[Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing

[Kanuri, Seshu \(Company IT\)]

Does 
anyone know an IP Phone or Device  that works with 
Asterisk as an announcement only device with a loud speaker and that 
is online forever and will not hungup for any reason and even if it hangsup, it 
will reboot itself to connect to a VOIP Server with a given configuration to 
receive the voice data stream. 
 
I have the requirement 
for a few hundred such devices for a conferencing application that keeps 
relaying a conference proceeding all the at a remote unattended location, 
without hanging up. If there is  a hangup of the phone it should reset 
itself again to rceive the stream again from the Asterisk 
server.
 
Any 
information on such a device is 
appreciated.
 
Seshu KanuriMorgan Stanley | 
Technology1633 Broadway | Floor 19New York, NY 10019Phone: +1 212 
537-2849[EMAIL PROTECTED]



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RE: [Asterisk-Users] Which is Better!

[Kanuri, Seshu \(Company IT\)]

I 
second that, except that if you have more money to spend and want a maintenance 
free device, go for Cisco gear. 
Qunitum needs a reboot once in a while. Cisco gear does 
not. There is some difference in call quality and call handling as 
well.
 
Seshu 
Kanuri
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Anders 
SvenssonSent: Tuesday, November 22, 2005 11:26 AMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] Which is Better!


We have tried both but 
given up hope about them. So now we only use Quintum DX series. Amazing 
machine
 
Anders Svensson Bobas 
Communication
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Goran DonevSent: den 22 november 2005 
16:41To: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Which is 
Better!
 
Which FXO gateway is better and has 
better sound quality.
 
AudioCodes?
 
Or 
 
Mediatrix.
 
Thanks for your 
input



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RE: [Asterisk-Users] Re: [OTAnn] Feedback

[Kanuri, Seshu \(Company IT\)]

I totally agree. But doe the Asterisk list servers have any such feature
to block the spam and delete the spamming users? I don't think so.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Tuesday, November 08, 2005 2:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: [OTAnn] Feedback

This is blatant spam.

Looking at: http://roomity.com/advertising.jsp
it looks like they have spammed at least 60,000 other mailing lists too.
WTF would I want to use their crappy video and flash ad spewing crappy
web interface that requires me to be online all the time over my awesome
ad-free threaded client?

If you run a mailing list or email server, it's time to firewall their
ass.

On Tue, Nov 08, 2005 at 10:16:18AM -0500, Steven said:
> I use a newsreader pointed at gmane.org.
> It is agregated and only uses my internet connection when I tell it
to.
> 
> 
> "shenanigans" <[EMAIL PROTECTED]> wrote in message 
> news:[EMAIL PROTECTED]
>> I was interested in getting feedback from current mail group users.
>> 
>> We have mirrored your mail list in a new application that provides a 
>> more aggregated and safe environment which utilizes the power of
broadband.
>> 
>> Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version 
>> adds broadcast video and social networking such as favorite authors 
>> and an html editor.
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RE: [Asterisk-Users] how to conferencd in Asterisk

[Kanuri, Seshu \(Company IT\)]

Nrk,

Do some googling and try to find all this info on the Wiki site for
Asterisk.

SK

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nr k
Sent: Sunday, November 06, 2005 5:20 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how to conferencd in Asterisk

Hi all


How ro enable conference in asterisk and also how to make call between
sccp device and sip device is there any special config in asterisk.

regards
ramakrishnan.n




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RE: [Asterisk-Users] Uninstall AMP

[Kanuri, Seshu \(Company IT\)]

I want to give the benefit of doubt to the suggestion as I think there
is a misunderstanding of the suggested method of removal of AMP.

I guess that he was suggesting to remove it from your linux installation
by using the rm -rf command as under

 cd /var/www
 rm -rf *

which will efectively remove all the web pages associated with the AMP
instalation.

-Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anders
Svensson
Sent: Friday, November 04, 2005 1:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Uninstall AMP

I agree. It is like we newbie's on Asterisk is just trouble for the list
members. Pity there is no newbie list. But all were newbie's in the
beginning and not so pompous as some on this list

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Ferrell
Sent: den 4 november 2005 19:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Uninstall AMP


Claudio Canseco wrote:
> Really is that the way to uninstall FOP and AMP?, thank you i've been 
> looking for an answer about it.
>  
> Regards
> Claudio.
> 

No Claudio,

That will wipe your system.  He's being a smartass.

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RE: [Asterisk-Users] Looking por a provider to work with asterisk

[Kanuri, Seshu \(Company IT\)]

You 
can also try http://www.terracall.com I have been using them with good results 
lately
 
Seshu 
Kanuri


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jason 
BrashearSent: Thursday, November 03, 2005 11:03 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Looking por 
a provider to work with asterisk


I know about 
broadvoice.com
But are they the only 
solution?
I want to have two lines with 
Asterisk.
This is just a home 
install.
Believe it or not I have been using 
Vonage for about 2 ½ years and now I want to get rid of them 
to
Use and learn 
Asterisk.
Any help would be 
appreciated.
-Jason



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RE: [Asterisk-Users] How to configure Asterisk through webmin

[Kanuri, Seshu \(Company IT\)]

The 
Thirdlane PBX Manager solution is just a few perl scripts. This is no better 
than what you can do by directly modifying the Asterisk Config files or many 
Open Source GUIs like Phonecall etc you have out there.
 
Infact 
Areski's A2Billing has a good extension configurator in the solution. So that 
may be something you can consider.
 
Seshu 
Kanuri


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
PikoroSent: Thursday, November 03, 2005 7:09 PMTo: 
[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [Asterisk-Users] How to configure Asterisk 
through webmin
I tried the third lane asterisk manager thingy for webmin and let me 
tell you, it did not work.  Only made things harder and i had to result to 
making the configuration by hand in order to get asterisk to work.  Going 
to email them today and ask for a refund.That webmin module by third 
lane looks like a good solution, but the thing i noticed by reading the manual 
was that there are quite a few references to "you'll have to change that in the 
config file" type lines.  Basically, it's good for creating extensions, but 
nothing more.AaronStefan-Michael. Guenther (in-put GbR) 
wrote: 

  On Thu, November 3, 2005 17:46, nr k said:

Hi all
I configured asterisk and webmin.i dont know how to
integrate webmin with asterisk and how to access
asterisk
through webmin.pls do the needful.

regards
ramakrishnan.n
  Asterisk is not managed through webmin. Webmin is a tool to help
administer the rest of the server.

 and Asterisk, too:

Have a look at THIRD LANE ASTERISK PBX MANAGER
http://www.thirdlane.com/opensource.htm#manager

Stefan
  



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[Asterisk-Users] RE: [Asterisk-biz] Asterisk as a Voice Conference Server

[Kanuri, Seshu \(Company IT\)]

Iain Barker Wrote:
-
>Our experience with over 10 or more participants 
>in a single Asterisk conference was that quality 
>degraded quite rapidly.
 
Is this really true as there were many in this list 
who had confirmed that they have used the conference 
bridge for a lot more connections than what you have
Suggested as the upper limit.

Logically the conference bridge should work at the 
same capacity as the number of calls Asterisk can 
handle in a given configuration.

Though your solution looks impressive and probably is
the best for upto 30 simultaneous calls, I am more
interested in knowing what it takes for Asterisk to be 
able to handle the 100 channels I need to run 
Simultaneously.

Seshu Kanuri



-Original Message-
From: Iain Barker [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, November 02, 2005 1:41 PM
To: Kanuri, Seshu (Company IT)
Subject: Re: [Asterisk-biz] Asterisk as a Voice Conference Server

Seshu,

Our experience with over 10 or more participants in a single Asterisk
conference was that quality degraded quite rapidly.

The solution was a dedicated hardware bridge for conference mixing

http://www.aastra.com/enterpriseip/pro_238.asp



Kanuri, Seshu (Company IT) wrote:
>
>I am working on a bid for a New York State requirement where we need to

>provide access to 100 Simulataneous Investors to get into a conference 
>with the Pension Funds Officer for discussions.
>
>As you might have guessed it, I am presenting an Asterisk enabled 
>Conference solution.
>
>One of the Bid requirement is to provide three verifiable references 
>who have implemented a similar voice conference solution for more or 
>less 100 simultaneous calls, with a possible recording of the entire
call.
>
>If anyone has implemented this on a commercial scale, I am looking for 
>referrals at this time, and a possible co-operation in future.
>
>I would appreciate if you can send me your name, contact Info, company 
>name and a one para description of the solution and the name/type of 
>client whom/where this solution is running at this time.
>
>A couple of minutes of your time is needed when the guys at Albany may 
>like to speak to you for a confirmation that Asterisk is real and it 
>can do the 100 people conference, what they are looking for.
>
>  
>

I do thousands of conferences a day using asterisk as the backend, most
are in the 5-50 user range, but many are in the 150+ range. (but, I use
app_conference, not app_meetme for them).

I can give you my contact information off-list if you want it.

-SteveK


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RE: [Asterisk-Users] inband dtmf on ploycom ip501?

[Kanuri, Seshu \(Company IT\)]

Add the line as under to your sip.conf entry.

Progressinband=no 
;progressinband=never   - is erroneous
 
Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Miller
Sent: Tuesday, November 01, 2005 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] inband dtmf on ploycom ip501?

Damon Estep wrote:
> Anyone have any hints on how to get the polycom ip501 to send dtmf 
> inband, our upstream providers require inband and the native rfc2833 
> format of the polycom does not work.

In order for inband to make it beyond Asterisk, you need to disable
rfc2833 control in the Polycom config file "sip.cfg" (i.e. via ftp
server).

tone.dtmf.rfc2833Control="0"

It appears that although Asterisk recognizes and uses inband dtmf
internally, rfc2833 is used on the external channel. I noticed this
behavior when remote IVR systems weren't acknowledging dtmf.

Regards,
Chris

Chris Miller
President - Rocket Scientist
ScratchSpace Inc.
(831) 621-7928
http://www.scratchspace.com
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[Asterisk-Users] Is anyone using OpenSer - A fork of SER?

[Kanuri, Seshu \(Company IT\)]

Folks!

I want to know if anyone in the list is using OpenSER, 
which appears to be a fork of SER. If so can you post
Your comments on its functionality?

The location where this is available is here:
http://openser.org/index.php#about

Some of the the features I am impressed with being... 
1)Programming command syntax, which was not available 
in SER.
2)Modular Architecture like Asterisk

A list of modules available are here:
http://openser.org/diffs-0.9.0.php

Seshu Kanuri


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RE: [Asterisk-Users] ANNOUNCEMENT : A2Billing - AreskiCC V3 new release

[Kanuri, Seshu \(Company IT\)]

Areski,
 
The featurelist in this version is Awsome. This will clearly and
absolutely make all those Closed Source Billing systems and so called
Soft Switches like Bicom's Switchware, obsolete.

Kudos for your effort and contribution to the Asterisk users.

This probaly is one of the most important application for Asterisk
similar to AMP, which will help users create a business out of the
Asterisk Box.

Seshu Kanuri
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski K
Sent: Wednesday, October 26, 2005 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ANNOUNCEMENT : A2Billing - AreskiCC V3 new
release

Dear Friends,


Great day for the callingcard-lovers !!!
I am pleased to release the version 3.0 of AreskiCC !!!
http://www.areski.net/a2billing/
http://www.voip-info.org/wiki/view/A2Billing


Little unexpected change, we got a new name... bit more serious
"A2Billing"
Many many features have been added, lot bugs solved and a bunch of good
enhancements made!


The key newest features :
- Full MYSQL support
- USE PHP-AGI LIB 2.14
- CALLERID SUPPORT AUTHENTICATION
- MUSICONHOLD CUSTOMIZATION BY DIALPREFIX
- UPLOADING TOOLS TO CONFIGURE MUSICONHOLD
- INVOICES PDF / HTML
- ADD NET REPORTING FROM ASTERISK-STAT
# calls compare
# monthly traffic
# daily load
- DEFAULT DIALING FOR RATECARD
- FAILOVER TRUNK DEFINITION
- FASTER RATECARD CREATION (range, interval)
- MONITOR CALLS & LISTEN BY SIMPLE CLICK TO THE CALL
- REDIAL FEATURE
- Register_global = Off :D
- Recurring service : Apply batch process of certain card.
- MENU CHOOSE THE LANGUAGE
- CONFIGURATION /etc/asterisk/a2billing.conf
- SUPPORT SEVERAL CONFIGURATION
DeadAGI(a2billing.php|%idconfig%)
- SEVERAL EXPIRATION CARD MODE
- VOUCHER SUPPORT
- DNID BILLING RULES SUPPORT (CHOOSE A PREFERENTIAL RATECARD
ACCORDING TO THE DNID)
- FULL CURRENCIES SUPPORT MANAGEMENT - USE WWW.OANDA.COM FOR
CURRENCIES VALUE/LIST
- and more...



Other good thing, we have an handbook :D It's covering mainly
installation for the moment but I will complete the part for the user
guideline pretty soon.
Any help (documenting, dev...) would be greatly appreciated, so if
someone is willing to help, please contact me  !!!


I am sure you will enjoy this new version!
Have fun and don't forget to send me some feedback, /Areski

- A2Billing (Asterisk to Billing)
http://www.areski.net/a2billing/



--
--
~ - Belaid Arezqui ( [EMAIL PROTECTED] )
   'v'- Cell Phone. : (+34) 650 78 43 55  (Spain, GMT+1hr)
  // \\
 /(   )\  - RACCOON TRAINER
  ^`~'^   - http://www.areski.net
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RE: [Asterisk-Users] Goiax.com DID not working anymore?

[Kanuri, Seshu \(Company IT\)]

[EMAIL PROTECTED] wrote
> It's a free service. It belongs on this list.
 
Olle is right. Even if it is a free service it does not belong here.
This forum is for any Asterisk related user issues, not some DID issue
of one of a hundred such service providers.

Take it off this list.

Seshu Kanuri


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RE: [Asterisk-Users] PA168S/AT320P

[Kanuri, Seshu \(Company IT\)]

have you configured the STUN server on the phone to any one of the
available stun servers like stun.xten.net?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Thursday, October 13, 2005 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PA168S/AT320P

Hi,
thanks to reply:
1)SIP
2)yes. I've used the original 1.46 for SIP protocol Also your solution
do not work.
Are 2 days that I'm trying configurations and googling for this problem,
but nothing!
Always: "LOG ON FAILED"
I've saw about problems with this phone, but my hope was that with the
new firmware something could be solved.

Thanks again.

2005/10/13, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]>:
> 1)What is the protocol you are using? SIP or IAX2?
> 2)Have you applied the correct firmware to the Phone?
>
> Pa168 phones are falwless when connecting to Asterisk.
>
> Start the configuration as asimple entry as under.
>
> I have added Port address and allowed codecs in the config below:
>
> [221]
> type=friend
> username=221
> secret=secret
> context=local
> host=dynamic
> dtmfmode=rfc2833
> nat=yes
> Port=5060
> Disallow=all
> Allow=g729
> Allow=ulaw
> Allow=gsm
>
> Seshu Kanuri
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of FaberK
> Sent: Thursday, October 13, 2005 9:35 AM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] PA168S/AT320P
>
> Hi all!
> I've got a problem with thia PA168S/AT320P telephone.
> I got 2 servers: one with SER and the other with Asterisk.
> All users are on SER and Asterisk is the gateway/voicemail.
> In these days I'm starting some tests using Asterisk accounts users.
> With this PA168S/AT320P, if I use it with a user from SER, it's ok but

> I can forget to use it with Asterisk users!!!
> I've also updated the firware at the 1.46 released the october 10th, 
> but nothing changed.
> These are my user settings:
> 
> [221]
> type=friend
> username=221
> secret=secret
> host=dynamic
> canreinvite=yes
> dtmfmode=rfc2833
> nat=yes
> context=local
> [EMAIL PROTECTED]
> callerid="221" <221>
> accountcode=221
> qualify=yes
> 
> Any ideas?
>
> Thanks to all.
> --
> .:FaberK:.
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>
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RE: [Asterisk-Users] PA168S/AT320P

[Kanuri, Seshu \(Company IT\)]

1)What is the protocol you are using? SIP or IAX2?
2)Have you applied the correct firmware to the Phone?

Pa168 phones are falwless when connecting to Asterisk.

Start the configuration as asimple entry as under. 

I have added Port address and allowed codecs in the config below:

[221]
type=friend
username=221
secret=secret
context=local
host=dynamic
dtmfmode=rfc2833
nat=yes
Port=5060
Disallow=all
Allow=g729
Allow=ulaw
Allow=gsm

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Thursday, October 13, 2005 9:35 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] PA168S/AT320P

Hi all!
I've got a problem with thia PA168S/AT320P telephone.
I got 2 servers: one with SER and the other with Asterisk.
All users are on SER and Asterisk is the gateway/voicemail.
In these days I'm starting some tests using Asterisk accounts users.
With this PA168S/AT320P, if I use it with a user from SER, it's ok but I
can forget to use it with Asterisk users!!!
I've also updated the firware at the 1.46 released the october 10th, but
nothing changed.
These are my user settings:

[221]
type=friend
username=221
secret=secret
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
nat=yes
context=local
[EMAIL PROTECTED]
callerid="221" <221>
accountcode=221
qualify=yes

Any ideas?

