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 <a...@avhan.com> wrote: > 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 <a...@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 >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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