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Hosted PBX use of FXO gateways as CPE


Carlos Montemayor
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Hi,

 

I was able to make work the HT503 from Grandstream as a FXO Gateway trunk for the PBX in the scenario where both are CPEs (in the same LAN). Now that I am mostly using the hosted version, I would like to keep using the FXO trunks. Surely somethings have to be adjusted. It is certainly not the same having the PBX outside of the LAN. I already tried to imagine things and made some attempts and I was able to receive calls, however, placing calls I was not. Has somebody already worked with this scenario? Can I get a guide or orientation on this?

 

Regards

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The easiest way would be to run the gateway on a public IP address. Then you can just set up a regular trunk to the PBX as if this would be a LAN deployment.

 

Of course you should make sure that the gateway accepts traffic only from the IP address of the PBX. But for most quality PBX, that is not a big problem. The good news here is that hosted PBX have a static IP address.

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Hi, thanks for answering.

 

The idea is to be able to scale up, I mean, to be able to sign small businesses and the most likely scenario (almost omnipresent) will be that there will be dynamic IP addresses on the customers side. I already have a handful of host accounts of the dynamic dns service and I can do port forwarding on the routers of the customers.

 

I just tried that scenario at my lab (an euphemism for my own home´s corner) and as I was saying earlier, I could receive calls but not to place outside calls. There must be some piece of configuration that I am missing. At the last attempt, I was able to place a call, but it was wrongly routed. Instead of going out (to the number I dialed) it went to an extension. The caller ID at the receiving extension showed me the ID of the trunk, as if the call was coming from the trunk, not the extension. What could it be?

 

Regards

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Well in the dial plan you can send a call to an extension. You might have see that already. That means that the FXO gateway needs to register to the PBX like an extension. Then when the PBX sends the call to the gateway, the question is where is the number to be dialed. Here the gateway must look into the To-header, not the Request-URI. Then his project might become possible.

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Hi,

 

Yes, I just saw that one of the options in the trunk section of a dialplan is "Call Extension". Following your line of thought, I guessed that I should place the extension number that corresponds to the FXO gateway in the replacement field. Also, I was able to register the FXO port as an extension of the domain (to my amasement). So, I did a test call. I found out that if what I dial gets me to the rule that sends me to the extension of referece, I would get a new dial tone. So, I dialed the number and bingo! it did go out through the fxo port and to the pstn. Even the caller ID was correct. Much obliged!!

 

Now I have a cuestion. In the configuration page of the fxo port, I have two fields for identity purposes. One is "SIP user ID" and the other is "Authenticate ID". In order to get the fxo port register as an extension, I had to put the extension number in both fields. First I had tried using extension@domain but that did not work. In order to find the proper domain (because the nature of a hosted system is to be multidomain) I had to define the domain of interest with the alias of "localhost". I imagine that such thing makes it like the default domain. Then, the fxo port found its way and registered as an extension. My concern is, how am I going to register a new fxo port to a new domain? I mean, the idea is that all domains could have their fxo port. What should I had write in each of those fields? I feel that we are very close to set up this correctly.

 

I very much appreicate your help since this feature is of vital importance to my project.

 

Regards and thanks.

 

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Hi,

 

This is looking very promising. I kept trying and I was able to get registration using the following setup:

 

SIP user ID: extension@domain

Authenticate ID: extension

 

It seems that I also needed to be patient. It takes it a bit to get registered. I will find out for sure when I attempt to add a second fxo port for a different domain.

 

One very interesting thing. With the new setup, the behavior changed a bit. Before, I had to dial something, anything, and I would get a second dial tone and then I had to dial the number from scratch. Now, with the new setup, I need to dial just once, the final destination number and what I hear is: first a very short dial tone, then a few seconds of silence and then the ringing of the call attempt. I believe this to be a more natural behavior, exception made for the few seconds of silence, but users could get used to it rather fast. Does this change in behavior tell you something?

 

Regards

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