Hopefully this is a simple question. How does a non-secure phone that is on a PBX connected to an asterisk over a SIP trunk communicate with a secure phone connected to the Asterisk server? I like to think that the secure call terminates on the Asterisk and the non-secure call is somehow joined to that call on the Asterisk. Same with the voice traffic, I'm once again thinking that the Asterisk proxies the calls. Just not sure. Thanks, Mitch
I'm in the process of testing a TLS/SRTP install. My experience is
improving with each new challenge, but this one is a great test of my 2
month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error
message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the
extensions and sip.conf files. *CLI> == Using SIP RTP CoS mark 5
The Asterisk Development Team has announced the release of Asterisk 1.4.40. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.40 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you! The following is a sample of the issues resolved in this release: * Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev,
jthurman, elguero, zerohalo. Patched by tilghman) * Resolve issue where re-transmissions of SUBSCRIBE could break…