[asterisk-users] Conference topology

[Dovid Bender]

Hi,

Does anyone have any guides, documents on best practice for "bridging"
multiple Asterisk boxes together so no matter what box a person lands on,
they can be on the same call? I assume the easiest would be to have one box
dial out to all other boxes and bridge them. For example If we have room
100 on Box A. We would initiate a call from A -> B and then from A -> C so
if this way if anyone talk on C it's heard on B and A and vice versa. Is
there any specific logic to how this is done? Do you always designate one
host to do all of the bridging or do you randomly select one box to do the
hosting? How do you plan for a failure if the main bridging host fails?

TIA.

Regards,

Dovid
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Re: [asterisk-users] Conference bridge recording file name

[Nick Olsen]

That did it! I had missed that option. Thanks for the assistance!

On Thu, Aug 26, 2021 at 9:50 AM Doug Lytle  wrote:

> According to the wiki, you can disable the timestamp
>
> record_file_timestamp
>
> Append the start time to the record_file name so that it is unique.
>
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge
>
> Doug
>
>
>
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Re: [asterisk-users] Conference bridge recording file name

[Doug Lytle]

According to the wiki, you can disable the timestamp

record_file_timestamp

Append the start time to the record_file name so that it is unique. 

https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge

Doug



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[asterisk-users] Conference bridge recording file name

[Nick Olsen]

Hello, I'm attempting to enable conference bridge recording.

I have it working, and I'm dynamically pushing the filename onto the bridge
via the set CONFBRIDGE commands. But it seems regardless of what name I
set, the actual filename is written as WHATIPROVIDED-uniqueid.wav.

Example, I use the following command to set the recording file prior to
calling confbridge in the dialplan. (Realtime dialplan)
Set
CONFBRIDGE(bridge,record_file)=/var/spool/asterisk/monitor/confbridge-NicksBridge-1234.wav

However, The file is actually written as
"confbridge-NicksBridge-1234-1629925359.wav"

I'm attempting to have a known name for the recording file that is NOT
unique as the next step is using the record_command to call a script that
runs after the conference to upload the file to a MySQL database. Is there
any way to get confbridge to not append the unique ID to the end of the
file name? Or perhaps a variable I can call in the record_command field to
predict what this number will be?
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[asterisk-users] Conference bridge profile does not exist

[Thomas]

Hello,

I have ab profile in /etc/asterisk/confbridge.conf but in my dialplan this 
profile is not found
I tried a lot, but did no solution.
What can be wrong?


[out_bridge]
type=bridge


exten => ,n,ConfBridge(${conf_room},out_bridge)

[Jun  5 19:27:09] WARNING[11008][C-0468]: app_confbridge.c:1612 
confbridge_exec: Conference bridge profile out_bridge does not exist

The Dialplan is in the same directory and changes will be working, so the 
directory for confbridge.conf should be OK.

version is Asterisk 11.13.1~dfsg-2+deb8u1 built by pbuilder @ compile on a 
x86_64 running Linux on 2016-10-24 19:32:53 UTC

best regards
Thomas

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Re: [asterisk-users] Conference Call like conference done by mobile!

[Matthew Jordan]

On Wed, Oct 5, 2016 at 11:46 PM, Mandar Khire  wrote:
> Hi,
> Thanks for reply.
> For use confbridge I follows link http://www.mytechrepublic.com/?p=418
> By it I manage to create Conference room & add members to it.
> But each member has to dial conference Number.
> In my scenario Only first person dial second person's number.
> Example:-
> If Person1 has 6001, Person2 6002, person3 has 6003 & so on,
> Then In confbridge as per given link example Person1 dial 1030, then person2
> dial 1030, then person3 dial 1030 & so on for conference call.
> But In my scenario Person1 dial 6002, then make it hold, then dial 6003 &
> then merge call.
> Is it depend on softphone functionality or we need to write something in
> some conf file?
> Can we do it some how?
> I tried it on mobile & I can make conference with 6 friends means total 7
> people talk to each other without dial any conference number.
>

There isn't anything in Asterisk, out of the box, that will do
*exactly* what you're describing.

You could create it, however, using ARI [1]. I'd create a special
bridge for users who dial into the system. When they're bridged with
other users, if they hit hold, I'd intercept the hold using the
HOLD_INTERCEPT [1] function, and hang up the hold initiator, keeping
the dialled party in the same bridge. When I get a new dial attempt
from the original caller, I'd put both the caller and the new callee
in the same bridge as the original callee.

This process could be repeated as many times as you want.

[1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
[2] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_HOLD_INTERCEPT

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Re: [asterisk-users] Conference Call like conference done by mobile!

[Mandar Khire]

Hi,
Thanks for reply.
For use confbridge I follows link http://www.mytechrepublic.com/?p=418
By it I manage to create Conference room & add members to it.
But each member has to dial conference Number.
In my scenario Only first person dial second person's number.
Example:-
If Person1 has 6001, Person2 6002, person3 has 6003 & so on,
Then In confbridge as per given link example Person1 dial 1030, then
person2 dial 1030, then person3 dial 1030 & so on for conference call.
But In my scenario Person1 dial 6002, then make it hold, then dial 6003 &
then merge call.
Is it depend on softphone functionality or we need to write something in
some conf file?
Can we do it some how?
I tried it on mobile & I can make conference with 6 friends means total 7
people talk to each other without dial any conference number.

Thanks.

Mandar P. Khire
+919769419340
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Re: [asterisk-users] Conference Call like conference done by mobile!

[Andrew Ruthven]

On Wed, 2016-10-05 at 17:34 +0530, Mandar Khire wrote:
> hi,
> I trying to solve one scenario:-
> As I can make call from mobile phone to my friend1. As he accept it,
> I put him on hold, & dial friend2.
> As he also accept it, I put him on hold & follow same procedure till
> friend6.
> The I click on 'Merge call' & I can talk to all 6 friends at a time &
> they can talk each other.
> Can I write This scene by dialplan?How?
> I used Confbridge but its different type of conference.
> Need help.
> Thanks.

Hi Mandar,

Check out the "addcaller" stuff here:

https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration

Essentially you'd have a dialplan where you can call another number
which is then added to the confbridge.

Cheers,
Andrew

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Re: [asterisk-users] Conference Call like conference done by mobile!

[Carlos Chavez]

On 10/5/16 7:04 AM, Mandar Khire wrote:


hi,
I trying to solve one scenario:-
As I can make call from mobile phone to my friend1. As he accept it, I 
put him on hold, & dial friend2.
As he also accept it, I put him on hold & follow same procedure till 
friend6.
The I click on 'Merge call' & I can talk to all 6 friends at a time & 
they can talk each other.


Can I write This scene by dialplan?How?

I used Confbridge but its different type of conference.

Need help.
Thanks.

What you are mentioning is a function of the phone an not of 
Asterisk.  The phone has to support all those channels and mix them 
locally.  Most phones only do three way calling but some can do more.  
What is the problem with dumping everyone into a Confbridge conference 
room?  Same result at the end.



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[asterisk-users] Conference Call like conference done by mobile!

[Mandar Khire]

hi,
I trying to solve one scenario:-
As I can make call from mobile phone to my friend1. As he accept it, I put
him on hold, & dial friend2.
As he also accept it, I put him on hold & follow same procedure till
friend6.
The I click on 'Merge call' & I can talk to all 6 friends at a time & they
can talk each other.

Can I write This scene by dialplan?How?

I used Confbridge but its different type of conference.

Need help.
Thanks.


Mandar P. Khire
+919769419340
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Re: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

[Kevin Long]

Thanks John,


For anyone reading this using FreePBX - simply switching the default conference 
app from MeetMe to ConfBridge seems to be a drastic improvement, have not 
stress tested but running a conf now with no stutter on Confbrdige app.

Cheers,

Kevin Long



> On Mar 9, 2016, at 12:17 PM, Tech Support <aster...@voipbusiness.us> wrote:
> 
> One of the things you can do is google "app_konference". It doesn't require
> a clock source and is a very good application. I've successfully been using
> it for years and have had no problem with 100+ users in a single conference.
> Regards;
> John V.  
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Long
> Sent: Wednesday, March 09, 2016 2:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] conference call stuttering / clocking issue (?) -
> ESXi virtual environment
> 
> 
> 
> Title says it all - for the time being I am stuck deploying Asterisk in ESXi
> . We are also looking at Proxmox for our next round of servers.. 
> 
> Everything works fine except conference calls - very stuttery , have tried a
> few different codecs.  I assume this is a granular clocking issue , and
> wondering if anyone has anything I could try to fix or mitigate the problem
> in ESXi environment .
> 
> We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8
> pjsip .
> 
> Thank you again,
> 
> 
> Kevin Long
> 
> 
> 
> 
> 
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Re: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

[Tech Support]

One of the things you can do is google "app_konference". It doesn't require
a clock source and is a very good application. I've successfully been using
it for years and have had no problem with 100+ users in a single conference.
Regards;
John V.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Long
Sent: Wednesday, March 09, 2016 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] conference call stuttering / clocking issue (?) -
ESXi virtual environment



Title says it all - for the time being I am stuck deploying Asterisk in ESXi
. We are also looking at Proxmox for our next round of servers.. 

Everything works fine except conference calls - very stuttery , have tried a
few different codecs.  I assume this is a granular clocking issue , and
wondering if anyone has anything I could try to fix or mitigate the problem
in ESXi environment .

We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8
pjsip .

Thank you again,


Kevin Long





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[asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

[Kevin Long]

Title says it all - for the time being I am stuck deploying Asterisk in ESXi . 
We are also looking at Proxmox for our next round of servers.. 

Everything works fine except conference calls - very stuttery , have tried a 
few different codecs.  I assume this is a granular clocking issue , and 
wondering if anyone has anything I could try to fix or mitigate the problem in 
ESXi environment .

We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8 
pjsip .

Thank you again,


Kevin Long





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[asterisk-users] Conference calls wont traverse my trunk

[DadoMaker]

My conference call wont go thru my SIP trunk.  I may be missing a dialplan
configuration setting as my PCM phone to phone calls go over the (GSM) tunk.


The server with the conference:
exten = 5777,1,GoTo(conf-confDemo,join,1)
[conf-confDemo]
exten = join,1,ConfBridge(confDemo/S/1)

The server from which some users dial in from:
exten = 5777,1,Dial(SIP/$EXTEN}@200_PBX)

Any insight appreciated.

Thanks,

Dado
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Re: [asterisk-users] Conference calls wont traverse my trunk

[Steve Totaro]

On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote:

 My conference call wont go thru my SIP trunk.  I may be missing a dialplan
 configuration setting as my PCM phone to phone calls go over the (GSM) tunk.


 The server with the conference:
 exten = 5777,1,GoTo(conf-confDemo,join,1)
 [conf-confDemo]
 exten = join,1,ConfBridge(confDemo/S/1)

 The server from which some users dial in from:
 exten = 5777,1,Dial(SIP/$EXTEN}@200_PBX)

 Any insight appreciated.

 Thanks,

 Dado


Dado, subject sounds like a personal problem.  Sorry couldn't resist.

How about some CLI debug info while trying a call?