Thanks to all.
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RE: [Asterisk-Users] Open Source Content Management System - Joomla

[Kanuri, Seshu \(Company IT\)]

You may also like to check the Asterisk management module developed for
Drupal at http://www.drupal.org

Check in the list of modules available.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goran
Skular
Sent: Tuesday, October 11, 2005 3:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Open Source Content Management System -
Joomla

Our Web is based on Mambo portal software and it is connected with our
Asterisk installation.

We wrote our own CDR rating engine and modules for Mambo. Also, you can
register for VoIP termination services inside mambo.. we wrote one
component and couple of modules.

So, when user create an account on mambo, asterisk account is also
created automaticly (if choosed... cron script every 10 minutes) ...

CDR rating is simple, but it works.. there is no fancy things in rating
like tariffs or similar... (at this moment)

You can check on www.slsolucije.hr .. it is on Croatian.. but.. 


>-Original Message-
>From: [EMAIL PROTECTED] [mailto:asterisk-users- 
>[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)
>Sent: Monday, October 10, 2005 11:21 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: RE: [Asterisk-Users] Open Source Content Management System - 
>Joomla
>
>>And how exactly is Asterisk relevant to a CMS? could you give a more 
>>specific example?
>
>This is relevant where Administrative users wanted to manage their 
>Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc
>
>Seshu
>
>
>NOTICE: If received in error, please destroy and notify sender.  Sender

>does not waive confidentiality or privilege, and use is prohibited.
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RE: [Asterisk-Users] Open Source Content Management System - Joomla

[Kanuri, Seshu \(Company IT\)]

Are has answered this question with an example, which shall statisfy the
curious. 

-Original Message-
Kanuri, Seshu (Company IT) wrote:
> This is relevant where Administrative users wanted to manage their 
> Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc
Now you have made me curious :-)

In what way could a CMS contribute to this? (just curious because I'm
using some CMS for other purposes)

Cheers


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RE: [Asterisk-Users] Open Source Content Management System - Joomla

[Kanuri, Seshu \(Company IT\)]

>And how exactly is Asterisk relevant to a CMS? could you give a more 
>specific example?

This is relevant where Administrative users wanted to manage their
Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc

Seshu


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[Asterisk-Users] Open Source Content Management System - Joomla

[Kanuri, Seshu \(Company IT\)]

There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL. The guys who
developed Mambo have a new product now - Joomla. I am using this and it
appears to be better than Mambo in many respects. Read the gist about
Joomla below.

-
If you've read anything at all about Content Management Systems (CMS),
you'll probably know at least three things: CMS are the most exciting
way to do business, CMS can be really, I mean really, complicated and
lastly Portals are absolutely, outrageously, often unaffordably
expensive. 

Joomla! is set to change all that ... Joomla! is different from the
normal models for portal software. For a start, it's not complicated.
Joomla! has been developed for the masses. It's licensed under the
GNU/GPL license, easy to install and administer and reliable. Joomla!
doesn't even require the user or administrator of the system to know
HTML to operate it once it's up and running.

http://www.joomla.org/

--

Seshu Kanuri


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[Asterisk-Users] Issue with terra-call today

[Kanuri, Seshu \(Company IT\)]

Looks like Terracall 
has not only reduced the rates but also reduced their ability to connect the 
calls to India. 
Today we are not 
able to make even one call, but the CDRs are still coming as connected and we 
are being charged.
 
Please note the 
request I sent below for the credit.

 
Adella,
 
My 
support people are trying to call India since morning and have not been 
able to connect. However I have been charged as under:


  
  
SIP
011918632210439
10/7/2005 7:58:50 AM
1:00
$0.08
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:58:31 AM
0:00
$0.00
I
Incoming Completed
  
SIP
011918632210439
10/7/2005 7:28:35 AM
1:00
$0.08
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:28:18 AM
0:00
$0.00
I
Incoming Completed
  
SIP
011914424332094
10/7/2005 7:26:00 AM
1:00
$0.08
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:25:40 AM
0:00
$0.00
I
Incoming Completed
  
SIP
7323874133
10/7/2005 7:24:03 AM
1:00
$0.02
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:23:40 AM
0:00
$0.00
I
Incoming Completed
  
1280642680
280642680
10/7/2005 7:15:43 AM
0:00
$0.00
I
Incoming Completed
  
SIP
011914424332094
10/7/2005 7:15:43 AM
0:00
$0.00
O
Outgoing No Answer
  
SIP
011914424332094
10/7/2005 7:14:31 AM
1:00
$0.08
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:14:17 AM
0:00
$0.00
I
Incoming Completed
  
SIP
011918632210439
10/7/2005 7:13:04 AM
1:00
$0.08
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:12:49 AM
0:00
$0.00
I
Incoming Completed
  
SIP
011914424332094
10/7/2005 7:11:05 AM
1:00
$0.08
O
Outgoing Completed
  
1280642680
280642680
10/7/2005 7:10:51 AM
0:00
$0.00
I
Incoming 
Completed
 
Please give credit for the 
above calls as we have not been able to connect at all.
 
Seshu 
Kanuri



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RE: [Asterisk-Users] Asterisk on windows

[Kanuri, Seshu \(Company IT\)]

The religious Zealot was catholic or more accurately speaking, a
Zehova's witness  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of canuck15
> 

So how does that explain muslims blowing themselves up and taking as
many non-believers with them as possible?  I don't see any of them
trying to convert anyone.  Is this a bug in Linux?

I'm not sure if this is a bit off topic but I apologize if it is. ;)
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RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

[Kanuri, Seshu \(Company IT\)]

Morgan,

For the most part people have either ignored or dismissed this company
Bicom Systems and it's products Pbxware and Switchware as they don't fit
into the same mould as Asterisk / Linux / APACHE/ Mysql /PHP /SugarCRM.
Bicom Systems want to ride on the Open source systems without giving
back anything to the community. 

[EMAIL PROTECTED], AMP, Phonecall, AreskiCC, ASTPP etc have a positive and
different approach and hence they are gaining popularity.

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morgan
Gilroy
Sent: Wednesday, September 28, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

Well I guess im the lucky one then.
I have had no problems with them at all and have been treated very well.

Anyone else had a good/bad experience with them?


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RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

[Kanuri, Seshu \(Company IT\)]

You are probably the guy whom they are using as reference point, to
screw others.

Why are they not delivering the software and support to several guys who
paid for the software.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morgan
Gilroy
Sent: Wednesday, September 28, 2005 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

Sorry?
I have used Bicom for a while and the support I get is top notch.
And the end product is working very well and does everything we have
asked of it.
Bicom have even added features for us with no charge and have
implemented them very fast.

Sure its not as flexible as plain asterisk but its defiantly a lot
faster and easier to setup.

And so far I have been very happy with it.

 > -Original Message-
 > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)  >
Sent: 27 September 2005 19:07  > To: Asterisk Users Mailing List -
Non-Commercial Discussion  > Subject: RE: [Asterisk-Users] Software only
Asterisk PBX (commercial)  >  > Don't you ever recommend Bicom as they
take your money and will never  > deliver a product that works.
 >
 > -Original Message-
 > From: [EMAIL PROTECTED]
 > [mailto:[EMAIL PROTECTED] On Behalf Of Morgan
> Gilroy  > Sent: Tuesday, September 27, 2005 1:20 PM  > To: Asterisk
Users Mailing List - Non-Commercial Discussion  > Subject: RE:
[Asterisk-Users] Software only Asterisk PBX (commercial)  >  > Also
check out http://www.bicom.us pretty expensive but if that's your  >
thing :)  >  >  > -Original Message-  >  > From:
[EMAIL PROTECTED]
[mailto:asterisk-users-
 > > [EMAIL PROTECTED] On Behalf Of Ronald Hartmann  > Sent: 27
> September 2005 16:47  > To: 'Asterisk Users Mailing List -  >
Non-Commercial Discussion'
 >  > Subject: RE: [Asterisk-Users] Software only Asterisk PBX
(commercial)
 > >  > >  > >Are there any switchvox/fonality type Asterisk based PBXs
> where I can  > >buy just the software?  I don't want to buy their  >
'bundles' that come  > >with junky PC hardware.  I just want their  >
software/GUI to run on my  > >hardware.
 >  > >
 >  >
 >  > Have a look at the AMP project
 >  >
 >  > http://sourceforge.net/projects/amportal
 >  >
 >  > ~ron
 >  >
 >  > >
 >  >
 >  >
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 >
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RE: [Asterisk-Users] Asterisk on windows

[Kanuri, Seshu \(Company IT\)]

Could not agree more with Matt. I have been a linux geek for a long time
and I would think twice before calling Windows a crap o/s as linux feels
crappier when it comes to usability, administration and the pain in
making it work the first time, with due respect to all those who are
contributing to the open source revolution.

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, September 28, 2005 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk on windows

>
> Personally, I could care less which O/S the stuff runs on as long as 
> it runs reliably, and the sys admin understands how to manage whatever

> sytem he/she is responsible for.
>
>

Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or so).   There is definately a
time and place for Windows.. I'm just not sure a real-time-VoIP server
is the time or place.Being semi-half serious about the GUI there
also.You install X on your Asterisk server and things will not be
happy either.


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RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

[Kanuri, Seshu \(Company IT\)]

Don't you ever recommend Bicom as they take your money and will never
deliver a product that works.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morgan
Gilroy
Sent: Tuesday, September 27, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial)

Also check out http://www.bicom.us pretty expensive but if that's your
thing :)

 > -Original Message-
 > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ronald Hartmann  > Sent: 27
September 2005 16:47  > To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
 > Subject: RE: [Asterisk-Users] Software only Asterisk PBX (commercial)
>  > >  > >Are there any switchvox/fonality type Asterisk based PBXs
where I can  > >buy just the software?  I don't want to buy their
'bundles' that come  > >with junky PC hardware.  I just want their
software/GUI to run on my  > >hardware.
 > >
 >
 > Have a look at the AMP project
 >
 > http://sourceforge.net/projects/amportal
 >
 > ~ron
 >
 > >
 >
 >
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RE: [Asterisk-Users] IAX2 hard phone

[Kanuri, Seshu \(Company IT\)]

Alberto,
 
PA168 
chip does not have Hold and Transfer features on it until firmware version 1.44. 
Atcom never claimed that 
these 
will work as the Pa168 firmware is still under development.
 
Yesterday I met Peter Sun, President and owner of Atcom 
China,  in New York. He is here to attend VON in 
Boston.
I have 
enquired about the fix for this and he said that Hold and Transfer are working 
with 1.45 Firmware. 
I 
mentined that this is not the case with the phones we tried to use 
here.
 
Peter 
mentioned that firmware version 1.46 is going to be relased this week, which 
will provide these features 
and 
also the Voice Mail Messages Indicator Led should work too.
 
I am 
waiting for this firmware to be released. I will forward this to you as soon as 
it is released. In the meantime you
can 
check for the updates at http://www.iareaphone.com under downloads, 
if this becomes available sooner.
 
Seshu 
Kanuri


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto 
RiscoSent: Tuesday, September 27, 2005 9:06 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] IAX2 hard 
phone


I purchased an IAX2 hardphone, X100 otherwise known as a 
Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, 
from a US retailer.  I was able to 
configure the phone to work with my Asterisk box, except the hold and transfer 
buttons do not work.  When you press the hold button, it rings endlessly, 
the transfer button, displays “transferring” but it does nothing.  Has 
anybody with these phones run into similar problems? Or can recommend a good 
functional IAX2 hard phone.
 
 
Thanks,
 
Alberto
 
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[Asterisk-Users] Asterisk Platform - Success Strories - iAreanet in the news.

[Kanuri, Seshu \(Company IT\)]

Very rarely we come across real success stories using Asterisk as a part
of a great solution, and when I see one, I want to share it with you.
Though it is not mentioned in the news item, it is a fact that Iareanet
uses Asterisk as the core for their messaging part of the solution and
today they are on the news with a 5 star rating.

Check the link below:
http://mobileoffice.about.com/od/connectingviatheinternet/gr/iareanet.ht
m

Folks here already may be aware of the Contact Center Solution Aheeva
(http://www.atelka.com) built around Asterisk which is a success too.

Iareanet product as I know of has been helping katrina victims and many
businesses as a disiater recovery-business continuancy solution.
 
Seshu Kanuri


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RE: [Asterisk-Users] USB Phones for use with Asterisk

[Kanuri, Seshu \(Company IT\)]

USB phone and NAT - What has USB Phpne got to do with NAT?

USB Phone is just a hardware piece that pipes the audio output from your
softphone.

Your softphone has to take care of that.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Callum
McGillivray
Sent: Thursday, September 15, 2005 8:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] USB Phones for use with Asterisk

Hi all,

I have a question that I was hoping someone could answer for me.

I would like to find a USB phone that works with Asterisk... preferably
one that does not have any issues with NAT.

Can anyone point me to something suitable ?  We are essentially planning
to do a beta for a few hundred customers on our network and would like
them to be able to call one another @ no charge.

Because these are going to be end users who have no idea about Voice /
VOIP stuff... it needs to be extremely easy to use.

Can someone point me in the right direction ?

Cheers,

Callum
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RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

[Kanuri, Seshu \(Company IT\)]

The article below, posted a while ago by a Wharton dude is very predictive and 
interesting on Skype's power.

http://www2.cio.com/higher/report3799.html

Some snippets from the article are pasted here:
--
Skype's potential has a few big investors placing bets. Speaking at the 
AlwaysOn Innovation Summit at Stanford University on July 20, Tim Draper, 
founder and managing director of venture capital firm Draper Fisher Jurvetson, 
said he invested about $10 million in Skype after meeting with Zennström. 
Draper also funded Hotmail, a web-based email service that was acquired by 
Microsoft on New Year's Eve in 1997.

Steve McGeown, director of product management at Sandvine, a Waterloo, 
Ontario-based company that makes equipment for broadband networks, says 
Draper's bet could pay off. McGeown's colleagues use Skype regularly when 
traveling and have been putting Skype numbers on their business cards-a sign of 
mass market appeal. "Skype doesn't have a mass market yet, but I did get my 
first business card with a Skype number on it," says McGeown.

Takeover Target?
-
What the future holds for Skype is unclear. Its options include:

Skype could emerge as a new communications platform that ties voice and video, 
not to mention millions of people together via handheld devices. Regardless, 
Dreze expects Skype to face increasing competition from the likes of Microsoft, 
Yahoo and Google. 

It could be acquired by an established communications or media company. Werbach 
says rumors about companies allegedly looking to buy Skype are a dime a dozen. 
Indeed, a report in CNET.com this week says that Rupert Murdoch's News Corp. 
allegedly discussed buying Skype for around $3 billion, but that talks broke 
down. The big question is valuing Skype. When Zennström was asked how much 
Skype is worth, the CEO replied, "We don't know. We're happy to keep this a 
private company."

The company could be pulled into a regulatory fray over issues such as 911 
calls as it grows into a de facto telecommunications company. 

Or it could remain an underground movement that continues to garner millions of 
users across the globe and is able to skirt regulatory concerns altogether. 

Another potential threat for Skype lies in cable and telecommunications 
companies that could block its use. However, Zennström says such a move would 
be counterproductive for the telecoms. "People use the Internet to get access 
to services," he notes. "If a telecommunications company is blocking access to 
those services it's telling people to go away." Given that fact, maybe it would 
make sense for a telecommunications carrier to simply buy Skype. "A takeover 
has potential, but the culture and the history of the company would be a 
challenge for most major telecom and media companies," says Werbach. "I would 
be surprised, but not shocked, if it was acquired." 

Seshu Kanuri


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RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

[Kanuri, Seshu \(Company IT\)]

> Skype will utilise PayPal for all payment systems.

What prevents Skype users using PayPal even without Ebay?

> Ebay doesn't have lots of users in territories that Skype does (Nordic
and South America for one).

Ebay is not interested in those territories anyway as they don't operate
Paypal in South America and even if someone wants to buy stuff there,
there is no way they can pay nor ship goods from US to Argentina.

>There'll be increased integration with Ebay and Skype for certain Ebay
sites.

There are opensource clients available to integarate, if that was the
reason. Azureus - BitTorrent Client, the most sought after Filesharing
software on Sourceforge that works on Linux could be a good candidate to
add Voice to it and make this a part of greater Ebay Solution.

But I agree that it is good for the Skype founders.

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Kennedy
Sent: Monday, September 12, 2005 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

On Mon, Sep 12, 2005 at 07:30:56PM +0100, Iqbal wrote:

> 70 million users, now how many of these are ALREADY ebay customers. 
> Google never made a succes out of any other thing other than search 
> and that will remain the case, companies never do, they are usually 
> good at what they started at, especially if they grow very big, then 
> they decided they may as well do everything, which is where their own 
> success starts to eat away at them.

Ebay doesn't have lots of users in territories that Skype does (Nordic
and South America for one).

Skype will utilise PayPal for all payment systems.

There'll be increased integration with Ebay and Skype for certain Ebay
sites.

Whether it's worth the money they paid, who knows? I'm sure the Skype
founders are happy, as are the investors who put in $35m.


