Hi,
We are moving away from OpenSIPS and would like to start testing Kamailio.
I really liked how Kamailio complies so clean and is configured easily in comparison with OpenSIPS monstrosity.
So to not waste time can anybody provide some practical info for...
Kamailio complilation and config examples (w/MySQL) to route Carrier SIP (DID incoming and outgoing PSTN termination) traffic to and from Asterisk PBXs.
Example current OpenSIPS setup using dynamic routing module:
1-. dr_rules have the complete DID for PBXs (for incoming traffic from Carriers to be proxied to the correct PBX).
2-. dr_rules have the partial DID for Carrier gateways (for example based on internatinal, state, etc. routing of outgoing traffic from PBXs to Carriers).
3-. dr_gateways have the IP numbers for PBXs.
4-. dr_gateways have the IP numbers for Carriers.
Any info and pointers appreciated.
Best regards, Gary
Hello,
installing latest kamailio stable version from git with mysql support is detailed at:
- http://www.kamailio.org/wiki/install/4.1.x/git
You will find also a kamailio-basic.cfg (along kamailio.cfg), which is simpler version.
There is also drouting module, but if you look for least cost routing solution, you can check also the lcr or carrierroute module.
A simpler solution can be combining mtree and dispatcher modules, like:
- mtree: used to map did to a dispatcher group id (or even to an IP address if you don't have multiple asterisks to try for same DID) - dispatcher: used to hold the list of IPs for asterisks, grouped per IPs of boxes able to handle same bunch of DIDs (dispatcher module has a complete cfg in the readme, good to start with)
Permissions module can be used if any extra IP checking is needed (dispatcher module can also test source IP addresses).
Cheers, Daniel
On 24/04/14 16:34, Gary Wallis wrote:
Hi,
We are moving away from OpenSIPS and would like to start testing Kamailio.
I really liked how Kamailio complies so clean and is configured easily in comparison with OpenSIPS monstrosity.
So to not waste time can anybody provide some practical info for...
Kamailio complilation and config examples (w/MySQL) to route Carrier SIP (DID incoming and outgoing PSTN termination) traffic to and from Asterisk PBXs.
Example current OpenSIPS setup using dynamic routing module:
1-. dr_rules have the complete DID for PBXs (for incoming traffic from Carriers to be proxied to the correct PBX).
2-. dr_rules have the partial DID for Carrier gateways (for example based on internatinal, state, etc. routing of outgoing traffic from PBXs to Carriers).
3-. dr_gateways have the IP numbers for PBXs.
4-. dr_gateways have the IP numbers for Carriers.
Any info and pointers appreciated.
Best regards, Gary
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
Anybody have a working kamailio.cfg file that uses the lcr module?
Or as Daniel has kindly pointed out a kamailio.cfg that uses mtree and dispatch modules?
Thanks!
--- Details for those interested (Excuse any lack of clarity due to incorrect use of any industry standard language. Please correct me if you have time.):
Big picture: Be able to provide only a few SIP gateway IPs to PSTN and DID Carriers. These SIP proxy gateways will then be able to service incoming and outgoing INVITE traffic for thousands of PBXs.
I am only interested in "switching" pure SIP traffic between PBXs and Carriers.
In this case everything (normally) starts with an INVITE. From a PBX or Carrier (DID provider in this case) that must have it's IP number in a table (sole auth method).
It also seems that we must "group" these gateways into PBXs and Carriers for more efficient use of prefixes (even though Carriers AND PBXs are just gateways identified by IP and associated with a prefix and other optional routing/switching decision making parameters).
The "switch" then must, based on the prefix (or DID), forward the INVITE from a PBX to a Carrier (in the PSTN termination provider role). Or, if the INVITE is from a Carrier, forward to the PBX.
I simplify greatly providing this traffic aggregation sip proxy "switch" example for others that may have similar needs in the future.
Best regards, Gary