27 Jul
2016
27 Jul
'16
9:21 a.m.
Hi Tickling,
Could you please share with us your working config?
With kind regards,
Jurijs
On Wed, Jul 27, 2016 at 8:54 AM, SamyGo <govoiper(a)gmail.com> wrote:
Hi Again,
You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then
depending upon how your clients will interact with asterisk you may or may
not need a media proxy, like RTPproxy. If asterisks can send/receive media
directly from the internet then its ok for now, else you definitely need to
have rtpproxy/rtpengine in there.
Regards,
Sammy
On Tue, Jul 26, 2016 at 10:29 PM, Tickling Contest <
tickling.contest(a)gmail.com> wrote:
With the help of members from this mailing list
(many thanks!), I finally
got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make
a call using the setup that looks like this:
[Kamailio 4.4.2]<->[Asterisk 13.7.2]
Kamailio manages REGISTERs, but also forwarding them to Asterisk.
I am able to make a call, but I get only one way audio or no audio
depending on which client made the call (SipDroid->Zoiper I hear one way
audio on Zoiper, but no audio if the call is made the other way). I noticed
that Kamailio forced direct media between the endpoints in this situation,
but my application really needs Asterisk to handle it.
How do I do this? Should I start by forwarding INVITEs to Asterisk? How
do I do that?
Any help is appreciated.
Thanks!
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