Re: [Serusers] call PSTN -> SIP - sr-users

19 May 2005

not sure I follow, assume you have no src_ip check, what happens, does 
ser send a request for authroisation to the gateway, if so , then before 
u send that request that where you add a if statement to check its a 
request from ur src_ip...

Iqbal

Kostas Marneris wrote:

...
 Thanks for your help Iqbal..

but I'm still experiencing problems :
I put the 'src_ip' in top of 'Call Processing System'

my conf :
	...
        # ---------------------------------------
        # Record Route Section
        # ---------------------------------------
	...
        # ---------------------------------------
        # Loose Route Section
        # ---------------------------------------
	...
        # ----------------------------------------
        # Call Type Processing System
        # ----------------------------------------
        if (src_ip ==62.103.4.155) {
                log(1,"PSTN-GW originated req...");
                break;
        };

        if (uri != myself) {
                log(1,"URI is not myself");
                route(1);
                break;
        };
	...
	bla-bla-bla

I just expected SER logged the mesg "PSTN-GW originated req..."
to /var/log/messages...
But nothing is logged there... :(

On the other hand I can see SIP mesgs on c5300 (debug ccsip all)
and I can verify these with tcpdump/5060 on SER box..

thank you anyway,
Kostas

Iqbal wrote:

 just accept the calls from that IP address, by
detecting src_ip=a.b.cd 
or adding into trusted table, and checking for allow_trusted.

The gateway should then send to ser, ser should then accept the call 
without calling any auth scenario

Iqbal

Kostas Marneris wrote:

 Hello,

I want to make a call from PSTN to a SIP phone.
I use a cisco 5300 for answering/handling the PSTN number
and then forward the call to a SER box (0.8.14) .

	POTS -> c5300 --> SER -> SIP softphone

My conf on c5300:
!
dial-peer voice 1003 voip
destination-pattern 12345
session protocol sipv2
session target sip-server
codec g711ulaw
!
sip-ua
sip-server ipv4:X.Y.Z.W
!

What is the appropriate configuration on ser.cfg in order to
handle this call and forward it to sip:userX@mydomain.com ??
Any hint about this ??

I've already done the opposite scenario (SIP->PSTN) :
ser.cfg:
      if (uri =~ "^sip:2*") {
              rewritehostport("1.2.3.4:5060");
              forward(uri:host, uri:port);
              break;
      };

thanks for any help,
Kostas

.