Hello there,
I'm evaluating if I can use dial-plan and dispatcher module to build a
kamailio routing server to route calls based on number dialed(sometimes the
number dialed can be Alphanumeric).
I'm expecting have lot of dial-plan entries, some of them with
manipulations, others without.
Anyone can see here any limitation on this implementation?
One thing that I have noticed is that dial-plan module enforces me to fill
the fields substr_exp and repl_exp even if i don't have any manipulations
to apply.
I'm also concerning about performance issues with lot of dialplan entries.
What are your thoughts about this solution?
Thank you
--
José Seabra
Hi all
Follow scenario
Class5 system [c5] --> Loadbalancer kamailio (dispatcher module) [lbl] ---> gateway kamailio [gw] --> carrier [carr]
I get Invites from [c5] with
Request ,To, from, contact, pid in national format 0794445566
[lbl] dispatches this to [gw]
For the [carr] I need international format.
So doing these transactions in [gw]
And sending to [carr] in international format
Request, to, from, contact, ... => 417794445566
Everything ok
Then I get a 100, 183 and even 200 from [carr]
Ack is coming from [c5] to [lbl] and [gw] - but then it stocks
The ACK is not sent to the [carr]
I kamailio log I see
DEBUG: RFC3261 transaction matching failed
DEBUG: t_lookup_request: no transaction found
So for me, the ACK cannot be assigned to a transaction and gets discarded by
if ( is_method("ACK") ) {
xlog(,"L_INFO", "WITHINDLG ACK - not loose route\n");
if ( t_check_trans() ) {
xlog(,"L_INFO", "WITHINDLG ACK - t_check_trans() \n");
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
xlog(,"L_INFO", "WITHINDLG ACK - not t_check_trans() DISCARD!!\n");
# ACK without matching transaction ... ignore and discard
route(NATMANAGE);
#t_relay();
#exit;
Any idea?
Problem with modifying the sip tags? Or problem with the dialog?
Thanks for helping
OIi
Hey all,
Back with more questions.
I'm using Kamailio to make an HTTP call to my API to perform authentication
and message routing. Currently, I'm trying to build up the post body that I
send to my API to make those decisions.
I've cherry picked a few of the headers that are important in my routing
decisions. But, ideally, I'd like to just iterate over all the SIP message
headers and append them as request parameters to my API call. Is this
possible with Kamailio? I've been looking through the docs and can't seem
to find a function to iterate the full list of headers.
Thanks in advance.
Best,
Colin
Good day! Can anyone point me to right direction? The question is - I
want to get list of registered users (to find out who is online at the
moment) from sip server. I'm using kamailio on server side and exosip
library on client side.
I tried to find something in RFC 3261 and failured... Can someone
help and tell what to do or where to look for the answer?
Hello there,
The Pseudo-Variablest hat store information about P-Asserted-Identity are
all as set null:
$pU=<null>
$pd=<null>
$pn=<null>
Is there any known reason for that?
The kamailio version that I'm using this 4.4.1
Thank you for you support
Best regards
--
José Seabra
I am interested in opinions and suggestions about load balancing with
Kamailio. I work for an ITSP that currently uses Oracle & Broadsoft, and I
am working to design and develop an open source solution using Kamailio
(Proxy, Registrar, LB) & other Application/Media servers for more
flexibility and freedom :) Thank you ALL for the work you have put in on
Kamailio!
After much reading and configuration I have a Kamailio 'proxy' setup, with
endpoints registered using the Registrar module, and calls being
sent/received to/from the PSTN. I am interested in separating the Load
Balancers from the Registrars & logic, for security and in order to be able
to scale appropriately. I will be using the dispatcher module for both
load balancers and proxies using the following architecture:
PublicIP = 5.5.5.5; Private IP = 192.168.1.0/24
USERS (Public Internet) ==> (public: 5.5.5.5) [ Kamailio (LoadBalancer,
Firewall, Sanity Checks) ] (core:192.168.1.2) ==> [ Kamailio Registrar,
Proxy, PSTN GW ] ==> AppServers or PSTN GW
(we can access our PSTN gateways Via our Core using Private IPs)
Questions:
1) Is it overkill to separate the LB & Proxy/Registrars?
2) Is this a common architecture & anyone configured this architecture
successfully?
Thanks in advance for your help!
*Daryn Johnson*
*Senior VoIP Engineer*
Hello,
Kamailio SIP Server v4.3.6 stable release is out.
This is a maintenance release of the previous stable branch, 4.3, that
includes fixes since release of v4.3.5. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.3.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.3.6 (or to 4.4.x series).
For more details about version 4.3.6 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2016/06/kamailio-v4-3-6-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Note: the latest stable branch is 4.4, at this moment with its latest
release v4.4.2. See more details about it at:
* https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/miconda - http://www.linkedin.com/in/miconda
Hi All,
I did not get any concrete information on whether PCSCF supports transport
mode ipsec in Kamailio release 4.4. It is mentioned that it is not
supported in release 4.0.
Could anyone please let me know the status on this?
Thanks,
Vikram
