voip callshop (voip-to-voip termination)

Hi, I have a callshop that I would like to run using voip instead of the PSTN connection. I bought an 4-fxs ports welltech gateway on which 4 analogue phones are connected on one side and the LAN port connected to an ADSL Broadband router for internet connectivity. I configured the box so that calls from that callshop will be terminating on my gateway (IP configured on the fxs gateway). My Cisco router(AS5350) is located in a data centre and one E1 of that AS5350 is connected to a switch using interface 1/0:15. I also terminate voip calls to voip partners (USA and Africa) using prefixes.
Now when calls arer coming from the callshop, they will be terminated ok if the destination number is the one configured to terminate through the E1 port connected to the switch. If I make a call that is terminated only through a voip partner, the call will fail with destination unknown.
It looks like the call will reach the cisco and could not be redirected to the voip partner for some reasons. I tried to use voip translation-rule called "tag" (giving a tag to those calls and direct them to a voip partner) but no chance.
The commands is like these:
  translation-rule 5
  rule 0 ^00 9900
  voip incoming translation-rule called 5
  dial-peer voice 12 voip
  destination pattern 99.T
  session target ipv4:
  voice-codec 1 (all codec listed with g731.x =prioritised)
  voip protocol being used on the welltech gateway are g323.1; g711A-law...
A friend of mine said that it was impossible to have voip call to be sent to  a voip dial-peer. A gatekeeper or a virtual switch must be installed or I should direct calls straight to my voip partners(billing will not be possible this way).
Can anyone help getting it right without gatekeeper or if any other device is required, what is it ?
Thanks a lot
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In regards to yourVoIP partners, what type of signaling are they using? What is at their remote end, type of router, phones and such?

Also need some more information on your config, possible to run and post the following:

show dial-peer voice summary

Not sure about being impossible, but a gatekeeper is very handy when using H.323, but it sounds more like a routing/digit translation problem. One question, when Welltech dials a call meant  for E1 what would be a typical digit pattern? Then what would the digit pattern be if same phone dialed to one of your VoIP partners?

Also have you thought about using SIP verses H.323?
Why don’t you program the call to go to the switch with a prefix, and then have it send it back out to the gateway with a different prefix? You can also use a crossover cable between the E1 ports to do effectively the same thing, but that ties up two E1 ports on the gateway which isn’t an optimal solution most of the time. Another solution would be to get an old PC and run OpenH323 Gatekeeper on it which is free, and would give you more flexibility, especially if you continue to expand your network with additional gateways.

The reason what you are trying to do doesn’t work, is the Cisco gateway won’t act as an H323 proxy, so you can not relay a call off it to another gateway. So you either have to fool the gateway into seeing the call as a VOIP call coming in and an incoming voice call or it will not work. You don’t know how many people I know that wish it would, but that is not the way Cisco designed their gateways to work.


In regards to your setup, a H.323 gateway is your best route to solve your problems from an immediate nature.

You could add rules that would add and strip digits and allow the call to be routed thru the current setup. This form of routing is the basics for what is called Tandem trunking. The drawback is that you are building a configuration that can become very complex and cumbersome to deploy.

In regards to SIP, I would look heavily at deploying that within your environment. The battle between which VoIP protocols to use has been going on for years. Being that my foundation in Convergence started in the voice industry, I was heavily biased towards the use of H.323, being the standard came from the ITU. In the early days, SIP was very primitive in its call control and implementation. Now years have passed and SIP has leapfrogged past H.323 in features and functionality. Also, H.323 is so tightly structured that it is very hard to for it to adapt to the fast moving nature of the IP telephony movement.

I believe that SIP is the standard to go with for various technical reasons, but more so that the industry has really geared toward the use of it as the de facto standard. This is evident if you look at the services and products being offered by such companies as Vonnage, Polycom, Cisco and countless others.

Also SIP will allow you to deploy a host of other services for end clients (additional revenues), such as various hard based and soft based phones; deploy additional technologies such as WiFi, Bluetooth and cellular based. By using SIP you can start looking at companies such as Vonnage and other SIP based carriers to terminate or originate calls. Addtionally by using SIP you can look at various ASP based applications you may be able to provide to your end clients.

I would look at using Sip Express Router to help deploying and managing SIP within your environment. Here is a link to start the SIP journey:


This is something that will require some planning and implementation, but the good news is the cost for the software is essentially free, SER can be deployed on many flavors of LINUX (Debian is my favorite distro for its stability). Mandrake would be next of choice due to its ease of use and ability to obtain support if purchased.

I hope this helps and let me know if you need additional help or information.

Kindest regards

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