Hello,
The database tables definition for 5.2.x mentions the lcr_id parameter
being of unsigned short type, while in the code I only see definitions
of type int.
Is it safe to assume the parameter is of type int, so the database
column can also be set to an integer type?
Regards,
Grant
Hello,
we ran into a trouble with sharing variable into reply route...
We are going to match asterisk's uniqueid with rtpengine records like this:
route[from_asterisk]{
$var(UniqueId)=$hdr(X-UniqueId);
$var(rtp_flags) = $var(rtp_flags) + "
label={uid:$var(UniqueId),stream:outgoing} record-call=on";
rtpengine_manage($var(rtp_flags));
}
reply_route[from_pstn]{
rtpengine_manage("replace-origin replace-session-connection ICE=remove
RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing");
}
and it doesn't work... The one possible way what we found its using $shv
instead of $var... But its unsecure way, cuz it may confus our bussines
logic...
Do you have any idea how to implement this?
Many thanks!
Hello,
I want to debug kamailio with gdb or try to get core.dump file but i failed. GDB wants to kamailio symbols. How to install them?
on Ubuntu - raspberry PI
Best regards.
Yasin CANER
Hi,
We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances
running on Ubuntu..
1. 2 Sip client registers successfully (shows Online) as 'bob(a)net1.test'
and 'alice(a)net1.test'
2. When we check server logs: Invite request shows 'bob@IP address' i.e. '
bob(a)10.0.0.10'
3. However, in HSS it is registered as 'bob(a)net1.test'.
4. When the request from SCSCF goes to ISCSF it gives Error: Users - 604
Does not exist anywhere - HSS User Unknown. As it compares 'bob(a)10.0.0.10'
with 'bob(a)net1.test'.
Please help and advise how everywhere (SCSCF, PCSCF) during
query/comparisons it can take 'bob(a)net1.test' and not 'bob(a)10.0.0.10'
Thanks and Regards,
Jack.
Dear All,
I am evaluating the kamailio proxy 5.2 server with rtpproxy on an internal network.
Everything are ok.
Now, I would like to evalute Kamailio proxy from outside (internet) but I have some difficulties to setup my PfSense firewall to enable inbound & outbound rules, network settings and kamailio.cfg.
The issues are :
Setup correctly Network files :
* /etc/hostname
127.0.0.1 localhost
ip_private codecsip.mydomain codecsip
ip_public codecsip.mydomain codecsip
* /etc/hosts file
codesip
* dns
* /etc/resolv.conf
nameserver 8.8.8.8Setup /etc/kamailio/kamailio.cfg :#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_NAT
#!define WITH_TLS
/* uncomment the next line to disable the auto discovery of local aliases
based on reverse DNS on IPs (default on) */
#auto_aliases=no
/* add local domain aliases */
alias="codecsip.mydomain"
#!ifdef WITH_NAT
# ----- rtpproxy params -----
# modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")
Setup etc/default/rtpproxy :
CONTROL_SOCK=udp:127.0.0.1:7722
EXTRA_OPTS="-l <IP-address>"
Where <IP-address> is the external IP address of your host.
Regarding the firewall rules, it will be great to identify which rules I need to create : Inbound and outbound rules and NAT 1:1 ?
If anyone have any information it could be great
Best regards,
Youssef
Hi everyone
I use kamailio 5.1 with rtpengine and I would like to set up a simple
warm standby or hot standby system for my sip server. on the web I
found various discussions and posts that led me to deepen them
the dispatcher, dmq, dmq_usrloc, db_cluster and p_usrloc modules. now,
though, I'm a little confused:
what approach do you recommend for this system?
I thought of using two identical servers:
Kam01 10.10.10.2
Kam02 10.10.10.3
Hostname cluster.mydomain.com
each with its own mariadb and implementing data replication (with the
dmq module? Or db_cluster and sqlops?). in this scenario is it
necessary to also use the dispatcher module or implement a high
available system with keepalives and floating ip?
thanks, Christian
new to the area and trying to setup Kamailio with Asterisk in a single
machine. All users will register to Kamailio's port and in case of need for
media, it will be forwarded to Asterisk, that is my intention. All of my
work is based on the following link
https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
Here is what i have done:
- Debian 8, 64 bit machine with mysql and odbc
-
*root@kamast: ~ $ lsb_release -a No LSB modules are available. Distributor
ID: Debian Description: Debian GNU/Linux 8.11 (jessie) Release:
8.11 Codename: jessie root@kamast: ~ $ uname -a Linux kamast
3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64 GNU/Linux
root@kamast: ~ $ *
- Kamailio 5.2 installed from Kamailio's deb repository
- Asterisk 13LTS installed from source
- Used the same passwords such as kamailiorw and asterisk_password,
since this is a test system, for proof of concept.
I did import to the mysql>asterisk database 3 users 2200, 2201 and 2202.
Then created in sip.conf the same 3 users with the same credentials. Then
on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account to a
softphone. Problems:
- Cannot see the users in the Asterisk's cli, sip show peers
- I can see users only in Kamailio with kamctl ul show
- A call between the extensions goes to voicemail. It never reaches the
other destination eg 2200 calls 2201 and in Asterisk's console i am getting
a message that 2201 is absent and it goes to voicemail. The same with any
other extension.
Attached you can find:
1. Kamailio.cfg
2. Asterisk's sip.conf
3. Asterisk's extension.conf
4. The import that i have done to mysql for the user creation.
I would appreciate if someone could point me to the error and help me fix
it please?
Hi,
Testing on latest master deb nightly build from last night I've noticed
that when $au is not available, $Au is being set to $fu (uri) instead of
$fU (username).
Docs: https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables
...
$Au - Acc username: username for accounting purposes. It's a selective
pseudo variable (inherited from acc module). It returns auth username ($au)
if exists or From username ($fU) otherwise.
...
I have the following log in several places in a test box:
xlog("L_NOTICE", "[DEBUG] Au=$Au au=$au aU=$aU fU=$fU fu=$fu - \n");
And when I run through them (on both request and reply routes) I get the
following:
Aug 8 14:37:43 csbc01 csbc[9478]: NOTICE: {1 21564 REGISTER
d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain
au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain -
Aug 8 14:37:43 csbc01 csbc[9479]: NOTICE: {1 21565 REGISTER
d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain
au=1002366 aU=1002366 fU=1002366 fu=sip:1002366@some.domain -
Aug 8 14:37:43 csbc01 csbc[9484]: NOTICE: {2 21565 REGISTER
d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null>
aU=<null> fU=1002366 fu=sip:1002366@some.domain -
Those 3 logs belong to the same flow from different places in the config
script.
I would expect $Au to never have a "@some.domain" in it, so I assume in
some place the replacement is not being done with $fU but with $fu.
@Devs: I'm happy to create a GH issue for tracking etc if required, please
let me know!
Cheers,
Joel.
Hi,
I want to show call logs in my SIP app. How can I do this with Kamailio? I
want to do it from the *kamailio.cfg *file. Is it possible to do it from
where? I guess Dialog module should be used for this, but can not figure
out how to do it. I want minimalist feature like who call whom with
timestamp, thats all. How can I do this?