Alexei:

Thanks for following up.  Let me know when the sound issues are resolved and 
I'll give it a try.  I'm assuming the sip numbers you provided are for a 
conference call, i.e., all attendees in the meeting call the same sip number 
and are bridged together.  Also I assume you are using Asterisk as the 
conference bridge.

I look forward to giving it a try.

Thanks so much.

Joseph

Joseph Karwat | CEO | 415-462-0263 | JellVideo | www.jellnet.com |  
JELL NETWORKS, INC. 
Enterprise Video Collaboration Solutions

Click Here to Register for your JellVideo Collaboration Account



-----Original Message-----
From: Alexei Fedotov [mailto:alexei.fedo...@gmail.com] 
Sent: Wednesday, September 05, 2012 4:42 AM
To: Joseph Karwat
Cc: openmeetings-user@incubator.apache.org; Timur Tleukenov
Subject: Re: SIP Integration

Hello Joseph, folks,

I'm back with instructions:

1. Enter to SIP test demo room.
2. Actions -> Show sip dialer.
3. Enter the number. The list of numbers is restricted to a few (the allowed 
number is +79139066442, you should enter it in the form 979139066442).
4. Call.

Our support got a call. We have tested sound today, it does not work right now 
(the call itself works). I'll ask Timur to fix the sound.


--
With best regards / с наилучшими пожеланиями, Alexei Fedotov / Алексей Федотов, 
http://dataved.ru/
+7 916 562 8095



