[asterisk-users] Regarding SIP-T/SIP-I support in Asterisk.

[Amit]

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 <amit at avhan.com 
> <mailto:amit at avhan.com>> 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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