I have a Kamailio and 2 asterisk servers. All users are created in both of
the asterisk servers. I am forwarding the registration to asterisk. The
problem is that it is always used on only one server from the list. Even if
one goes to shutdown, then there is not any registration sent to the
available server. Even if *some *of the extensions can be seen registered
in both of the asterisk's, if the secondary goes down, then there are no
services for the phones.
I am attaching the kamailio.cfg. My dispatch list is:
1 sip:192.168.0.100:5080 0 0 maxload=20
2 sip:192.168.0.101:5080 0 0 maxload=20
In both of the asterisk servers i am using sip.conf to create users and s
sip trunk for the Kamailio. Nothing special about it.
I am looking to find what is wrong with my config and i cannot
loadbalance/failover to the asterisk servers.
I have recently installed Siremis
And have set up a domain pointing to it, how do I configure it so registration requests to this domain are passed thru to the actual PBX
Adam
Hi
We have a question. We are preparing a setup to bridge the SIP call
from Kamailio IMS to PSTN, but we are puzzling how we can present the
P-Asserted-Identity in the SIP INVITE.
By default, the PAI is in sip:user@domain, but we believed we should
use<tel> format in the PAI when bridge to PSTN.
question is
1. should we have two PAI headers?, or
2. should we paste <tel> format in PAI User Part and domain in PAI
Host Part?, or
3. Should we only use one PAI header with tel format?
In our test setup, we are using FHoSS, and the IMPU sip:user@domain
have below entry in "IMPUs from Implicit-Set"
- sip:user@domain
- tel:number
Question is how we can extract the number from the HSS and presented in PAI?
- RBK
Trying to setup a lab where i would like tohave:
-kamailio (5.3 from deb repo) in front to load balance. Keep registrations.
IP: 192.168.1.100
-Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will
serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to
Kamailio in sip.conf
-Asterisk-2 (16 LTS). IP: 192.168.1.102. Identical config to Asterisk-1.
Tried to follow this
https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb,
but it is quite old and it does jot apply to newer versions of Asterisk and
Kamailio.
Up to jow i have the Kamailio setup and users can register on Kamailio
following the basic setup. I would like advice on setup and configuration,
to integrate Kamailio with Asterisk boxes
Some advice please?
Sincerely yours,
John
Trying to setup a lab where i would have:
-kamailio in front to load balance. Keep registrations. IP: 192.168.1.100
-Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will
serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to
Kamailio in sip.conf
-Asterisk-2 (16. IP: 192.168.1.102. Identical config to Asterisk-1.
Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.
0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to
newer versions of Asterisk and Kamailio.
Up to jow i have the Kamailio setup and users can register on Kamailio
following the basic setup. I would like advice on setup and configuration,
to integrate Kamailio with Asterisk boxes
Trying to setup a lab where i would have:
-kamailio in front to load balance. Keep registrations. IP: 192.168.1.218
-Asterisk _1. IP: 192.169.1.232.It will serve as media server for IVR, VM,
etc
-Asterisk-2. IP: 192.168.1.233. Identical config to Asterisk-1.
Tried to follow this
https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb,
but it is quite old and it does jot apply to newer versions of Asterisk and
Kamailio.
Up to jow i have the
I would like advice on setup and configuration,
Trying to setup Kamailio with Freeswitch in a Debian machine as in the
http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc, but
Kamailio fails to start and gives a ton of errors.
Before of the alterations Kamailio and Siremis start correctly and i can
easily create users through Siremis. After the changes to include
Freeswitch's blocks, it fails to start. All run in the same machine with IP
192.168.1.213.
I am attaching kamailio.cfg (named as test1.txt) if someone could spot the
error, please. Also i am attaching the kamailio.cfg(Original1.txt) when it
was working fine with Siremis.
Hi
During the testing with Kamailio IMS, we found that SUBSCRIBE and
NOTIFY message flow between UE and P-CSCF is shown below.
SUBSCRIBE and its response: 192.168.1.102:9101 (port-c) ->
192.168.2.66:6101 (port-s)
NOTIFY: 192.168.2.66:5101 (port-c) -> 192.168.1.102:9100 (port-s)
based on the sip security, is it the expected behavior that they are
using port-c and port-s?
The reason why we asked is because we found that the NOTIFY did not
properly propagate to the UE due to different port addresses.
Does anyone have this experience? and how we can resolve it?
- RBK