Thanks,
Steve T
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Re: [asterisk-users] Conference calls wont traverse my trunk

[DadoMaker]

The cogerence works but doesnt go over my trunk. Its bypassing and the
codec is PCM of phone.  But in phone to phone call, the rtp traverses the
trunk and I capture gsm packets to verify.

The sip debug for conf call setup and leave:
*CLI   == Using SIP RTP CoS mark 5
-- Executing [5777@public:1] Goto(SIP/127.0.0.1-0012,
conf-confDemo,join,1) in new stack
-- Goto (conf-confDemo,join,1)
-- Executing [join@conf-confDemo:1]
ConfBridge(SIP/127.0.0.1-0012, 1) in new stack
0x7f006c015150 -- Probation passed - setting RTP source address to
192.168.100.100:4002
-- SIP/127.0.0.1-0012 Playing 'conf-onlyperson.ulaw' (language
'en')
-- SIP/127.0.0.1-0012 Playing 'confbridge-join.ulaw' (language
'en')
-- Bridge/0x7f0058001af8-input Playing 'confbridge-join.slin'
(language 'en')
  == Using SIP RTP CoS mark 5
-- Executing [5777@default:1] Goto(SIP/5700-0013,
conf-confDemo,join,1) in new stack
-- Goto (conf-confDemo,join,1)
-- Executing [join@conf-confDemo:1] ConfBridge(SIP/5700-0013,
1) in new stack
0x7f006c031d90 -- Probation passed - setting RTP source address to
127.0.0.1:4004
-- SIP/5700-0013 Playing 'confbridge-join.ulaw' (language 'en')
0x7f006c031d90 -- Switching RTP source address to 192.168.1.10:4004
-- Bridge/0x7f0058001af8-input Playing 'confbridge-join.slin'
(language 'en')
-- Bridge/0x7f0058001af8-input Playing 'confbridge-leave.slin'
(language 'en')
-- Bridge/0x7f0058001af8-input Playing 'confbridge-leave.slin'
(language 'en')

Thanks,
Dado


On Thu, Jun 27, 2013 at 10:36 AM, Steve Totaro 
stot...@totarotechnologies.com wrote:




 On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote:

 My conference call wont go thru my SIP trunk.  I may be missing a
 dialplan configuration setting as my PCM phone to phone calls go over the
 (GSM) tunk.


 The server with the conference:
 exten = 5777,1,GoTo(conf-confDemo,join,1)
 [conf-confDemo]
 exten = join,1,ConfBridge(confDemo/S/1)

 The server from which some users dial in from:
 exten = 5777,1,Dial(SIP/$EXTEN}@200_PBX)

 Any insight appreciated.

 Thanks,

 Dado


 Dado, subject sounds like a personal problem.  Sorry couldn't resist.

 How about some CLI debug info while trying a call?

 Thanks,
 Steve T

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Re: [asterisk-users] Conference calls wont traverse my trunk

[DadoMaker]

Found a syntax err in my dialplan on the far side Asterisk config.
Thanks,
Dado


On Thu, Jun 27, 2013 at 10:41 AM, DadoMaker dadoma...@gmail.com wrote:

 The cogerence works but doesnt go over my trunk. Its bypassing and the
 codec is PCM of phone.  But in phone to phone call, the rtp traverses the
 trunk and I capture gsm packets to verify.

 The sip debug for conf call setup and leave:
 *CLI   == Using SIP RTP CoS mark 5
 -- Executing [5777@public:1] Goto(SIP/127.0.0.1-0012,
 conf-confDemo,join,1) in new stack
 -- Goto (conf-confDemo,join,1)
 -- Executing [join@conf-confDemo:1]
 ConfBridge(SIP/127.0.0.1-0012, 1) in new stack
 0x7f006c015150 -- Probation passed - setting RTP source address
 to 192.168.100.100:4002
 -- SIP/127.0.0.1-0012 Playing 'conf-onlyperson.ulaw' (language
 'en')
 -- SIP/127.0.0.1-0012 Playing 'confbridge-join.ulaw' (language
 'en')
 -- Bridge/0x7f0058001af8-input Playing 'confbridge-join.slin'
 (language 'en')
   == Using SIP RTP CoS mark 5
 -- Executing [5777@default:1] Goto(SIP/5700-0013,
 conf-confDemo,join,1) in new stack
 -- Goto (conf-confDemo,join,1)
 -- Executing [join@conf-confDemo:1] ConfBridge(SIP/5700-0013,
 1) in new stack
 0x7f006c031d90 -- Probation passed - setting RTP source address
 to 127.0.0.1:4004
 -- SIP/5700-0013 Playing 'confbridge-join.ulaw' (language 'en')
 0x7f006c031d90 -- Switching RTP source address to
 192.168.1.10:4004
 -- Bridge/0x7f0058001af8-input Playing 'confbridge-join.slin'
 (language 'en')
 -- Bridge/0x7f0058001af8-input Playing 'confbridge-leave.slin'
 (language 'en')
 -- Bridge/0x7f0058001af8-input Playing 'confbridge-leave.slin'
 (language 'en')

 Thanks,
 Dado


 On Thu, Jun 27, 2013 at 10:36 AM, Steve Totaro 
 stot...@totarotechnologies.com wrote:




 On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote:

 My conference call wont go thru my SIP trunk.  I may be missing a
 dialplan configuration setting as my PCM phone to phone calls go over the
 (GSM) tunk.


 The server with the conference:
 exten = 5777,1,GoTo(conf-confDemo,join,1)
 [conf-confDemo]
 exten = join,1,ConfBridge(confDemo/S/1)

 The server from which some users dial in from:
 exten = 5777,1,Dial(SIP/$EXTEN}@200_PBX)

 Any insight appreciated.

 Thanks,

 Dado


 Dado, subject sounds like a personal problem.  Sorry couldn't resist.

 How about some CLI debug info while trying a call?

 Thanks,
 Steve T

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Re: [asterisk-users] Conference solution to handle 10, 000 participants - possible at all?

[Sammy Govind]

Hi,

I'd been thinking about such a huge conferencing system for about last few
months. Like Steve suggested, my concept is almost similar but instead of
making a central hub conference junction between multiple Conferences I was
thinking of making a peer2peer runtime connection between conferences hosted
on multiple servers.

All the asterisks are load balanced by a super node which will be
OpenSIPS/Sip proxy.

Any conference participant call will first land on SIP proxy where Prosy
will do some required resgiteration of the participant, decide if the
required conference server is full or not- If not route the call to
previously used server else route the call to newer server and send a
trigger to new asterisk server to bridge with the older server's conference.
--
Regards,
Sammy

On Tue, Oct 18, 2011 at 6:08 AM, Steve Edwards asterisk@sedwards.comwrote:

 On Mon, 17 Oct 2011, VisionVoIP wrote:

  A client is asking to setup an asterisk based conferencing solution which
 could handle 10,000 participants (in one single conference or combined in
 multiple conferences), and later could be scaled to handle up to 50,000
 participants. All callers will be over SIP, using g711.


 If you scour the archives, you'll find discussion about this kind of thing
 several years ago, and then again sometime in the last 6 months. Googling
 about a bit should also yield relevant references.

 The OP built a system where NASCAR fans could call into conferences and
 listen to the cockpit chatter of the car of their choice.

 His system handled around 6,000 callers, but could be scaled higher.

 Think of a tree where 1 system hosts the conference. All 'callers' to this
 host are the next level of Asterisk systems. Add additional layers to build
 out to the number of real callers you want on an individual server.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000


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[asterisk-users] Conference solution to handle 10, 000 participants - possible at all?

[VisionVoIP]

Hello list,

A client is asking to setup an asterisk based conferencing solution 
which could handle 10,000 participants (in one single conference or 
combined in multiple conferences), and later could be scaled to handle 
up to 50,000 participants. All callers will be over SIP, using g711.


I have designed and implemented MeetMe based conferencing solutions 
before but nowhere near to this scale. Is it possible to do this at all 
using Asterisk? If yes, wow, but how?


Even a powerful server will only handle up to a 1000 concurrent g711 
calls, maybe a few hundred more. Going beyond that will require scaling 
to another server. But since MeetMe will be running on the first server, 
how will users from the second server join it?


Any ideas, suggestions and experience to share?

--

Zeeshan A Zakaria

PBX - visionvoip.com
Blog - ilovetovoip.com


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Re: [asterisk-users] Conference solution to handle 10, 000 participants - possible at all?

[Steve Edwards]

On Mon, 17 Oct 2011, VisionVoIP wrote:

A client is asking to setup an asterisk based conferencing solution 
which could handle 10,000 participants (in one single conference or 
combined in multiple conferences), and later could be scaled to handle 
up to 50,000 participants. All callers will be over SIP, using g711.


If you scour the archives, you'll find discussion about this kind of thing 
several years ago, and then again sometime in the last 6 months. Googling 
about a bit should also yield relevant references.


The OP built a system where NASCAR fans could call into conferences and 
listen to the cockpit chatter of the car of their choice.


His system handled around 6,000 callers, but could be scaled higher.

Think of a tree where 1 system hosts the conference. All 'callers' to this 
host are the next level of Asterisk systems. Add additional layers to 
build out to the number of real callers you want on an individual server.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Conference calls through web-interface with moderation using Asterisk?

[Alec Taylor]

Good Morning,

I have been researching this for a while, basically I'd like to have a
website with the following functionality:
• One-click call-in to show (after setting username, best-case
scenario: sign-in through Drupal, use that name for conference-call)
• Web-interface only (Flash/Flex, Javascript/JQuery or Java), without
any additional software/addons/plugins to install
• Moderation: host of conference call can quieten/mute or even kick
people from the conference call if they're being rowdy

So far I have setup an IceCAST server, broadcasting through edcast in
an mp3 stream. Viewers of my website can now listen-in on the /radio/
sub-page.

How do I setup the aforementioned [3] features using Asterisk? — Do I
need [Free, Open-Source] products other than Asterisk to get this
done, if so, which?

Thanks for all suggestions,

Alec Taylor

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Re: [asterisk-users] Conference feature

[Satish Barot]

Would this be of any help to you?
http://lists.digium.com/pipermail/asterisk-users/2011-June/263339.html


[SATISH]
Mumbai, India.

On Mon, Jun 27, 2011 at 7:14 AM, Rafael dos Santos Saraiva 
rafaels...@gmail.com wrote:

 I am referring to 3-way conference

 Att,
 Rafael Saraiva



 2011/6/26 Flavio Miranda flaviormira...@hotmail.com


 Very simple..

 Just edit the meetme.conf in /etc/asterisk like this :
 [rooms]

 conf = 888

 And then, in /etc/asterisk/ extensions.conf , put something like that:

 [conference]

 exten = 888,1,Set(CHANNEL(language)=pt_BR)if you have pt_BR audio
 exten = 888,n,MeetMe(888,pdM)
 exten = 888,n,Playback(vm-goodbye)
 exten = 888,n,Hangup

 When an user call 888 he will be in a conference  room.

 I hope it  help!


 Att,

 Flavio Roberto Miranda
 MSN:flaviormira...@hotmail.com
 Skype: flaviormiranda

 --
 Date: Sun, 26 Jun 2011 22:25:00 -0300
 From: rafaels...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Conference feature

 Hi

 How to create the conference feature in Asterisk?

