My call flow is:

Softphone --> TLS ---> Kamailio --> UDP --> ASTERISK --> PSTN

I want to use tcpops module between the softphone and kamailio.

• enable tcpops : no problem

• disable tcpops : two cases 
• cancel or bye from softphone: no problem.
• cancel or bye to softphone via kamailio: how disable tcpops? I can't use $avp(bye_conid) because it is asterisk thant sending the sip message. Can we use $avp(caller_conid)? In my case, the softphone is always at the origin of the call establishment.

Regards

Abdoul.