didlogic

Setting up a trunk to a SIP call provider in Asterisk can be a pain. Even for me – and I do it a lot!

When we set up a SIP call provider in Asterisk (a SIP trunk) and send calls to them our Asterisk server will send the call provider a SIP INVITE. Their system could respond in many different ways – decline the call, ask for user info, silently drop the call, process the call ….

Sometimes the call provider will send back information in the SIP reply as to why the call has failed. Maybe you have no credit, maybe the codec is not supported, maybe you have the number in the wrong format … but often they will not, they will just not process the call and send back a generic response.

At this point you need some help from the call provider. You need them to tell you why the call is not completing correctly.

Recently I was helping a customer set up a trunk to didlogic. His Asterisk server was sending the call to didlogic, but their system was responding with a fairly generic reply, indicating the SIP credentials were incorrect.

They seemed either unable or unwilling to investigate why this was. Even after being provided with SIP traces they showed no interest, just replying with generic suggestions that were no help.

It turned out that creating a new SIP account in the didlogic web portal resolved the issue. The old SIP credentials didn’t work, the new SIP credentials did.

I get asked a lot for call provider recommendations, and this is the only experience I have with didlogic, but based on it I couldn’t recommend them.

If you are having issues sending SIP calls to them (and getting just s SIP 407 back) maybe try creating a new set of SIP credentials in the didlogic portal.

Update 30/5/2014

I had a response to this post from DIDLOGIC which sounds encouraging. This was the reply –

Hello Matt,
Thanks very much for taking the time to detail these issues. It has been a couple of months and we have made our maximum effort to get a handle on the situation with authentication errors related to various flavors of Asterisk.
We apologize for causing your such inconvenience. Please be advised that there is now much more scope for troubleshooting these rare occurrences, as we have invested heavily in improving our support and customer service.

In case you ever run into such difficulty again, please send a short message to [email protected] and CC the engineer assigned to your account. All system builders and VoIP consultants integrating our SIP trunking solutions for their clients get a dedicated point of contact and such trivial issues are dealt with immediately.

Thanks again for trying our services.

3 thoughts on “didlogic

  1. sam

    hi Matt,
    Good to read your experience. Any other Sip providers you recommend. Specifically for getting local tel numbers in West Africa. Nigeria, Ghana, Ivory Coast. Deallogic is the only company. Trying to find out the limitation on their sip channels for incoming calls. It does not state the flat rate channels for each of the countries just uk, germany etc. Emailed customer services 3 times .. over the past 2 weeks but no reply.
    Any recommendation on other companies I can speak with would be great! Been searching for several weeks now.
    Thanks
    Sam

  2. Din

    hi Matt,
    It would be good if you can make a write-up on what each means, in a general sense. (i.e) “decline the call, ask for user info, silently drop the call, process the call ….”.
    Maybe do an inbound call setup for different types of trunks. (i,e) other than the free ones.
    Thanks
    Din

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