 Thank's

 Att,
 Rafael Saraiva


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[asterisk-users] Conference feature

[Rafael dos Santos Saraiva]

Hi

How to create the conference feature in Asterisk?

Thank's

Att,
Rafael Saraiva
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Re: [asterisk-users] Conference feature

[Steve Edwards]

On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote:


How to create the conference feature in Asterisk?


RTM, keeping your eyes open for references to 'meetme.'

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Conference feature

[Flavio Miranda]

Very simple..
Just edit the meetme.conf in /etc/asterisk like this :[rooms]
conf = 888
And then, in /etc/asterisk/ extensions.conf , put something like that:
[conference]
exten = 888,1,Set(CHANNEL(language)=pt_BR)if you have pt_BR audioexten = 
888,n,MeetMe(888,pdM)exten = 888,n,Playback(vm-goodbye)exten = 888,n,Hangup
When an user call 888 he will be in a conference  room.
I hope it  help!
 Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

Date: Sun, 26 Jun 2011 22:25:00 -0300
From: rafaels...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Conference feature

Hi
How to create the conference feature in Asterisk?
Thank'sAtt,Rafael Saraiva


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Re: [asterisk-users] Conference feature

[Rafael dos Santos Saraiva]

I am referring to 3-way conference

Att,
Rafael Saraiva



2011/6/26 Flavio Miranda flaviormira...@hotmail.com


 Very simple..

 Just edit the meetme.conf in /etc/asterisk like this :
 [rooms]

 conf = 888

 And then, in /etc/asterisk/ extensions.conf , put something like that:

 [conference]

 exten = 888,1,Set(CHANNEL(language)=pt_BR)if you have pt_BR audio
 exten = 888,n,MeetMe(888,pdM)
 exten = 888,n,Playback(vm-goodbye)
 exten = 888,n,Hangup

 When an user call 888 he will be in a conference  room.

 I hope it  help!


 Att,

 Flavio Roberto Miranda
 MSN:flaviormira...@hotmail.com
 Skype: flaviormiranda

 --
 Date: Sun, 26 Jun 2011 22:25:00 -0300
 From: rafaels...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Conference feature

 Hi

 How to create the conference feature in Asterisk?

 Thank's

 Att,
 Rafael Saraiva


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Re: [asterisk-users] Conference feature

[Steve Edwards]

On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote:


I am referring to 3-way conference


With a little reading, you would discover that meetme can handle lots of 
participants.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] Conference feature

[John Novack]

Steve Edwards wrote:

On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote:


I am referring to 3-way conference


With a little reading, you would discover that meetme can handle lots 
of participants.


For those who know Telephony, 3 way conference and meet me conference 
are NOT the same.


Someone needs to RTM on telephony!

John Novack

--

Dog is my Co-pilot


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Re: [asterisk-users] Conference feature

[C F]

Does asterisk support it?

On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva
rafaels...@gmail.com wrote:
 Hi
 How to create the conference feature in Asterisk?
 Thank's
 Att,
 Rafael Saraiva

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Re: [asterisk-users] Conference feature

[Alex Balashov]

I am given to understand that it does not.

On 06/27/2011 12:13 AM, C F wrote:


Does asterisk support it?

On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva
rafaels...@gmail.com  wrote:

Hi
How to create the conference feature in Asterisk?
Thank's
Att,
Rafael Saraiva

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260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Conference feature

[Faisal Hanif]

If you can explain a bit more what exactly you need?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, June 27, 2011 9:16 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Conference feature

I am given to understand that it does not.

On 06/27/2011 12:13 AM, C F wrote:

 Does asterisk support it?

 On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva 
 rafaels...@gmail.com  wrote:
 Hi
 How to create the conference feature in Asterisk?
 Thank's
 Att,
 Rafael Saraiva

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260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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[asterisk-users] CONFERENCE CONFIGURATION REQUIRE

[mahesh katta]

Hi all,

I am using asterisk1.2(vicidial). I am using like pbx . In this how can I
confugure the internal conference calls. suppose I have A,B,C,D,E users
these all peoples should be internal conferece . for them i was give
101,102,103,104,105 extensions. For this scenario what can I do exact
configuration in dialplan and any to edit confugration files please help me
.
and how can they cut the conference of after concall.

Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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[asterisk-users] conference room ideas

[Dean Collins]

Some neat conference room ideas that would be great to see incorporated
into asterisk conference.

 

https://imeet.com/support 

 

 

Cheers,

Dean

 

 

 

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[asterisk-users] Conference Meetme

[torintino1]

How many simultaneous conference meetme setups can be supported in the same 
time on Asterisk, and what are the corresponding server's specs for this.

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Re: [asterisk-users] Conference Meetme

[Steve Edwards]

On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:

 How many simultaneous conference meetme setups can be supported in the 
 same time on Asterisk, and what are the corresponding server's specs for 
 this.

How long is a piece of string?

0) A better subject yields better answers

1) A more detailed question yields a more detailed answer.

A reasonably configured Asterisk server can handle XXX callers with Y 
callers in each of Z conferences.

-- 
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Conference Meetme

[Zeeshan Zakaria]

Last year I did a lab test for a customer who wanted conferencing solution
for his organization, on a 2 x dual core xeon with 4GB type server, which
had 120 zap channels and I put all the channels in mutiple conferences, from
4 to 20 users per conference and let it running for two weeks. Munin graph
showed that CPU load was only 6 to 7 percent during this period, no
conference dropped and asterisk didn't crash, and I occasionally used it to
make calls and run other processes, and no call quality issues. Now that
server is in production and customer is happy with it. I don't know about
SIP which will use more processing, but I am sure a decent server of today
can handle a good number of conferences with a good number of users each.
What numbers you are looking for?

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-04-14 12:56 PM, Steve Edwards asterisk@sedwards.com wrote:

On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:

 How many simultaneous conference meetme setups...
How long is a piece of string?

0) A better subject yields better answers

1) A more detailed question yields a more detailed answer.

A reasonably configured Asterisk server can handle XXX callers with Y
callers in each of Z conferences.

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Conference Meetme

[torintino1]

I need the server to handle about 300 - 400 simultaneous meetme conferences, 
5-10 participants in each,

Actually I need to know, if I will get an IBM X3650 M2, QuadCore, 4-6 GB RAM, 
8MB cache,
how many simultaneous meetme conferences I can operate on a this server.

Thanks



From: Zeeshan Zakaria 
Sent: Wednesday, April 14, 2010 8:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: Re: [asterisk-users] Conference Meetme


Last year I did a lab test for a customer who wanted conferencing solution for 
his organization, on a 2 x dual core xeon with 4GB type server, which had 120 
zap channels and I put all the channels in mutiple conferences, from 4 to 20 
users per conference and let it running for two weeks. Munin graph showed that 
CPU load was only 6 to 7 percent during this period, no conference dropped and 
asterisk didn't crash, and I occasionally used it to make calls and run other 
processes, and no call quality issues. Now that server is in production and 
customer is happy with it. I don't know about SIP which will use more 
processing, but I am sure a decent server of today can handle a good number of 
conferences with a good number of users each. What numbers you are looking for?

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.


  On 2010-04-14 12:56 PM, Steve Edwards asterisk@sedwards.com wrote:


  On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:

   How many simultaneous conference meetme setups...

  How long is a piece of string?

  0) A better subject yields better answers

  1) A more detailed question yields a more detailed answer.

  A reasonably configured Asterisk server can handle XXX callers with Y
  callers in each of Z conferences.

  --
  Thanks in advance,
  -
  Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
  Newline  Fax: +1-760-731-3000

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[asterisk-users] [Conference] Audio/Video

[Stéphane Bauland]

Hi guys,

I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.

So each ppl will have a audio and video stream.

I'm wondering if you know a way to do this with asterisk or if it's
supported ?

If it is, i'm asking you about some documentation or related article (if
you know ones) where i could find more informations.

Else, do you know any other way to do this ?

Best regards,

-- 
Stéphane Bauland

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Re: [asterisk-users] Conference Meetme

[Philipp von Klitzing]

Hi!

 I need the server to handle about 300 - 400 simultaneous meetme 
 conferences, 5-10 participants in each, 
 
 Actually I need to know, if I will get an IBM X3650 M2,QuadCore, 4-6
 GB RAM, 8MB cache, how many simultaneous meetme conferences I can
 operate on a this server. 

There is no simple answer for you - look here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+dimensioning

Philipp


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Re: [asterisk-users] [Conference] Audio/Video

[Stéphane Bauland]

Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
 On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
 Hi guys,

 I'm planning of creating a speech/video conference application. This
 application will provide a system to see/listen to each personn present
 in the conference.

 So each ppl will have a audio and video stream.

 I'm wondering if you know a way to do this with asterisk or if it's
 supported ?

 If it is, i'm asking you about some documentation or related article (if
 you know ones) where i could find more informations.

 Else, do you know any other way to do this ?

 Best regards,


 Would love to see a _working_ video conf.
 afaicr it's currently vapor-ware
 Are you thinking of letting asterisk doing video multiplexing?
 Or are you aiming just for a conference with a small number of
 participants?

 hw


We (cause we are a team) are planning of doing a multi user conference
software at a end school project.

The way we go is, we are looking throught jungle (xmpp ext for jabber)
to create conference between many people. We don't want to set a 
limitation about how many participant of a conference).

But right now, i'm discovering asterisk, and i need some informations
from people like you that know the soft and his capatibilities...

So i think yes, we want to do video multiplexing.

Do you think a software like that could use asterisk as a backend ?

And, do you know any other software that is doing the same thing using
asterisk ?

-- 
Stéphane Bauland

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Re: [asterisk-users] [Conference] Audio/Video

[David Backeberg]

On Wed, Apr 14, 2010 at 4:55 PM, Stéphane Bauland baula...@epitech.net wrote:
 I'm planning of creating a speech/video conference application. This
 application will provide a system to see/listen to each personn present
 in the conference.
 Else, do you know any other way to do this ?

http://en.wikipedia.org/wiki/CU-SeeMe
it was kindof a solved problem,
but that's not really around anymore.

these days, ichat and google chat and Ekiga do one-on-one chat well.

The problem is n-to-n chat.

Take a look at openmcu, and good luck.

Unfortunately, the products that work well AND are turnkey generally
require money, ranging from a little to literally millions for a
full-featured Cisco telepresence solution.

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Re: [asterisk-users] [Conference] Audio/Video

[Jamie A. Stapleton]

http://www.projectdiastar.org/ looks promising...

On Apr 14, 2010, at 7:04 PM, Stéphane Bauland wrote:

Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
Hi guys,

I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.

So each ppl will have a audio and video stream.

I'm wondering if you know a way to do this with asterisk or if it's
supported ?

If it is, i'm asking you about some documentation or related article (if
you know ones) where i could find more informations.

Else, do you know any other way to do this ?

Best regards,


Would love to see a _working_ video conf.
afaicr it's currently vapor-ware
Are you thinking of letting asterisk doing video multiplexing?
Or are you aiming just for a conference with a small number of
participants?

hw


We (cause we are a team) are planning of doing a multi user conference
software at a end school project.

