Is there a way to obtain the effect of msg_apply_changes in a branch route?
I want to do:
route[BRANCHMANAGE] {
rtpengine_manage()
msg_apply_changes()
// further mangle the SDP with textops for
// obtuse UAs
}
The reason for this, is that after forking, I have some UAs that are
extremely picky about SDP, and I need to mangle the body to make them
happy. (4000 char limit, 68 attribute limit on SDP body - ever seen
that? Compared with the absolutely gigantic OFFERs from WebRTC
signalling libraries...)
Now rtpengine_manage() and textops are working on separate copies of
the msg body and the results don't stack correctly.
Cheers
Anthony Alba
Thanks for the link.
One issue I've noticed:
If you have an empty comment line (just # on a single line), then the next line is wrongly highlighted as comments.
For example:
#
loadmodule "db_postgres.so"
> i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode.
> its great!
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Hi Patrick,
better to contact our sr-users list with the usage related questions, added to CC.
Have a look to the SDP of the SIP packets to see if it contains the correct IP would be one idea to debug this further.
Feel free to ask again on sr-users after you have got more details.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-dev <sr-dev-bounces(a)lists.kamailio.org> On Behalf Of Patrick Leybag
Sent: Wednesday, November 25, 2020 6:26 AM
To: sr-dev(a)lists.kamailio.org
Subject: [sr-dev] kamailio SIP and RTP proxy
Hi, Can someone help me?
I self host a kamailio using my raspberry pi as a load balancer for my two asterisk servers and get a did number. when I call to my DID number it points to my kamailio and kamailio will distribute to asterisk server but the call has no audio. I tried port forwarding ports 5060 for SIP and 10000-20000 for RTP but it still does not work.
Any help is much appreciated. Thank you in advance
Hi,
I am having a problem on fail overing my master kamailio server and slave
kamailio server.
I use kamailio as a load balancer for my asterisk servers. And it is
working with calls, but when I added failover with keepalived to the slave,
there is no audio during calls.
Thank you,
Rolly
Hello,
I have a kamailio server running behind HAProxy with proxy protocol v2 enabled.
In Kamailio I have set the parameter tcp_accept_haproxy=yes and loaded tcpops module.
UEs are registered using TLS and kamailio sees that the message has received from their real ip address + port and not HAProxy ip + port.
When UE A calls UE B, kamailio is trying to reach UE B using his real ip address and port instead of HAProxy IP address + port.
I know I can get the tcp ip and port of HAProxy using $tcp(c_si) and $tcp(c_sp) but I can’t make it work.
What is the right way to do this? How should I use these variables properly in order to establish the call successfully?
Thanks,
Joey.
