27 Jul
2016
27 Jul
'16
5:29 a.m.
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!
27 Jul
27 Jul
8:54 a.m.
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!
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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!
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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5:15 p.m.
On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo wrote:
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.
I'd suggest to use rtpengine for all calls, it fixes most problems and
uses nearly no resources (with the kernel plugin)
10:40 p.m.
I added the #!define WITH_NAT option, and now the call can only be made one
way. RTPProxy was started like so:
$ rtpproxy -l 192.168.1.101 -s udp:localhost:7722 -u kamailio
root@kamailioA:~# netstat -pln | egrep "kamailio|rtpproxy"
tcp 0 0 192.168.1.101:5060 0.0.0.0:* LISTEN
10112/kamailio
tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN
10112/kamailio
udp 0 0 192.168.1.101:5060 0.0.0.0:*
10081/kamailio
udp 0 0 127.0.0.1:5060 0.0.0.0:*
10081/kamailio
udp 0 0 127.0.0.1:7722 0.0.0.0:*
10042/rtpproxy
raw 0 0 0.0.0.0:255 0.0.0.0:* 7
10081/kamailio
unix 2 [ ACC ] STREAM LISTENING 33357 10102/kamailio
/var/run/kamailio//kamailio_ctl
My full config is at
https://gist.github.com/ticklingcontest/e315972c80c82f6dfa23920c7725d60b
BTW, my entire setup, kamailio, asterisk and the phones etc. are in one
private network. I think setting realtime endpoint with "direct_media=no"
is pointless as all of these interactions are fronted by Kamailio.
What's going on here? Any help is appreciated.
On Wed, Jul 27, 2016 at 10:15 AM, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo
wrote:
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.
I'd suggest to use rtpengine for all calls, it fixes most problems and
uses nearly no resources (with the kernel plugin)
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
29 Jul
29 Jul
1:24 a.m.
Hello,
I didn't follow your previous tread, but I suppose you use Kamailio as
'frontdoors' gate and route all your calls to external network using
Kamailio. In that way you'd better use RTPEngine (
https://github.com/sipwise/rtpengine) installed on Kamailio machine running
with external/internal interfaces.
Easiest thing is to put inside LOCATION route something like that
if route(FROMASTERISK) {
rtpengine_manage(force trust-address direction=internal
direction=external);
}
else {
rtpengine_manage(force trust-address direction=external
direction=internal);
}
(in route[FROMASTERISK] put a check to be sure call is comeing from your
asterisk)
and also - yes, define WITH_NAT if you're using standart configuration
2016-07-27 22:40 GMT+03:00 Tickling Contest <tickling.contest(a)gmail.com>om>:
I added the #!define WITH_NAT option, and now the call
can only be made
one way. RTPProxy was started like so:
$ rtpproxy -l 192.168.1.101 -s udp:localhost:7722 -u kamailio
root@kamailioA:~# netstat -pln | egrep "kamailio|rtpproxy"
tcp 0 0 192.168.1.101:5060 0.0.0.0:*
LISTEN 10112/kamailio
tcp 0 0 127.0.0.1:5060 0.0.0.0:*
LISTEN 10112/kamailio
udp 0 0 192.168.1.101:5060 0.0.0.0:*
10081/kamailio
udp 0 0 127.0.0.1:5060 0.0.0.0:*
10081/kamailio
udp 0 0 127.0.0.1:7722 0.0.0.0:*
10042/rtpproxy
raw 0 0 0.0.0.0:255 0.0.0.0:* 7
10081/kamailio
unix 2 [ ACC ] STREAM LISTENING 33357 10102/kamailio
/var/run/kamailio//kamailio_ctl
My full config is at
https://gist.github.com/ticklingcontest/e315972c80c82f6dfa23920c7725d60b
BTW, my entire setup, kamailio, asterisk and the phones etc. are in one
private network. I think setting realtime endpoint with "direct_media=no"
is pointless as all of these interactions are fronted by Kamailio.
What's going on here? Any help is appreciated.
On Wed, Jul 27, 2016 at 10:15 AM, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
--
Alexandru Covalschi
VoIP engineer and system administrator
tel: +37367398493
On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo
wrote:
_______________________________________________
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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.
I'd suggest to use rtpengine for all calls, it fixes most problems and
uses nearly no resources (with the kernel plugin)
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3043
days inactive
3044
days old
5 comments
5 participants
participants (5)
-
Alexandru Covalschi -
Daniel Tryba -
Jurijs Ivolga -
SamyGo -
Tickling Contest