Steve


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RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

[Kanuri, Seshu \(Company IT\)]

Ebay is seething with rage as yahoo took 40% stake in Alibaba in China.
They want to counter that by acquring a large user base in Asia, so that
their experience in Japan (Where ebay was kicked out completely by local
players) is not repeated in China and India.

This is an example of how you could make mistakes, if you don't keep
your cool and learn to count from 1 to 10 before you prss the trigger on
the gun.

Seshu Kanuri

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Monday, September 12, 2005 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

Seems to me that the intellectual property (the network + encryption +
client) is really the crown jewel here and an Ebay/Skype client is a no
brainer. In fact, I can see a scenario where they drop or severly
de-ephasize the voice part for their ends - maybe reduced to a "Powered
by Skype" blurb on the client skin (just like you see "Powered by AOL
search"
or other claptrap on CNN) 

The reality will be how well Ebay handles this. They will probably
bumble it. 

Funny thing is, for the $2.6 bil they spent for IP+client list, they
chould have designed and implemented their *own* stuff coupled with the
mother of all marketing campaigns (think global: tons of Skype users are
Asians - any Asian users on the list care to comment about how this is
being percieved in
Asia?) 

May You Live In Interesting Times.


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RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

[Kanuri, Seshu \(Company IT\)]

That theory sounds great on paper, but consider some of the factors
below.

1) On the web there are no loyalties. Today it is Skype but tomorrow
certainly belongs to GoogleTalk, if they can put Encryption. How
difficult is that to do? Whose quality is better? Skype or Googletalk? I
read that already 10% of the Skype users have migrated to Googletalk and
the numbers are soaring. Googletalk is still in Beta.

2)Paying users of payPal are hardly 5% of the 75 Million that they are
boasting of. And the numbers are coming down as PayPal has come to be
known as a 'PreyPal' of late. Many have stopped using this when the
charges have started ripping them off at 6% of the transfer amount for
International payments (even between canada and USA) and about 3.5%
within USA.  

3)What is the percentage of paying users of Skype? 1%  - 2%, may be.
These customers may already be Ebay's customers.

Ebay seems to be going like Time Warner buying AOL because it's user
base. There are several similarities in this deal.

Skype is cool, it has good stuff in it. How much effort does it take for
Google to add these features and make the googlers start using Google
Talk - a month or may be two.

My 0.02 cents

Seshu Kanuri


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Monday, September 12, 2005 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

While ostensibly people see this purchase as Ebay being a "me-too" VoIP
player with GoogleTalk and Microsoft buying, what's it called, Teleo? ,
I think there's a deeper plan. Let's look at the numbers:

-Huge installed base, like 30 to 70 million users (those are the numbers
I hear kicked around) -Secure, encrypted P2P client that works great
through firewalls -Name brand recognition. Even my mother in law has
heard of Skype and she is a techno-illiterate.
-Installed base loyalty. People who use Skype, love it. 
-Oh yeah, it does voice, too. (sarcasm on purpose: voice is only
partially the point)

So what Ebay could do with this is use the Skype client as an Ebay
portal that they control. Why, do you ask, would they want to do this?
Several
reasons:

-Break away from browsers with security flaws and inherent
social-engineering tactics like phishing. Nobody controls the WWW. Ebay
now controls the Skype network, controls the encryption, everything.
Less fraud for Ebay to deal with, assuming they repackage the client
into "Ebay 2.0 - download now for free!"

-Paypal integration - back to voice, wouldn't it be nice to pay for your
phone bill (a Skype account -> regular PSTN) through PayPal? 

-Repositioning of Ebay as an online solution provider once the "Ebay
Way" is fleshed out, much as Google has moved beyond search into "holy
crap" stuff like video on demand, mapping, etc. In the end, isn't it a
good business plan to become an indispensable, comprehensive middleman
instead of a one trick pony? Voice is just one aspect of it. 

-30-70 million users ain't bad. They may be Ebayers - they may not.
However, this gives Ebay a huge "installed base" that they can play with
before they bring new products and services to the general market.
Skype users are crazy about Skype, and they will download any client
that Ebay shoves down their throat, as long as it's free. Later, once
the service is worked out and debugged, *then* they will do
micropayments for the service (via PayPal, of course) 

still, 2.6 bil is a crazy price and I agree that they paid too much. I
would have loved to be a fly on the wall and see the other offers and
who was in the running. The Skype guys have made no secret that the
whole reason that they created Skype in the first place was to make
money and to be bought out. I'm sure there is champagne and hookers
galore at the Skype offices today. 



-Original Message-
From: Dean Collins [mailto:[EMAIL PROTECTED]
Sent: Monday, September 12, 2005 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion


This is the most insane decision taken by eBay ever.

If I was an investor I would be devastated.

There is nothing wrong with a company starting up a voice over ip
service but this isn't what eBay should be doing.

Companies need to understand they are stewards for their investors money
not the ultimate owners and that if they cant deliver ROI on this money
then it is their duty to return the money via dividends or share buy
backs.

eBay is insane and this is the beginning of the end for them.


Cheers,
Dean




> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of Cory Andrews
> Sent: Monday, 12 September 2005 1:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion
> 
> Good news for service providers in my opinion

RE: [Asterisk-Users] Skype purchased by Ebay 2.6 Billion

[Kanuri, Seshu \(Company IT\)]

As an ex 7th employee of PayPal before it was sold to ebay and having
gone through Ebay's carenot attitude to it's customers, let alone
investors, I concur with Dean's comments on ebay's future.

If you still have ebay's shares in your closet, get them out now.

Seshu Kanuri

 
Dean Collins Wrote:
-Original Message-
This is the most insane decision taken by eBay ever.

If I was an investor I would be devastated.

There is nothing wrong with a company starting up a voice over ip
service but this isn't what eBay should be doing.

Companies need to understand they are stewards for their investors money
not the ultimate owners and that if they cant deliver ROI on this money
then it is their duty to return the money via dividends or share buy
backs.

eBay is insane and this is the beginning of the end for them.

Cheers,
Dean


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RE: [Asterisk-Users] AGI + Ruby

[Kanuri, Seshu \(Company IT\)]

What can RAGI do additionally that AGI or FastAgi and DeadAgi cannot do
which is already available under Asterisk?

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of joe
heitzeberg
Sent: Sunday, September 11, 2005 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI + Ruby

Hi,

We have created RAGI (Ruby Asterisk Gateway Interface) for the open
source community so that Ruby and Ruby on Rails can be used to easily
and effeciently create Asterisk-based applications.  Examples:  IVR,
call center apps, Asterisk management consoles, etc.

RAGI includes a set of objects to interface over AGI to Asterisk for
handling inbound calls and outbound dialing, and includes a server
component, documentation and a sample apps to get you going quickly.

Please see: http://ragi.sourceforge.net/

The prelimenary release is available now on
https://sourceforge.net/projects/ragi

We welcome input and development participation in the effort.


thanks,
Joe Heitzeberg
SNAPVINE



On 8/24/05, Innocent Evil <[EMAIL PROTECTED]> wrote:
> I would like to write AGI script in Ruby Would anybody please show me 
> right direction..
> 
> 
> Thanks___
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RE: [Asterisk-Users] AMP 1.10.009 released!

[Kanuri, Seshu \(Company IT\)]

Good job Coalescent Systems. Good Job Jason Becker and Ryan Courtnage.
You guys rock. 

Your open source AMP GUI is more feature rich and much more desirable
than some closed source garbage out there in the market.

Keep up the goodwork.

Seshu Kanuri





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Courtnage
Sent: Friday, September 09, 2005 4:42 PM
To: asterisk-users@lists.digium.com;
[EMAIL PROTECTED]
Subject: [Asterisk-Users] AMP 1.10.009 released!

Hello all,

Asterisk Management Portal 1.10.009 has now been released.  This
exciting new version has several notable additions (listed below).

The AMP homepage is http://amp.coalescentsystems.ca.  Here you'll find
links to the download, install guide, and documentation wiki.

As usual, please use amportal-users mailing list for discussions about
AMP: https://sourceforge.net/mail/?group_id=121515


AMP 1.10.009 changes:

- Optional separation of Devices and Users.  Devices are endpoints (ie:
phones), and can be Fixed (to a user), or Adhoc.  Users are extensions,
with options like voicemail.  A user can log in to one or more Adhoc
devices by dialing *11, and log-off by dialing *12.

- "Custom" device technology support - this means devices that are not
configured directly in AMP's admin can still be used (ie: SCCP)

- Asterisk Recording Interface (ARI).  ARI is a php interface to
Voicemail and Monitor recordings. (written by littlejohnconsulting.com)

- RingGroups now use strategies: Ring All (default), Hunt, Memory Hunt

- DID Routes re-written as Inbound Routing.  This allows for DID
specific fax emails and call answering options.

- Queues can now play a "welcome" message to callers upon joining.

- HINT priorities for FIXED devices

- Interface translated to French, German, Italian, Spanish

- FOP .21

- FOP button layout can now be sorted by last name or extension number


Regards
--
Ryan Courtnage
Director & CTO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.2790
www.coalescentsystems.ca

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[Asterisk-Users] IP PBX Market Share and Growth

[Kanuri, Seshu \(Company IT\)]

Have a 
look at the article appearing in Globalsources, pasted below, that 
highlights the growing marketshare for IP PBX.
 
http://www.globalsources.com/gsol/GeneralManager?design=clean&language=en&page=showarticle&action="">
 
Seshu 
Kanuri
 



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RE: [Asterisk-Users] Oracle Realtime Driver and CDR Logger

[Kanuri, Seshu \(Company IT\)]

And to add to what Kevin said, we don't want any closed source stuff, be
it a database module or a device driver, to be a part of Asterisk as a
standard module, for obvious reasons.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, September 01, 2005 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Oracle Realtime Driver and CDR Logger

Chris Deserva wrote:

> I have written it in C++, because I used an OCI interface library 
> (ORAPP). I want to post it opensource so that I could get help in its 
> development and testing, and be a part of Asterisk modules.

You cannot make this open source. The Oracle client libraries are not
license-compatible with open source licenses, so it's not legal for you
to distribute code which links to them and is open source. Obviously
that also means it cannot be part of Asterisk as a standard module.


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RE: [Asterisk-Users] VoIP service recommendation

[Kanuri, Seshu \(Company IT\)]

I will not use any VOIP service that requires a large 
upfront payment, in this case a 1 year service charge


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Mag 
GamSent: Wednesday, August 31, 2005 10:57 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] VoIP service recommendation
Okay! What do you guys think about SunRocket (https://www.sunrocket.com/sign_up/linkshare_entry.do?plan=y&partner=ls&siteID=ydmf4rFDNTw-WD0UFulSHtqwjMnOUau4yg 
)Should I go for this?
On 8/31/05, Bjørn Ove 
Kristiansen <[EMAIL PROTECTED]> 
wrote:

  
  You can give www.broadvoice.com a try. 
  Only been using them for a few weeks, but their prices are 
  decent.
   
  Bjorn
   
  
  
  
  
  Fra: [EMAIL PROTECTED] [mailto: 
  [EMAIL PROTECTED]] På vegne av Mag GamSendt: 31. august 2005 15:17Til: asterisk-users@lists.digium.comEmne: [Asterisk-Users] VoIP service 
  recommendation
   
  I am planning to sign up for a VoIP service in the 
  U.S. Can anyone recommend anything cheap, reliable and good quality? I want to 
  use it for my primary house phone (I also own a cell phone). I also 
  want the service to be asterisk friendly so I can play with it 
  :-)Thanks in advance.
  --No 
  virus found in this incoming message.Checked by AVG 
  Anti-Virus.Version: 7.0.344 / Virus Database: 267.10.17/85 - Release Date: 
  30.08.2005
  --No virus found in this outgoing message.Checked by 
  AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.10.17/85 - Release 
  Date: 
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RE: [Asterisk-Users] (no subject)

[Kanuri, Seshu \(Company IT\)]

I use BINK to burn ISO Images and it works great.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, August 30, 2005 11:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] (no subject)

On Tue, Aug 30, 2005 at 05:27:37PM -0400, Mark Phillips wrote:
> Sounds to me like you copied the file to a disk rather than burn an 
> ISO image.  A common mistake folks make especially if they've never 
> done an iso before.

But then also wrote:

> 
> What tools are you using? I prefer k3b. It rocks

But also complicates the procedure when you want a simple ISO image
burning.
Hence the confusion with burning of the disk's files.

  cdrecord dev=whatever iso.image && eject

-- 
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http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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RE: [Asterisk-Users] IAX2 with g729 ATA Device

[Kanuri, Seshu \(Company IT\)]

Atcom's ATA AG168 can do this. Contact the US distributor
http://www.iareaphone.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Henderson
Sent: Monday, August 22, 2005 8:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 with g729 ATA Device

I am trying to find an ATA that will provice IAX2 and g729.  I have not
had much luck, I am hoping someone here might have some ideas.


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RE: [Asterisk-Users] [OT] Looking for Web based SIP endpoint

[Kanuri, Seshu \(Company IT\)]

Isnt Firefly and for that matter any other  IAX2 Softphone an IAX2
Endpoint in real sense? 

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Walsh
Sent: Saturday, August 20, 2005 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] [OT] Looking for Web based SIP endpoint

Matt Riddell [EMAIL PROTECTED] wrote:
> Jason Becker wrote:
> > https://sip-communicator.dev.java.net/
> > 
> > Don't know the current state of functionality with Asterisk. I 
> > couldn't get it to work many months ago - even with help from the
developer.
> >
> Any reason you are looking for SIP and not IAX?
>
Is there an IAX alternative that you'd recommend?

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RE: [Asterisk-Users] PhoneCALL v2.6.1 - Released

[Kanuri, Seshu \(Company IT\)]

Dustin,

It is pretty amazing, that you PhoneCALL has so many features
incorporated into a GUI of the tool, that needs little manual
modifications to the Asterisk config files.

I am sure that this will make all those closed source Commercial GUIs
redundant in near future.

Kudos and keep up the good work. 

Seshu Kanuri

Dustin Wildes wrote:
> Hello All!
> 
> Just a notice that our PHP/Smarty-based GPL version of PhoneCALL 
> version
> 2.6.1 has been released, and is the current stable release.
> http://www.vecsector.com/phonecall
> 
> We're always looking for feedback/testers to help us enhance it and 
> make it even easier for everyone to use.  The current version is 
> designed around the advanced Asterisk user, and we are working on a 
> more 'restrictive' model for different types of users in the system, 
> for
> example:
> 1)  User-based logins so users can control their phone options (like 
> DND, Call CellPhone, Text Message) or update their name, email
> 2)  Admin-based logins that control the general 'call flow' - but not 
> administer any of the scripts/macros and can only see the information 
> for the tenant they are assigned.
> 3)  Site-Admin has full access to all accounts/scripts, etc...  like 
> root account (current setup)
> 
> We're taking feature requests, and all feedback is welcome.
> Thanks!
> 
> 
> 
> Dustin Wildes
> VecSector, LLC
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-- 

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http://www.g7ltt.com
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RE: [Asterisk-Users] Features you'd like to see in a GUI?

[Kanuri, Seshu \(Company IT\)]

As this thread has come into the open, my suggestion is to get at least
5 references for Swithware and 5 references for Pbxware from Bicom
Systems, and speak to all of them and decide which way to go.

I can probably give a couple of references you can speak to, besides
myself on the usability of Pbxware and Switchware.

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, August 04, 2005 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Features you'd like to see in a GUI?

Considering I'm in the market to outfit 5 offices with an integrated
phone system, I'm interested in hearing a response from Bicom on these
topics.  I like the "look" of PBXware.  I may be interested in getting
more technical details.  I like the Bicom business model better than the
Fonality business model.

One major issue for me as the end user is that I already have 5 Polycom
phones.  Does PBWware support Polycom phones yet?


At 12:34 PM 8/4/2005, you wrote:
>Senad,
>
>I don't want to take this conversation much further and send a laundry
>list of issues we faced with Switchware to this forum, and the myriads
>of bugs we have wrestled with in the past 18 months, as that list is
too
>long and this forum is not for that purpose.
>
> >In addition, our clients we deal on regular basis do
> > not "disappear". Each and every client is looked after by us.
> > They do not, "install" the software try it and then come back
> > after several weeks/months and start the install process all
> > over again... That is not a workable model for anyone.
>
>The answer to your above comments is already known to you and it is
>already mentioned in Para-1. As you already knew, the two install
>efforts by me, one in September-October 2004 and  One in
>November-December 2004 have failed to install a working software. On
>several occasions when I have reverted back to you on giving me a
>working version, your answer ( as well as Stephen Wingfield's ) was
that
>Bicom don't have one ready.
>
>Infact, as I remember this correctly, Switchware was taken out of your
>product list due to these issues. Moreover is it ethical or legitimate
>to sell a software that has to be installed only by your technical team
>and works only on a specific hardware for a given network card and for
a
>given IP. You don't even have a tarball to download and install till
>today, let alone a CD image?. This is not the way products are sold,
>where everything is closed, even the features of Asterisk that are
>available otherwise.
>
> > Our "installed" clients certainly do not try to look
> > into our source code and then say "it is not working".
>
>I don't understand what you mean by looking into your source code, but
>if you mean that "once Switchware is installed, live with what Bicom
>allows you to do with Asterisk, and dont ask questions like 'Why Cannot
>I use  4 Digit extensions' or 'Why cant I use ASTCC with Switchware' or
>'Why cannot I create more than X number of channels'" or millions of
>other such questions, my answer is that this is totally out of line
with
>the purpose of Asterisk as an Open Source software and if Digium knew
>what is going on, probably you are going to have legal issues as far as
>the GPL of the code is concerned, in the same way Sysmaster did with
>their SM7000 products.
>
>Your Channel Locking and per channel pricing poilicy I am sure will put
>Oracle and Larry Ellsion to shame.
>
>I feel that Bicom should have spent more time making Switchware or
>PBXware work cleanly, rather than spending most of the time in
>copyprotecting and closed sourcing Asterisk so that if the customer has
>to move the installation to another server, they have to call you for a
>new license code and pay you for the installation, rather than doing it
>himself.
>
>This is against the open source philosophy we are all trying to benefit
>from.
>
>Seshu Kanuri

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RE: [Asterisk-Users] Features you'd like to see in a GUI?

