Hi Dhaval,
If you capture and share SIP traces for inbound and outbound calls
separately, experts on this list can guide to achieve objective.
You can enable SIP trace on asterisk by executing following command in
Asterisk console
*sip set debug on*
*Thanks & Regards,*
Amit Patkar
On 3/12/2014 11:17 PM, DHAVAL INDRODIYA wrote:
Thanks Amit,
I want following scenario.
INCOMINGCALL ---> MSC (SIP-T) ----> PBX (Asterisk)
OUTGOINGCALL ---> PBX (Asterisk) (SIP) to (SIP-T) ---> Aircel MSC
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.
please correct me if can achieve this functionality.
Thanks
Dhaval
On Wed, Mar 12, 2014 at 6:15 PM, Amit <[email protected]
<mailto:[email protected]>> wrote:
Hi Dhaval,
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.
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.
*Regards,*
Amit Patkar
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