<DIV>
<DIV>Hi,</DIV>
<DIV> </DIV>
<DIV>Trying to set up a SIP environment with Asterisk - T1/PRI on Cisco 2811 - PSTN (Teltone TSP). SIP call looks working and I am able to make call from SIP phones to PSTN analog phones. But I have troubles with the Call from PSTN to the SIP phones. I guess the problem is the dial plan.</DIV>
<DIV> </DIV>
<DIV>I am using four digits extensions (2301, 2302,..., 2xxx) on the Asterisk for the SIP phones and the full PSTN digits are 7 digits starting with 222. I.e. My analog phones are 2223301, and 2223401. When dialing from the analog to the SIP phones, I dialed full number 2222301. I can see the gateway and the Asterisk server were receiving the calls, but the Asterisk does not know who is 2222301, he only knows 2301, so replied with 503 mesg.</DIV>
<DIV> </DIV>
<DIV>Potentially, I think two ways to solve the problem, first, on the Cisco router, strip the 222 before sending the call to the Asterisk, Second, on the asterisk, strip 222 before sending out to the SIP phones. My questions are, does anyone know how to strip the numbers on the router, and can this be done globally on the Asterisk, and how? </DIV>
<DIV> </DIV>
<DIV>Thanks in advance,</DIV>
<DIV>Dave</DIV>
<DIV> </DIV>
<DIV> </DIV></DIV><p>
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