[Kanuri, Seshu \(Company IT\)]

Senad,

I don't want to take this conversation much further and send a laundry
list of issues we faced with Switchware to this forum, and the myriads
of bugs we have wrestled with in the past 18 months, as that list is too
long and this forum is not for that purpose. 

>In addition, our clients we deal on regular basis do 
> not "disappear". Each and every client is looked after by us. 
> They do not, "install" the software try it and then come back 
> after several weeks/months and start the install process all 
> over again... That is not a workable model for anyone.

The answer to your above comments is already known to you and it is
already mentioned in Para-1. As you already knew, the two install
efforts by me, one in September-October 2004 and  One in
November-December 2004 have failed to install a working software. On
several occasions when I have reverted back to you on giving me a
working version, your answer ( as well as Stephen Wingfield's ) was that
Bicom don't have one ready. 

Infact, as I remember this correctly, Switchware was taken out of your
product list due to these issues. Moreover is it ethical or legitimate
to sell a software that has to be installed only by your technical team
and works only on a specific hardware for a given network card and for a
given IP. You don't even have a tarball to download and install till
today, let alone a CD image?. This is not the way products are sold,
where everything is closed, even the features of Asterisk that are
available otherwise.

> Our "installed" clients certainly do not try to look 
> into our source code and then say "it is not working".

I don't understand what you mean by looking into your source code, but
if you mean that "once Switchware is installed, live with what Bicom
allows you to do with Asterisk, and dont ask questions like 'Why Cannot
I use  4 Digit extensions' or 'Why cant I use ASTCC with Switchware' or
'Why cannot I create more than X number of channels'" or millions of
other such questions, my answer is that this is totally out of line with
the purpose of Asterisk as an Open Source software and if Digium knew
what is going on, probably you are going to have legal issues as far as
the GPL of the code is concerned, in the same way Sysmaster did with
their SM7000 products.

Your Channel Locking and per channel pricing poilicy I am sure will put
Oracle and Larry Ellsion to shame.

I feel that Bicom should have spent more time making Switchware or
PBXware work cleanly, rather than spending most of the time in
copyprotecting and closed sourcing Asterisk so that if the customer has
to move the installation to another server, they have to call you for a
new license code and pay you for the installation, rather than doing it
himself. 

This is against the open source philosophy we are all trying to benefit
from.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad J
Sent: Thursday, August 04, 2005 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Features you'd like to see in a GUI?

> If your intention is just to build a GUI for Asterisk, read no 
> further.
> If your desire is to build something more purposeful, your best bet 
> would be to see the existing commercial GUI/HostedPBX offerings like 
> Pbxware and Switchware from bicomsystems.com ( 
> http://www.bicomsystems.com) and Thirdlane Technologies 
> (http://www.thirdlane.com/opensource.htm)
> and
> the Open Source software like AMP and try to emulate (or preferably 
> improve upon) them.
>
> My suggestion is to create a "VOIP Business in a Box System" that has 
> inter-alia following list of modules:
>
> 1) GUI To configure Administer Asterisk Extensions across many servers
> 2) Postpaid and Prepaid Billing modules with realtime call progress 
> detection and call cut-off 3)CRM Module for customers to register and 
> provide their information for recurring billing.
> 4)Web based conference room management module 5)Web based click to 
> dial and callback module
>
> Many of these modules are already available on Open Source like 
> SugarCRM, AsreskiCC etc., and [EMAIL PROTECTED] CD contains AMP and 
> SugarCRM at this time, besides other Open Source utilities like 
> PhpMyAdmin.
>
> Here is the bottomline:
> --
> The real need is for a commercially deployable solution that can 
> create a business, without too many additions to it.
>
> Bicom Systems has promised for too long that their Pbxware and 
> Switchware can fullfill this need to create a business, but they never

> deliverd their promise. PBXware and Switchware have been a total and 
> expensive disappointment to me and for the few who invested in them.


I would not normally reply on a post like this, but since you are
publicly expressing your opinion on products and services that we offer,
I must defend our years of work.

We have PBXware and SWITCHware deployed at many locations around the

RE: [Asterisk-Users] Features you'd like to see in a GUI?

[Kanuri, Seshu \(Company IT\)]

Sherwood,

Your intentions are noble and your desire to build this, fullfills an
immediate need for business.

If your intention is just to build a GUI for Asterisk, read no further.
If your desire is to build something more purposeful, your best bet
would be to see the existing commercial GUI/HostedPBX offerings like
Pbxware and Switchware from bicomsystems.com 
( http://www.bicomsystems.com) 
and Thirdlane Technologies (http://www.thirdlane.com/opensource.htm) and
the Open Source software like AMP and try to emulate (or preferably
improve upon) them.

My suggestion is to create a "VOIP Business in a Box System" that has
inter-alia following list of modules:

1) GUI To configure Administer Asterisk Extensions across many servers
2) Postpaid and Prepaid Billing modules with realtime call progress
detection and call cut-off
3)CRM Module for customers to register and provide their information for
recurring billing.
4)Web based conference room management module
5)Web based click to dial and callback module

Many of these modules are already available on Open Source like
SugarCRM, AsreskiCC etc., and [EMAIL PROTECTED] CD contains AMP and SugarCRM
at this time, besides other Open Source utilities like PhpMyAdmin.

Here is the bottomline:
--
The real need is for a commercially deployable solution that can create
a business, without too many additions to it.

Bicom Systems has promised for too long that their Pbxware and
Switchware can fullfill this need to create a business, but they never
deliverd their promise. PBXware and Switchware have been a total and
expensive disappointment to me and for the few who invested in them.

The story is similar with Thirdlane Technologies who promised a good
Asterisk Management Interface but they have not been able to deliver a
mature GUI to manage Asterisk, let alone deploying a Hosted PBX.

AMP fullfills this promise partially and I am impressed with it's
richness of features, to create a Open Source Hosted PBX where you don't
need to bill the customer.

AreskiCC Provides a module to run a Calling Card Business but not as a
Hosted PBX .

AMP Gives a nice Gui and can be used as a Corporate PBX but not as a
Commercial Hosted PBX as it does not have a billing system.

In summary a system that is a blend of AMP + AreskiCC + SugarCRM would
be a good mix of ingredients to build what I would call a "Business in a
Box" solution. There is money to make in such solutions.

The Business in a box solution could be any one of the following, with
the ingredient modules being varied.

1)Asterisk Calling Card Business in a Box
2)Asterisk Call Center Business in a Box
3)Asterisk Business Messaging Business in a Box
4)Asterisk Hosted Telephone and PBX business in a Box
  Etc etc.

My .02 cents on this.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Thursday, August 04, 2005 9:02 AM
To: 'Matt Florell'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Features you'd like to see in a GUI?


->Is this going to be a configuration utility or an end-user utility?

Both, there will eventually be functionality for per-user permissions,
however right now this is for the administrator to add/remove "accounts"
in SIP and IAX, eventually other trunks, voicemail configurations, per
user configuration of options, etc...think Asterisk @ Home's combination
of tools, all in one place. 

Perhaps it would be better to show what's going on already, here's what
I have currently working in my employer's system (an ITSP):

Editing of individual extensions/trunks in SIP and IAX Voicemail Account
/ Option Editing CDR Searching Provisioning File Editing for devices
User configuration editing (extra table setup for Account options)
Account Management (Using SIP for end user accounts, interface merges
voicemail, sip, provisioning, and user options on one page) System
Statistics (concurrent calls, channels, memory usage, load, etc...)

Current future plans:
Specific SIP/IAX Channel Information
Allowing dropping of a channel (this is due to the occasional need for
dropping hung channels, which can really rack up fees on a system) SIP
PRUNE REALTIME PEER  SIP SHOW PEER  [load] The above two
options for IAX as well.
Voicemail Greetings webinterface (not hard, just not finished yet)
Voicemail Webinterface (really fairly easy, just not done)
Extensions.conf editing Logging Interface Log viewing & Management


As you can see, this is a fairly indepth project. Until I actually had
to use Asterisk on a large scale (1000+ concurrent users), I hadn't
thought of a lot of these features. The more I use it the more I come
across little things that would be really useful. Thanks to Digium for
offering the commandline option "asterisk -rx commandtoexecute"!

Aside from these planned features, are there any other arcane options
that you'd like to see? I'm definitely trying to get as many of the CLI
c

RE: [Asterisk-Users] Full T38 sip Faxing now Available

[Kanuri, Seshu \(Company IT\)]

Michael,

Here are some of the reactions to your original post on the T38 FAX
thingy:

1) Why the big secret?  Why not post your solution to the list?

2) It's probably just another one of those nasty closed source add-ons
for sale.

3) I'm guessing it has nothing to do with *. Probably MAX TNT, Cisco, or
some other soft-switch. In other words, nothing new. 

You have not answered any of those questions, but here you are accusing
someone as un-professional. Why don't you answer their questions first
and then we all can decide if your post was a spam or not.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Wednesday, August 03, 2005 1:47 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Full T38 sip Faxing now Available

Why do you put me down? I have not done a thing to you and I'm not a
spammer. Please stop this activity It's not professional. If I were to
give you bad service please feel free to comment negatively but I've
never dealt with you nor do you have an account with us.

Sincerely

Michael D. Schelin


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[Asterisk-Users] Oh323 Module - Not Loading Error - Unregistered channel type 'Modem'

[Kanuri, Seshu \(Company IT\)]

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk.

Oh323 Module compiled without errors. But When I try to stary Asterisk
with the Oh323.so file in the modules folder, Asterisk is dying with the
following error.


 [chan_oh323.so]Aug  2 14:08:14 NOTICE[18873]: res_musiconhold.c:490
monmp3thread: Request to schedule in the past?!?!
 => (InAccess Networks OpenH323 Channel Driver)
  == Parsing '/etc/asterisk/rtp.conf': Found
  == Parsing '/etc/asterisk/oh323.conf': Found Aug  2 14:08:15
NOTICE[18873]: chan_oh323.c:4855 reload_config: Ignoring unknown H.323
[general] keyword 'connectPort', line 25.
Aug  2 14:08:15 NOTICE[18873]: chan_oh323.c:4855 reload_config: Ignoring
unknown H.323 [general] keyword 'silenceSuppression', line 58.
Aug  2 14:08:15 NOTICE[18873]: res_musiconhold.c:490 monmp3thread:
Request to schedule in the past?!?!
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5, PWlib v1.6.6
[1]WrapGatekeeperServer::WrapGatekeeperServer: Creating new gatekeeper.
Aug  2 14:08:15 ERROR[18873]: chan_oh323.c:5378 load_module: H.323
listener creation failed.
  == Unregistered channel type 'Modem'
Aug  2 14:08:15 WARNING[18873]: loader.c:403 __load_resource:
chan_oh323.so: load_module failed, returning -1
  == Cleaning up OpenH323 channel driver.
Aug  2 14:08:15 NOTICE[18873]: res_musiconhold.c:490 monmp3thread:
Request to schedule in the past?!?!
[1]WrapGatekeeperServer::WrapGatekeeperServer: Destroying gatekeeper.
Aug  2 14:08:15 WARNING[18873]: loader.c:543 load_modules: Loading
module chan_oh323.so failed!



I would appreciate if somebody can give a clue to fix this error.

Seshu Kanuri


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RE: [Asterisk-Users] Asterisk PSTN connectivity

[Kanuri, Seshu \(Company IT\)]

Use Google extensively and the WIKI site here  http://www.voip-info.org/wiki-Asterisk , 
till you become familiar with the architecture of Asterisk. probably for a 
couple of months. 
 
You can come back here if you still have any questions 
at that time and all the member here would be happy to answer all your 
questions.
 
Seshu



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Nil 
SSent: Tuesday, August 02, 2005 5:07 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk 
PSTN connectivity

Hello Everybody,
 
I am a new user in this group.
I have installed asterisk on my test linux machine and setup the call 
from one asterisk user to another asterisk user successfully. It is working 
great.
 
Now i want to setup the call from one asterisk user to any PSTN user in the 
world or vice versa. Could you please help me out in this?
Please guide me how to do this as I am  completely unaware of this 
Asteris PSTN connectivity.
Please suggest me some configuration steps also.
 
Waiting for positive reply.
 
Thanks.Nil
 
 
 


Do you Yahoo!?Yahoo! 
Mail - You care about security. So do we.



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[Asterisk-Users] IAX Devices Recommendation

[Kanuri, Seshu \(Company IT\)]

With due respect to Digium and Mark Spencer and the greatest protocol he
defined, I have used IAXY and I regret to say that IAXY at $99 is plain
garbage compared to the $49 ATA made by ATCOM. 

Try the ATCOM AG168  sold as ATA-100 by iareaphone.com. This has an
additional lifeline port and gives the best bang for the buck.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Graham
Pearson
Sent: Monday, August 01, 2005 3:07 PM
To: Asterisk Users
Subject: [Asterisk-Users] IAX Devices Recommendation

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure through various routers than a SIP
Device. I just started to look at the Digium IAXy Single FXS Adapter but
unable to find a Telephone that supports the IAX Protocol. Any
Recommendations or is the Digium FXS Adapter the way to go.


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[Asterisk-Users] IAX Devices Recommendation

[Kanuri, Seshu \(Company IT\)]

Graham,

Digium IAX2 FXS unit called IAXY is just no good. I would say that it is
garbage. 

Try the IAX2 ATA ( AG168 sold as Netweb ATA-100) with a life line port
made by Atcom and available from http://www.iareaphone.com

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Graham
Pearson
Sent: Monday, August 01, 2005 3:07 PM
To: Asterisk Users
Subject: [Asterisk-Users] IAX Devices Recommendation

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure through various routers than a SIP
Device. I just started to look at the Digium IAXy Single FXS Adapter but
unable to find a Telephone that supports the IAX Protocol. Any
Recommendations or is the Digium FXS Adapter the way to go.


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RE: [Asterisk-Users] Re: IAX Devices Recommendation

[Kanuri, Seshu \(Company IT\)]

You may have bought the Chinese Versions and hence the problem in slow
response.

Have you tried the US versions available from http://www.iareaphone.com
?

-S


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Redstone
Sent: Tuesday, August 02, 2005 12:32 AM
To: Asterisk User
Subject: [Asterisk-Users] Re: IAX Devices Recommendation

Hi

We purchased the AT320-EE IAXtalk phone from www.iaxtalk.com which
ocnnects to our own asterisk server.

Good value, a little tricky to set up - the instructions they supply to
which they give you a link on their web site are OK, but their are some
gaps which the asterisk wiki pages fill well - cannot find this at the
moment but it explains how to do resets.

IN summary you buy the phone and then upload the firmware for IAX2
protocol. 
Configuration is via web browser which works well. Automaticlaly logs
in.

Works well. Slightly slower to respond than (say)  firefly softphone
which we use for most users - the hardphone is for reception and as
backup in case of computer failure.

Paul Redstone
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RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

[Kanuri, Seshu (Company IT)]

The download link is in the url pasted in the email.

You can test it from here. Click on the first link:

http://www.geocities.com/babarnazmi/

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M
Sent: Thursday, July 28, 2005 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to
Call)

On Thu, 2005-07-28 at 10:48 -0400, Kanuri, Seshu (Company IT) wrote:
> Try babar nazmi's IAX web phone. This does not have G729 or G723 but 
> it has high bit rate codecs.
>  
> http://www.geocities.com/babarnazmi/


Have you the url where can I download it?

I need to test it.

>  
> We at iareanet use this product as part of our virtual office solution
where remote customers dial in and dial out using the IAX SoftPhone.
>  
> Seshu
> 
> 
> 
> __
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Walid 
> Azab
> Sent: Thursday, July 28, 2005 10:07 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to
> Call)
> 
> 
> Hi,
>  
> I appreciate it if someone knows what is available for SIP web phones 
> out there. I am interested in putting a soft phone on a website that 
> registers with Asterisk using SIP. Then, when someone uses it, it 
> directly calls into an asterisk call queue..
>  
>  
> Any ideas?
> 
> __
> 
> NOTICE: If received in error, please destroy and notify sender.
> Sender does not waive confidentiality or privilege, and use is 
> prohibited.
> 
> 
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

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RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

[Kanuri, Seshu (Company IT)]

Try babar nazmi's IAX web phone. This does not have 
G729 or G723 but it has high bit rate codecs.
 
http://www.geocities.com/babarnazmi/
 
We at iareanet use this product as part of our virtual office solution where remote customers dial in and dial out 
using the IAX SoftPhone.
 