The way we go is, we are looking throught jungle (xmpp ext for jabber)
to create conference between many people. We don't want to set a
limitation about how many participant of a conference).

But right now, i'm discovering asterisk, and i need some informations
from people like you that know the soft and his capatibilities...

So i think yes, we want to do video multiplexing.

Do you think a software like that could use asterisk as a backend ?

And, do you know any other software that is doing the same thing using
asterisk ?

--
Stéphane Bauland

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Jamie A. Stapleton
CBSi - Connecting your problems with solutions.
Telephone:  (804) 412-1601
Facsimile:  (804) 412-1611
VideoConf:  callto:jstapleton.computer-business.com

Meet me on LinkedInhttp://www.linkedin.com/in/jstapleton

Have I exceeded your expectations?  Please share your experience with our 
Founder, Fred W. Brumbaughmailto:fbrumba...@computer-business.com

LEGAL DISCLAIMER
The information transmitted is intended solely for the individual or entity to 
which it is addressed and may contain confidential and/or privileged material. 
Any review, retransmission, dissemination or other use of or taking action in 
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contact the sender and delete the material from any computer.


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Re: [asterisk-users] [Conference] Audio/Video

[Ioan Indreias]

We have used with success BBB (BigBlueButton - open source -
http://bigbluebutton.org) and I recommend to try their demo in order
to see if this solution gives all you need.

Voice conf is based on Asterisk.

HTH,
Ioan Indreias
www.modulo.ro

On Thu, Apr 15, 2010 at 2:04 AM, Stéphane Bauland baula...@epitech.net wrote:
 Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
 On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
 Hi guys,

 I'm planning of creating a speech/video conference application. This
 application will provide a system to see/listen to each personn present
 in the conference.

 So each ppl will have a audio and video stream.

 I'm wondering if you know a way to do this with asterisk or if it's
 supported ?

 If it is, i'm asking you about some documentation or related article (if
 you know ones) where i could find more informations.

 Else, do you know any other way to do this ?

 Best regards,


 Would love to see a _working_ video conf.
 afaicr it's currently vapor-ware
 Are you thinking of letting asterisk doing video multiplexing?
 Or are you aiming just for a conference with a small number of
 participants?

 hw


 We (cause we are a team) are planning of doing a multi user conference
 software at a end school project.

 The way we go is, we are looking throught jungle (xmpp ext for jabber)
 to create conference between many people. We don't want to set a
 limitation about how many participant of a conference).

 But right now, i'm discovering asterisk, and i need some informations
 from people like you that know the soft and his capatibilities...

 So i think yes, we want to do video multiplexing.

 Do you think a software like that could use asterisk as a backend ?

 And, do you know any other software that is doing the same thing using
 asterisk ?

 --
 Stéphane Bauland


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[asterisk-users] Conference Calling

[Faheem]

Hey All,
I want to implement a conference calling scenario.
Conference Call Procedure:User1 dial the User2. When call is connected put the 
current call on Hold and dial User3. When the call is connected between User1 
and User3 join the User2 in a conference room!How I can implement this 
scenario. What are generic steps to do so! 
Thanks=Muhammad Faheem  




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Re: [asterisk-users] Conference Calling

[Tri Tu]

Here is where to get you start with this.

http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO

-Tri





From: Faheem faheem_...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Sat, February 27, 2010 12:08:24 PM
Subject: [asterisk-users] Conference Calling




Hey All,

I want to implement a conference calling scenario.

Conference Call Procedure:
User1 dial the User2. When call is connected put the current call on Hold and 
dial User3. When the call is connected between User1 and User3 join the User2 
in a conference room!
How I can implement this scenario. What are generic steps to do so! Thanks
=
Muhammad Faheem 


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Re: [asterisk-users] Conference Calling

[meetmecall]

Muhammad

It is not really your scenario but the scenario to setup a conference  
call with three numbers could be to generate two call files that  
points to a local channel/a context/extension that route the leg into  
the conference room and have your own leg routed into the conference  
room after the input is done This not the solution but one of the many  
possible.


enter the numbers for setting up the conference call like  
number1*number2   (check Read() cmd for storing input into a  
variable)


split the input in seperated numbers See 
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables

generate the call files for setting up the connection. Point to a  
context, extension, priority to route the lef into a conference room.  
See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out


move the call files to /var/spool/asterisk/outgoing (check System()  
cmd )


have your own leg routed into the conference room  (check Goto() cmd )

Have a nice chat with the three of you ;-)

Erik



On 27 feb 2010, at 21:08, Faheem wrote:



Hey All,

I want to implement a conference calling scenario.

Conference Call Procedure:
User1 dial the User2. When call is connected put the current call on  
Hold and dial User3. When the call is connected between User1 and  
User3 join the User2 in a conference room!


How I can implement this scenario. What are generic steps to do so!  
Thanks


=

Muhammad Faheem




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Re: [asterisk-users] Conference problem

[Cristi Iconaru]

The CM is sending the BYE messages.
 
Any ideas?
 
Christian

--- On Wed, 4/22/09, Martin asteriskl...@callthem.info wrote:


From: Martin asteriskl...@callthem.info
Subject: Re: [asterisk-users] Conference problem
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Wednesday, April 22, 2009, 8:08 PM


run a sip debug and check whether it's asterisk disconnecting the
calls (usually a SIP BYE message)
or whether Asterisk is getting the disconnect from your Cisco GW

Martin

On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru
cristi_icon...@yahoo.com wrote:
 Hello all,

 I have some issues with the MeetMe application.

 The working topology is as follows. The Asterisk (1.4.22-rc5) is connected
 through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco
 Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are
 forwarded to Asterisk by the CM.

 The problem is that some users who are calling in from PSTN are getting
 disconnected from the conference room after a period of time. They can get
 in but after a while suddenly they are disconnected. The funny thing is that
 on the Asterisk CLI/logs no errors/retrans/etc. appeared.

 The Asterisk has no Zaptel hardware. All the necesary modules are installed.

 Thanks,
 Christian

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[asterisk-users] Conference problem

[Cristi Iconaru]

Hello all,
 
I have some issues with the MeetMe application.
 
The working topology is as follows. The Asterisk (1.4.22-rc5) is connected 
through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice 
Gateway. The Gateway is connected to PSTN through a PRI. The calls are 
forwarded to Asterisk by the CM.
 
The problem is that some users who are calling in from PSTN are getting 
disconnected from the conference room after a period of time. They can get in 
but after a while suddenly they are disconnected. The funny thing is that on 
the Asterisk CLI/logs no errors/retrans/etc. appeared.
 
The Asterisk has no Zaptel hardware. All the necesary modules are installed.
 
Thanks,
Christian


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Re: [asterisk-users] Conference problem

[Martin]

run a sip debug and check whether it's asterisk disconnecting the
calls (usually a SIP BYE message)
or whether Asterisk is getting the disconnect from your Cisco GW

Martin

On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru
cristi_icon...@yahoo.com wrote:
 Hello all,

 I have some issues with the MeetMe application.

 The working topology is as follows. The Asterisk (1.4.22-rc5) is connected
 through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco
 Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are
 forwarded to Asterisk by the CM.

 The problem is that some users who are calling in from PSTN are getting
 disconnected from the conference room after a period of time. They can get
 in but after a while suddenly they are disconnected. The funny thing is that
 on the Asterisk CLI/logs no errors/retrans/etc. appeared.

 The Asterisk has no Zaptel hardware. All the necesary modules are installed.

 Thanks,
 Christian

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[asterisk-users] conference calling

[Danny Nicholas]

Greetings listers.

 I'm running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones.  My outgoing connections are Zapata using a TDM401P.
For the most part I can make and receive calls fine except for these 3
issues:

1.   When I call an external conference, the call never bridges and
hangs up after 60-90 seconds.

2.   When I call another number there is a 2-4 second delay before the
callee can hear me.

3.   When I call an external conference and connect, the others cannot
hear me.

 

Zapata.conf

[trunkgroups]

 

[channels]

;context=from-zaptel

;context=line1

busydetect=yes

callprogress=yes

busycount=4

hanguponpolarityswitch=yes

answeronpolarityswitch=yes

usecallingpres=yes

priindication=outofband

pritimer=t305,5

signalling=fxs_ks

wink=50

useincomingcalleridonzaptransfer=yes

echocancel=yes

echocancelwhenbridged=yes

faxdetect=yes

rxgain=1.0

txgain=21.0

callgroup=1

group=1

usecallerid=yes

callerid=asreceived

cidstart=ring

hidecallerid=no

immediate=no

pickupgroup=1

;context=incoming

channel = 1-4

 

Sip.conf

[general]

srvlookup=yes ;allows DNS lookups of server names

naxexpirey=180

defaultexpirey=160

context=default ; Default context for incoming calls

allowoverlap=no ; Disable overlap dialing support. (Default is yes)

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)

tos_sip=cs3

tos_audio=ef

 

; bindport is the local UDP port that Asterisk will

; listen on

bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls

limitonpeers=yes

notifyringing=yes

rtupdate=yes[authentication]

 

[104]

type=peer

context=phones

host=dynamic

fromuser=104

secret=xx

canreinvite=update

directrtpsetup=no

call-limit=3

nat=yes

qualify=yes

register=no

session-timers=accept

session-expires=90

session-minse=120

session-refresher=uac

register = 104:xx...@xx.com/104

defaultip=192.168.xx.xxx

mailbox=104

disallow=all

allow=ulaw,alaw

artcachefriends=yes

notifyhold=yes

incominglimit=1

call-limit=3

 

Other information will be provided as asked for.  

 

Thanks in advance for any help you can provide.

 

Danny Nicholas

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Re: [asterisk-users] conference calling

[Martin]

Turn off callprogres=yes or have it configured properly.
It should fix your problem.

regards
Martin

On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas da...@debsinc.com wrote:
 Greetings listers.

  I’m running asterisk 1.4.21.2 on SUSE 11.0 using
 Polycom 501 phones.  My outgoing connections are Zapata using a TDM401P.
 For the most part I can make and receive calls fine except for these 3
 issues:

 1.   When I call an external conference, the call never bridges and
 hangs up after 60-90 seconds.

 2.   When I call another number there is a 2-4 second delay before the
 callee can hear me.

 3.   When I call an external conference and connect, the others cannot
 hear me.



 Zapata.conf

 [trunkgroups]



 [channels]

 ;context=from-zaptel

 ;context=line1

 busydetect=yes

 callprogress=yes

 busycount=4

 hanguponpolarityswitch=yes

 answeronpolarityswitch=yes

 usecallingpres=yes

 priindication=outofband

 pritimer=t305,5

 signalling=fxs_ks

 wink=50

 useincomingcalleridonzaptransfer=yes

 echocancel=yes

 echocancelwhenbridged=yes

 faxdetect=yes

 rxgain=1.0

 txgain=21.0

 callgroup=1

 group=1

 usecallerid=yes

 callerid=asreceived

 cidstart=ring

 hidecallerid=no

 immediate=no

 pickupgroup=1

 ;context=incoming

 channel = 1-4



 Sip.conf

 [general]

 srvlookup=yes ;allows DNS lookups of server names

 naxexpirey=180

 defaultexpirey=160

 context=default ; Default context for incoming calls

 allowoverlap=no ; Disable overlap dialing support. (Default is yes)

 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)

 tos_sip=cs3

 tos_audio=ef



 ; bindport is the local UDP port that Asterisk will

 ; listen on

 bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all)

 srvlookup=yes ; Enable DNS SRV lookups on outbound calls

 limitonpeers=yes

 notifyringing=yes

 rtupdate=yes[authentication]



 [104]

 type=peer

 context=phones

 host=dynamic

 fromuser=104

 secret=xx

 canreinvite=update

 directrtpsetup=no

 call-limit=3

 nat=yes

 qualify=yes

 register=no

 session-timers=accept

 session-expires=90

 session-minse=120

 session-refresher=uac

 register = 104:xx...@xx.com/104

 defaultip=192.168.xx.xxx

 mailbox=104

 disallow=all

 allow=ulaw,alaw

 artcachefriends=yes

 notifyhold=yes

 incominglimit=1

 call-limit=3



 Other information will be provided as asked for.



