Missing voicemail prompt beginning

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Asterisk Users 9 Comments

Hello,

I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like “number 12345 not available” I was only
hearing “345 not available”. Verbose level 5 on the asterisk console didn’t
give me any hint on this, it only shows that playback of the prompt started
correctly in every test case. Any hints on how I can debug this? I think
it’s some problem on my local configuration, I doubt it’s a problem with my
SIP provider or mobile phone provider, they are both very reliable (Sipgate
and T-Mobile).

Thanks for any hint!

Best regards
Stefan

9 thoughts on - Missing voicemail prompt beginning

  • Stefan at WPF wrote:

    User answer(500) or wait(1) before the audio prompts.

    Example:

    exten => s,1,Answer(500)
    exten => s,n,Voicemail({$ARG1}@sip,u)
    exten => s,n,Hangup()

    Doug

  • First of all, thank you for your reply, however I see two problems with
    this solution:
    1) I think sometimes even more than a second from the beginning of the
    prompt is missing, so I have to set a larger value, meaning in cases where
    nothing of the prompt was missing, the calling person listens to a pause of
    some seconds.
    2) Your solutions handles the symptoms of the problem, I’d like to fix the
    root cause of this problem.

    Any ideas on number 2, fixing / finding the root cause of this problem?
    Thanks 🙂

    2012/6/17 Doug Lytle

  • Stefan at WPF wrote:

    The root cause of the problem (Most likely) is that the channel hadn’t
    be answered. A wait, allows the channel to be established and audio to
    pass.

    Doug

  • 2012/6/17 Doug Lytle

    Which end do you mean with “channel not answered”? The asterisk end or
    mobile phone end of the channel? Also I am confused that it sometimes work
    and sometimes it does not? ): I tried with Wait(1) and Answer(1000),
    unfortunately both didn’t change things – sometimes the complete prompt is
    there, sometimes the beginning is missing ):

    Are there any relevant logs for these things / how to check what the
    problem is without trying settings? Thanks 🙂

  • Stefan at WPF wrote:

    The Asterisk side. If the answer didn’t fix the issue, then my guess is
    that it’s on the cellular provider’s side (Which isn’t unheard of).

    Doug

  • Hmm, I tried calling myself (the asterisk voicemail) from another SIP
    provider, same problem. What always works reliable is using and calling the
    voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably
    hear the complete prompt. Doesn’t this contradict the assumption that the
    problem is on the mobile phone side?

    2012/6/17 Doug Lytle

  • Please excuse the top post, I’m on my phone.

    Before we have a better idea of what’s going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt.

    Also, if you can clarify the infrastructure setup as well, that would be helpful.

    Thanks,

  • Thank you Warren,

    I will temporarily skip this step, as I don’t have the problem anymore,
    though I don’t know why (for that and learning purposes the logs maybe
    would be still useful).
    I found some different settings for Asterisk and Sipgate (actually I found
    the settings for private users on the Sipgate website, before that I found
    the settings for business customers and assumed there wouldn’t be a
    difference).
    When I had the problem, my sip.conf looked like this:

    [general]

    Now my sip.conf looks like this (source:
    http://www.sipgate.de/faq/index.php?do=displayArticle&article=540&id=257):
    (I have commented the additions / changes)

    The dialplan in both cases was this:

    (I left out the Dial command for testing purposes after I found the
    voicemail prompt problems)

    If anyone has an idea why it now works without problems, please let me
    know for learning purposes. I still have to read up on the options. When I
    have more time I will probably also set the old settings again to learn how
    I could have identified the problem.

    2012/6/17 Warren Selby

  • Sorry for the second mail, about the infrastructure:
    phone – asterisk – HW firewall including NAT – Sipgate SIP Provider

    About Software:
    Asterisk 1.8.13.0 running on Raspbian Debian Linux (http://www.raspbian.org/,
    Raspbian includes up to date Asterisk paackages while the normal Raspberry
    Pi Debian does not) running on a Raspberry Pi http://www.raspberrypi.org/:-)

    2012/6/17 Stefan at WPF