Seshu



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Walid 
AzabSent: Thursday, July 28, 2005 10:07 AMTo: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] SIP WEB Phone (Wanna implement Click to 
Call)

Hi,
 
I appreciate it if 
someone knows what is available for SIP web phones out there. I am interested in 
putting a soft phone on a website that registers with Asterisk using SIP. Then, 
when someone uses it, it directly calls into an asterisk call queue.. 

 
 
Any 
ideas?



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[Asterisk-Users] H323 Configuration file

[Kanuri, Seshu (Company IT)]

Folks!

I would appreciate if someone could send me a simple working h323
configuration file oh323.conf that is part of [EMAIL PROTECTED]
installation.

I have tried to use the oh323.conf content listed on WIKI but it is just
not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot
register. I need a working example of this file for similar phone.

Seshu


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RE: [Asterisk-Users] Soft Phone

[Kanuri, Seshu (Company IT)]

Title: Soft Phone




Firefly Third Party version beats all others. 



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave 
MorrowSent: Friday, July 22, 2005 4:12 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Soft Phone

Can anyone recommend a good soft phone that's easy to 
configure under Asterisk and works well on a typical Windows XP 
system?
David A. Morrow Technical Systems Lead Autodata 
Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: 
(519) 951-6079 Fax: (519) 
451-6615  
< Poor planning on your part does 
not necessarily constitute an emergency on my part! > 
This message has originated from Autodata Solutions. 
The attached material is the Confidential and Proprietary Information of 
Autodata Solutions. This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or entity to whom 
they are addressed. If you have received this email in error please delete this 
message and notify the Autodata system administrator at [EMAIL PROTECTED] 



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[Asterisk-Users] Freshtel.net - Spamming?

[Kanuri, Seshu (Company IT)]

I agree with Brian! Robert's post is off topic or 
may be just a marketing effort, to push their site.
 
Anyone who wants freshtel.net for US/Canada calling 
at 6.9 Cents a minute, raise their hands?

...

I see none

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Wednesday, July 20, 2005 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Firefly 3rd party - it hangs on
"Initialising"and exits with error

Robert Webb wrote:
>>
> 
> 
> Yeah... Try the web site for the writers of Firefly:
> 
> http://www.freshtel.net/
> 
> This is an Asterisk USer list. Not a Firefly list.

And please note that in general members of the list dislike List Police
even more than they do off-topic posters.

B.


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RE: [Asterisk-Users] Best VoIP provider

[Kanuri, Seshu (Company IT)]

What happens if you never received the first email from Nufone when you
signup? Is there someway to get this information from the web site? I
don't even see a download area for such information.

Can someone please send me nufone server address off list?

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent: Tuesday, July 19, 2005 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Best VoIP provider

rofl.. 
nufone sends you configuration information via email after you sign up
for an account..


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RE: [Asterisk-Users] SIP Phones with Asterisk

[Kanuri, Seshu (Company IT)]

What is the Model you have? ML220(newer model, Supports SIP and H323) or  
ML210A (older model, supports only H323 I guess)?

The manual at the link below is self explanatory. Just provide your Server IP, 
Account and Pin/Password for your Asterisk Box and you are ready to go.

Seshu


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francisco Paulo 
Mateus Nascimento Adriano
Sent: Tuesday, July 19, 2005 10:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Phones with Asterisk

Hi,

I have a bunch of NetPhones that I have bought from MeritCall some time ago for 
their service. How can I use this phones (supposed SIP phones) to integrate 
with a Asterisk Setup.

I have seen a manual for a similar one but I don´t know If mine are hardcoded 
in some way. This devices are used by MeritCall , MamaKall, Vivophone etc.

I found the manual for an exact same type here:

http://www.konceptusa.com/downloads/Netphone-KE1020A_KE1021A%20User%20Manual.pdf#search='netphone%20configuration%20for%20sip'


Thanks in advance,

Francisco
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RE: [Asterisk-Users] Best VoIP provider

[Kanuri, Seshu (Company IT)]

It does not look like Nufone is still in business, judging from the
content on their site, which is very little. There is not even a
configuration document to download, to connect to their network.

The rates file is only for US/Canada calling. No international 
rates on this rates.csv file.

I have signed up with a $5.00 account with them way back in November
2004. After signup, I havent received any email or anything of that
sort, explaining to me how to connect to their network. 

The only email address I see on their site is [EMAIL PROTECTED], there is
no support related contact information on the site, which does not
inspire much confidence.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Tuesday, July 19, 2005 12:33 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Best VoIP provider

That's odd -- they used to be here: http://www.nufone.net/rates.csv

Of course, you can't rely on that.

> -Original Message-
> From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 19, 2005 6:13 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Best VoIP provider
> 
> 
> Madhawa Jayanath wrote:
> 
> > o Bernie,
> > 1) best results www.nufone.net
> > 2) low cost www.voipjet.com
> 
> Anyone able to find NuFone's rates? I have been looking for them on 
> their site. I need international rates and UK Mobile.
> 
> --
> Chris Mason
> NetConcepts
> (264) 497-5670 Fax: (264) 497-8463
> Int:  (305) 704-7249 Fax: (815)301-9759
> Cell: 264-235-5670
> Yahoo IM: [EMAIL PROTECTED]


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RE: [Asterisk-Users] [Aserisk-Users]no audio inside the net

[Kanuri, Seshu (Company IT)]

1) reinvite=yes is incorrect syntax? Check the info here:
http://voip-info.org/wiki-Asterisk+sip+canreinvite

You can keep canrenvite=yes, but why do you want that?

;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go
directly from
; the caller to the callee.  Some
devices do not
; support this (especially if one of
them is
; behind a NAT). So use canreinvite=no

2) Use nat=yes or nat=auto for correct evaluation of NAT by Asterisk.
 
3)  qualify=yes may be used as qualify=800

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sistemista
WebSolvingJaa
Sent: Friday, July 15, 2005 12:08 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [Aserisk-Users]no audio inside the net

Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i don't
know its brand. with this sip.conf :

[general]
 port = 5060
 bindaddr = 192.168.100.229
 context =  default ;x changed from default to sip  localnet =
192.168.100.0/24  srvlookup = yes  allow=all


 [2001] ;grandstream 2
 type=friend
 username=2001
 secret=1945
 canreinvite=yes
 reinvite=yes
 host=dynamic
 dtmfmode=rfc2833
 qualify=yes
 ;mailbox=2001
 nat=1
 allow=all

 [2002] ; soft phone
 type=friend
 username=2002
 secret=1945
 canreinvite=yes
 reinvite=yes
 host=dynamic
 dtmfmode=rfc2833
 qualify=200
 mailbox=2002
 nat=1
 allow=all

 [2010]; wi-fi phone
 type=friend
 username=2010
 secret=1945
 nat=1
 host=dynamic
 dtmfmode=rfc2833
 canreinvite=yes
 reinvite=yes
 qualify=200
 allow=all

 [2011] ; ip-phone no brand
 type=friend
 username=2011
 secret=1945
 nat=1
 host=dynamic
 dtmfmode=rfc2833
 canreinvite=yes
 reinvite=yes
 qualify=yes
 allow=all

 [2012] ;grandstream1
 type=friend
 username=2012
 secret=1945
 nat=1
 host=dynamic
 dtmfmode=rfc2833
 canreinvite=yes
 reinvite=yes
 qualify=yes
 allow=all

*

and with this extensions.conf file:

[general]

static=yes
writeprotect=yes
autofallthrough=yes

[globals]
CONSOLE=Console/dsp ; Console interface for
demo
CONSOLE=Zap/1
CONSOLE=Phone/phone0
IAXINFO=guest   ; IAXtel
username/password
TRUNK=Zap/g2; Trunk interface

TRUNKMSD=1  ; MSD digits to strip
(usually 1 or 0)

[dundi-e164-local]

include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
switch => DUNDi/e164

[dundi-e164-lookup]

include => dundi-e164-local
include => dundi-e164-switch

[macro-dundi-e164]

exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

[iaxtel700]
exten =>
_91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])

[trunkint]

exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld]

exten => _91NXXNXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunklocal]

exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunktollfree]

exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[international]

ignorepat => 9
include => longdistance
include => trunkint

[longdistance]

ignorepat => 9
include => local
include => trunkld

[local]

ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider


[macro-stdexten];

exten => s,1,Dial(${ARG2},20)   ; Ring
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1})   ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start

exten => s-BUSY,1,Voicemail(b${ARG1})   ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)  ;
Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain

[demo]

exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,SetVar(TIMEOUT(digit)=5)

RE: [Asterisk-Users] OT: DS3 -> VoIP Hardware Recommendations

[Kanuri, Seshu (Company IT)]

TNT is the worst piece of garbage that has ever been sold in the name of
a VOIP Switch.
This stuff is avialble on ebay for a fraction of it's sticker price, if
you dare to bid on it.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Wednesday, July 13, 2005 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: DS3 -> VoIP Hardware Recommendations

Tom wrote:
> At 10:06 AM 7/13/2005, you wrote:
> 
>> Hello all,
>>  We are looking for some hardware requirements/recommendations to be 
>> able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 
>> would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We 
>> would then need to convert those calls into G729 SIP VoIP calls to 
>> send to our asterisk box over ethernet. Since everything is going 
>> in/out of asterisk is 729, and no features are needed, I think it can

>> handle the routing. If not, I can whip up a SER box.
>>
>>  We currently have a Cisco 7206VXR (1 voice resource) and a Cisco 
>> AS5300 (120 voice resources). The DS3 will also have SS7 signaling on
it.
>>
>> Recommendations/comments/concerns/rants are graciously welcomed.
> 
> 
> Lucent TNT
> 

Lucent's goto sleep after 5 min of no activity. At least the one they
had here did. They have boycotted them.

But thanks for suggestion.

-Matthew


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RE: [Asterisk-Users] SIP PHONE

[Kanuri, Seshu (Company IT)]

Try www.SIPphone.com or www.terracall.com
 
Seshu


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Ellafi 
FituriSent: Tuesday, July 05, 2005 2:07 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] SIP PHONE



  Hi All,
   
   
  I just got a SIP phone and I would like to know where I could find 
  service?
  Please helpThank you very much for your help


Yahoo! SportsRekindle 
the Rivalries. Sign up for Fantasy Football 



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RE: [Asterisk-Users] wi-fi phone advice

[Kanuri, Seshu (Company IT)]

The price of $39 for a WIFI SIP Phone sounds goofy to me. 
I am not sure if you will be able to buy anything that is 
close to a working wired SIP Phone, let alone a WIFI one.

I have a couple of Zyxel phones and they cost 5 times this price.

It looks like the makers of Hop1502 are trying to 
generate some prospective interest in their phone through 
a "creative advertising".

Seshu

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of chawki
hammoud
Sent: Thursday, June 30, 2005 4:35 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] wi-fi phone advice

Hi:

I want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so
I can make voip calls.
 
please send me your recomendation about what wi-fi phone I should be
looking for. Anybody tried the
HOP1502 Wi-Fi IP phone. Its listed price $39.

Regards;
Chawki




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RE: [Asterisk-Users] WiFi IP Phones

[Kanuri, Seshu (Company IT)]

How About "Pro Reliable" before we look for "Anti Explosive" WIFI
phones. 
Has anyone got recommendations on WIFI Phones that work with Asterisk?

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Wednesday, June 15, 2005 4:53 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] WiFi IP Phones

I know... The term "anti explosive" is new to me.. I never heard of it
but a possible client is asking for that exactly since the phones are
going to be used in oil refinary and r&d platforms using voip over
satelite connections...

What do you think?

BTW, how is voip over satelite? I know you have the usual 500 ms lag for
up and down stream delays but hows quality?


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RE: [Asterisk-Users] Asterisk connecting remote villages in westernUganda

[Kanuri, Seshu (Company IT)]

Mark,

This is a wonderful thing to do for underserved societies like Uganda.

The datasheet you have provided and the layout could be the model for
many other developing societies both In Africa as well as central and
South America.

Kudos to Inveneo.org under your able leadership. Keep up the good work.

Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Summer
Sent: Monday, June 13, 2005 1:56 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Asterisk connecting remote villages in
westernUganda

Hi,

I though some of you on this list might be interested in what Inveneo is
doing in Uganda. We are a San Francisco based non-profit organization
that builds rugged, low-cost, highly reliable and open- source
communications systems for under-served communities around the world. We
have just completed our first installation in western Uganda, Africa.

The system is up and running since this past Wednesday (June 8th). We
have installed 5 units, 4 of which are in villages with with no access
to power. The system provides Internet access and phone capabilities to
the users. Phone calls among the connected villages are free of charge,
with the ability to place and receive calls to / from  the Ugandan phone
network and voice mail boxes for each station. The systems are linked
using 802.11 WiFi links.

For more information please have a look at the following links:

For more detailed information and pictures of the Uganda deployment:

http://www.inveneo.org/?q=uganda

For more information about the solution we have built and implemented,
here is a link to our PDF datasheet:

http://www.inveneo.org/download/inveneoDatasheet.pdf

And of course our website:

http://www.inveneo.org/


Thank you!

Mark


Mark Summer
co-founder, Inveneo
web:   http://www.inveneo.org
phone: +1-415-901-1969 x 1200
FWD:   603303
cell:  +1-415-867-9751
email: [EMAIL PROTECTED]

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RE: [Asterisk-Users] Re: POLYCOM IP 500 Setup

[Kanuri, Seshu (Company IT)]

[EMAIL PROTECTED] will 
not be able to configure polycom500 phones.
 
You need to add this entry in sip.conf manually with 
one additional line as under:
 
progressinband=no
 
Seshu
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Noah 
MillerSent: Monday, June 13, 2005 9:44 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: POLYCOM 
IP 500 Setup


Hi Matt -


  Hello, I just wiped out my old asterisk install and installed 
  Asterisk 
  at Home.  I was quickly 
  able to get my Digium TDM422P working, 2 POTS 
  lines, 2 phones.  I also 
  got X-Lite working as a SIP extension.  
  I then 
  tried to setup my Polycom IP 500, and this was not so 
easy...
  
  Using AMP I created SIP extension 205 to be used with my Polycom phone. 
   
  I setup username = 205, secret = 123, context = 
  from-internal.
  
  I setup my phone to have a static IP address, then pointed my web 
  browser at it, to setup my phone.
  I setup Sip Conf with: Address = "IP of * server",  Server1 = "IP of * 
  Server"
  Under Registration, I setup: Identification: Address = "IP of * 
  Server" 
  , Auth User ID = 205,  
  Auth Password = 123, Server1: Address = "IP of * 
  server"

For your phone-specific file, address isn't the asterisk address, it is the 
sip address of the phone - you can just use "205".


- Noah








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[Asterisk-Users] Unable to register Zyxel WIFI Phone as SIP Client to Asterisk

[Kanuri, Seshu (Company IT)]

Folks!
 
I have this 
expensive gizmo Zyxel-2000 WIFI  Wireless Phone that can run SIP 
protocol. 
 
I have configured 
this to my Asterisk as a SIP client but cannot register at the 
server.
 
I have a basic 
configuration entry in sip.conf  and I am running it having the client 
connected 
with a Dynamic DHCP 
address. My Asterisk server is running fine and it has several SIP and IAX2 
clients. No problem there
 
I have used the 
following options in sip.conf as trial and error in various 
combinations
 
nat=yes
host=dynamic
canreinvite=no
defaulthost=xx.xx.xx.xx
 
Asterisk sees the 
phone trying to connect but it cannot authenticate with the 
Login/Pass
 
Does anyone have a 
working configuration for the Phone as well as sip.conf entry? 

 
If not any 
suggestions
 
Thanks
 
Seshu 
KanuriMorgan Stanley | Technology1633 
Broadway | Floor 19New York, NY 10019Phone: +1 212 537-2849[EMAIL PROTECTED]
 




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RE: [Asterisk-Users] Asterisk Live! CF

[Kanuri, Seshu (Company IT)]

Abel,

I am working on Intel boards only.

I have tried VIA boards and I do not recommend anyone to work on VIA
boards for a production system. The reasons for this being that there
are just way too many issues with these boards, gcc being just one of
them. The main issue is Interrupt Conflicts and incompatibility for many
accessories.

The link below has more information on these problems:
http://pcbuyersguide.com/hardware/motherboards/VIA-Problems.html

A few more snippets are here

http://www.georgebreese.com/net/software/

Kris probably will answer your other question.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of abel
Sent: Thursday, June 09, 2005 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Live! CF

Seshu,
Are you working on a VIA based motherboard?
I am working on a VIA based motherboard.
Andy Powell (author of Asterisk Live! distro) tells me that VIA is not
quite good when emulating i686 behavoir and since his distro is compiled
for i686...
We are trying to confirm that but may be interesting to know about your
setup and how is Kristian's distro compiled.