 Thanks in advance for any help you can provide.



 Danny Nicholas

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[asterisk-users] conference function problems

[Rilawich Ango]

The CLI shows zap is necessary for conference recording.  Can I enable
conference recording if using ztdummy or dahdi, how?  ango

-- Executing [...@owt_meetme:4] MeetMe(SIP/3601-c80b4520,
5599|rcixMP) in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '5599'
[Mar 31 17:57:39] WARNING[22242]: app_meetme.c:2375
find_conf_realtime: No Zap channel available for conference, user
introduction disabled
[Mar 31 17:57:39] WARNING[22242]: app_meetme.c:2381
find_conf_realtime: No Zap channel available for conference,
conference recording disabled
-- SIP/3601-c80b4520 Playing 'conf-getpin' (language 'en')

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Re: [asterisk-users] conference and wifi phones

[randulo]

On Tue, Mar 24, 2009 at 2:14 AM, Michael Graves mgra...@mstvp.com wrote:
 Amen to that! Unles you have some compelling reason for VoWifi it's not
 worthy of consideration. Especially for SOHO or small biz use. Too
 costly to do well.

I have never understood why anyone would use wifi just to get cordless
facility when DECT works so much better.

 DECT, in contrast, is solid and not overly expensive. SIP/DECT systems
 like the snom m3 and Siemens S675/S685 give you all the benefits of a
 SIP phone with the reliability and mobility of DECT. I much prefer this
 to an ATA and analog DECT phone.

I have been using a S675IP with two handsets in our office for over 6
months. Each handset has its own directory, can read a single RSS feed
(title only), has 6 SIP providers AND the base connects to PSTN and
has built-in voicemail. The phone is a geeks dream with all those
possibilities and the base handles two simultaneous VoIP calls.

The lack of a mute button has been ranted about often on the
conference, and a few things can get on your nerves, like filling out
a whole directory entry that is then erased if you backspace one too
many times (Doesn't anyone test the firmware before this kind of
device is sent out? This has to be easy to fox!)

I don't think these phones will outlast a Polycom but they have a
great feature set, rock solid DECT even on low power setting and very
good battery life.

/r

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Re: [asterisk-users] conference and wifi phones

[Gordon Henderson]

On Mon, 23 Mar 2009, Kelvin Chan wrote:


One of our local companies here in the UK are trialling a new conference
phone - the Konftel 300IP SIP however it's still as expensive as a
Polycom, but that might be the $/£ exchange - might be cheaper where you
are?


It seems like an interesting product. Compared to Polycom 7000, it's 
roughly $400(list price) cheaper. Just out of curiosity, the product is 
made in Sweden. It should be cheaper in UK. How much are they selling 
over there?


http://www.provu.co.uk/konftel_300IP.html

or £499.99

A Polycom IP4000 from another UK disty is £525.00, so not much in it.

http://www.voipon.co.uk/polycom-soundstation-ip4000-p-253.html

The 7000 is £683:

http://www.voipon.co.uk/polycom-soundstation-ip7000-p-921.html


For wifi phone, I tried Linksys iPhone. It works well but lacks a
cradle. My users often forget to charge it when they leave for the day
and come back to a dead wifi phone for the next morning.

Any good recommendations?


DECT not Wi-Fi?

Can you get the Siemens range over there?

DECT is not wifi. It is just another digital wireless protocol built for 
wireless phones.


I'm not sure what do you mean by Siemens range.


I was a bit too terse there... I mean Why not use DECT rather than 
Wi-Fi.


My Wi-Fi experience is only with 2 phones - The UT Starcom F1000G and my 
Nokia E90. The F1000G is rubbish. Actually, it was OK until I upgraded it, 
then it became rubbish. The real issue is that it doesn't have a built-in 
web browser, so to get it to latch-on to a public Wi-Fi access point is 
almost impossible as all the ones here (UK) needs some sort of 
registration system which needs a web browser to work.


The Nokia E90 is better in that respect, but at the end of the day, 
they're still Wi-Fi and Wi-Fi is basically a reasnable effort transport 
mechanism as far as I'm concerend.. It's trivially easy to swamp the air 
with uploads/downloads, etc. from other PCs, making VoIP over Wi-Fi 
problematic at best...


Siemens make a good range of VoIP ready DECT phone systems - their 
Gigaset range. Base stations have an Ethernet port and off you go. There 
are also repeaters to extend the range too - although I've only ever used 
a Snom repeater (with Siemens phones!). I do not recomend the snom M3 
range of DECT phones. All the ones I've installed have been returned...


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Re: [asterisk-users] conference and wifi phones

[Michael Graves]

On Tue, 24 Mar 2009 01:51:36 + (UTC), Jeff LaCoursiere wrote:


On Mon, 23 Mar 2009, Michael Graves wrote:

 On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:

 Siemens make a range of DECT handsets under the Gigaset model range.

 Yes they shit all over every wifi handset I have ever used.

 Dect is way better.


 Amen to that! Unles you have some compelling reason for VoWifi it's not
 worthy of consideration. Especially for SOHO or small biz use. Too
 costly to do well.

Roaming through hotspots is a point taken - you cannot exactly take your 
base to Starbucks.  But I have yet to run across a phone that does this 
well.


 DECT, in contrast, is solid and not overly expensive. SIP/DECT systems
 like the snom m3 and Siemens S675/S685 give you all the benefits of a
 SIP phone with the reliability and mobility of DECT. I much prefer this
 to an ATA and analog DECT phone.


Not that I am arguing - simply curious - what do you see as the advantages 
of a SIP phone over ATA/analog DECT?

I described this elsewhere previously. Check this out:

http://www.smallnetbuilder.com/content/view/30614/84/

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
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Re: [asterisk-users] conference and wifi phones

[Steve Gladden]

I REALLY like the Snom M3 DECT SIP base.
You can have up to 3 simultaneous calls through the base
and you can have up to 8 phones registered with it.
It's all web managed as well as http/s provisionable and has
this nice phone to line matrix where you can set which phones
ring on inbound calls and what outgoing 'account' a handset will use.
So per handset you can pick the outbound identity/account and you
can also take inbound calls from 1-8 SIP accounts!
And you can call (intercom) phone to phone,
and transfer etc.

And it works with just about ANY DECT(GAP) phone.
We use some Phillips handsets with it that cost $20 each and have
excellent talk time and very long range albeit a bit cheap feeling
and sounding as compared to a $400 Polycom KIRK handset
Which we are also using with this SNOM DECT gateway :-)

Snom's M3 DECT handset is very light and inexpensive feeling like a home
grade WiFi Phone.. which I DONT like but it performs very well for the
price has a color (passive matrix) screen and is rather full featured.

I seriously don't get why they won't build a wifi/dect phone of the same
rugged construction quality as even the most basic of cell phones.
Hopefully that will change some day :-)
Even very inexpensive cell phones are of MUCH more rugged design.

I just don't like these cheap feeling light soft plastic phones that I
sometimes improperly refer to as 'cheap skype toys'.
I was a PAP2 + cordless phone user until I got that SNOM M3 system.

I also use a snom DECT repeater outdoors on an outdoor antenna so I get
about 1/2 mile range on the DECT Phones and it works inside most of the
neighbor's houses up to about 1/4 mile away from home.

I still use a PAP2 to run a (FREE) payphone that sits on the back patio
for people to play with when they come over for parties.
And I use them anywher a POTS style interface is desired.
:-)








 On Mon, 23 Mar 2009, Michael Graves wrote:

 On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:

 Siemens make a range of DECT handsets under the Gigaset model range.

 Yes they shit all over every wifi handset I have ever used.

 Dect is way better.


 Amen to that! Unles you have some compelling reason for VoWifi it's not
 worthy of consideration. Especially for SOHO or small biz use. Too
 costly to do well.

 Roaming through hotspots is a point taken - you cannot exactly take your
 base to Starbucks.  But I have yet to run across a phone that does this
 well.


 DECT, in contrast, is solid and not overly expensive. SIP/DECT systems
 like the snom m3 and Siemens S675/S685 give you all the benefits of a
 SIP phone with the reliability and mobility of DECT. I much prefer this
 to an ATA and analog DECT phone.


 Not that I am arguing - simply curious - what do you see as the advantages
 of a SIP phone over ATA/analog DECT?

 Cheers,

 j


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Re: [asterisk-users] conference and wifi phones

[Christian Victor]

2009/3/24 Steve Gladden aster...@michiganbroadband.com

 I REALLY like the Snom M3 DECT SIP base.


Yeah - it's such a pitty that you always have to buy it bundled with one of
these crappy handsets. Or is there a way to get only the base that I don't
know?

Chris
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Re: [asterisk-users] conference and wifi phones

[Frank Bulk]

In a SOHO environment I would agree with you, but not if your coverage area
needs to be tens of thousands of square feet.  Deploying a complete overlay
wireless infrastructure doesn't make sense and is another infrastructure to
manage and maintain.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of randulo
Sent: Tuesday, March 24, 2009 1:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] conference and wifi phones

snip

I have never understood why anyone would use wifi just to get cordless
facility when DECT works so much better.

snip

/r

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Re: [asterisk-users] conference and wifi phones

[Jon Pounder]

Frank Bulk wrote:
 In a SOHO environment I would agree with you, but not if your coverage area
 needs to be tens of thousands of square feet.  Deploying a complete overlay
 wireless infrastructure doesn't make sense and is another infrastructure to
 manage and maintain.
   

did you think about your numbers before posting this ? what access point 
does not have a 50ft radius ?
100x100 ft is 1square feet = a single AP for 10's of 1000's of 
square feet, hardly a huge undertaking.

Dect is not going to let you roam into another network or hotspot and 
still work, nor is it going to support the myriad of other devices that 
work on wifi networks.
 Frank

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of randulo
 Sent: Tuesday, March 24, 2009 1:53 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] conference and wifi phones

 snip

 I have never understood why anyone would use wifi just to get cordless
 facility when DECT works so much better.

 snip

 /r

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Re: [asterisk-users] conference and wifi phones

[Darwin O. Solano]

ok


-Original Message-
From: Frank Bulk frnk...@iname.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] conference and wifi phones
Date: Tue, 24 Mar 2009 21:38:34 -0500


In a SOHO environment I would agree with you, but not if your coverage area
needs to be tens of thousands of square feet.  Deploying a complete overlay
wireless infrastructure doesn't make sense and is another infrastructure to
manage and maintain.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of randulo
Sent: Tuesday, March 24, 2009 1:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] conference and wifi phones

snip

I have never understood why anyone would use wifi just to get cordless
facility when DECT works so much better.

snip

/r

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[asterisk-users] conference and wifi phones

[Kelvin Chan]

Hi guys,

I'm looking for a affordable conference phone and a wifi phone that has a 
cradle. 

