I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to ...
I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually). I can connect to it, make a call to other endpoint (namely, my android ...
Having successfully configured and maintained few Asterisk based installations, I have now been provided a task to configure FreeSwitch SIP server. ISO downloaded from ...
Suppose I have Freeswitch, which has SIP gateway configured. So, the local users of this Freeswitch can call outside via <extension name="russia"> <condition field="destination_number" ...
I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, ...
my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...