985
handle_response_peerpoke: Peer '3002' is now Reachable. (4ms / 2000ms)
-- Executing [85115@bbb-voip:1] MeetMe("SIP/3002-086b4c70",

"85115|cdMsT") in new stack
-- Created MeetMe conference 1023 for conference '85115'

-- <SIP/3002-086b4c70> Playing 'conf-onlyperson' (language 'en')
-- Started music on hold, class 'default', on SIP/3002-086b4c70

-- Registered SIP 'user1' at 127.0.0.1 port 5061
-- Saved useragent "Zoiper rev.5415" for peer user1

[Nov 4 18:47:43] NOTICE[16473]: chan_sip.c:12985
handle_response_peerpoke: Peer 'user1' is now Reachable. (3ms /
2000ms)
-- Executing [85115@bbb-conference:1] AGI("SIP/user1-086b33b8",
"agi://localhost/findConference?conference=85115") in new stack
-- AGI Script agi://localhost/findConference?conference=85115
completed, returning 0
-- Executing [85115@bbb-conference:2] GotoIf("SIP/user1-086b33b8",
"1?valid:invalid") in new stack
-- Goto (bbb-conference,85115,3)
-- Executing [85115@bbb-conference:3]
Playback("SIP/user1-086b33b8", "conf-placeintoconf") in new stack
-- <SIP/user1-086b33b8> Playing 'conf-placeintoconf' (language 'en')
-- Executing [85115@bbb-conference:4] MeetMe("SIP/user1-086b33b8",

"85115|cdMsT") in new stack

-- <SIP/user1-086b33b8> Playing 'conf-onlyone' (language 'en')
-- Stopped music on hold on SIP/3002-086b4c70

[Richard Alam]

Hi Hugo,

For now you have to add them manually.

On the next iteration, we'll figure out the best and easier way to add users.

Richard

[Hugo Flambó]

Ok thanks again!!!

I'll be following the project with attention!;)

BR

Hugo