[OpenSIPS-Users] OpenSIPS handling B2B features

[Rick Wilson]

Asterisk has other timing sources from pthreads and the file system.  
The file system interface seems to work pretty well on my vm.

I do not use it very much though.  So, your mileage may vary.

On 1/27/2011 4:52 PM, Dave Singer wrote:
> Toyima,
>
> Asterisk is a pain for making conference work because of its
> dependence on the dahdi kernel driver for timing.
> I like the FreeSwitch implementation. FreeSwitch has a very active irc
> at irc.freenode.net.
>
> Dave
>
> On Thu, Jan 27, 2011 at 12:21 AM, Toyima Dias<toyimads at gmail.com>  wrote:
>> Anca,
>>
>> What conference system would you recommend for me? Asterisk? SEMS? may be some licensed software could be ok for me...i would like to heard some recommendations
>>
>> Regards
>>
>> 2011/1/26 Anca Vamanu<anca at opensips.org>
>>> Hi Toyima,
>>>
>>> On 01/25/2011 06:38 PM, Toyima Dias wrote:
>>>> Thanks Anca,
>>>> To make transfers possibles, i would use REFER messages, and for this i would need to use B2BUA module of opensips, rigth?
>>> VoIP phones can handle REFER messages, the problem is with the gateways, and this is why sometimes it is required to handle the REFER at the server and instead of forwarding it, replace it with connecting the party in a call to the new party from the server.
>>>> What about conferences? For conferences is the same procedure (REFER messages), but always handling a media b2bua, rigth?
>>> For conferences it is not enough to handle the signaling, you also need a conference bridge to distribute the media to more parties.
>>>
>>> Regards,
>>>
>>> --
>>> Anca Vamanu
>>> www.voice-system.ro
>>>
>>>
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