<div dir="ltr">Thanks Amit,<div><br></div><div>I want following scenario.</div><div><br></div><div><span style="font-family:arial,sans-serif;font-size:13px">INCOMINGCALL ---> MSC (SIP-T) ----> PBX (Asterisk)</span><br style="font-family:arial,sans-serif;font-size:13px">
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">OUTGOINGCALL ---> PBX (Asterisk) (SIP) to (SIP-T) ---> Aircel MSC </div></div><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px">I understood that via Dial-plan we can achieve and get extra parameters values. But what about RTP fields as per my analysis ISUP packets are not sending RTP/AVP they are sending multipart data.</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">please correct me if can achieve this functionality.</div><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px">Thanks</div><div style="font-family:arial,sans-serif;font-size:13px">Dhaval</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Mar 12, 2014 at 6:15 PM, Amit <span dir="ltr"><<a href="mailto:amit@avhan.com" target="_blank">amit@avhan.com</a>></span> wrote:<br>
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<div><font color="#000000">Hi Dhaval,<br>
<br>
Theoretically, Asterisk can support SIP-I / SIP-T. Since
protocols provide additional information and controls, you will
not get those benefits. You will have to write dial plan
functions to extract addition information exposed by SIP-I /
SIP-T.<br>
Though, I have not tested it with Asterisk, I have successfully
deployed application on other SIP platforms and interoperability
with SIP-I/SIP-T was not an issue.<br>
<br>
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<p><font color="#000000"><b><span lang="EN-US">Regards,</span></b></font><span lang="EN-US"><font color="#000000"><u></u><u></u><br>
Amit Patkar</font><u></u><u></u></span></p>
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