Hi,
Can anybody tell some success story about using OpenSER as
Asterisk/CallWeaver frontend? I tried to googling first but where are no
detailed useful stories :(
Now I have CallWeaver (Asterisk fork) installed on Linux server which
connected to 2 different providers via one wan interface but 2 ip
addresses from one network, for example:
eth0 - 10.0.0.2/24 -> provider1(10.0.2.1)
- 10.0.0.3/24 -> provider1(10.0.3.1)
So, I need send SIP requests to provider1 from address 10.0.0.2/24 and
to provider2 from 10.0.0.3/24. This must be done because my internet
provider must account trafic on router 10.0.0.1 for 2 VoIP providers
separately and it's billing system can do it only by different source ip
addresses and can't do it by destination ip addresses.
I can separate SIP requests between 2 ip addresses with linux kernel
2.6.18 or higher by using ip_nat_sip iptables module and MARK/SNAT. But
this approach can't be used in OpenVZ virtial environment in which
CallWeaver must be used.
Is it possible to send SIP requests from CallWeaver via OpenSER and
modify them with OpenSER to replace source addresses in UDP headers and
SIP headers? How can I say CallWeaver to send SIP requests via OpenSER?
How can I implement this routing logic in OpenSER? Can anybody give me a
simple example?
--
Thanks,
Eugene Prokopiev
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