Polycom seems to make pretty nice conf phones but the price is a bit crazy for 
us. I saw the recommendation with ATA plus an analog Polycom phone but I do 
prefer a SIP phone. All because it's just too difficult to pull a phone cable 
into the current conference room. Is there any cheaper SIP solutions out there?

For wifi phone, I tried Linksys iPhone. It works well but lacks a cradle. My 
users often forget to charge it when they leave for the day and come back to a 
dead wifi phone for the next morning.

Any good recommendations?

Cheers,

Kelvin Chan   | Positronics Ent.
Product Development   |
  | unit 272
604-628-9330 (direct) | 8128 128th St.
604-585-2...@104 (main)   | Surrey, BC
604-585-3056 (fax)| Canada, V3W 1R1



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Re: [asterisk-users] conference and wifi phones

[Jeff LaCoursiere]

On Mon, 23 Mar 2009, Kelvin Chan wrote:

 Hi guys,

 I'm looking for a affordable conference phone and a wifi phone that has a 
 cradle.

 Polycom seems to make pretty nice conf phones but the price is a bit 
 crazy for us. I saw the recommendation with ATA plus an analog Polycom 
 phone but I do prefer a SIP phone. All because it's just too difficult 
 to pull a phone cable into the current conference room. Is there any 
 cheaper SIP solutions out there?

Why not have the ATA in the same room as the conference phone?  I did this 
for a client.  The ATA is bolted to the underside of the conference table 
and plugs into the ethernet switch (also bolted to the underside of the 
conference table), and a simple RJ11 patch cable runs up to the legacy 
Polycom conference phone.


 For wifi phone, I tried Linksys iPhone. It works well but lacks a 
 cradle. My users often forget to charge it when they leave for the day 
 and come back to a dead wifi phone for the next morning.

I still don't get the market for this kind of phone.  DECT cordless phones 
can be had for $20 with very long battery life and range, and if the base 
is plugged into an ATA you have your wireless SIP phone.

Long live the PAP2T...

j


 Any good recommendations?

 Cheers,

 Kelvin Chan   | Positronics Ent.
 Product Development   |
  | unit 272
 604-628-9330 (direct) | 8128 128th St.
 604-585-2...@104 (main)   | Surrey, BC
 604-585-3056 (fax)| Canada, V3W 1R1



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Re: [asterisk-users] conference and wifi phones

[Gordon Henderson]

On Mon, 23 Mar 2009, Kelvin Chan wrote:


Hi guys,

I'm looking for a affordable conference phone and a wifi phone that has 
a cradle.


Polycom seems to make pretty nice conf phones but the price is a bit 
crazy for us. I saw the recommendation with ATA plus an analog Polycom 
phone but I do prefer a SIP phone. All because it's just too difficult 
to pull a phone cable into the current conference room. Is there any 
cheaper SIP solutions out there?


One of our local companies here in the UK are trialling a new conference 
phone - the Konftel 300IP SIP however it's still as expensive as a 
Polycom, but that might be the $/£ exchange - might be cheaper where you 
are?


For wifi phone, I tried Linksys iPhone. It works well but lacks a 
cradle. My users often forget to charge it when they leave for the day 
and come back to a dead wifi phone for the next morning.


Any good recommendations?


DECT not Wi-Fi?

Can you get the Siemens range over there?

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Re: [asterisk-users] conference and wifi phones

[Kelvin Chan]

 
 
  For wifi phone, I tried Linksys iPhone. It works well but lacks a
  cradle. My users often forget to charge it when they leave for the day
  and come back to a dead wifi phone for the next morning.
 
 I still don't get the market for this kind of phone.  DECT cordless phones
 can be had for $20 with very long battery life and range, and if the base
 is plugged into an ATA you have your wireless SIP phone.
 
 Long live the PAP2T...
 

There are couple markets for Wifi phones out there.
For frequent travellers, wifi phone is small and convenient. Sit down and login 
to hotel's wifi network and you are back in the office. No more longD charges.

For large corporate whose building extends more than a few rooms, wifi phone 
allows someone, tech support folks mainly, to use his phone just about anywhere 
in the building. You can probably argue corporate this size usually provide 
employees a business cell but we are not going there. :)

But for what I want to do, PTP2T + DECT is indeed the best solution. Can't beat 
that on price and battery life. And find phone button will definitely be used 
quite frequently! :)

Cheers,


Kelvin Chan   | Positronics Ent.
Product Development   |
  | unit 272
604-628-9330 (direct) | 8128 128th St.
604-585-2...@104 (main)   | Surrey, BC
604-585-3056 (fax)| Canada, V3W 1R1

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Re: [asterisk-users] conference and wifi phones

[Kelvin Chan]

 One of our local companies here in the UK are trialling a new conference
 phone - the Konftel 300IP SIP however it's still as expensive as a
 Polycom, but that might be the $/£ exchange - might be cheaper where you
 are?

It seems like an interesting product. Compared to Polycom 7000, it's roughly 
$400(list price) cheaper.
Just out of curiosity, the product is made in Sweden. It should be cheaper in 
UK. How much are they selling over there?

 
  For wifi phone, I tried Linksys iPhone. It works well but lacks a
  cradle. My users often forget to charge it when they leave for the day
  and come back to a dead wifi phone for the next morning.
 
  Any good recommendations?
 
 DECT not Wi-Fi?
 
 Can you get the Siemens range over there?
 
DECT is not wifi. It is just another digital wireless protocol built for 
wireless phones. 

I'm not sure what do you mean by Siemens range.


Kelvin Chan   | Positronics Ent.
Product Development   |
  | unit 272
604-628-9330 (direct) | 8128 128th St.
604-585-2...@104 (main)   | Surrey, BC
604-585-3056 (fax)| Canada, V3W 1R1



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Re: [asterisk-users] conference and wifi phones

[Dean Collins]

Siemens make a range of DECT handsets under the Gigaset model range.

Yes they shit all over every wifi handset I have ever used.

Dect is way better.

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelvin Chan
Sent: Monday, March 23, 2009 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] conference and wifi phones

 One of our local companies here in the UK are trialling a new conference
 phone - the Konftel 300IP SIP however it's still as expensive as a
 Polycom, but that might be the $/£ exchange - might be cheaper where you
 are?

It seems like an interesting product. Compared to Polycom 7000, it's roughly 
$400(list price) cheaper.
Just out of curiosity, the product is made in Sweden. It should be cheaper in 
UK. How much are they selling over there?

 
  For wifi phone, I tried Linksys iPhone. It works well but lacks a
  cradle. My users often forget to charge it when they leave for the day
  and come back to a dead wifi phone for the next morning.
 
  Any good recommendations?
 
 DECT not Wi-Fi?
 
 Can you get the Siemens range over there?
 
DECT is not wifi. It is just another digital wireless protocol built for 
wireless phones. 

I'm not sure what do you mean by Siemens range.


Kelvin Chan   | Positronics Ent.
Product Development   |
  | unit 272
604-628-9330 (direct) | 8128 128th St.
604-585-2...@104 (main)   | Surrey, BC
604-585-3056 (fax)| Canada, V3W 1R1



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Re: [asterisk-users] conference and wifi phones

[Michael Graves]

On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:

Siemens make a range of DECT handsets under the Gigaset model range.

Yes they shit all over every wifi handset I have ever used.

Dect is way better.


Amen to that! Unles you have some compelling reason for VoWifi it's not
worthy of consideration. Especially for SOHO or small biz use. Too
costly to do well.

DECT, in contrast, is solid and not overly expensive. SIP/DECT systems
like the snom m3 and Siemens S675/S685 give you all the benefits of a
SIP phone with the reliability and mobility of DECT. I much prefer this
to an ATA and analog DECT phone.

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
fwd 54245




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Re: [asterisk-users] conference and wifi phones

[Jeff LaCoursiere]

On Mon, 23 Mar 2009, Michael Graves wrote:

 On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:

 Siemens make a range of DECT handsets under the Gigaset model range.

 Yes they shit all over every wifi handset I have ever used.

 Dect is way better.


 Amen to that! Unles you have some compelling reason for VoWifi it's not
 worthy of consideration. Especially for SOHO or small biz use. Too
 costly to do well.

Roaming through hotspots is a point taken - you cannot exactly take your 
base to Starbucks.  But I have yet to run across a phone that does this 
well.


 DECT, in contrast, is solid and not overly expensive. SIP/DECT systems
 like the snom m3 and Siemens S675/S685 give you all the benefits of a
 SIP phone with the reliability and mobility of DECT. I much prefer this
 to an ATA and analog DECT phone.


Not that I am arguing - simply curious - what do you see as the advantages 
of a SIP phone over ATA/analog DECT?

Cheers,

j


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Re: [asterisk-users] Conference with an AGI inside Queue for password change

[David fire]

2008/12/19 Rajkumar S rajkum...@gmail.com

 Hi,

 I have a typical call center with queues and agents added via
 AddQueueMember. One of my requirement is to implement a forgot
 password function. If a caller does not remember the password, he
 calls up an unauthenticated line and the agent manually authenticates
 him. Then the caller should have a provision to reset his password.
 The requirement is that the agent should not know the new password of
 caller.

 One possible solution to this is for the agent to call an agi into
 conference with the call after caller has been verified. The agi will
 prompt for the password which the caller will type in his keypad.
 Although the agent will hear the password prompt, he cannot overhear
 the DTMF digits typed by caller.

 Can this be implemented in asterisk? I have looked but did not find
 any hints. Is there a better solution to the problem I am having?

 Thanks for reading and any replies.

 raj

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maybe a simpler solution is set some variables to the caller channel trasfer
to extencion where asterisk ask for the password put it in the data base and
then transfer back to the agent.
this is not so dificult to implement.

you can use the mysql function or you can make a webservice and use CURL
where you just put a url whit all the info.
the variables in the caller channel are for tell asterisk where tos end the
call back and the caller user to use in the mysql or webservices.
David

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Re: [asterisk-users] Conference with an AGI inside Queue for password change

[David fire]

maybe a simpler solution is set some variables to the caller channel trasfer
to extencion where asterisk ask for the password put it in the data base and
then transfer back to the agent.
this is not so dificult to implement.

you can use the mysql function or you can make a webservice and use CURL
where you just put a url whit all the info.
the variables in the caller channel are for tell asterisk where tos end the
call back and the caller user to use in the mysql or webservices.
David

2008/12/19 Rajkumar S rajkum...@gmail.com

 Hi,

 I have a typical call center with queues and agents added via
 AddQueueMember. One of my requirement is to implement a forgot
 password function. If a caller does not remember the password, he
 calls up an unauthenticated line and the agent manually authenticates
 him. Then the caller should have a provision to reset his password.
 The requirement is that the agent should not know the new password of
 caller.

