Return-Path: X-Original-To: apmail-openmeetings-user-archive@www.apache.org Delivered-To: apmail-openmeetings-user-archive@www.apache.org Received: from mail.apache.org (hermes.apache.org [140.211.11.3]) by minotaur.apache.org (Postfix) with SMTP id 5EB14E47B for ; Wed, 13 Feb 2013 23:03:59 +0000 (UTC) Received: (qmail 82867 invoked by uid 500); 13 Feb 2013 23:03:59 -0000 Delivered-To: apmail-openmeetings-user-archive@openmeetings.apache.org Received: (qmail 82835 invoked by uid 500); 13 Feb 2013 23:03:59 -0000 Mailing-List: contact user-help@openmeetings.apache.org; run by ezmlm Precedence: bulk List-Help: List-Unsubscribe: List-Post: List-Id: Reply-To: user@openmeetings.apache.org Delivered-To: mailing list user@openmeetings.apache.org Received: (qmail 82827 invoked by uid 99); 13 Feb 2013 23:03:59 -0000 Received: from athena.apache.org (HELO athena.apache.org) (140.211.11.136) by apache.org (qpsmtpd/0.29) with ESMTP; Wed, 13 Feb 2013 23:03:59 +0000 X-ASF-Spam-Status: No, hits=1.8 required=5.0 tests=FREEMAIL_ENVFROM_END_DIGIT,HTML_MESSAGE,RCVD_IN_DNSWL_LOW,SPF_PASS,WEIRD_PORT X-Spam-Check-By: apache.org Received-SPF: pass (athena.apache.org: domain of solomax666@gmail.com designates 209.85.210.48 as permitted sender) Received: from [209.85.210.48] (HELO mail-da0-f48.google.com) (209.85.210.48) by apache.org (qpsmtpd/0.29) with ESMTP; Wed, 13 Feb 2013 23:03:54 +0000 Received: by mail-da0-f48.google.com with SMTP id v40so758559dad.35 for ; Wed, 13 Feb 2013 15:03:34 -0800 (PST) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20120113; h=mime-version:x-received:in-reply-to:references:date:message-id :subject:from:to:content-type; bh=rymoGLEFPfCLjq+oK7qboY8XSH3/rSz1ioYRxK5Ix7U=; b=yiSVzbFXjg0XwCwIZS+Qcc5JvKLTPCqKn7OjT0yP9tIwRshr4gmOdGcRuqz9ntgrsW UQTaI0KeiNoNJzirIOkKcaDYX3K8WPdOwBxQFKgVYslgPuTqkZCllanygxTR0pcgz2Xn VF5i2BVcc39DxU57nGqs3ZwYGnu3wVznPnlL5pvZo+7VGwyS/05r+ErJdtUEKr9i8BSx A9vkkp0ZTSvUNziq7BK4lVt53xjDdDZdtNSEnQXU1Co25/6g3xVaKBaqiYagFZ3jB7wU YPkB4hrZXkdxj7yiVIAUlx8SVab1dGD/tZjUsCzZ+FoyQ4y5OGSMkM4SFiSLx3r9S/4j XuwA== MIME-Version: 1.0 X-Received: by 10.66.89.199 with SMTP id bq7mr68402134pab.26.1360796609760; Wed, 13 Feb 2013 15:03:29 -0800 (PST) Received: by 10.66.16.72 with HTTP; Wed, 13 Feb 2013 15:03:29 -0800 (PST) In-Reply-To: <511C0DFC.3080208@gmail.com> References: <5106EF5D.5030601@telenet.be> <2B93465B6F724E45BCF490E30DA2FF6701FCBAFA@INFHYDEX10MB2.corp.infotech-enterprises.com> <5106F125.4020502@telenet.be> <51079B9A.8050108@telenet.be> <51079E2D.1080004@telenet.be> <2B93465B6F724E45BCF490E30DA2FF6701FCBDE2@INFHYDEX10MB2.corp.infotech-enterprises.com> <51098E31.7010604@telenet.be> <2B93465B6F724E45BCF490E30DA2FF6701FCCB04@INFHYDEX10MB2.corp.infotech-enterprises.com> <51099473.9010101@telenet.be> <510A41DC.3000806@telenet.be> <5115805D.1010707@telenet.be> <84A5A60234E2484892D29C78CC4414FD20E8DA@SBS-MAIL.datus.local> <511C0DFC.3080208@gmail.com> Date: Thu, 14 Feb 2013 06:03:29 +0700 Message-ID: Subject: Re: SIP connectivity From: Maxim Solodovnik To: Openmeetings user-list Content-Type: multipart/alternative; boundary=f46d042ef591894fcb04d5a32626 X-Virus-Checked: Checked by ClamAV on apache.org --f46d042ef591894fcb04d5a32626 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: quoted-printable I never tried asterisk integration in 2.0, it was reported to be very unstable 2.1 can be downloaded from here https://builds.apache.org/job/openmeetings/= (it is not released yet) On Thu, Feb 14, 2013 at 5:04 AM, Bakko wrote: > Hello, > > two questions. > > Asterisk integration working on openmeetings 2.0? > > Where can I download 2.1 version > > Thank you > > Regards > > > El 13/02/2013 14:59, Naderi, Sascha escribi=C3=B3: > > Dear all, **** > > **** > > **** > > **** > > i have tested the asterisk sip integration as documented with the most re= cent instruction > (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it work= s > just fine.**** > > **** > > The only thing i am missing is a way to get this working when > i choose to rename the openmeetings context from > http://yourcorp.com:5080/openmeetings to > http://yourcorp.com:5080/yourmeetings **** > > **** > > Which settings do i have to modify > so that red5sip functions even if the context name is changed?