On Mon, 6 Jun 2005 16:41:43 -0400, Kanuri, Seshu (Company IT) wrote
> Kristian,
> 
> I am talking about your distro, that does not seem to be able to boot 
> when I have mounted (if that is the right word) the CF  into my Dell 
> Server and tried to boot from it as the only IDE drive available.
> 
> The Linux just does not kick in.
> 
> If you want to debug this I can Fedex to you, my 800MB CF disk with 
> your distro on it, you for your R&D.
> 
> Seshu
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kristian

> Kielhofner
> Sent: Monday, June 06, 2005 3:36 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk Live! CF
> 
> abel wrote:
> > My theory is that the 64 MB image is built with a specific hdd form 
> > factor and when burning onto a different size CF it is mapped 
> > differently and it does not work.
> > On the other hand, you always can find out how the device is beeing 
> > seen by the system and customize the binary image accordingly.
> > Other software prepared to be run from CF (I recall WISP, the LEAF 
> > branch for wireless routers) have a final step which takes the 
> > software already compiled and 'packages' it to build the disk image.
> > I would be extremely happy if I could download the code tree and run

> > that final step by myself to get the disk image that suits my needs.
> > Second best would be to get the source tree and compile all the 
> > stuff to get that point.
> > Is that possible? Is the code available in the way I need for this
> operation? 
> > TIA.
> 
> abel,
> 
>   This is simply untrue.  My distro's (AstLinux) 32mb CF images
work on 
> anything...
> 
> http://www.kriscompanies.com/modules.php?name=Content&pa=showpage&pid=
> 3
> 
> --
> Kristian Kielhofner
> 
>  
> NOTICE: If received in error, please destroy and notify sender.  
> Sender does
not waive confidentiality or privilege, and use is prohibited. 
>  
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RE: [Asterisk-Users] format g729 and Voxee.com

[Kanuri, Seshu (Company IT)]

Voxee will not accept any calls that are not in G729. 
You need G729 codec on your Asterisk. 
Period.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, June 09, 2005 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Todd Reese
Subject: Re: [Asterisk-Users] format g729 and Voxee.com


> I have just signed up with Voxee.com and have attached my Asterisk 
> server to dial them via  IAX2.
> 
> Below is the start of the log which dials the number  and promply 
> hangs up when the call is answered, with the logs saying that the 
> channel is not compatiable.
> 
> I have traced this down to the g.729 codec which I don't have 
> installed.  Any ideas on how to force that the codec not be used?
> 
> BTW,  I have disallow=all and allow only the codecs that I want to use

> in both iax.conf and sip.conf.
> 
> Best Regards,
> 
> Todd Reese 

 
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RE: [Asterisk-Users] IP PHONE iareaphone x100, tested??

[Kanuri, Seshu (Company IT)]

 
Your ringtone seem to have gone bad. You have to upload a new ringtone
file to correct your phone problem.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Wednesday, June 08, 2005 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP PHONE iareaphone x100, tested??

On 6/8/05, Jorge Ortega Perez <[EMAIL PROTECTED]> wrote:
> Good day, im gonna buy 2 IP phones to test Asterisk, i don't wanna 
> spend too much $$$ on then, so i was looking at the internet and i 
> read a lot, the cheapest are the Grandstream BudgetTone but some 
> reviews of this list says they are not so good ... so i found 
> iareaphones but i can't find reviews about them, i would like to know 
> if someone has experience with them, at their site the phone seems to 
> be done to work for Asterisk ... but im not gonna buy something 
> without a good review ...

I got one a few weeks back.  It's cheap.  It has a strange annoying
dialtone, it doesn't have north american ring patterns that I could get
working, and it has what sounds like bad feedback on the speakerphone
that you can hear at both ends of the connection.  I wouldn't have
bought it if I had known.

Chris
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RE: [Asterisk-Users] IP PHONE iareaphone x100, tested??

[Kanuri, Seshu (Company IT)]

 
This Phone is same as Netweb-X100 made by Yuxin. These phones are
reliable. It has PA168 Chipset.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Brown
Sent: Wednesday, June 08, 2005 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] IP PHONE iareaphone x100, tested??

Hi,
I have used the Budgetone 102's extensively on Asterisk and found then
quite reliable as long as you update the firmware.
The GXP2000 is quite a mess at the moment as the current firmware does
not support 3 quarters of the advertised functions and codec support is
extremely limited. I have tested the unreleased latest firmware update
for the GXP2000 and it's an incredible difference though I still have
many phone hangs and crappy sound quality even with the new firmware.
Though this could all be sorted with the new official firmware rumoured
to be released any day now.

Mark



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Ortega Perez
Sent: 08 June 2005 21:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IP PHONE iareaphone x100, tested??

Good day, im gonna buy 2 IP phones to test Asterisk, i don't wanna spend
too much $$$ on then, so i was looking at the internet and i read a lot,
the cheapest are the Grandstream BudgetTone but some reviews of this
list says they are not so good ... so i found iareaphones but i can't
find reviews about them, i would like to know if someone has experience
with them, at their

site the phone seems to be done to
work for Asterisk ... but im not gonna buy something without a good
review ...

This is the web:

http://www.iareaphone.com/Hardware.htm

I really appreciate any help, thank you.

Jorge Ortega.

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RE: [Asterisk-Users] Asterisk Live! CF

[Kanuri, Seshu (Company IT)]

Kristian,

I am talking about your distro, that does not seem to be able to boot
when I have mounted (if that is the right word) the CF  into my Dell
Server and tried to boot from it as the only IDE drive available.

The Linux just does not kick in.

If you want to debug this I can Fedex to you, my 800MB CF disk with your
distro on it, you for your R&D.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Monday, June 06, 2005 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Live! CF

abel wrote:
> My theory is that the 64 MB image is built with a specific hdd form 
> factor and when burning onto a different size CF it is mapped 
> differently and it does not work.
> On the other hand, you always can find out how the device is beeing 
> seen by the system and customize the binary image accordingly.
> Other software prepared to be run from CF (I recall WISP, the LEAF 
> branch for wireless routers) have a final step which takes the 
> software already compiled and 'packages' it to build the disk image.
> I would be extremely happy if I could download the code tree and run 
> that final step by myself to get the disk image that suits my needs.
> Second best would be to get the source tree and compile all the stuff 
> to get that point.
> Is that possible? Is the code available in the way I need for this
operation? 
> TIA.

abel,

This is simply untrue.  My distro's (AstLinux) 32mb CF images
work on anything...

http://www.kriscompanies.com/modules.php?name=Content&pa=showpage&pid=3

--
Kristian Kielhofner 

 
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RE: [Asterisk-Users] CLUELESS NEWBIE needs help making an outboundsip call to PSTN

[Kanuri, Seshu (Company IT)]

Steve,
 
1) go to /etc/asterisk
 2) open modules.conf for editing using vi
 3) add this line:
noload=pbx_wilcalu.so

 4) Save the file
 5) Restart asterisk

 Lightup the candles, open the  Cabernet Savignon ( or whatever your
prefernce) and call your girlfriend.

;)

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Sent: Monday, June 06, 2005 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CLUELESS NEWBIE needs help making an
outboundsip call to PSTN


Still just simply want to be able to make an outbound sip provider call
from asterisk that's all :-) Kinda like that guy that wants to call
his girlfriend
I'm getting lonely here.


Ok completely started over

Installed CVS-HEAD

zaptel seems to compile ok (see lots of warnings)
libpri seems to compile ok

zaptel and ztdummy load ok after compile

asterisk builds ok but exits with this error at runtime:

  [pbx_wilcalu.so]Jun  6 14:32:54 WARNING[27986]: loader.c
:310 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so:
undefined 
symbol: ast_p
thread_create
Jun  6 14:33:49 WARNING[27986]: loader.c:518 load_modules
: Loading module pbx_wilcalu.so failed!

asterisk will not run!

I have no idea what this means or how to deal with it. any help is much 
appreciated!

Asterisk version: Vontage:/etc/asterisk# asterisk -V
Asterisk CVS-HEAD

umm not really informative there :-) I downloaded and built it 
June, 6 
2:45PM Eastern STD time (US)

Here's some more info about my system just in case it is userful:

Stable compiles & runs OK.

Thanks!

Steve

Vontage:/usr/src/asterisk# cd /proc/version
-bash: cd: /proc/version: Not a directory
Vontage:/usr/src/asterisk# cat /proc/version ; cat /proc/cpuinfo
Linux version 2.4.27-2-386 ([EMAIL PROTECTED]) (gcc 
version 3.3.5 (Debian 1:3.3.5-6)) #1 Thu Jan 20 10:55:08 JST 2005
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 6
model   : 5
model name  : Pentium II (Deschutes)
stepping: 2
cpu MHz : 448.976
cache size  : 512 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
mca 
cmov pat pse36 mmx fxsr
bogomips: 894.56

Vontage:/usr/src/asterisk# free
  total   used   free sharedbuffers
cached
Mem:321924 284556  37368  0  76080
149288
-/+ buffers/cache:  59188 262736
Swap:   373928  0 373928
Vontage:/usr/src/asterisk# df
Filesystem   1K-blocks  Used Available Use% Mounted on
/dev/hda3  7661260   1131756   6140332  16% /
tmpfs   160960 0160960   0% /dev/shm
/dev/hda190297  5768 79712   7% /boot
Vontage:/usr/src/asterisk#
Vontage:/usr/src/asterisk# lsmod
Module  Size  Used byNot tainted
ztdummy 1688   0  (unused)
zaptel219616   0  [ztdummy]
af_packet  11048   1  (autoclean)
usb-uhci   19504   0  [ztdummy]
usbcore52268   1  [usb-uhci]
ymfpci 39144   0
ac97_codec 11252   0  [ymfpci]
soundcore   3268   2  [ymfpci]
ide-scsi8272   0
scsi_mod   86052   1  [ide-scsi]
8139too12328   1
mii 1952   0  [8139too]
crc32   2848   0  [8139too]
agpgart39108   0  (unused)
dm-mod 36120   0  (unused)
ide-cd 27072   0
cdrom  26212   0  [ide-cd]
rtc 5768   0  (autoclean)
ext3   65388   1  (autoclean)
jbd34628   1  (autoclean) [ext3]
ide-detect   288   0  (autoclean) (unused)
piix7784   1  (autoclean)
ide-disk   12448   3  (autoclean)
ide-core   91832   3  (autoclean) [ide-scsi ide-cd
ide-detect 
piix ide-disk]
unix   12752   9  (autoclean) 

 
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RE: [Asterisk-Users] Re: Polycom 500...

[Kanuri, Seshu (Company IT)]

I have never used inband. I always used 
rfc2833.
 
This problem is seen even if you are using 
dtmf=rfc2833. 
 
the line progressinband=no,  fixes 
that.
 
Seshu


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Noah 
MillerSent: Monday, June 06, 2005 3:07 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: Polycom 
500...



  Are you using inband DTMF? There are other options but I don't know 
  much 
  about the polycom phones. I have noticed that sometimes when 
  accessing 
  voicemail, it will 'miss' some dtmf tones if they are too short. 
  This 
  doesn't explain the number changing, unless your dial plan is putting 
  in 
  the leading zero.
If you are using a version of CVS HEAD from April 2005 or later, you should 
definitely use dtmf=rfc2833 rather than dtmf=inband.  In fact, just use 
rfc2833 - it works with all versions of asterisk that I've tested, but inband 
only works with certain versions.

Also, you may want to check the digit map (the pattern recognition when you 
dial).  It is in sip.cfg.  I know that by default it treats numbers 
that start in "11" specially, but it shouldn't really transpose numbers.  
That's weird.




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RE: [Asterisk-Users] Asterisk Live! CF

[Kanuri, Seshu (Company IT)]

Abel,

In have the same issue when I have burned the image to an 800MB CF Disk.
All it displays is GRUB CLI in a continuous stream.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of abel
Sent: Monday, June 06, 2005 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk Live! CF

I downloaded teh Asterisk live CF distribution (http://www.automated.it/
asterisk/asterisk-cf.htm) and I cannot make it work. 
I suspect the problem is the distro is a binary image to be burned in a
64 MB CF while I am trying to burn it in a 256 MB CF. 
Does anybody know haw to burn a working 256 MB CF? 
BTW the symptom is: the word GRUB is displayed and nothing else happens
when trying to boot up. 
TIA for your help.

abel
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RE: [Asterisk-Users] How to make Polycom phones work with Asterisk asaSIP Client?

[Kanuri, Seshu (Company IT)]

Wiley,

There are a couple of issues that we saw  while not using this option.

1) sip authentication failures as Asterisk is not able to reach Polycom
phones.

A typical problem description is here:
http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht
ml

2) DTMF issues for Transfers, Hold or simply to dial extensions. This
problem is more pronounced when you are using dtmfmode=inband

Check the Links here:

http://lists.digium.com/pipermail/asterisk-users/2005-March/092401.html

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg94489.htm
l


You can do a test of this using [EMAIL PROTECTED] version 1.0 if you have a
Polycom phone at your disposal. I ran this test as under:

1) Configure a Polycom Phone as SIP extension under [EMAIL PROTECTED]
2) Try to make a call - It will not register with Asterisk as
[EMAIL PROTECTED] does not use this option in it's sip configuration and
even if you add this manually in the sip_additional.conf, AMP will
overwrite(remove) this line from the sip_additional.conf

To make Polycom phone work with Asterisk, add the sip.conf entry into
the main sip.conf, using progressinband=no option. Polycom phone will
immediately connect.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Monday, June 06, 2005 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to make Polycom phones work with
Asterisk asaSIP Client?

Seshu,

I have Polycom IP500s and I have never had to set that parameter to make
them work with Asterisk.  I have used various versions of the BootROM
and sip.ld without any issue.

What problem are you specifically addressing?

Thanks,
Wiley


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri,
Seshu (Company IT)
Sent: Monday, June 06, 2005 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to make Polycom phones work with Asterisk
as aSIP Client?

Folks!

If you are using Polycom Phones, any model - 500, 300, 400, 600 etc,
please rememeber to add this line to your sip.conf entry.

Progressinband=no

Without this line, these phones may not work. Probably this one line may
fix most of the problems users are reporting on this forum about
Polycoms.

Seshu Kanuri

 
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[Asterisk-Users] How to make Polycom phones work with Asterisk as a SIP Client?

[Kanuri, Seshu (Company IT)]

Folks!

If you are using Polycom Phones, any model - 500, 300, 400, 600 etc,
please rememeber to add this line to your sip.conf entry.

Progressinband=no

Without this line, these phones may not work. Probably this one line may
fix most of the problems users are reporting on this forum about
Polycoms.

Seshu Kanuri 

 
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RE: R: [Asterisk-Users] AT-320 + supervised transfer

[Kanuri, Seshu (Company IT)]

Remove the Tthr options. You don't need any of them in the dial string for 
AT320s

Seshu

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Wednesday, June 01, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: R: R: [Asterisk-Users] AT-320 + supervised transfer

This is what happen when i call a peer that not answer:

   -- Executing Dial("SIP/401-4de6", "SIP/402|60|Thtr") in new stack
-- Called 402
-- SIP/402-fa23 is ringing
-- SIP/402-fa23 answered SIP/401-4de6
-- Attempting native bridge of SIP/401-4de6 and SIP/402-fa23
-- Started music on hold, class 'default', on SIP/401-4de6
-- Playing 'pbx-transfer' (language 'it')
-- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/406|60|Tthr") in new 
stack
-- Called 406
-- SIP/406-aa46 is ringing
Warning, flexibel rate not heavily tested!
Jun  1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable to 
create channel Local/[EMAIL PROTECTED]/n do you have chan_local?
-- Stopped music on hold on SIP/401-4de6
  == Spawn extension (local, 406, 1) exited non-zero on 'Local/[EMAIL 
PROTECTED],2'
-- Playing 'beeperr' (language 'it')
  == Spawn extension (local, 402, 1) exited non-zero on 'SIP/401-4de6'

It could some extensions.conf problem ?

Thanks 

-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill
Inviato: mercoledì 1 giugno 2005 14.20
A: asterisk-users@lists.digium.com
Oggetto: Re: R: [Asterisk-Users] AT-320 + supervised transfer

On Wednesday 01 June 2005 13:04, Giordano Grandis wrote:
> Ok, thanks for all.
> Just a thingh: how do u set DTMF on your phones ?

We have them set to RFC2833. 

I think I've noticed some cases where the remote party hears the tones, but 
it's not an issue that bothers me :)

Cheers,
Gavin.
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RE: [Asterisk-Users] IAX2 analog telephone adapter

[Kanuri, Seshu (Company IT)]

Try ATCOM's AG168V  available from US 
Distributor http://www.iareaphone.com
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
DineshSent: Wednesday, June 01, 2005 9:40 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] IAX2 analog 
telephone adapter 


Hello 
All,
 
I am looking for a IAX2 analog 
telephone adapter, just want to ask your views on which ones are bad, good and 
the best.
 
