[Asterisk-Dev] SIP channels not cleared

[Kevin P. Fleming]

Chee Foong wrote:

> My vendor's system (Pactolus) sends a BYE for INVITE sent by Asterisk before
> sending OK to asterisk to terminate a call. This caused SIP channels is
> Asterisk not being cleared up (described in my previous post). I have looked
> at chan_sip.c, I am thinking of modify the handle_request_bye to accomodate
> my vendor's system but I am not sure how to do it because I am not
> proficient in C programming.

This it not valid SIP behavior, and it will not be easy to make chan_sip 
deal with it. BYE can only be sent by the callee after it has sent '200 
OK' _AND_ received an 'ACK' from the caller.

Your vendor's system is broken and will not interoperate with RFC 
conforming SIP implementations. At best, Asterisk could ignore the BYE, 
but it certainly will not cancel an outbound call because the callee 
sent a BYE before accepting the call.