Archives : May-2016
all,I have a intriguing issue that the RFC is not really clear about. Sometimes call hang-up on 45min mark because no-one refresh the call ( far-end hangup)On both good and bad calls:1)We initiate an invite2)200OK is answered as refresher=UAS3)Send AC..
I am trying to secure SIP session with TLS on Asterisk Server 1.8. Ikeep getter an error, == Problem setting up ssl connection: error:14094418:SSLroutines:SSL3_READ_BYTES:tlsv1 alert unknown ca[2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:F..
Have a strange issue at a customer, they went and replaced all of their old PoE switches with brand new HPE 5130 EI Switch Series.Their PBX has been up and stable for several years with no recent changes, but since they change the switches they are hav..
It looks as though something might be going wrong in the AGI script itself.Did you use a proper AGI library, or a quick-and-dirty homebrew solution?(There is little virtue in walking all the way to the tool shed to fetch a chisel, if you know the screwdri..
I have an Asterisk 13.8.2, which is supposed to be only a client to an encrypted SIP service. All local phones are connected via UDP.Since I cant use PJSIP (see my mailing list post from yesterday), Itried configuring chan_sip to work that way. My setti..
2016-05-03 16:43 GMT+02:00 Matt Fredrickson :OKYes, I think issue must come from incorrect Audiocodes settings. Requiring T.38 settings within first INVITE seems very unusual.Thank you very much fo..