24 Apr
2014
24 Apr
'14
5:34 p.m.
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
24 Apr
24 Apr
6:15 p.m.
New subject: Minimalist Kamilio configuration example using dynamic routing module
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
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
25 Apr
25 Apr
6:01 p.m.
New subject: Anybody have a configuration file using lcr or mtree and dispatch?
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
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participants (2)
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Daniel-Constantin Mierla -
Gary Wallis