everyone,I am sending out a multicast page using the following in my dialplan:Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q)Everything works great, but I had a question about SIP and SDP:Should I be seeing a SIP/..
Author : Matthew Murphy
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everyone,I upgraded from Asterisk 13.5.0 to 13.7.0 and I am having database connection problems. I am doing Asterisk realtime with PJSIP 2.4.5 and it works perfectly in 13.5.0. But now I am losing my database connection (running on a virtual box) ..
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everyone,I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type pjsip show endpoints at the CLI, I get No Objects Found.However, if I request information on a specific endpoint, (for example: pjsip show endpoint 101) then I get ..
Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing c..
Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried sett..