Announcement

Collapse
No announcement yet.

Configuring 871w To Work With FreePBX Voice Server

Collapse
X
  • Filter
  • Time
  • Show
Clear All
new posts

  • Configuring 871w To Work With FreePBX Voice Server

    Hello, folks...

    I actually went to the "happy router" site first due to not realizing David had moved everything over to Petri. Knowing is half the battle I believe G.I. Joe once said...

    Anyway, after spending a month fighting with the "FreePBX" Voice Server Application, I've finally gotten it to run using an old Pentium II processor with 128MB of memory and calls are "very" clear. I'm currently running it behind a CISCO 871w router I've had for a few years (I pm'd David a lot a few years ago when I first started learning CISCO IOS). If you're wondering why I used such an "underpowered box," this was my first attempt, so nothing lavish was required (i.e., dual core processors, 8GB memory, etc...).

    The problem I initially ran into was that I put the PBX server behind an old (yet solid) Linksys WRV54g router; everything ran fine, to include zero conversation dropouts. If I put the box behind the 871w (using exact same Peer Details Configs) conversation on my two configured trunks dropped. The issue turned out be the "sip_nat.config" file in FreePBX. Apparently, the 871w is considered to run a "symmetrical firewall" whereas the Linksys didn't (non-symmetrical). Yeah, jury is out on that explanation

    Anyhow, below are the steps I took to correct the issue:

    Sip_nat.conf:

    nat=yes
    canreinvite=no
    srvlookup=yes
    externip=74.xxx.xxx.xxx
    localnet=172.16.3.0/255.255.255.0

    Initially, I had "nat," "externip," "externrefresh," and "localnet" in this directory, but once I removed "externrefresh" and added the above settings to sip_nat.config, calls stopped dropping.

    Here's the process I followed (CISCO 871w router was used for the below port forwarding):

    Step 1 - Create Access List (Port Range Forwarding Method):

    access-list 130 permit udp any any range 8000 8766
    access-list 130 permit udp any any range 10000 20000
    NOTE: Ports 5000 - 5084 are open/transmitting "without" any direct configuration; if needed, just add the range as an additional access-list statement:

    access-list 130 permit udp any any range 5000 5084 (optional)

    I found one of David's old posts that shed some light on using the "port range forwarding" option. Man, that cuts down on typing, I'm here to tell you

    Step 2 - Create Route-Map:

    route-map freepbx permit 1
    match ip address 130 (This line refers to the access-list created in the step 1)

    Step 3 - Apply "One-to-One" NAT Mapping With Route-Map:

    ip nat inside source static [internal ip] [external ip] route-map freepbx

    Here's a better example of the above:

    ip nat inside source static 172.16.3.4 74.55.43.3 route-map freepbx

    The above statement tells the router to create a "one-to-one" mapping from the "internal ip address" of your PBX server to your "external ip address" assigned to you by your ISP, along with saying that any VoIP related issue is to be handled by the route-map statement (freepbx)

    Next are the trunk settings seen below (CallCentric for Stateside Calls/VoipTalk for UK Calls):

    CallCentric Trunk/Peer Details
    call-limit=5
    username=818xxxx
    type=peer
    secret=[password]
    qualify=yes
    nat=yes
    insecure=invite,port
    host=voiptalk.org
    fromuser=818xxxx
    fromdomain=voiptalk.org
    outboundproxy=nat.voiptalk.org
    dtmfmode=inband
    disallow=all
    context=from-trunk
    canreinvite=no
    authuser=818xxxx
    allow=ulaw,alaw,g729,gsm
    CC Dial Patterns-1NXXXXXXXX, 011.

    VoipTalk Trunk/Peer Details:

    username=1777xxxxxxx
    type=peer
    secret=[password]
    qualify=yes
    nat=yes
    insecure=invite,port
    host=callcentric.com
    fromuser=1777xxxxxxx
    fromdomain=callcentric.com
    dtmfmode=inband
    disallow=all
    context=from-trunk
    canreinvite=no
    authuser=1777xxxxxxx
    allow=ulaw,alaw,g729,gsm
    VoipTalk Dial Pattern - 001. , 0|Z. , 044+XXXXXXXXXX

    I figured starting with a Linux distro application such as this would be a good way to "ease" into the voice certification track after I finish my CCNP. I've been management for so long, I'm literally having to "kick dust" off areas of my grey matter to remember my CCNA IOS commands *heh* Being I've passed the switching (1yr ago), I just need to get past "Route" and "TShoot," and then it's off to the voice side...

    Let me know if anyone finds this useful. While I'm nowhere near an expert, the last month has broadened my experience with telephony and I found "plenty" of resources to assist...



    Jay Johnson
    Certified: CCNA, Security+, Network+, Server+, I.T. Project+
    Last edited by jay.johnson; 8th November 2010, 03:33.
Working...
X