[Users] Remote PSTN gateway

Klaus Darilion klaus.mailinglists at pernau.at
Mon Nov 28 11:29:36 CET 2005

Yes, it is possible.

Christoffer Soop wrote:
> Hi there!
> I would like to implement the following scenario using OpenSER:
> - when a SIP client sends a request consisting of only a simple phone
> number (on the form +CC-AAA-NNNNNN, with or without plus sign, hyphens,
> possible spaces etc where C, A and N are simple numbers) OpenSER should
> route the request to another, remote SIP-proxy acting as a PSTN gateway.

If (uri =~ "\+?[0-9]@yourdomain") {

probably you have to adpot the regexp to fit your needs (to also mathc 
white space...)

> - either the same credentials should be used or preferebly credentials
> dependent on the domain name of the callee for all requests to the
> SIP-proxy

if your SIP should use digest authentication to the PSTN GW you need the 
UAC module.

> - after the session with the remote proxy has been set up OpenSER either
> works as proxy between the client and the remote proxy, or better
> transfers the session to the client

If you use record_route your proxy stays inbetween the client and the 
PSTN GW. If not, your proxy stays out of the folowing conversations. But 
you will loose the ability to account BYE messages, and will have 
problem when doing NAT traversal.

> - all requests for a client not on the form of a phone number (i.e. with
> a fully qualified domain name should be routed to the correct location
> as decided by SRV-records etc.


if (uri =~ "\+?[0-9]@yourdomain") {
   t_relay(); /* +cc.. to GW */
} else if {
  (uri =~ ".+ at yourdomain") {
   t_relay(); /* local users */
} else if {
   t_relay(); /* outbound */

> Is this possible?  Any hints on how to accomplish this?
> Sincerely,
> 	Kris
> PS/Please note that I am new to OpenSER, SIP and SIP terminology...
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