Archives : May-2022
We have a customer where their switch sends pidf+xml presence information in the SIP INVITE message.Does Asterisk process this pidf+xml information?Does it store this in a channel variable that a dial plan could access?If not, does it store present t..
Just ran into another weird issue… In Swiss Telephone Interconnection, ptime=20 is a requirement. So on our SBC we enforce the presence of ptime=20 to avoid issues. I have an asterisk with chan_sip in the LAB which behaves weirdly… Inbound SDP au..
Hi. Im trying to use different logging verbosity levels to get dialplan output into different log files, and theres clearly something I havent understood about how Asterisk does this… I have the following in /etc/asterisk/logger.conf: [logfiles] logtest.verbos..
I have come over a codec negotiation issue. A (asterisk) is sending in INVITE containing * opus (type 107) * g722 * alaw (type 8) B answers with 183 containing SDP * alaw a=sendrecv B then answer the call with 200 and NO SDP I suppose that result..