**** > > ** ** > > > Regards > Sascha Naderi > > > ------------------------------ > *Von:* Maxim Solodovnik [solomax666@gmail.com] > *Gesendet:* Samstag, 9. Februar 2013 02:32 > *Bis:* Bart Coninckx > *Cc:* user > *Betreff:* Re: SIP connectivity > > All tables are created by OM automatically > On Feb 9, 2013 5:46 AM, "Bart Coninckx" wrote: > >> May I add that a portion is missing, since one explains how to >> configure Asterisk for Realtime, but one does not stipulate how to creat= e >> the necessary tables. >> It's in my CentOS docs however (which I hope to post shortly). >> >> BC >> >> On 01/31/13 13:05, Maxim Solodovnik wrote: >> >> Hello Bart, >> >> I just take a look at your URL ... >> OM does not create/use sipfriends DB table (at least from version 2.1) >> only meetme table is used >> >> so I'm afraid there is nothing to change here >> >> Here is the most recent instruction: >> http://openmeetings.apache.org/red5sip-integration_2.1.html >> >> Will ask our SIP guru to review it one more time :) >> >> >> >> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik = wrote: >> >>> OK will add it and notify you >>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" >>> wrote: >>> >>>> It is for Asterisk 11 - don't know for other versions. You probably >>>> have no issues because of the 1.8 version. To be sure the .sql files i= n the >>>> Asterisk source should be compared across versions. >>>> >>>> this one is missing: >>>> >>>> `useragent` varchar(20) DEFAULT NULL, >>>> >>>> complete list (I think) is on: >>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table= +structure >>>> >>>> >>>> If I bump into others, I'll report ASAP, >>>> >>>> >>>> BC >>>> >>>> >>>> >>>> On 01/31/13 06:21, Maxim Solodovnik wrote: >>>> >>>> Is the OM meetme table incomplete? >>>> My asterisk reports no issues :( >>>> >>>> could you provide me with missing fields and I'll add it. >>>> My purpose was to create table with required fields only. >>>> >>>> >>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx < >>>> bart.coninckx@telenet.be> wrote: >>>> >>>>> Openmeetings installed them for me, that's why I ended up with >>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a g= ood >>>>> idea to have 'em removed from the install procedure. >>>>> >>>>> BC >>>>> >>>>> >>>>> On 01/30/13 22:30, Jeff Clay wrote: >>>>> >>>>> Bart, >>>>> >>>>> >>>>> >>>>> If you look in the source directory of your asterisk tar file, under >>>>> contrib/realtime/mysql you=E2=80=99ll find the .sql files required fo= r all the >>>>> realtime drivers. I never thought to use the ones with OM. >>>>> >>>>> >>>>> >>>>> Jeff Clay >>>>> >>>>> Network Administrator >>>>> >>>>> Infotech Enterprises America >>>>> >>>>> 870-215-5506 >>>>> >>>>> Ext. 1506 >>>>> >>>>> >>>>> >>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be] >>>>> >>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM >>>>> *To:* user@openmeetings.apache.org >>>>> *Cc:* Jeff Clay >>>>> *Subject:* Re: SIP connectivity >>>>> >>>>> >>>>> >>>>> Well, >>>>> >>>>> I might have found one difference though: >>>>> >>>>> >>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+tabl= e+structure >>>>> dictates how the table should look like. I obviously used the one in = the >>>>> openmeetings mysql database, but this one seems to miss the table >>>>> "useragent". I discovered this because it showed up in the logfiles. >>>>> >>>>> BC >>>>> >>>>> On 01/29/13 14:41, Jeff Clay wrote: >>>>> >>>>> Bart, >>>>> >>>>> >>>>> >>>>> From an asterisk configuration standpoint there are very few >>>>> differences between 1.8.x and 11.x. If memory serves, the only major >>>>> changes that I ran into (in my production environment) was changes to= SIP >>>>> NAT values and the behavior of app_page() now uses confbridge instead= of >>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a majo= r >>>>> overhauling. There were of course many other changes and bug fixes, y= ou can >>>>> skim through the change log for full details, but I think that was th= e jist >>>>> of it. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Jeff Clay >>>>> >>>>> Network Administrator >>>>> >>>>> Infotech Enterprises America >>>>> >>>>> 870-215-5506 >>>>> >>>>> Ext. 1506 >>>>> >>>>> >>>>> >>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be] >>>>> >>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM >>>>> *To:* Maxim Solodovnik >>>>> *Cc:* user >>>>> *Subject:* Re: SIP connectivity >>>>> >>>>> >>>>> >>>>> I see - I'm willing to try the 11 version in the next fiew days if >>>>> desired. >>>>> >>>>> BC >>>>> >>>>> >>>>> On 01/29/13 10:57, Maxim Solodovnik wrote: >>>>> >>>>> I test the integration using >>>>> >>>>> Asterisk 1.8.13.1 (Ubuntu 12.10) >>>>> >>>>> >>>>> >>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx < >>>>> bart.coninckx@telenet.be> wrote: >>>>> >>>>> That is amazing - I initially tried to do the same thing by using the >>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server. >>>>> >>>>> Are you guys using Asterisk 11? This version is the newest LTS versio= n >>>>> and has the best video capabilities. >>>>> >>>>> Cheers, >>>>> >>>>> BC >>>>> >>>>> >>>>> On 01/29/13 02:44, Maxim Solodovnik wrote: >>>>> >>>>> red5sip will create special OM user in the room: "SIP Transport" >>>>> >>>>> after that you can call to the OM room using SIP hard or soft phone. >>>>> >>>>> >>>>> >>>>> We are currently testing it and trying to add video capabilities ... >>>>> >>>>> >>>>> >>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx < >>>>> bart.coninckx@telenet.be> wrote: >>>>> >>>>> Hi Jeff, >>>>> >>>>> In fact, I saw both pages, but none explain what they set up to do, >>>>> just a bunch of command line instructions are given. >>>>> Your "OM will create a meetme meeting as configured in the realtime >>>>> meetme database" actually says it all in one go :-) >>>>> >>>>> cheers, >>>>> >>>>> BC >>>>> >>>>> >>>>> >>>>> >>>>> On 01/28/13 22:38, Jeff Clay wrote: >>>>> >>>>> Bart, >>>>> >>>>> OM will create a meetme meeting as configured in the realtime meetme >>>>> database. Have you read this page >>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integrati= on.html ? You might also check out >>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume >>>>> this is the one you're already referring to. >>>>> >>>>> Jeff Clay >>>>> Network Administrator >>>>> Infotech Enterprises America >>>>> 870-215-5506 >>>>> Ext. 1506 >>>>> >>>>> -----Original Message----- >>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be] >>>>> Sent: Monday, January 28, 2013 3:36 PM >>>>> To: user@openmeetings.apache.org >>>>> Subject: SIP connectivity >>>>> >>>>> Hi, >>>>> >>>>> I noticed some documentation on how to connect OM with a SIP proxy or >>>>> server, more particularly with the MeetMe application in Asterisk. >>>>> >>>>> The exact goal or purpose is not mentionned however. Will OM callout >>>>> to a MeetMe conference? Or is it the other way round? >>>>> >>>>> >>>>> Cheers, >>>>> >>>>> Bc >>>>> >>>>> ________________________________ >>>>> >>>>> DISCLAIMER: >>>>> >>>>> This email may contain confidential information and is intended only >>>>> for the use of the specific individual(s) to which it is addressed. I= f you >>>>> are not the intended recipient of this email, you are hereby notified= that >>>>> any unauthorized use, dissemination or copying of this email or the >>>>> information contained in it or attached to it is strictly prohibited.= If >>>>> you received this message in error, please immediately notify the sen= der at >>>>> Infotech and delete the original message. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> WBR >>>>> Maxim aka solomax >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> WBR >>>>> Maxim aka solomax >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> -- >>>> WBR >>>> Maxim aka solomax >>>> >>>> >>>> >> >> >> -- >> WBR >> Maxim aka solomax >> >> >> > --=20 WBR Maxim aka solomax --f46d042ef591894fcb04d5a32626 Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: quoted-printable
I never tried asterisk integration in 2.0, it was reported= to be very unstable

