MOH problems if calls from outside

Discussion in 'General' started by MST, Jul 1, 2010.

  1. MST

    MST

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    Looks like this is a bug affected 1.6 and was not fixed in 2.0 CR

    To make it short: when I place someone on hold the music is perfect inside of a network - so far ok. When someone calls from outside using SIP, PSTN, etc trunks the MOH gets choppy
    creepy, etc......

    I have tried 3 new installation 2 X Elastix 1.6 and 1x 2.0 and all of them have the same problem.

    I have seen 6 or 7 posts but none of resolution works so far.

    Experts is there any final and official fix for that problem? This is one of the most important Elastix' features, and really need that working.


    Thank You
     
  2. jgutierrez

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    And do you have the same audio issues when you are talking?
     
  3. MST

    MST

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    No there is no issue at all. All 3 default song that are included in Elastix are ok
    - no choppy at all. When use Elastix and MOH no matter using pSTN or Voip Provider still choppy sound and clients are complaining when they are on hold.

    The same think when use VOIP ICON ....

    The same think is with 1.6 as 2.0


    Besides that I have QOS set up for VOIP in Cisco devices and it was tested under heavy DATA LOAD so it is not QOS problem. As I see all files are GSM and WAV and here I think is the problem. I susspect all songs attached by default to Elastix but they are ok when I play them in Win OS crappy OS so it has to be something wrong with Elastix.
     
  4. dicko

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    Or perhaps your moh files are encoded badly.

    for example:


    file /var/lib/asterisk/mohmp3/fpm-sunshine.wav

    will return

    /var/lib/asterisk/mohmp3/fpm-sunshine.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz


    Your uploaded files should reflect identical encoding whether encapsulated or not, just the same as the distributed files, and of course they should be readable by asterisk.


    FWIW it is always better to use the un-encapsulated .wav encoding as they will not need transcoding.

    dicko

    p.s.

    Please remember that linux is case sensitive so .WAV files are not the same as .wav files and the name should reflect what is in that file, and .GSM files will not be recognized at all, rename them to .gsm and make sure they are actually gsm encoded.
     
  5. Bob

    Bob

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    First of all the musiconhold function is almost purely a function of Asterisk. So I would be looking at issues with Asterisk 1.4 and 1.6, particularly bug reports on the Asterisk system. To the best of my knowledge the Elastix base just supplies files supplied by Asterisk.

    Secondly, of all the installations that we have done of 1.4 and 1.6 Elastix, the music on hold has never been an issue, so as a wide spread issue I don't think that it is a bug that has to be fixed (although I have mentioned to look at the bug reports on Asterisk as there may be a special condition where it appears).

    One of the biggest variables that does come into the equation that is a more likely the culprit is the hardware that does have a reasonable input into the quality, in particular in the following areas:-
    Music On Hold
    Conference Rooms
    and I am sure there are others...

    Now I am assuming that you are using a TDM400 for your PSTN, if not then you need to get a card to provide a reliable timing source.

    On some hardware, you don't need the card, but experience has shown that the majority of hardware, both low cost and some higher end systems, cannot provide a reliable time source in motherboard hardware, and the card (even a non utilised TDM400/410 with a single fxs or Fxo port) is a valuable investment in even SIP only based systems.

    Now even if you have got a card in your system, you could still be having an issue with interrupts. Some of the motherboards, especially with some of the SATA controllers, have implemented "hacks" to improve their performance (especially to improve their scores in speed tests), especially in the disk access. E.g. some of the BIOS settings for Sata such as Enhanced, combined or Auto modes are areas worth playing with.

    Also have a read of dahdi_test and zttest (for the older systems) and look for a consistent 99.97%+. If it drops below or varies constantly below this, then you need to look closer at your hardware.....

    This is even more important if you are using X100p (or their look alike) or using external PSTN units (e.g. SPA3102, SPA400 etc)....or GSM gateways...

    Regards
    Bob
     
  6. MST

    MST

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    Thank You Gentlemen That give me better idea of what comes from and to whom belongs to.

    There is something that I don't fully understand:

    "Now I am assuming that you are using a TDM400 for your PSTN, if not then you need to get a card to provide a reliable timing source. "

    If I don't have any PSTN card in a server that means I don't have any realiable timing source..... and that means more trouble ? What about If I have only VOIP provider (none of any PSTN cards) and use stricly all calls throught VOIP provider? Is that mean that I need to have PSTN card even if don't have to use it in order to have MOH working correctly?

    I thought inserting internal_timing=yes into asterisk.conf would fix a lot of sound problems without having any PSTN card in a server.

    P.S We use DELL 2650 with no SATA HDDs ....

    Regards,

    MST
     
  7. Bob

    Bob

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    You have misread a little bit.....

    As a summary there are a lot of systems out there that provide a stable enough timing source for Asterisk functions without a hardware card such as a TDM400/410. I have implemented a few.

    However, there are also a reasonable percentage that cannot provide a reliable timing source, and need some sort of hardware to provide this timing source.

    The way I have always looked at it, is that I am providing a IP PBX system which is a critical part of their business, and I want the installation to be faultless, no second guessing whether timing is issue. For the minimal cost (as opposed to the main server/machine used) of adding a card (which naturally you incorporate in the cost) it is a nice simple equation. Of course you have some hardware that offers no space for a card, and here you take your chances. That's why even fanless or slimline hardware chosen, I always look for a model that will take at least one card.

    Even for systems unseen (e.g. someone else has implemented), I always have a tdm400 with me, but as I said, run the tests I mentioned in the previous post, this will give you an idea of how stable your hardware timing is.

    This is not a slight on your hardware, the Dell is a good machine, but it can be possible that the timing source that Asterisk uses on this hardware is stable for all other functionality (e.g. Windows, Linux), but when it comes to a voice application it might not be enough....

    Regards

    Bob
     
  8. MST

    MST

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    Thank You Bob.

    Now it's clear for me.

    Cheers,

    MST
     

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