Thank you ! It work for me.
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Kamailio (SER) - Development M
As guessed before, you are using forward() in your config, which does stateless
forwarding, next is the part of the config showing it:
```
...
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
t_check_trans();
Closed #1841.
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sr-dev@l
[kamailio.txt](https://github.com/kamailio/kamailio/files/2854168/kamailio.txt)
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This is my config.
#!KAMAILIO
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_PRESENCE
#!define WITH_ACCDB
#
# Kamailio (OpenSER) SIP Server v5.2 - default configuration script
# - web: https://www.kamailio.org
# - git: https://github.com/kamailio/kamailio
#
# D
Are you using t_relay() to send out the INVITE? Not using forward(), right?
If using t_relay(), I need all log messages printed by Kamailio with debug=3 in
kamailio.cfg from the moment of receiving the first INVITE till processing the
CANCEL. Attributes in CANCEL match the INVITE, so the other p
[root@kamailio log]# ngrep -d any -qt -W byline "sip" port 5060
interface: any
filter: ( port 5060 ) and (ip || ip6)
match: sip
U 2019/02/11 10:04:27.082271 214.22.198.51:5060 -> 38.149.172.163:5060 #1
INVITE sip:100@38.149.172.163:5060 SIP/2.0.
Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36
Can you get the incoming and outgoing SIP traffic for such case on kamailio
server? Either pcap or ngrep output, e.g.:
```
ngrep -d any -qt -W byline "sip" port 5060
```
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https
Hello,
I'm trying to set up kamailio as a simple load balancer proxy between VOIP
Providers
and a cluster of asterisk servers.
I have started with the default kamailio.cfg.
At the end, i got this up and running, but with one exception: if incoming
call is not answered by the asterisk end (i.e.
