Dialing Out Issue

MG_732_08873

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#1
Newbie question, I think.

We just set-up Elastic PBX and everything is working fine except from time to time when we try to dial out (local number or a national number) we get the annoying "Due to area code changes" unable-to-connect message. Except after several consecutive tries this problem ceases to exist.

My question is what is causing this random error (when there shouldn't be one) and how can we resolve it so that we can dial any number at any time without it giving this unable-to-connect error.


Any help or direction would be greatly appreciated, thanks ahead of time guys.
 

dicko

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#2
Welcome here!

can you give us some more information as to what trunks you use and the providers that you connect to on them?
 

MG_732_08873

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#3
We have the Digium TDM410 FXO card and 4 Analog PSTN line. Our outbound route dial pattern is 1NXXNXXXXXX and NXXNXXXXXX.

We use the SIP for internal communications (which works fine) and additionally we have a PSTN line through a (Verizon Analog Lines).

I hope this helps. Thank you.
 

dicko

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#4
As a start, I would add a trunk dial rule

1+NXXNXXXXXX

to your trunks (presumable ZAP/g0) which will replace NXXNXXXXXX with 1NXXNXXXXXX before sending them to your carrier (eleven digits is usually preferred and will likely become mandatory in the near future) .

Is the Verizon trunk on the TDM card or separate and possible included in the outbound route?
 

MG_732_08873

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#5
First off, thank you for the suggestion. After our tech department tried, it seems the problem wasn't resolved but thank you regardless.

As for as the Verizon trunk, we don't have a verizon trunk, we only have a TDM card and a analog trunk. And Verizon is the entity that services our analog PSTN line.

Thank you.
 

dicko

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#6
Look at the logs to see if the digits were actually sent to the trunk correctly.
Have your tech department listen on the individual trunks (= pstn lines in this context) preferable with a "butt set" and use there ears to see if the notice anything abnormal on the incomplete calls, like delayed dialtone presentation or MWI tones, missing "most significant digits" significant echo or unbalanced Transmit/Recieve levels.
 

MG_732_08873

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#7
Thanks a lot, I'm going to have the tech department look into that. And from searching the web in general, it doesn't seem to be a common problem, and the fact that its not a constant problem with us makes it harder to pin-point the problem.
 

dicko

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#8
Unfortunately when you try to debug analog lines, you have to resort to analog debugging methods.
But don't worry we'll get you there!
 

ramoncio

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#9
You can enabe debug in asterisk console and check into the logs for the problem.
Tell them to write down the exact time when the problem happens again so you can find what you want easier. If you set a high enough debug level you should see what the problem is or maybe you can put the relevant part here so we can help you.
But watch out, and remember to disable debug after you solve the problem, or you can get really big log files.
 

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