On Fri, Aug 24, 2012 at 8:33 PM, Joseph Karwat <jkar...@jellnet.com> wrote:
>
> Alexei:
>
>
>
> Thanks for the added permissions.  But this installation does not have SIP 
> enabled.  I checked the configuration table and SIP in not enabled as far as 
> I could tell.  This whole thread was to find an example of an OM installation 
> using SIP.  As I missing something?  It seems like much effort was put into 
> the code to enable SIP, I’d like to see if SIP handles the audio any better.
>
>
>
> Does anyone know of a OM site that has SIP enabled?
>
>
>
> Thanks,
>
>
>
> Joseph
>
>
>
> Joseph Karwat | CEO | 415-462-0263 | JellVideo | www.jellnet.com |
>
> Jell Networks, Inc.
>
> Enterprise Video Collaboration Solutions
>
>
>
> Click Here to Register for your JellVideo Collaboration Account
>
>
>
>
>
> From: Alexei Fedotov [mailto:alexei.fedo...@gmail.com]
> Sent: Friday, August 24, 2012 12:33 AM
> To: Joseph Karwat; Денис Кандров
> Cc: openmeetings-user@incubator.apache.org
> Subject: Re: SIP Integration
>
>
>
> I've added maximum permissions.
>
>
>
> Denis,
>
> could you provide some instructions on how to use SIP on demo.dataved.ru?
>
>
> --
> With best regards / с наилучшими пожеланиями, Alexei Fedotov / Алексей 
> Федотов, http://dataved.ru/
> +7 916 562 8095
>
>
> On Thu, Aug 23, 2012 at 6:51 PM, Joseph Karwat <jkar...@jellnet.com> wrote:
>
> Alexei:
>
>
>
> I’m not seeing anything different in the “upgraded” account.  Where is the 
> SIP Menu item you mentioned?
>
>
>
> How do I set-up a conference using SIP?  Or does it happen automatically.
>
>
>
> Thanks,
>
>
>
> Joseph
>
>
>
>
>
>
>
> Joseph Karwat | CEO | 415-462-0263 | JellVideo | www.jellnet.com |
>
> Jell Networks, Inc.
>
> Enterprise Video Collaboration Solutions
>
>
>
> Click Here to Register for your JellVideo Collaboration Account
>
>
>
>
>
> From: Alexei Fedotov [mailto:alexei.fedo...@gmail.com]
> Sent: Thursday, August 23, 2012 3:59 AM
> To: Joseph Karwat; openmeetings-user@incubator.apache.org
> Subject: Re: SIP Integration
>
>
>
> > Does the upgraded account use SIP automatically for the audio stream or do 
> > I need to do something special to make it work.  I’m not seeing anything 
> > different than what I saw before when I logged on.
>
> I believe you should be able to make SIP calls via menu.
>
> > For SIP what package are you using. Asterisk?
>
> Yes, Asterisk is used as an end-point to red5sip
>
>
>
> --
> With best regards / с наилучшими пожеланиями, Alexei Fedotov / Алексей 
> Федотов, http://dataved.ru/
> +7 916 562 8095
>
> On Wed, Aug 22, 2012 at 8:42 PM, Joseph Karwat <jkar...@jellnet.com> wrote:
>
> Alexei:
>
>
>
> Thanks.  A couple questions:
>
>
>
> ·        Does the upgraded account use SIP automatically for the audio stream 
> or do I need to do something special to make it work.  I’m not seeing 
> anything different than what I saw before when I logged on.
>
> ·        For SIP what package are you using. Asterisk?
>
>
>
> Thanks,
>
>
>
> Joseph
>
> Joseph Karwat | CEO | 415-462-0263 | JellVideo | www.jellnet.com |
>
> Jell Networks, Inc.
>
> Enterprise Video Collaboration Solutions
>
>
>
> Click Here to Register for your JellVideo Collaboration Account
>
>
>
>
>
> From: Alexei Fedotov [mailto:alexei.fedo...@gmail.com]
> Sent: Wednesday, August 22, 2012 3:38 AM
> To: Joseph Karwat
> Subject: Re: SIP Integration
>
>
>
> Hello Joseph,
>
>
>
> I've upgraded your account.
>
>
>
>
> --
> With best regards / с наилучшими пожеланиями, Alexei Fedotov / Алексей 
> Федотов, http://dataved.ru/
> +7 916 562 8095
>
> On Tue, Aug 21, 2012 at 8:06 PM, Joseph Karwat <jkar...@jellnet.com> wrote:
>
> Alexi:
>
>
>
> If you have SIP working.  I’d like to give it a try.  Can you give me rights. 
>  I believe I signed up on your OM under jkar...@jellnet.com.
>
>
>
> Thanks,
>
>
>
> Joseph
>
>
>
> Joseph Karwat | CEO | 415-462-0263 | JellVideo | www.jellnet.com |
>
> Jell Networks, Inc.
>
> Enterprise Video Collaboration Solutions
>
>
>
> Click Here to Register for your JellVideo Collaboration Account
>
>
>
>
>
> From: Alexei Fedotov [mailto:alexei.fedo...@gmail.com]
> Sent: Tuesday, August 21, 2012 12:24 AM
>
>
> To: openmeetings-user@incubator.apache.org
> Subject: Re: SIP Integration
>
>
>
> demo.dataved.ru should have the integration
>
> I can upgrade your account with moderator permissions to test it
>
> please note, phone calls are not free
>
>
> --
> With best regards / с наилучшими пожеланиями, Alexei Fedotov / Алексей 
> Федотов, http://dataved.ru/
> +7 916 562 8095
>
> On Tue, Aug 21, 2012 at 11:16 AM, seba.wag...@gmail.com 
> <seba.wag...@gmail.com> wrote:
>
> Alexei has a demo server working with the SIP integration. I think they are 
> preparing the demo server for a presentation currently.
>
> Sebastian
>
>
>
> 2012/8/20 George Kirkham <gkirk...@co2crc.com.au>
>
> Joseph,
>
>
>
> You asked about “successfully integrated OM with SIP, preferably Asterisk”, I 
> agree that would be nice but I have not been able to understand enough to get 
> SIP working.
>
>
>
> The echo you speak about is an issue and will remain an issue unless all your 
> meeting participants are using echo cancelling speaker phones or headphones.
>
>
>
> Greatest cause of echo for me is people with laptops who are using inbuilt 
> speakers and microphone, any sound from their speakers just feeds straight 
> back into their microphone, causing so much echo problems.  But when everyone 
> is using echo cancelling speaker phones or headphones all is great.
>
>
>
> Thanks,
>
>
>
> George Kirkham
>
>
>
>
>
>
>
> From: Joseph Karwat [mailto:jkar...@jellnet.com]
> Sent: Tuesday, 21 August 2012 4:19 AM
> To: openmeetings-user@incubator.apache.org
> Subject: SIP Integration
>
>
>
> If anyone has successfully integrated OM with SIP, preferably Asterisk, I’d 
> be interested in your results.  Would it be possible to set-up vchat with me 
> so I can see how it works.  Does it deal well with eliminating the echo on 
> the audio.
>
>
>
> Thanks,
>
>
>
> Joseph
>
>
>
> Joseph Karwat | CEO | 415-462-0263 | JellVideo | www.jellnet.com |
>
> Jell Networks, Inc.
>
> Enterprise Video Collaboration Solutions
>
>
>
>
>
>
>
>
>
> --
> Sebastian Wagner
> https://twitter.com/#!/dead_lock
> http://www.webbase-design.de
> http://www.wagner-sebastian.com
> seba.wag...@gmail.com
>
>
>
>
>
>
>
>

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