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[dhaval.it01034 at gmai...]

Hello Group Members,


I have one question regarding SIP-I/SIP-T support in any of Asterisk versions.


We have client which send SIP-I/SIP-T request can asterisk handle it and serve as a normal SIP call.


As per mine analysis SIP-I/SIP-T are variant of SIP protocol with adding of ISUP/SS7 packets to original SIP request.



If we want to support it then how do we implement it and support it with asterisk . is there any open-source package or tool available to communicate and works as SIP-T to SIP and SIP to SIP-T gateway. I got a reference from kamailio which have SIPT module in latest version is anyone had worked or having an idea regarding this module and its operations .


Hope any one worked and having some idea


Any help appreciated



Thanks


Dhaval

[amit at avhan.com]

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.

212 Clean Clean false false false false EN-IN X-NONE X-NONE MicrosoftInternetExplorer4 <![endif]--> <![endif]--> /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman","serif";} <![endif]--> <![endif]--> <![endif]-->
Regards,
Amit Patkar

[dhaval.it01034 at gmai...]

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@avhan.com (amit@avhan.com)> wrote:

[mitul at enterux.in]

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@gmail.com (dhaval.it01034@gmail.com)> wrote:

[amit at avhan.com]

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

212 Clean Clean false false false false EN-IN X-NONE X-NONE MicrosoftInternetExplorer4 <![endif]--> <![endif]--> /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin:0cm; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman","serif";} <![endif]--> <![endif]--> <![endif]-->
Thanks & Regards,
Amit Patkar


On 3/12/2014 11:17 PM, DHAVAL INDRODIYA wrote:

[dhaval.it01034 at gmai...]

Amit,


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.


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 .


Once I will integrate to our provider I will definitely check and share with experts here.












On Thu, Mar 13, 2014 at 11:13 AM, Amit <amit@avhan.com (amit@avhan.com)> wrote: