Weirdest Outbound Call Problem Ever

Discussion in 'General' started by ericbringas, Nov 16, 2009.

  1. ericbringas

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    I recently bought a preconfigured elastix box (Elastix ELX-025 Sangoma 4 FXO, 1 FXS EC) and I am having this big issue about outbound calls. the box is connected to 3 POTS lines. Our tel. provider is At&T. I am located in San Diego, CA. I use aastra 6731i phones.

    First after installing onsite, Outbound calls is a hit and miss. First it will go through without any problem, but then the next time I dial out, it just fails. Our tel. provider say "your call cannot be process at this time" or some other messages. I should add Incoming calls is no problem at all.

    I brought back the box to our Lab and the box just worked out fine. Calling out is solid. No fails. After few days of testing I re-deployed the system back to the client and AGAIN outbound calls are failing.

    Looking back at this event, It seems the difference bet. our office setup and the client setup is the telephone line. What puzzles me more is our tel. lines and the client lines are using the same telephone provider.

    So the big question? Why would my system work on one location, and it won't on another location?
    At this point I kinda dumbfounded on what the issue is. Is this a DTMF issue?

    Has someone ever experience such a problem? If you do please give me some ideas. At this point I would try everything.

    Thanks
    Eric
     
  2. telecomtechnician

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    Hi there, please do the following:

    1) Check that the three lines have dial tone from the wall jack. Use an analog phone to test them. Check the wiring, connections and jacks.
    2) Verify that the settings in the trunk configurations related to line groups are correctly configured. I assume that the three zap lines belong to the same group.
    3) Offhook three extensions and dial an outbound call on each. This three calls should be done simultaneasly.This test is very important.


    All these hints should tell you were the problem is (cabling, hardware, configuration).

    Good luck

    David Medina
     
  3. ericbringas

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    Thanks David

    I should have add. We bypass inside wiring and actually connected elastix right through the MPOE (minimum point of entry) of our provider . we have verified all lines have dial tone. And we did use all the three lines at the same time. But then we make the second call then it all fails. I can hear the voice message (coming from my provider) "sorry please make your call later".

    Another test we did is to connect analog phone on each of the lines and we can do multiple dialling out with no problem.

    I also need to mention. We have an old PBX system connected to the same lines and it doesn't have any problem calling out.

    rgds
     
  4. telecomtechnician

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    Check this

    Well, it seems a dtmf problem related to the elastix server, have you changed the settings? there are a lot of options to play with.

    My brainstorming produced this questions:

    Could it be a wrong setting in the aastra phone?

    Does the failure occur on the same phone or randomly?

    Does the failure occur randomly on any of the lines or in one in particular?


    Waiting for your comments

    David Medina
     
  5. ericbringas

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    Re:Check this

    I have 4 phones. it's unlikely that all fails? the failures are on random phones and on random lines. it seems that the aasta are all set to dtmf 8322.

    I have not played around dtmf option on the elastix server. I don't even know where to start really.

    thanks again david
     
  6. dicko

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    Re:Check this

    Given your two situations, where one location works and one doesn't. . .

    First off having identified that the rather woolly description of "your call cannot be process at this time" or the less likely "sorry please make your call later", is actually coming from the telco and not your own box (the exact wording would help but watch the asterisk CLI to verify), I would suggest ,to start with,that it's time to go "back to basics", i,e stick a "butt set" on the lines and monitor the behavior, there are an almost infinite set of USOC's (Universal Service Order Codes) that can be applied by the same provider to different lines, even out of the same CO (Central Office) the wrong admixture can spoil your whole day, only they can be authoritative about the USOC's and only they seem to regard those settings as state secrets, ( go try and get them from your telco!), I would suspect "far end disconnect" problems causing delays in trunks being stable/available, incorrect rx/tx gain settings (and thusly possible DTMF mis-detection) or even trunk mis-wiring (split/reversed pairs etc.)

    Analog trunks cannot use rfc8322, it will be translated into inband DTMF audio as asterisk trans-codes to ulaw g711, which is what you HAVE to use on your trunks, so don't bother chasing that red herring.

    JM2CWAE

    dicko
     
  7. blangys

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    Re:Check this

    I've had this issue with other systems, such as the Toshiba CIX. Analog lines can vary greatly from place to place and that definitely can impact how your dtmf is interpreted. The phone company will generally come out and test the signal level for you if you complain about it.
     
  8. ericbringas

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    Re:Check this

    These are all possible solutions which I'm going to try. Can some please point me to any procedure to play around dtmfmode on the elastix?

    thanks
     
  9. dicko

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    Re:Check this

    again,

    .
    .
    Analog trunks cannot use rfc8322, it will be translated into inband DTMF audio as asterisk trans-codes to ulaw g711, which is what you HAVE to use on your trunks, so don't bother chasing that red herring.

    In other words ALL DTMF will be translated to INBAND signalling on your analog trunks, effectively and tranparently by Asterisk, dtmfmode is NOT applicable.

    But feel free to peruse http://www.voip-info.org/wiki/view/Asterisk+DTMF if you have time, it will explain in more depth what you can and can't do with DTMF on analog trunks.

    dicko
     
  10. rollinsolo

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    Re:Check this

    Another suggestion would be to port those numbers over to a SIP provider and send that card back for a refund unless you want to keep one line and the card for a fax server. set up a few temp DID's with the SIP provider and call ATT and have them foward all calls to the DID's until they can port them over. It's dirty but your dealing with a business and they cant be down.
     
  11. ericbringas

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    Re:Check this

    Just an FYI about this problem. The solution was simple. Adding "w" prefix did it for me.
     

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