The only two "custom" SIP headers you can set currently are VXML_URL and Alert-Info:
You would be better served by looking at SER (http://www.iptel.org/ser/) for the type of functionality you're after.
JT
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:
[EMAIL PROTECTED];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.
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