Hi everyone,

It has been while I’v not written anything here but this week I encountered a weird behavior that I want to share with you.

It started with the wish to set up a coexistence between a SIP telephony provider, a Skype for Business and an Alcatel OXE. As the Alcatel OXE was not having SIP board so we had to use an E1 trunk. To do so, we chose an SBC Audiocodes Mediant 800C.

This is just the beginning of the story because the customer wanted to give the choice to his users to use the Alcatel or Skype for Business at any time so we had to use call forking…

Then the nightmare started…

To do call forking with an Audiocodes, you have to set up two IP-To-IP routing rules and the first one has to be configured like this :

  • Alternative Route Options = Route Row
  • Group Policy = Forking

Like below :

As this rule is supposed to route calls to Alcatel, the Destination Type is “gateway”.

The second IP-To-IP routing rule will have to be configured like this :

  • Alternative Route Options = Group Member Ignore Inputs
  • Group Policy = Sequential

Like below :

As this rule is supposed to route calls to Skype for business, the Destination Type is “IP Group”, the Destination IP Group is “IPG_SFB” and the Destination SIP Interface is “SIP_SFB”.

And “voilà” !!! So we took a phone and tried to call a number that is available both on Skype and Alcatel but only the Alcatel ring….

Having a look at the Syslog from the Audiocodes, we saw only one call to the Alcatel which was the case but going through the log, we saw that error after the INVITE :

We can see that it lacks of UDP ports… After a call to an Audiocodes trainer, he told us that Alternate routing and Call Forking with E1 need UDP ports to create the second call session.

After we enabled UDP ports on the SIP provider interface, the error disappeared and calls arrived both on Skype and Alcatel phone.

Below is the trace of the two calls :

To summarize, if you want to do call forking or alternative routing with an Audiocodes Mediant, SIP PBX and E1 PBX, enable UDP ports on your SIP interfaces.

Hope this will save you time 😉

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