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Adding A Cellular Trunk To A Home Office VOIP System: Part 2

In Use: Cell-to-Hosted PBX

Once configured for use with OnSIP, I used the MV-370 as my principal means to call the UK from my US cell for two months. I used two-stage dialing to reach people not in the speed dial table.

The cellular signal in my office registered between 16 and 21 according to the MV-370’s Mobile Status menu, which is within acceptable limits. Once connected, the call quality was as good as a typical cell call, which makes perfect sense.

I occasionally had some trouble making connections through the gateway. In the case where I had programmed a speed dial into a cell phone memory, I found that sometimes the extra digit passed to the gateway would be ignored. When this happened, I just had to manually press the digit and the call would proceed.

I suspect that this had more to do with the default duration of DTMF tones on my Blackberry 8100 than the gateway itself. Both “in-band” and “RFC2833” types of DTMF work with OnSIP if I used the G.711 codec. Switching to the low bandwidth G.729 codec forces the use of RFC2833 type DTMF.

The gateway provides a “mobile dtmf debounce” adjustment setting (Figure 5) to overcome just such trouble. By setting this to 120 ms, passing DTMF was a little more reliable.

Figure 5: DTMF debounce

Figure 5: DTMF debounce

There was also a little extra latency to calls placed through the gateway compared to dialing the UK directly. This makes sense, since the call is passing through significantly more processing by being bridged into the IP domain.

The effect of the added latency was not so bad as to be a problem. But it might keep me from considering such a gateway as a primary trunk. At no point did anyone I called think that I was using anything more than a direct cellular call.

Configuration #2: Use With A Local Asterisk Server

When I finally had some time to experiment with the gateway, I configured it to access my local Asterisk server as the default realm with the hosted PBX as a secondary realm (Figure 6).

Figure 6: Registered with local Asterisk server

Figure 6: Registered with local Asterisk server

When more than one account is registered, the gateway will answer incoming calls on all accounts, but only pass outgoing calls to the default account. So my Asterisk server needed to be the default account or I would not be able to pass IP calls out to the cellular network.

Defining Routes: LAN-to-Mobile

Previously my routing had been established for calls originating on the cellular network and being passed into the IP domain. Now I needed routing for calls passing from LAN to Mobile.

Figure 7: MV-370 LAN-To-Mobile routing table

Figure 7: MV-370 LAN-To-Mobile routing table

The LAN To Mobile routing table on the gateway (Figure 7) allows you to restrict acceptance of calls by originating URL, IP address, or IP address range. It also provides a means of restricting where you might call out to, perhaps restricting access to overseas calls, for example. In my case, the routing could be wide open since I was comfortable that any call originating on my home office LAN would be legitimate.

This Post Has 3 Comments
  1. Hi. I’m using mv-372 and am consirned about latency (200-300ms).
    Another issue is hangup. When a hangup mobile phone conection to mv-372, asterisk still rings for about 3-5 seconds.

    Could anyone help me?

  2. I am facing echo issues and i have tried a lot with different values right now i am using the following values Under mobile setetingsd

    voip tx gain = 9
    Voip rx gain=11
    codexc tx gain= 6
    codec rx gain=6
    Do you have the same issues?
    Please help if you find some changes can reslove the echo issue.

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