Thanks in 
advance,
 
Dinesh 
BirlasekaranNetwork Engineer,ComIT, Institute of Molecular and Cell 
Biology61 Biopolis Drive, 
Singapore 
138673HP : 92962676 DID : 65869804 Fax : 67791117 Email : [EMAIL PROTECTED]WWW: www.imcb.a-star.edu.sg
 




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RE: [Asterisk-Users] Asterisk@Home 1.1b1 has been released

[Kanuri, Seshu (Company IT)]

I don't see the SugarCRM being part of the install. 
How do you activate this?

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 31, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] [EMAIL PROTECTED] 1.1b1 has been released

We have replaced the simple contact management system in [EMAIL PROTECTED]
with SugarCRM a full CRM system. This might seem like over kill for a
home PBX but Sugar has the best contact management we have seen. With
click to dial functionality and the ability to import data from other
contact managers it's a great fit for [EMAIL PROTECTED]

We have also added new version of the usual Asterisk software AMP and
Flash operator panel.

Download from http://asteriskathome.sourceforge.net

For support please read the [EMAIL PROTECTED] Handbook
http://asteriskathome.sourceforge.net/handbook/index.html

and use our support forum at
http://sourceforge.net/forum/?group_id=123387




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RE: [Asterisk-Users] AreskiCC - DOES IT REALLY WORK??????

[Kanuri, Seshu (Company IT)]

Robson,
 
This is a Frames probelm. Areskicc uses a frame on the 
left and this frame may be showing your Login page in it again, instead of the 
menu. If this is the case, please check the "Stupids Guide to AreskiCC" on the 
WIKI
 
Seshu 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
[EMAIL PROTECTED]Sent: Tuesday, May 31, 2005 11:27 
AMTo: asterisk-users@lists.digium.comSubject: 
[Asterisk-Users] AreskiCC - DOES IT REALLY WORK??


Hi all,
 
I am quite disappointed at the 
application AreskiCC. I have installed everything following the instructions 
 but the thing doesn’t want to work.
 
First of all, when I start the 
index.php page, any name/password logs in.
After the login it takes me to a 
page with a single option “LOGOUT”
 
We are monitoring the database and 
it seems like the application doesn’t connect to 
it.
 
Does anybody in this have made this 
work? Can someone help me please??
 
Thanks,
 
Robson




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RE: [Asterisk-Users] static database config gui

[Kanuri, Seshu (Company IT)]

Chris,

This is a good beginning. Though this does sound useful to me, I would
also like to mention that there are  some database management tools
already available out there - phpMyAdmin for example, where you can
modify everything visually and insert/update/delete Asterisk table data
over the web. 

There is another application 'phpconfig' which is part of [EMAIL PROTECTED]
- which can modify the config files in text mode directly. What would be
useful is to modify this application, to make it write the changes to
the mysql database also and synchronize the two sets of config data.
This provides flexibility for the user to edit the data .

AMP GUI does this in updating the tables and writing the same to
additional config files ( but not the other way, i.e if you change the
config files in text mode, it does not update the database, when you
open the gui), that are included in the default config files.

Seshu


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Thursday, May 26, 2005 3:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] static database config gui

I threw together a web gui for the static database configuration over
the last couple of days.

I built it using mod perl and the template toolkit.  If enough people
show an interest in this I'll put up a distribution, although it could
take a few days.

The interface is as generic as possible so you can throw pretty much any
asterisk .conf file in and it works.  The interface assumes you already
know how to edit the config files.  The database schema is the same as
on the wiki.

I'm working on making it a multi user interface.  So that you can have
multiple end users with their own copies of the config files all on the
same server.  The separation will be done through a naming
convention that will be applied appropriately.   A kind of asterisk
virtual hosting.

I have a demo setup at the following url:

http://catalog1.paymentonline.com/voip/demo/index.html


One note on the gui.  The numbers on the very left are the order of the
statements in the config file.  For extensions, when you change the
location of an extension priority the system will automatically renumber
the order and the dialplan automatically.  To insert a new priority in
the middle of an extension, use a number with a fraction. 
When you add, delete, or update the system will automatically renumber
everything.

For example if you have the following extension:

exten => 999,1,Answer
exten => 999,2,Dial
exten => 999,3,Hangup

And you want to insert a new priority after 1, add the new priority as
1.5 which when added would give you something like this:

exten => 999,1,Answer
exten => 999,2,Ringing
exten => 999,3,Dial
exten => 999,4,Hangup


Chris
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RE: [Asterisk-Users] What does Asterisk need in the way of a GUI?

[Kanuri, Seshu (Company IT)]

Let me recollect what I needed:

1) We need a TRUNK Configurator that can easily create multiple SIP/IAX
trunks and assign them to the dialing contexts. Current GUI tolls are
not doing this properly.

2) We need DUNDI configurator for Inter-Server access

3) We need an Accounting Module that can apply the rate plan to the
calls and generate call accounting records.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mitchel
Constantin
Sent: Wednesday, May 25, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] What does Asterisk need in the way of a GUI?

We are two programmers who are passionate for Asterisk and we will be
dedicating the next three months towards programming for Asterisk and
would like to get some input from everyone on what they feel Asterisk is
lacking or needs based on what is not currently a part of it or
available through third parties. Hopefully, by asking up front we won't
be wasting our time on something nobody wants or needs.

Specifically I am asking in the way of GUI's (web-based or not), not in
backend programming as Mark and others have that well under control!

Thank you for your suggestions,
Mitchel & Tom 

 
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[Asterisk-Users] oh323 problems - Solved

[Kanuri, Seshu (Company IT)]

For the benefit of everyone, having H323 Configuration problem due to
H245 Tunnel, check the h323 Config embeded at the end. Comment the
offending line as under:
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
-Original Message-
From: Tola Ogunsan [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, May 25, 2005 1:03 PM
To: Kanuri, Seshu (Company IT)
Subject: RE: oh323 problems

Thanks a million Seshu, it worked like a champ.  Thanks

>From: "Kanuri, Seshu (Company IT)" <[EMAIL PROTECTED]>
>To: "Tola Ogunsan" <[EMAIL PROTECTED]>
>Subject: RE: oh323 problems
>Date: Tue, 24 May 2005 10:11:16 -0400
>
>To start with:
>
>1)Disable H245 tunneling.
>2)Dial(Zap/g2/${EXTEN})  - change this to Dial(Zap/g2/${EXTEN}, 30)
>
>Try this and see how that changes
>
>Seshu
>
>-Original Message-
>
extensions.conf
[OH323]
exten => _4420X.,1,Answer
exten => _4420X.,2,Wait(2)
exten => _4420X.,3,Dial(Zap/g2/${EXTEN}) 
exten => _4420X.,4,Congestion
xten => _4420X.,5,Hangup
>


>--
>H323.conf
>--
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
connectPort=1720
;
; Configure TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=3
;tcpEnd=65000
;
; Configure UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
;   "rtp.conf"
;
udpStart=1
udpEnd=3
;udpEnd=65000
;
; Enable fast start (yes,no).
;
;fastStart=no
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
h245inSetup=yes
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
;jitterMax=500
jitterMax=1000
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
;
;ipTos=reliability
ipTos=reliability
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only trace info for OpenH323 is logged in libTraceFile.
;
;wrapLibTraceLevel=1
;libTraceLevel=2
wrapLibTraceLevel=5
libTraceLevel=5
;libTraceFile=stdout
libTraceFile=/var/log/asterisk/chan_oh323.log

; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   ,
;   ,
;   GKID:
;
;gatekeeper=192.168.1.2
;gatekeeper=DISCOVER
;gatekeeper=209.3.12.49
gatekeeper=DISABLE
;
; Set the gatekeeper password
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout
;
;gatekeeperTTL=600
;
; Set the mode for sending user-input
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;
;userInputMode=TONE
userInputMode=TONE
; AMA flags (default, omit, billing, documentation)
;
amaFlags=billing
;
; Account code
;
;accountCode=H323
accountcode=H323
;
; Set the default context of H.323 calls.
;
;context=voip-h323
;context=2003 234 80
context=OH323
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;context=23480T
;alias=68.50.129.5

;
; Aliases/prefixes routed in "all-aliases" context.
;
;context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes

;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

;context=newwave
;alias=216.206.188.200
;alias=newwave

;context=IPCBgw1
;alias=195.239.25.20

;context=IPCBgw2
;alias=195.239.25.21

;context=IPCBnetSoftSwitch
;alias=216.206.188.200

;Navao - Sola Omidiran
;context=80066
alias=213.255.198.11

RE: [Asterisk-Users] origination providers

[Kanuri, Seshu (Company IT)]

Mike,

> Many of the providers I've tried contacting either 
> won't call me back, or want me to sign an NDA just 
> to get a rate quote, or some other bullshit. 

Assuming that you will need about 12 to 24 simulataneous calls on each
DID you want to run, and you are using Ulaw to get these calls, what is
the bandwidth that the DID provider has to give you, apart from the DID
service?

Ulaw needing 64 kbps per line, needs 1.2 mbps for 20 simultaneous calls.


Assuming a Data T1 costs about $500 bucks a month and assuming that you
need/use the DID for only 8 hours a day at that rate, it costs about
$100 per month in data bandwidth alone.

Who will pay for this, If it is not Democracynow who is footing the
bill?

Seshu



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mike
castleman
Sent: Tuesday, May 24, 2005 3:00 PM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] origination providers

hi folks,

Has anyone found a good (and, ideally, cheap -- we don't really want any
per-minute charges) origination provider which can handle a moderate
number of simultaneous incoming calls (to the same, single DID)?

Many of the providers I've tried contacting either won't call me back,
or want me to sign an NDA just to get a rate quote, or some other
bullshit. Most of the providers whose rates are plainly posted on their
website have a limit of at most 4 or 6 simultaneous calls, which is not
likely to be enough for the application I'm considering.

You can reply off-list or on-list, as you prefer.

many thanks,
mike

--
mike castleman
network / systems administrator
democracy now!
mailto:[EMAIL PROTECTED]
tel:+1-212-431-9090 (office)
tel:+1-646-382-7220 (mobile) 

 
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RE: [Asterisk-Users] Windows IAX Softphone

[Kanuri, Seshu (Company IT)]

Title: Message


FireFly is the best of the IAX softphones. Other 
softphones do not work as good as FireFly. DIAX has many bugs still. DIAX 
Softphone disconnects with Windows DLL errors everytime there is a problem in 
the call like Asterisk Channel Not available etc.
 
Seshu


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Adam 
RobinsSent: Monday, May 23, 2005 5:11 PMTo: 
[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: RE: [Asterisk-Users] Windows IAX 
Softphone

Try DIAX.  Works just fine!
 
http://www.laser.com/dante/diax/diax.html


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Jeromy 
GrimmettSent: Monday, May 23, 2005 12:09 PMTo: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] Windows IAX Softphone

Is there a softphone 
for windows that supports IAX?
 
I can't seem to find 
anything out there...maybe im looking in the wrong places...
 
Jeromy 
Grimmett
VoipEmpire.com
[EMAIL PROTECTED]
The contents of this email 
message and any attachments are confidential and are intended solely for 
addressee. The information may also be legally privileged. This transmission is 
sent in trust, for the sole purpose of delivery to the intended recipient. If 
you have received this transmission in error, any use, reproduction or 
dissemination of this transmission is strictly prohibited. If you are not the 
intended recipient, please immediately notify the sender by reply email and 
delete this message and its attachments, if any.




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RE: [Asterisk-Users] Call forwarding...

[Kanuri, Seshu (Company IT)]

Try changing SetCIDNum SetCallerID and use to SetCIDName as under:

Ex:
---
exten => s, 1, SetCallerID(${CALLERIDNUM})
exten => s, 2, SetCIDName(${CALLERIDNAME})
exten => s, 3, Dial(${ARG2}/${ARG1},${RINGSECS})
exten => s, 4, Voicemail(u${ARG1})
exten => s, 5, Hangup
exten => s, 101, Voicemail(b${ARG1})
exten => s, 102, Hangup
 
Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Benson
Sent: Wednesday, May 18, 2005 6:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call forwarding...

Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...

Hi,

I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.

I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).

I tried this (from voip-info.org wiki)...

exten => 1234,1,dial(sip/1234,20)
exten => 1234,2,playback(pls-wait-connect-call)
exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 1234,4,SetCIDNum(0${CALLERIDNUM}) exten =>
1234,5,dial(${TRUNK}c/9871234321,20,r)
exten => 1234,6,SetCIDNum(${NewCaller})
exten => 1234,7,voicemail2([EMAIL PROTECTED]) exten =>
1234,101,voicemail2([EMAIL PROTECTED])
exten => 1234,102,hangup

Mine looks like this...

exten => 08700688nnn,1,Dial(SIP/operator,1,t)
exten => 08700688nnn,2,playback(pls-wait-connect-call)
exten => 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten => 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten => 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten => 08700688nnn,6,SetCIDNum(${NewCaller})
exten => 08700688nnn,7,Voicemail(u100)
exten => 08700688nnn,8,Hangup()
exten => 08700688nnn,101,Voicemail(b100) exten =>
08700688nnn,102,Hangup()

(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...

   -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"c/07961106nnn|20|r") in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)

Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...

   -- Nobody picked up in 1000 ms
   -- Executing Playback("IAX2/[EMAIL PROTECTED]:4569-1",
"pls-wait-connect-call") in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File
pls-wait-connect-call does not exist in any format May 18 10:20:26
WARNING[24416]: file.c:790 ast_streamfile: Unable to open
pls-wait-connect-call (format ilbc): No such file or directory May 18
10:20:26 WARNING[24416]: app_playback.c:83 playback_exec: 
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for
pls-wait-connect-call
   -- Executing SetVar("IAX2/[EMAIL PROTECTED]:4569-1",
"NewCaller=01202843nnn") in new stack
   -- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]:4569-1",
"001202843nnn") in new stack
   -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)
   -- Executing SetCIDNum("IAX2/[EMAIL PROTECTED]:4569-1",
"01202843nnn") in new stack
   -- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569-1",
"u100") in new stack
   -- Playing '/var/spool/asterisk/voicemail/default/100/unavail' 
(language 'en')

Again - I'm not worried about the audio file warning - I can fix that
later... I guess this is the important bit...

   -- Executing Dial("IAX2/[EMAIL PROTECTED]:4569-1",
"/07961106nnn|20|r") in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)

The call then drops into voicemail...

I've tried various permuations but still no call is made to the mobile
number. Any ideas?

Cheers,

Mark

I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work... 

 
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RE: [Asterisk-Users] Asterisk and a D/42NS

[Kanuri, Seshu (Company IT)]

> Forget the dialogic, the drivers are old and not free and almost no-
one is using them.  

I concur that view. Don't even touch Dialogic garbage. I have see people
spending months without being able to make them work. Dump them into
your incinerator and turn the knob from 'Medium' to 'High'. 

You would be happier buying either a Varion card (
http://www.govarion.com) or a Digium card (http://www.digium.com) at a
fraction of the price and a fraction of the time required to configure.

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tim panton
Sent: Tuesday, May 17, 2005 4:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and a D/42NS


On 17 May 2005, at 02:21, Corey Hickey wrote:

> Hello,
>
> The company I work for deploys and manages telecom hardware for
> small- to
> medium-sized businesses. My boss has asked me to investigate Asterisk 
> as a possible PBX for deploying to customers along with IP phones. The

> general layout would be:
>
>  -- --
> |trunk |-- |  LAN,|
> | (T1/analogs/ |  ===> | Asterisk | => |IP phones |
> | etc.)|--  --
>  --
>
> I'm building a test machine right now to experiment with using one of 
> our analog lines. We have two spare Dialogic D/42NS cards, and I was 
> hoping I would make one of them work. Has anyone tried that model? I 
> haven't found any information, good or bad.
>
> The supported hardware list on asterisk.org has the D/41JCT-LS. Is 
> that a very similar card? Could I "pretend" the D/42NS is a 
> D/41JCT-LS?

The advice I was given when I started down the same line a year ago was:
Forget the dialogic, the drivers are old and not free and almost no- one
is using them. You are better off just buying a well supported card
(ideally from Digium).

There are technical and legal reasons why the Dialogics don't fit into
the Asterisk world view - as such even if you do get it to work it won't
be representative of what you'd actually install for a customer.

What I did was to build a system that was 100% voip using spare old
hardware and played with it until I was comfortable with Asterisk, then
splashed out on an E1 card and some dedicated hardware.

Turned out fine for us.

Tim

>
> Thanks,
> Corey
>
> 

 
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RE: [Asterisk-Users] How to connect two Asterisk servers

[Kanuri, Seshu (Company IT)]

Frank,

Your solution is not clear to me. Can you tell me what Step 2 will do?

[general] 
register => [EMAIL PROTECTED]

How will it resolve the name obelix as an authenticated user, assuming
that asterix is reolved using dns? 

Seshu



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frank
Becker
Sent: Saturday, May 14, 2005 5:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to connect two Asterisk servers

> Pls I want to know how to connect two Asterisk servers with sip,on the

> voip-info.org the iax details exist but the sip there is nothing about

> its details,pls any one can help.

