One reason I didn't use sems is that it doesn't support G729 as a
media server. Did it change?
Reda
On 20 mars 2012, at 10:36, Juha Heinanen wrote:
> Reda Aouad writes:
>
>> But servers cost money to buy, host, and maintain.
>> This is why I would really love to see such functionality in Kamai
Thank you all for the answers. Finally, there is not a very big problem
if one side is disconnected because the other side will end dialog when
phone will be hang up
have a good day :)
--
Inutile d'imprimer ce mail
Le -10/01/-28163 22:59, Reda Aouad a écrit :
Not also that the recommend se
Reda Aouad writes:
> But servers cost money to buy, host, and maintain.
> This is why I would really love to see such functionality in Kamailio.
fine with kamailio if you don't have any media based services
(voicemail, announcements, conferencing, etc.), where you would need
something like sems a
Thanks Juha for the idea.
But servers cost money to buy, host, and maintain.
This is why I would really love to see such functionality in Kamailio.
May I suggest that for the next release ?
Reda
On Tue, Mar 20, 2012 at 10:12, Juha Heinanen wrote:
> Reda Aouad writes:
>
> > One solution if y
Not also that the recommend session timer value that UACs generally use by
default is 1800 seconds, or 30 minutes. Just in case you may accept it. To
me, it's not acceptable when customers are billed.
However, the risk is not so big. If one of the UACs is suddenly
disconnected, the other party wil
Reda Aouad writes:
> One solution if you really need to solve your problem would be to put a
> B2BUA in the SIP path, such as Asterisk or FreeSwitch. They enforce a
> maximum session timer which UACs can use to ping themselves every now and
> then, and Asterisk can even ping the UACs and terminate
Hello,
In SIP, session timers can be used to periodically ping the UAC (using a
re-INVITE or UPDATE) to know if it's alive or not. Then action can be taken
- terminating the call.
Kamailio has the SST (SIP Session Timer) module which only enforces a
minimum session timer value for UACs, but not a
Hi,
Yes that is the behaviour when the media isn't flowing through a regulatory
tool (in-terms it sees the media and know call is actually going on
rtpproxy/media-proxy) but in the absence of any such tool SIP server is
not aware that the call-media is still in progress or is dead ! so it
always
i guess it should time out...the other end...since it has no way of knowing
that the other end is no more present...
Regards,
Vineet Menon
On 20 March 2012 11:30, Rabary wrote:
> Hi mailing,
>
> Newbie to kamailio, I follow this tuto http://nil.uniza.sk/sip/**
> kamailio/adding-mysql-suppor