 One possible solution to this is for the agent to call an agi into
 conference with the call after caller has been verified. The agi will
 prompt for the password which the caller will type in his keypad.
 Although the agent will hear the password prompt, he cannot overhear
 the DTMF digits typed by caller.

 Can this be implemented in asterisk? I have looked but did not find
 any hints. Is there a better solution to the problem I am having?

 Thanks for reading and any replies.

 raj

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[asterisk-users] Conference with an AGI inside Queue for password change

[Rajkumar S]

Hi,

I have a typical call center with queues and agents added via
AddQueueMember. One of my requirement is to implement a forgot
password function. If a caller does not remember the password, he
calls up an unauthenticated line and the agent manually authenticates
him. Then the caller should have a provision to reset his password.
The requirement is that the agent should not know the new password of
caller.

One possible solution to this is for the agent to call an agi into
conference with the call after caller has been verified. The agi will
prompt for the password which the caller will type in his keypad.
Although the agent will hear the password prompt, he cannot overhear
the DTMF digits typed by caller.

Can this be implemented in asterisk? I have looked but did not find
any hints. Is there a better solution to the problem I am having?

Thanks for reading and any replies.

raj

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Re: [asterisk-users] conference bridge

[Alex Balashov]

Nhadie Ramos wrote:
 Hi,
 
 How can i setup conference when i have 2 asterisk servers?
 my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV 
 just for redundancy (not really high availability). i have a web 
 interface, wherein i can create extension, conference etc.
 
 adding extension is ok, even if ext1 is registered on Asterisk 1 and 
 ext2 is registered on asterisk 2 they will still be able to call each 
 other, but on the conference, e.g. when ext1 dials conference no. 1000 
 and ext 2 dials conf 1000 also, they will be connected to two different 
 conference room. my meetme is also setup on realtime. how can i set it 
 up in such a way ext on registered on different asterisk server can 
 connect to the same conference room.

Build a SIP trunk between them, and have an extension in a dedicated 
dial plan context on one of them (the one that will host the shared 
conference room) that automatically dumps the caller into the MeetMe 
room when dialed from the other Asterisk server.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] conference bridge

[Nhadie Ramos]

Hi,

How can i setup conference when i have 2 asterisk servers?
my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just 
for redundancy (not really high availability). i have a web interface, wherein 
i can create extension, conference etc.

adding extension is ok, even if ext1 is registered on Asterisk 1 and ext2 is 
registered on asterisk 2 they will still be able to call each other, but on the 
conference, e.g. when ext1 dials conference no. 1000 and ext 2 dials conf 1000 
also, they will be connected to two different conference room. my meetme is 
also setup on realtime. how can i set it up in such a way ext on registered on 
different asterisk server can connect to the same conference room.

Regrdas,
Nhadie



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[asterisk-users] Conference Hangup

[Enrico Pasqualotto]

Hi all, I have a question on asterisk conference.
Now I use appl Meetme with option A  x because when a marked person 
hangup I want to close all the conference.
But what I have to do if I want two marked person and kill the 
conference when one of two hangup?


Is possible?

Thanks. Enrico.
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
http://www.linkedin.com/in/epasqualotto


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[asterisk-users] Conference Call Dial-Out to a participant

[broadband Voice]

I have created a conference call solution for a client and works fine. The
next challenge is to let the conference dial out the participant instead.
Has anyone done this before or know the function to achieve this? Thanks.
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Re: [asterisk-users] Conference Call Dial-Out to a participant

[Tilghman Lesher]

On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
 I have created a conference call solution for a client and works fine. The
 next challenge is to let the conference dial out the participant instead.
 Has anyone done this before or know the function to achieve this? Thanks.

Please see sample.call in the root directory.

-- 
Tilghman

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Re: [asterisk-users] Conference Call Dial-Out to a participant

[broadband Voice]

I looked through /etc/asterisk and could not find the folder sampl.call.

On 11/18/07, Tilghman Lesher [EMAIL PROTECTED] wrote:

 On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
  I have created a conference call solution for a client and works fine.
 The
  next challenge is to let the conference dial out the participant
 instead.
  Has anyone done this before or know the function to achieve this?
 Thanks.

 Please see sample.call in the root directory.

 --
 Tilghman

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Re: [asterisk-users] Conference Call Dial-Out to a participant

[Yann JOUANIN]

You can find it enclosed

 



sample.call
Description: Binary data
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Re: [asterisk-users] Conference Call Dial-Out to a participant

[Brett Crapser]

On Sun, 18 Nov 2007, broadband Voice wrote:
 I looked through /etc/asterisk and could not find the folder sampl.call.

 On 11/18/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
 On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
 I have created a conference call solution for a client and works fine.
 The next challenge is to let the conference dial out the participant
 instead.
 Has anyone done this before or know the function to achieve this?
 Thanks.

 Please see sample.call in the root directory.

 --
 Tilghman

  /usr/src/asterisk/sample.call

Shows a sample of using call files for the
   /var/spool/asterisk/outgoing  directory.

Brett

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Re: [asterisk-users] Conference Call Dial-Out to a participant

[Eric ManxPower Wieling]

broadband Voice wrote:
 I looked through /etc/asterisk and could not find the folder sampl.call.

That is the Asterisk configuration directory.  You are looking for the 
Asterisk SOURCE CODE directory.  If you installed from a package (.deb, 
.rpm, etc) then you will have to contact the packager to find out where 
sample.call is located for your package.

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Re: [asterisk-users] Conference Call Dial-Out to a participant

[Tzafrir Cohen]

On Sun, Nov 18, 2007 at 01:37:00PM -0600, Eric ManxPower Wieling wrote:
 broadband Voice wrote:
  I looked through /etc/asterisk and could not find the folder sampl.call.
 
 That is the Asterisk configuration directory.  You are looking for the 
 Asterisk SOURCE CODE directory.  If you installed from a package (.deb, 
 .rpm, etc) then you will have to contact the packager to find out where 
 sample.call is located for your package.

In the debian asterisk package:

  /usr/share/doc/asterisk/examples/sample.call

as expected.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Conference rooms

[Fabio Cappelletti]

I all,
I have a question about the  use of conference rooms: can I, with a Voip
telephone or softphone call some other telephone and invite them in a
conference room? I read a lot of documentations about asterisk, but i
can't find any example !

Thanks, best regard

Fabio


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Re: [asterisk-users] Conference rooms

[map]

Hi Fabio,

Once you have an Asterisk box that have a conference room configured and a
VoIP phone the  supports forward you can easily forward your guests to the
conference room.
Moreover you can create a conference room extension available, via password,
from the PSTN  line.

Hope this can help you.


On Nov 13, 2007 3:38 PM, Fabio Cappelletti [EMAIL PROTECTED] wrote:

 I all,
 I have a question about the  use of conference rooms: can I, with a Voip
 telephone or softphone call some other telephone and invite them in a
 conference room? I read a lot of documentations about asterisk, but i
 can't find any example !

 Thanks, best regard

 Fabio


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[asterisk-users] Conference Calls with single-line SIP

[Zaheer Master]

Hi all,
If I have 2 single-line SIP phones, I can still do a conference call using
Asterisk, right? For example, two people in my office are on the call, along
with 1 other person at a remote site.

Regards,
Zaheer


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Re: [asterisk-users] Conference Calls with single-line SIP

[Nick Brown]

Yes, that will work fine Zaheer.

On 16/10/07 1:32 AM, Zaheer Master [EMAIL PROTECTED] wrote:

 Hi all,
 If I have 2 single-line SIP phones, I can still do a conference call using
 Asterisk, right? For example, two people in my office are on the call, along
 with 1 other person at a remote site.
 
 Regards,
 Zaheer
 
 
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[asterisk-users] Conference call today at 12:30 PM EDT

[randulo]

Hey folks,

Here's your chance to report in about Astricon, ask or answer general
asterisk questions, talk about your asterisk-related (or voip-related)
projects, sites, work, anything. We interested and listening. We have
a great core group on these conferences, even though Indiana is
disproportionately represented for some reason :)

This conference is NOT limited to developers or gurus, anyone
interested in VOIP and asterisk is welcome to join anytime.

Let's talk! http://www.VoipUsersConference.org

You don't have to register now, you can call in on any phone (or via asterisk):

Call (724) 444-7444
Enter 22622# then 1# or your PIN # if you have one.

Asterisk instructions for a painless dialplan experience are here:

http://www.voipusersconference.org/asterisktalkshoecallinsetup.htm

Last but not least, Windows and Mac users can use the built in SIP
client from Talkshoe.com to call in with a single click. Batteries not
included.

rr

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Re: [asterisk-users] Conference bridge.

[Alex Balashov]

On Thu, 13 Sep 2007, Paul Hales wrote:

 On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote:
 Any recommendations for an affordable SIP conference bridge unit?  I mean
 one that isn't crappy;  something where the duplex and cancellation
 functions that are traditionally built into such devices actually work.

 Do you want something cheap or something that works?

   True, true.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] Conference bridge.

[Alex Balashov]

Any recommendations for an affordable SIP conference bridge unit?  I mean 
one that isn't crappy;  something where the duplex and cancellation 
functions that are traditionally built into such devices actually work.

I am referring to something that looks like this . . .

http://www.hardware.com/products/cnet/I212272.jpg

But not necessarily that.

--
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Conference bridge.

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Alex Balashov wrote:
 Any recommendations for an affordable SIP conference bridge unit?  I mean 
 one that isn't crappy;  something where the duplex and cancellation 
 functions that are traditionally built into such devices actually work.

Most people tend to go for the polycom kit.

- --
Kind Regards,

Matt Riddell
Director
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http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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Re: [asterisk-users] Conference bridge.

[Paul Hales]

On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote:
 Any recommendations for an affordable SIP conference bridge unit?  I mean 
 one that isn't crappy;  something where the duplex and cancellation 
 functions that are traditionally built into such devices actually work.

Do you want something cheap or something that works?

You can't have both.

PaulH


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Re: [asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT

[randulo]

ENUM and ISN
You may be interested to know that John Todd was kind enough to come
by at the last minute and give us a thorough grounding in ENUM and
expand our knowledge about http://Freenum.org where you should run,
not walk, to get yourself an ISN (ITAD Subscriber Number).

You can listen to or download an mp3 of John Todd's talk or any other
conference recording on one of these pages:

http://www.voipusersconference.org/topics.php - topic agenda, download
links and player
or
http://www.voipusersconference.org/astusers.htm - Flash player for recordings
or
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622

If any of you have guest suggestions or if you have something you
would like to come and tell us about, please contact me.

On 9/6/07, randulo [EMAIL PROTECTED] wrote:
 FRIDAY September 7th at 12:30 PM EDT

 http://www.asteriskusersconference.org for more information on how to
 listen, talk, or both :)

 This week, ENUM is the main subject, although our friends at e164.org
 haven't been able to talk to us as planned. Come on by and share what
 you know about ENUM or ask questions.

 Also, during Astricon, we are hoping people will call us with reports,
 either live or recorded and maybe someone will have some video?

