SIP2Megaco Busy
"Jati Kalingga Praja" <[email protected]>
| Newsgroups | gmane.ietf.megaco |
|---|---|
| Message-ID | <[email protected]> |
Hi,, I'm new to this mailing list. Right now, i'm studying about interworking between megaco and sip. I simulated a call from SIP phone to Analog phone which was busy. Here i attach the captured data using ethereal, you can view the callflow by writing "*(megaco and ip.addr==10.14.32.186) or (sip and ip.addr==10.14.32.185) or rtp*" in filter textbox. I have a few questions about that captured data : 1. Why did MGC create a context and the terminations for the analog phone and etc (captured data No. 2079 - ~)? Isn't it busy? 2. if it was so, why did in no 2114, the MGC send modify (cg/dt) to MG? 3. Why did the MGC create 2 context? first, when the sip phone send the INVITE; second, when the analog phone onhook,,, I hope my questions are all clear to be understood,,,because i'm not too good in english. Hopefully, you guys can help me. Regards, Jati _______________________________________________ Megaco mailing list [email protected] https://www1.ietf.org/mailman/listinfo/megaco
4577020ke4580220busy.pcap
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