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From Alexei Fedotov <alexei.fedo...@gmail.com>
Subject Re: SIP connectivity
Date Thu, 14 Feb 2013 10:27:50 GMT
Hello folks,

Timur succesfully demonstrated Asterisk integration with Openmeetings 1.9
after three months if work at the end of 2011.

We failed to make the same things working for om 2.0 by spending two weeks
because integration success strongly deoends on asterisk & flash  versions,
network & asterisk configuration.

Another two months and I've seen SIP working again, though reliability and
sound quality were not good enough.

All are welcome to test the cuurent trunk and provide more details because
these two types of problems are hard to localise.
14.02.2013 3:04 пользователь "Maxim Solodovnik" <solomax666@gmail.com>
написал:

> 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 <asannucci@gmail.com> 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 escribió:
>>
>>  Dear all, ****
>>
>>  ****
>>
>>  ****
>>
>>  ****
>>
>> i have tested the asterisk sip integration as documented with the most recent instruction
>> (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works
>> 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" <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 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 <solomax666@gmail.com>wrote:
>>>
>>>> OK will add it and notify you
>>>>  On Jan 31, 2013 5:05 PM, "Bart Coninckx" <bart.coninckx@telenet.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 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 good
>>>>>> 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’ll find the .sql files required for
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<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+table+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
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.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<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
>>>>>> 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.
>>>>>>
>>>>>>
>>>>>>
>>>>>> 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-integration.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.
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.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>  --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>
>>>
>>>  --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>
>
>
> --
> WBR
> Maxim aka solomax
>

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