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5 Responses

  1. shane
    shane 2012/12/20 at 11:24 pm | | Reply

    Works for me just fine! I have been trolling the net for weeks looking for a solution when i came across your solution. Thanks!

  2. Rafa
    Rafa 2013/05/14 at 6:04 am | | Reply

    wonderful, the only guide that helped me to make it work. thanks a million!!!

  3. Red unsavory agents
    Red unsavory agents 2014/10/03 at 4:59 pm | | Reply

    Now help us with multiline VoiP desktop phones, and having multiple

    4 of: Cisco SPA525G2

    There are four areas to specify ports:

    * SIP tab (TCP ports, 2 fields for range)
    * SIP tab (RTP ports, 2 fields for range ; 1 field for ext begin port range)
    * each Line tab (EXT SIP, SIP)
    * each Line tab (registration and/or proxy, FDQN (colon port), color port is optional)

    Assume 3 IP-PBX, NO local SIP server [neither behind nor in front of NAT]:

    * anveo
    * vitelity

    After building list of three respective ports sets for each device and creating NAT rules (auto fw rules) audio tends to drop out (inbound only) at 26 second [into call] and regular intervals. RTP “packet timing” set to 0.005 (smaller bits lost = less perceptible grammar foibles)

    2.1.4-RELEASE (i386)
    built on Fri Jun 20 12:59:29 EDT 2014
    FreeBSD 8.3-RELEASE-p16

    siproxd shows in packagemanager for pfSense 1.2.1


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