Its quite simple:
Server2 (name obelix) should register at server1 (name asterix)

1. Enter in asterix: /etc/sip.conf - Server 1
[obelix]
;secret=
username=obelix
from_user=obelix
type=friend
context=default
host=dynamic
nat=no

2. in obelix: /etc/sip.conf enter 
the following in the section  - server 2 
[general] 
register => [EMAIL PROTECTED]

3. in obelix: To forward a call to asterix 
   simply use the following:  - Server 2

exten => s,1,Dial(SIP/[EMAIL PROTECTED],30,Ttr)


Hope this helps

Frank
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RE: [Asterisk-Users] SIP Gerenal settings conufsion

[Kanuri, Seshu (Company IT)]

I want to add to this question as this seems relevant here. My question
corresponds to sip.conf, iax.conf and [globals] in extensions.conf. My
questions are as under:

1) Can we have two general sections - one in sip.conf and another one in
one of the included additional sip extension files, like the example
below? How will the following affect the calls?

Ex: 
;sip.conf 
[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
context = from-sip-internal; Default SIP context
#include sip_nat.conf
#include sip_additional.conf

;sip_nat.conf 
[general]
port = 5060   ; Port to bind to (SIP is 5060)
host=dynamic
nat=yes

;sip_additional.conf 
[general]
port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
allow=g729
nat=never


;extensions.conf
[globals]
; globals 
VOICEPULSE => IAX2/xyzd:[EMAIL PROTECTED]
PSTN => Zap/1
CALLER_ID=7328106707
; Which phones to ring for various users
RING_Seshu => SIP/6005
MAX_RINGTIME => 15
RINGSECS => 20
LONGTIMEOUT => 60
#include extensions_additional.conf
/* End of Extensions.conf */

/* Beginning of extensions_additional.conf */
;extensions_additional.conf
[globals]
RINGTIMER = 30
REGTIME = 7:55-17:05
REGDAYS = mon-fri
RECORDEXTEN = ""
PARKNOTIFY = SIP/200
OUT_3 = ZAP/g0
OUT_2 = IAX2/nypbx
OUT_1 = SIP/SIP_USA
FAX_RX_EMAIL = [EMAIL PROTECTED]


Are the above valid? If so how will they affect the values of different
parameters when the user contects of a specific  file is being served?

Seshu Kanuri 

 
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RE: [Asterisk-Users] Astlinux & AMP

[Kanuri, Seshu (Company IT)]

What do you mean "Requires PHP+pear+php/mysql. But Will run as CGI. I
have had it working with php. So apache is not required."

To make PHP work, Apache is required anyway as a web server. Is in't it?

Seshu

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Thursday, May 12, 2005 3:14 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Astlinux & AMP

On Thu, May 12, 2005 at 04:12:58PM +1000, Rob Thomas wrote:
>   Needs apache + php (+30 odd mb)

Requires PHP+pear+php/mysql. But Will run as CGI. I have had it working
with php. So apache is not required.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend 

 
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RE: [Asterisk-Users] * Server

[Kanuri, Seshu (Company IT)]

Title: * Server


http://www.iareaphone.com sells 
these
 
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Montague, 
ClarenceSent: Thursday, May 12, 2005 1:30 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] * 
Server

Any reviews/comments out there on this server? Looks 
solid.. But would like to know if anyone has purchased one of these before. Any 
other companies out there offer pre-built * servers that someone would like to 
comment on? 
http://www.thevoipconnection.com/store/catalog/product_16214_VS1trade_Asterisk_Voice_Server.html 





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RE: [Asterisk-Users] Voice Recognition - Cases of success

[Kanuri, Seshu (Company IT)]

Interactive Intelligence has a commercial Speech recognition API for
this purpose.

Check http://www.inin.com

Or the specific Vocalite engine page at:

http://www.inin.com/Products/vocalite/vocalite.asp


Seshu Kanuri 

 
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[Asterisk-Users] GSM gateway for Asterisk

[Kanuri, Seshu (Company IT)]

Folks!

I am looking at a couple of models of Fixed GSM Gateways for the Purpose of 
VOIP connectivity and specifically to work with Asterisk. I found that  these 
can be imported into USA for about $99.99 or about that. This is a one channel 
unit just like tellular, one of them has GPRS.

FCT11M:
1)freq: GSM network,900/1800/1900Mhz,
2)provides reversal signal for payphone/billing
3)supports PBX and VOIP
4)for voice (no fax)
5)battery(optional)
6)can be used for the remote area where signal is weak.
 
FCT11G:
1)freq: GSM network,900/1800/1900Mhz,
2)provides reversal signal for payphone/billing
3)supports PBX and VOIP
4)for voice and GPRS (no fax)
5)battery(optional)
6)can be used for the remote area where signal is weak.

I am pasting an image of the network diagram here:

Specifications in text are below. I would appreciate for any feedback of their 
usability.

Seshu



Description:
---
This unit can conveniently access to the available GSM system network. This 
system
possesses such a high receiving sensitivity and a large transmitting power that 
it expandsthe
effective coverage of the cellular network to a larger geographic area (upto 15 
miles). 

The unit has been extensively used in the fixed access to the cellular network 
to solve the wired communications problems in the rural areas. It can also be 
used to develop fast radio public telephone services to satisfy the 
communications for the time being and work as the CO relay tosimplify the 
registration s and lower the cost. Furthermore it can meet the requirement of 
mobile communications onboard vehicles, ships, trains, etc. All these enlarge 
the number of the network subscribers considerably so that it can utilize the 
resources better. General Instructions How to link with a charger, Office PBX 
and VOIP 

Main functions
--
Payphone
Caller ID
Pin number locked (Optional)
Block prefix number (Optional)
Support OfficePBX
Support VOIP
Office PBX
VOIP

Description/ Unit Specifications

UP MHz 890~915 1710~1755 1850~1910 
WorkingFrequency 
DOWN MHz 935~960 1805~1850 1930~1990
Transmitting power dBm 33
Receiving sensitivity dBm -104
Atmosphere Kpa 86~106
Power Specifications
Power mode: AC to DC
a. Switch adaptor (without battery) 110-220V to 5V or 7-12V, 50/60Hz, 1.25A
b. Switch adaptor (with Ni-MH battery) 110- 220V to 7.5V, 50/60Hz, 1.0A
Backup battery:
Standby: 20Hrs(Appr.) Continued Talking: 3Hrs(Appr.) 
Note:
a. The battery will give the power when the normal power is off, and the 
battery power will be off when the normal
power is On.
B. The battery is for back up power only, It is not designed for normal power 
use.

Quick Installation
1. Take off the cover of the SIM holder,
then put in a SIM card into the holder.
Receiving sensitivity dBm -104
2. Plug in a phone into the phone socket RJ-11
3. a. Install the antenna first, please screw the antenna tightly into the 
connector,
and put the antenna in the purpose place.
   b. Connect the power, and put power switch ON.
Power Specifications
Power mode: AC to DC
a. Switch adaptor (without battery) 110-220V to 5V or 7-12V, 50/60Hz, 1.25A
b. Switch adaptor (with Ni-MH battery) 110- 220V to 7.5V, 50/60Hz, 1.0A

Antenna information
---
Frequency range: A:890 960MHz B:1710 1880MHz
Banwidth: A:70MHz B:170MHz
Gain: 2.15dBi or 5.5dBi (optional)
Impedance: 50Ù
Max Power: 50W
Connector Type: SMA
Size: Longth:30cm , 60cm and 100cm (optional)
Weight: 120g

Other Specifications

Plastic cover: light blue or black
Size 183mm 124mm 32mm(l\w h)
G.Weight(complete set) 1.2Kg
Circumstances: a temperature -20 ~50 b relative humidity 5%~95%

Switches on the Box:
---
ANT ON OFF
SET WORK
LOAD RJ11
Antenna
Power
Switch for power Switch for set or work
Set for factory only by now
Please put the switch on work
Load/USB for the factory only
Rj11 for phone line
Gain: 2.15dBi or 5.5dBi (optional)
Impedance: 50Ù
Max Power: 50W
Connector Type: SMA
Size: Longth:30cm , 60cm and 100cm (optional)
Weight: 120g 

 
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RE: [Asterisk-Users] Sip or IAX2 eb Client

[Kanuri, Seshu (Company IT)]

Hi Tim,

I am interested in your half finished Java AIX2 App. Can you send me the
source code?

Seshu Kanuri
212-537-2849

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tim panton
Sent: Wednesday, May 11, 2005 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sip or IAX2 eb Client


On 11 May 2005, at 12:41, Matt Riddell wrote:

> Anton Krall wrote:
>
>> Is there any good IAX2 or SIP free web client? Im looking for 
>> something open source or free preferably IAX2 for integrating into a 
>> web site...
>> Any leads?
>>
>
> Sounds like you're looking for the IAXClient libaries.  There are  
> many examples within it.
>
> This includes IAXCom, IAX2 ActiveX control etc.
>

Depending on how much of a hurry you are in,
I have a half complete java IAX2 client that might be of use.

Tim. 

 
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[Asterisk-Users] AreskiCC - Install Problems

[Kanuri, Seshu (Company IT)]

Nabeel,
 
I am trying to install AreskiCC 
and I get the following errors. 
 
Warning: pg_pconnect(): 
Unable to connect to PostgreSQL server: could not connect to server: Connection 
refused Is the server running on host localhost and accepting TCP/IP connections 
on port 5432? . in 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 
68Database error: Link-ID == false, pconnect 
failedPostgreSQL Error: 0 ()Warning: pg_pconnect(): 
Unable to connect to PostgreSQL server: could not connect to server: Connection 
refused Is the server running on host localhost and accepting TCP/IP connections 
on port 5432? . in 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 
68Database error: Link-ID == false, pconnect 
failedPostgreSQL Error: 0 ()Warning: 
pg_errormessage(): supplied argument is not a valid PostgreSQL link resource in 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 
101Warning: Cannot modify header information - headers 
already sent by (output started at 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in 
/var/www/html/areskicc/lib/module.access.php on line 
66Warning: Cannot modify header information - headers 
already sent by (output started at 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in 
/var/www/html/areskicc/lib/module.access.php on line 
67
Can you guide me as 
to what and where may be the problem?
 
Seshu 
KanuriMorgan Stanley | Technology1633 
Broadway | Floor 19New York, NY 10019Phone: +1 212 537-2849[EMAIL PROTECTED]
 




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[Asterisk-Users] AreskiCC Install Problems

[Kanuri, Seshu (Company IT)]

I have followed the 
Idiots' guide for installation, but still could not make it 
work.
 
When I try to login 
at the web page coming from /var/www/html/areski , I get the following 
errors:
 
Can some body give me some hints where and 
what to check for this error?. I am looking for info on the changes we have to 
make for 
1) the database 
name
2) user 
name
3) 
password
4)connection name 
(server running postgresql)
 
in all the files 
involved in the application, so that it works.
 
Seshu 

---
Warning: pg_pconnect(): Unable to 
connect to PostgreSQL server: could not connect to server: Connection refused Is 
the server running on host localhost and accepting TCP/IP connections on port 
5432? . in /var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on 
line 68Database error: Link-ID == false, 
pconnect failedPostgreSQL Error: 0 ()Warning: 
pg_pconnect(): Unable to connect to PostgreSQL server: could not connect to 
server: Connection refused Is the server running on host localhost and accepting 
TCP/IP connections on port 5432? . in 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 
68Database error: Link-ID == false, pconnect 
failedPostgreSQL Error: 0 ()Warning: 
pg_errormessage(): supplied argument is not a valid PostgreSQL link resource in 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php on line 
101Warning: Cannot modify header information - headers 
already sent by (output started at 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in 
/var/www/html/areskicc/lib/module.access.php on line 
66Warning: Cannot modify header information - headers 
already sent by (output started at 
/var/www/html/areskicc/lib/DB-modules/phplib_postgres.php:68) in 
/var/www/html/areskicc/lib/module.access.php on line 
67
 




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RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems

[Kanuri, Seshu (Company IT)]

Alex,
 
Asterisk does not have a Outbound SIP Proxy. 
Remove any Proxy configuration from your Phone. I guess that part is called 
Registrar Server.
 
Omit that information here and it should 
work.
 
Seshu
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
ScheerschmidtSent: Monday, May 09, 2005 2:18 PMTo: 
'Thore'; 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: RE: [Asterisk-Users] Zyxel 2000W (WI-FI) 
Problems

Oh yeah, i forgot, do you hav installed the latest firmware 
? If not, download it and install.
My config (Zyxel 
phone):
 



  
  

  SIP PROXY
  



  
  

  
  

  

  
SIP URI
sip:  @ 10.0.0.10 : 
  5060 


  
  
SIP Server 
  Address
  
  
SIP Server 
  Port
  
  
Registrar Server 
  Address
  
  
Registrar Server 
  Port
  
  
Register Expiry 
  Time(sec.)
  
  
OPTIONS Interval 
  Timer
  
  
Session Expiry 
  Time(sec.)
  
  
Display 
Name
   
   
  

  

  
Authentication

  

  

  
Registrar 
  Username
   
   
  
Registrar 
  Password
   
   
  

  

  
Registration 
  Status
Registered



 



  
  

  PHONE SETTINGS
  



  
  

  

  
Default Voice 
  Codec
G.729, 8kG.711u, 
64kG.711a, 
64k
  
Speaking 
  Volume(-14~14)
-14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314  
  
Listening 
  Volume(-14~14)
-14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314  
  
RTP Port

  
Jitter 
  Buffer
Small  Medium 
   Large 
 
  
Voice Frames per 
  Packet
Small  Medium 
   Large 
 
  
DTMF 
Relay
disableinband(RFC2833)outband
  
DTMF 
  Payload(0~127)
  
  



  
  

   
 
 
Regards,
Alexander
 
 From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
ThoreSent: Monday, 09 May 2005 10:20To: 
ASTERIKSSubject: [Asterisk-Users] Zyxel 2000W (WI-FI) 
Problems


Hi!
 
Then 
I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer 
the phone I am ringing.
It 
works fine if I call the 2000W from other 
phones.
 
I 
have tried many sip settings. I use this 
now:
[205]
type=friend
username=205
secret=passwd205
callerid="Zyxel" 
<205>
host=dynamic
context=local
nat=yes
canreinvite=no
disallow=all
allow=g729
dtmfmode=rfc2833
 
 
 
 
Sip 
debug:
 headers, 0 
lines
Retransmitting #4 
(NAT):
SIP/2.0 407 Proxy Authentication 
Required
Via: 
SIP/2.0/UDP 
192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060
From: 
;tag=C8355813679C716AFCA
To: 
;tag=as3bcc72b4
Call-ID: 
[EMAIL PROTECTED]
CSeq: 
1 INVITE
User-Agent: Asterisk 
PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
REFER
Contact: 

Proxy-Authenticate: Digest realm="pbx.com", 
nonce="1bed12f1"
Content-Length: 0
 
 
 to 
60.64.250.254:5060
Retransmitting #5 
(NAT):
SIP/2.0 407 Proxy Authentication 
Required
Via: 
SIP/2.0/UDP 
192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060
From: 
;tag=C8355813679C716AFCA
To: 
;tag=as3bcc72b4
Call-ID: 
[EMAIL PROTECTED]
CSeq: 
1 INVITE
User-Agent: Asterisk 
PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
REFER
Contact: 

Proxy-Authenticate: Digest realm="pbx.com", 
nonce="1bed12f1"
Content-Length: 0 





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RE: [Asterisk-Users] Connecting 20+ asterisk servers together

[Kanuri, Seshu (Company IT)]

Vikram,

Instead of trying to be over-ambitious and try to connect 20 Asterisk
boxes together, why don't you try top connect three (3) of them together
first.

There may lie a plausible solution for you. If this is done, you may go
and string four of them together and so on and so forth.

Take the first step now.

Seshu Kanuri

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Sent: Monday, May 09, 2005 6:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Connecting 20+ asterisk servers together

I have 20+ asterisk servers and need to network them together so a phone
on any of the servers can call a phone on any other server without any
trouble.

I can think of IAX trunks between every server. So every server will
have an IAX trunk to every server and then prefix bases routing in the
dialplan for each server (I can give a number to each server and use
that as a prefix for that server). But I think this is a maintainance
nightmare and also a very bad approch does anyone have any better ideas,
Also should the phones be able to send rtp between each other or only
through the Asterisk server since if its through the asterisk server and
say an IAX trunk then the max number of calls can be controlled right. 

Can dundi or the switch statement help me get out of this mess ? 

 
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RE: [Asterisk-Users] FXO ATA?

[Kanuri, Seshu (Company IT)]

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty
Shackleford
Sent: Friday, May 06, 2005 2:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FXO ATA?

> Why not go with Multitech?   They are expensive, but great units.

Because they are ridiculously expensive. 
It is true that Multitech's VOIP gear is very good stuff. I've used it
and it "just works". But apparently, their marketing people haven't been
paying attention to the market and they are still using pricing that
reflects the market 5 years ago.

/End Quote/

I totall agree with that comment. Multitech is just a rip-off, when you
compare the products with others existing in the market.

Seshu 

 
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[Asterisk-Users] RE: [Asterisk-biz] voip VPN solution requirement

[Kanuri, Seshu (Company IT)]

[EMAIL PROTECTED] Wrote:
--
-Original Message-
>Check the IareaNet solution at http://www.iaraenet.net
>
>
I hope the "IareaNet solution" works better than the above link.

Paul,

Sorry for the typo. Try http://www.iareanet.net

Seshu 

 
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