2.1 can be downloaded from he= re=C2=A0https://bui= lds.apache.org/job/openmeetings/=C2=A0(it is not released yet)


On Thu,= Feb 14, 2013 at 5:04 AM, Bakko <asannucci@gmail.com> wrot= e:
=20 =20 =20
Hello,

two questions.

Asterisk integration working on openmeetings 2.0?

Where can I download 2.1 version

Thank you

Regards


El 13/02/2013 14:59, Naderi, Sascha escribi=C3=B3:
=20 =20

= Dear= =C2=A0all, =20

= =C2=A0

= =C2=A0

= =C2=A0

= i=C2=A0have=C2=A0tested=C2=A0the=C2=A0asterisk=C2=A0sip=C2=A0integration=C2= =A0as=C2=A0documented=C2=A0with=C2=A0the=C2=A0most=C2=A0recent=C2=A0instruc= tion (http://openmeetings.apache.org/red5sip-integr= ation_2.1.html)=C2=A0and=C2=A0it=C2=A0works just fine.

=C2=A0

The=C2=A0only=C2=A0thing i am=C2=A0missing=C2=A0is a=C2=A0way=C2=A0to=C2=A0get=C2=A0this= =C2=A0working=C2=A0when i=C2=A0choose=C2=A0to=C2=A0rename=C2=A0the=C2=A0ope= nmeetings=C2=A0context=C2=A0from http://yourcorp.com:5080/openmeeting= s=C2=A0=C2=A0to http://yourcorp.com:5080/yourmeeting= s

=C2=A0

Which=C2=A0settings do i=C2=A0have=C2=A0to=C2=A0modify so=C2=A0that=C2=A0red5sip=C2= =A0functions=C2=A0even=C2=A0if=C2=A0the=C2=A0context=C2=A0name=C2=A0is changed?

=C2=A0=

=C2=A0

Sascha Naderi


Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Samstag, 9. Februar 2013 02:32
Bis: Bart Coninckx
Cc: user
Betreff: Re: SIP connectivity

All tables are created by OM automatically

On Feb 9, 2013 5:46 AM, "Bart Coninckx" <bart.coninckx@telenet.be> wrote:
May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables.
It's in my CentOS docs however (which I hope to pos= t shortly).

BC

On 01/31/13 13:05, Maxim Solodovnik wrote:
Hello Bart,

I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used

so I'm afraid there is nothing to change her= e

Here is the most recent instruction:

Will ask our SIP guru to review it one more time :)



On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <solomax666@gmail.com> wrote:

OK will add it and notify you

On Jan 31, 2013 5:05 PM, "Bart Coninckx" <bart.coninckx@tel= enet.be> wrote:
It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions.

this one is missing:

`useragent` varchar(20) DEFAULT =
NULL,

complete list (I think)  is on:

https://wiki.asterisk.org/wiki/display/=
AST/SIP+Realtime%2C+MySQL+table+structure

If I bump into others, I'll repor= t ASAP,


BC



On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(

could you provide me with missing fields and I'll add it.
My=C2=A0purpose=C2=A0was to cr= eate table with required fields only.


On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <bart.coninckx@telenet.be> wrote:
Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure.

BC


On 01/30/13 22:30, Jeff Clay wrote:

Bart,

=C2=A0

If you look in the source directory of your asterisk tar file, under contrib/realtime/= mysql you=E2=80=99ll fi= nd the .sql files required for all the realtime drivers. I never thought to use the ones with OM.

=C2=A0

Jeff Clay

Network Administrator

Infotech Enterprises America

870-215-5506

Ext. 1506

=C2=A0

From: Bar= t Coninckx [mailto:bart.coninckx= @telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetin= gs.apache.org
Cc: Jeff Clay
Subject: Re: SIP connectivity

=C2=A0

Well,

I might have found one difference though:

https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%= 2C+MySQL+table+structure=C2=A0 dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent&q= uot;. I discovered this because it showed up in the logfiles.

BC

On 01/29/13 14:41, Jeff Clay wrote:

Bart,

=C2=A0

From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.

=C2=A0

=C2=A0

=C2=A0

Jeff Clay

Network Administrator

Infotech Enterprises America

870-215-5506

Ext. 1506

=C2=A0

From: Bar= t Coninckx [mailto:bart.coninckx= @telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity

=C2=A0

I see - I'm willing to try the 11 version in the next fiew days if desired.

BC


On 01/29/13 10:57, Maxim Solodovnik wrote:

I test the integration using=C2=A0

Asterisk 1.8.13.1 (Ubuntu 12.10)

=C2=A0

On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <bart.coninckx@tel= enet.be> wrote:

That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.

Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.

Cheers,

BC


On 01/29/13 02:44, Maxim Solodovnik wrote:

red5sip will create special OM user in the room: "SIP Transport"

after that you can call to the OM room using SIP hard or soft phone.

=C2=A0

We are currently testing it and trying to add video capabilities ...

=C2=A0

On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <bart.coninckx@tel= enet.be> wrote:

Hi Jeff,

In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM wil= l create a meetme meeting as configured in the realtime meetme database" actually says it all in one go =C2=A0:-)

cheers,

BC




On 01/28/13 22:38, Jeff Clay wrote:

Bart,

OM will create a meetme meeting as configured in the realtime meetme database. =C2=A0Have you re= ad this page =C2=A0<= a href=3D"https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integ= ration.html" target=3D"_blank">https://cwiki.apache.org/OPENMEETINGS/openme= etings-asterisk-integration.html =C2=A0? =C2=A0 Yo= u might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.

Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506

-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telene= t.be]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apac= he.org
Subject: SIP connectivity

Hi,

I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.

The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?


Cheers,

Bc

________________________________

DISCLAIMER:

This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.

=C2=A0



=C2=A0

--
WBR<= br> Maxim aka solomax

=C2=A0



=C2=A0

--
WBR
Maxim aka solomax

=C2=A0

=C2=A0





--
WBR
Maxim aka solomax




--
WBR
Maxim aka solomax





--
WBR
Maxim= aka solomax
--f46d042ef591894fcb04d5a32626--