<div dir="ltr"><div><div><div>Amit,<br><br></div>I know how to play with SIP in asterisk and other tools . I want to know weather asterisk natively support or is there any extra patch or any workaround for SIP-T/SIP-I.<br>
<br></div>Regarding packets and other things I am still not integrating it . I am searching some open-source tool which can send generate this type of packets and structure .<br><br></div>Once I will integrate to our provider I will definitely check and share with experts here.<br>
<br><br><div><div><br><br><div><br><br></div></div></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Mar 13, 2014 at 11:13 AM, Amit <span dir="ltr"><<a href="mailto:amit@avhan.com" target="_blank">amit@avhan.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<div><font color="#000000">Hi Dhaval,<br>
<br>
If you capture and share SIP traces for inbound and outbound
calls separately, experts on this list can guide to achieve
objective.<br>
You can enable SIP trace on asterisk by executing following
command in Asterisk console<br>
<big><b>sip set debug on</b></big><br>
<br>
</font>
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<p><font color="#000000"><b><span lang="EN-US">Thanks
& Regards,</span></b><span lang="EN-US"><u></u><u></u><br>
Amit Patkar<u></u><u></u></span></font></p>
<br>
<span lang="EN-US"><u></u><u></u></span>
</div>
</div><div><div class="h5">
On 3/12/2014 11:17 PM, DHAVAL INDRODIYA wrote:<br>
</div></div></div><div><div class="h5">
<blockquote type="cite">
<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>
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<br>
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<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>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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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"><br>
Amit Patkar</font></span></p>
<span lang="EN-US"></span> </div>
</div>
</div>
<br>
</div>
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