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 To :  Paul Chitescu <paulc@v...>
 From :  andrzej.ciupek@a...
 Subject :  Re: [yate] sip cancel close all the session
 Date :  Mon, 25 Jun 2012 13:46:43 +0200

It is all made by INVITE:

We receive Invite from Yate:
CSeq: 540030 INVITE

SIP Proxy Receive it and send back INVITE to dst PSTN number:
CSeq: 695 INVITE

SIP Proxy receive Trying, Session Progress, Ringing, and after timeout  
of 15sec (this is set by Us, to limit time of follow-me, after 15sec  
it start to send next INVITE CSeq: 784 INVITE) SIP Proxy send:

CSeq: 695 CANCEL

It cause that Incomming INVITE CSeq: 540030 INVITE is disconnected.


> Hi!
> You need a capture to analyze what really happens.
> Does the SIP proxy fork the INVITE? Note that SIP forking is for simultaneous
> calls, not sequential. A sequential fork would need some clever SIP tricks to
> make sure at least one early dialog remains alive at all times. Unless both
> UAC and UAS support reliable 1xx provisional even this would be unreliable.
> Paul
> On Monday 25 June 2012 01:56:28 pm andrzej.ciupek@a... wrote:
>> Hello
>> I have, a scenario:
>> Incomming call from PSTN goes to Yate, that send Invite to My SIP Proxy,
>> SIP Proxy create follow-me to 3 PSTN numbers One by One. That calls
>> goes Out to PSTN by the same Yate. After first no answered call (1st
>> from follow-me list), but according to the RFC 3261 the Sip Proxy sent
>> the 'CANCEL' message in order to disconnect the first callee's leg
>> correctly before starting to send the 'INVITE' messages to the second
>> callee (second PSTN number form follow-me list). After that Yate
>> closes all the sessions after the 'CANCEL' message to the first callee
>> (1st from the follow-me list).
>> 1. --->Caller--->PSTN(Cisco)---->Yate-->SIP Proxy
>> 2. Sip Proxy get list of numbers from follow-me
>> 3. SIP proxy make call to PSTN number 1st from list
>> 4. ->SIP Proxy-> get callee number ---> Yate ---> PSTN(cisco)--->
>> 5. the callee is noanswered, so SIP Proxy send CANCEL after timeout of
>> first follow-me to Yate, to disconnect the first callee's leg,
>> 6. ->SIP Proxy-> get 2nd callee number ---> Yate ---> PSTN(cisco)--->
>> But in point number 5, Yate Disconnect Caller session too.
>> Is there a way to solve this trouble ?
>> The configuration at Yate is Cisco SLT + MGCP.
>> Greetings
>> Andrzej