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

[Mitul Limbani]

You can achieve this by setting relevant sip flags in the dialplan back and
forth.

Mitul
On Mar 12, 2014 11:18 PM, "DHAVAL INDRODIYA" <dhaval.it01034 at gmail.com>
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> 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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>
>
>
> --
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