 The IRC channel on Freenode.net is #asterisk_users_conference

 Past conference recordings:  http://www.asteriskusersconference.org/topics.php

 rr


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[asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT

[randulo]

FRIDAY September 7th at 12:30 PM EDT

http://www.asteriskusersconference.org for more information on how to
listen, talk, or both :)

This week, ENUM is the main subject, although our friends at e164.org
haven't been able to talk to us as planned. Come on by and share what
you know about ENUM or ask questions.

Also, during Astricon, we are hoping people will call us with reports,
either live or recorded and maybe someone will have some video?

The IRC channel on Freenode.net is #asterisk_users_conference

Past conference recordings:  http://www.asteriskusersconference.org/topics.php

rr

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[asterisk-users] Friday Aug 10 @ 12:30 PM EDT - Asterisk Users Conference

[randulo]

This week, the second part of connecting to the outside world using
TDM, ATA and even... IAX hardphones with compilable software.

More on topics and guests:

 http://groups.google.com/group/asterisk-users-conference

Instructions:

 http://www.AsteriskUsersConference.org

IRC on freenode.net: #asterisk-users-conference

/r

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[asterisk-users] Friday Aug 10th Asterisk Users Conference at 12:30 PM EDT

[randulo]

This Friday, part II of TDM solutions including ATA that do IAX and
SIP without opening the box and installing a card. Your experience in
this area would be appreciated.

You can find us here:

 http://www.AsteriskUsersConference.org

Also, a Google group has been created for discussions and scheduling
of the conferences. If you feel like this is of interest, please join
us:

http://groups.google.com/group/asterisk-users-conference

I hope we can make this a good way for you to know if  topic of
interest to you comes up. In the future, we'd like to get people using
ENUM and DUNDI to contribute their experience.

Please consider joining us.

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Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[randulo]

Steve,


On 8/3/07, Steve Totaro [EMAIL PROTECTED] wrote:
 I just tried to call in after creating an account.

 After the call connects, enter the show id: 22622# and your_PIN#

 I dial in and am asked for the podcast id, I enter 22622# and am told
 that my passcode is not correct. I also tried just entering my passcode
 but got the same error message.

 What am I doing wrong?

Nothing. What time did you do this? Are you sure the conference was
on? If it isn't live, you can't get in.

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Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[randulo]

Hi folks,

The August 3 edition of our Friday conference call and podcast kicks
off in just over and hour. I know the list isn't delivering properly
but if a few people  get this it'll be better than none.

We are going to be talking today about TDM inside and outside the box.
I own some antiiquated X100P FXO and a couple of TDM400p with the FXS
modules. This is how our company's litle pbx talks to two incoming
POTS lines and three regular phones connected to it. It also has a
long list of IAX and SIP providers connecting it to the rest of the
world. I am currently in the US so I use one of my 800 numbers to take
control of the asterisk box in Paris and make local calls in France
for a few pennies a minute. We also can send and receive SMS and of
course receive vmail via email.

But enough about me. What are you doing about connecting? And more to
today's point, what ATA are you using to connect without opening the
box and installing hardware?

Digium makes the IAXy, Sipura (whatever the name is today) has several
SIP models,  Grandcoughstream as well. What else is out there and
how well do they work?

Join us:

 http://AsteriskUsersConference.org

As Matt said somewhere, this conference is like a forum. It's a chance
for you to give back some of the valuable information and experience
into the community without writing a line of code. I've been using
asterisk for a few years and while I don't write code for it, I've
experimented a lot with lots of hardware and a long list of providers.
I've had time to learn a lot about the real world of all this stuff
and I'm willing to share what I know. How about you?



On 7/29/07, randulo [EMAIL PROTECTED] wrote:
 Hi,

 I am going to be on the road for the next few days and with the
 variable delay on the mailing list, I am posting this now, 4 days
 before the conference. If you haven't yet listened or participated,
 please consider doing it. We have a great kernel of people at all
 levels of expertise and ideas and questions can be kicked around
 immediately (well, there's a few milliseconds lag).

 This Friday we'll be talking about TDM solutions including ATA that do
 IAX and SIP without opening the box and installing a card. Your
 experience in this area would be appreciated. If you sell these
 solutions come over and pimp them.

 You can find us here:

  http://AsteriskUsersConference.org

 At this site there are three main conference pages, how to listen or
 participate, a player page for the archived recordings and a page with
 the extension for a SIP connection to the conference bridge. There are
 also two links to other pages, a related blog and AsteriskTV which
 will be getting more and better content and more formats due to the
 issue of Flash not being compatible with 64-bit systems. I'm working
 on this now and hope to have that done by mid September. If anyone
 knows how to convert mp3 to oog on a FreeBSD system, let me know. The
 video issues are going to be more complicated so if you have
 suggestions, please post them or email them to me.

 Thanks to the numerous people who have been supportive of these
 efforts including Mark Spencer and the guys at Digium.

 randy


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Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[Steve Totaro]

I just tried to call in after creating an account.

After the call connects, enter the show id: 22622# and your_PIN#

I dial in and am asked for the podcast id, I enter 22622# and am told 
that my passcode is not correct. I also tried just entering my passcode 
but got the same error message.

What am I doing wrong?

Thanks,
Steve

randulo wrote:
 Hi folks,

 The August 3 edition of our Friday conference call and podcast kicks
 off in just over and hour. I know the list isn't delivering properly
 but if a few people  get this it'll be better than none.

 We are going to be talking today about TDM inside and outside the box.
 I own some antiiquated X100P FXO and a couple of TDM400p with the FXS
 modules. This is how our company's litle pbx talks to two incoming
 POTS lines and three regular phones connected to it. It also has a
 long list of IAX and SIP providers connecting it to the rest of the
 world. I am currently in the US so I use one of my 800 numbers to take
 control of the asterisk box in Paris and make local calls in France
 for a few pennies a minute. We also can send and receive SMS and of
 course receive vmail via email.

 But enough about me. What are you doing about connecting? And more to
 today's point, what ATA are you using to connect without opening the
 box and installing hardware?

 Digium makes the IAXy, Sipura (whatever the name is today) has several
 SIP models,  Grandcoughstream as well. What else is out there and
 how well do they work?

 Join us:

  http://AsteriskUsersConference.org

 As Matt said somewhere, this conference is like a forum. It's a chance
 for you to give back some of the valuable information and experience
 into the community without writing a line of code. I've been using
 asterisk for a few years and while I don't write code for it, I've
 experimented a lot with lots of hardware and a long list of providers.
 I've had time to learn a lot about the real world of all this stuff
 and I'm willing to share what I know. How about you?



 On 7/29/07, randulo [EMAIL PROTECTED] wrote:
   
 Hi,

 I am going to be on the road for the next few days and with the
 variable delay on the mailing list, I am posting this now, 4 days
 before the conference. If you haven't yet listened or participated,
 please consider doing it. We have a great kernel of people at all
 levels of expertise and ideas and questions can be kicked around
 immediately (well, there's a few milliseconds lag).

 This Friday we'll be talking about TDM solutions including ATA that do
 IAX and SIP without opening the box and installing a card. Your
 experience in this area would be appreciated. If you sell these
 solutions come over and pimp them.

 You can find us here:

  http://AsteriskUsersConference.org

 At this site there are three main conference pages, how to listen or
 participate, a player page for the archived recordings and a page with
 the extension for a SIP connection to the conference bridge. There are
 also two links to other pages, a related blog and AsteriskTV which
 will be getting more and better content and more formats due to the
 issue of Flash not being compatible with 64-bit systems. I'm working
 on this now and hope to have that done by mid September. If anyone
 knows how to convert mp3 to oog on a FreeBSD system, let me know. The
 video issues are going to be more complicated so if you have
 suggestions, please post them or email them to me.

 Thanks to the numerous people who have been supportive of these
 efforts including Mark Spencer and the guys at Digium.

 randy

 

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[asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[randulo]

Hi,

I am going to be on the road for the next few days and with the
variable delay on the mailing list, I am posting this now, 4 days
before the conference. If you haven't yet listened or participated,
please consider doing it. We have a great kernel of people at all
levels of expertise and ideas and questions can be kicked around
immediately (well, there's a few milliseconds lag).

This Friday we'll be talking about TDM solutions including ATA that do
IAX and SIP without opening the box and installing a card. Your
experience in this area would be appreciated. If you sell these
solutions come over and pimp them.

You can find us here:

 http://AsteriskUsersConference.org

At this site there are three main conference pages, how to listen or
participate, a player page for the archived recordings and a page with
the extension for a SIP connection to the conference bridge. There are
also two links to other pages, a related blog and AsteriskTV which
will be getting more and better content and more formats due to the
issue of Flash not being compatible with 64-bit systems. I'm working
on this now and hope to have that done by mid September. If anyone
knows how to convert mp3 to oog on a FreeBSD system, let me know. The
video issues are going to be more complicated so if you have
suggestions, please post them or email them to me.

Thanks to the numerous people who have been supportive of these
efforts including Mark Spencer and the guys at Digium.

randy

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Re: [asterisk-users] Asterisk Users Conference Friday at 12:30 PM EDT

[randulo]

On 7/27/07, dave cantera [EMAIL PROTECTED] wrote:
 randulo,
 I could not get into the conference today...  the SIP line was busy, no
 matter what I do, the website thinks I'm not logged in and gives me the
 login page.  after I login, anything I want to do brings me back to the
 login page... so I tried to re-setup the account thinking I wasn't
 logging in, and the user name was taken  so I know I'm signed up.

 Dave,

I answered privately to get more details, but if anyone is having SIP
problems, for reference:
I have logged in successfully using asterisk hundreds of times. That
info is at http://x2z.eu

I've also used numerous SIP clients including the Java one called
ShoePhone built in to the conference interface (Win/Mac only) and
X-Lite, Gizmo project, Idefisk/Zoiper and some group meet freeware for
the Mac, so it wouldn't seem to be a problem  on the SIP server side.

I know tzafrir has had problems with the SIP and we can't figure out
why it doesn't work for him. However, IIRC, his issue is not an
apparent busy signal, but an auth problem. That busy signal means you
are not reaching the server (unless you see other messages). You can
call the SIP server anytime and it will always answer. You can not
however enter a conference until the host is there.

hth,

randy

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[asterisk-users] Asterisk Users Conference Friday at 12:30 PM EDT

[randulo]

You can listen or join the Asterisk Users Conference Fridays at  12:30 PM
EDT

Today's subject suggestions:

FAX capabilities, what's your solution?
Multiple asterisk server implimentation: ENUM, DUNDI or even two servers
connected
Your subjects?

Share your ideas, ask your questions!

See  http://x2z.eu  for instructions on how to join or listen

irc://irc.freenode.net/asterisk-users-conference

Note that the SIP channel will only be open from about 12:20PM EDT.
Testing before then will give you the message your PIN is not valid but if
it answers, you're good.

; SIP call
; exten = AUC,1,Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#))

If you would like to talk about services or products your company provides
and answer users' questions, contact me off list. Anyone is welcome to be a
guest and answer users' questions.

Previous guests have been Teliax, Lumenvox, Digium (duh!), Trixbox,
Adhearsion

Listen to the archived recordings here:

 http://x2z.eu/astusers.htm

The Asterisk Users Conference is independently run and has nothing to do
contractually or financially with Digium who owns the Asterisk trademark.
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