[Asterisk-Users] Advanced SIP management

[Cerrajetto]

Hello:

I would like to use Asterisk as a redirect/proxy sip server to route SIP
calls on a sip header/parameter basis.

I've tried some things successfully:

- SIP registration from clients.

- On-the-fly compression for wan VoIP transfers:
  SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711

- Sending custom parameters in URI:
  exten => 1,1,Setvar,VXML_URL=var1=value1
  exten => 1,2,Dial,sip/192.168.0.1

I need to know if there is some possibilities for:


1) Automatic SIP header and/or uri-parameter propagation

I need to propagate custom **headers** and/or **uri parameters** from source
to destination.
Does Asterisk propagate this additional info automatically?


2) Automatic SIP headers and/or uri-parameters propagation via IAX

Yes, in wan communications I use 2 asterisk servers:

SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711

Does Asterisk propagate custom headers/uri_parameters through IAX?


3) Custom header/parameter extraction

I need to be capable for **extracting** those values in extensions.conf and,
then, **decide** how to route.

For example, if the client_1 sends the following URI:

  222 at 192.168.200.200;sender=server01

to Asterisk, can Asterisk extract the variable "sender=server01" and,
therefore, decide what to do?:

  exten => 222,1,GotoIf,"$[${sender} = server01]?default|1|1:default|1|6";


4) Custom header/parameter insertion

OK, I can insert parameters in URIs:

  exten => 1,1,Setvar,VXML_URL=var1=value1
  exten => 1,2,Dial,sip/192.168.0.1

But, can I insert directly custom headers?


Maybe I'm confused and Asterisk cannot do what I need ... Then, I would
appreciate what could be the good direction.


Thank you very much,

Mark Cerrajetto.