[Serusers] call PSTN -> SIP

[Iqbal]

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 at 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
>>>
>>>
>>>      
>>>
>>
>>    
>>
>
>
>.
>
>  
>