Im trying to use Asterisk 13.9.1 with Homer SIP Capture Server.My hep.conf Asterisk configuration is:[general]enabled = yes capture_address220.127.116.11:9060;capture_password = foo capture_id = 2464SIP Signaling work correctly but no RTCP STATS arr..
,My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP Users secret. But the voip engineer before me didnt save / documented those password. Now the servers hardware is begin to broke, it hangs a lot, and have a lot..
I am using Asterisk 13.9.1 and want to catch AttendedTransfer, but it is not fired at all.Thank ..
We use Asterisk extensively for conferencing – for the last 8 years or so this has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 350 concurrent users. Right around that number DAHDI hits a hard coded memory limit and ki..
Im playing with the optional URL parameter of the Dial() command, which will also be sent to the called party upon successful connection, if the channel technology supports the sending of URLs in this way.Basically, the following asterisk dialp..