no audio on voicemail number

Discussion in 'Cisco' started by tg, May 19, 2009.

  1. tg

    tg Guest

    cisco 2651XM router
    IOS: c2600-adventerprisek9-mz.124-15.T8.bin
    sip trunk working and registered to sipgate.co.uk
    7940G ip phone

    I have a problem with my ip phone connection in that I can't get any
    audio when calling my voicemail number, yet all other numbers work fine.
    sipgate voixemail is 50000 from my sipgate phone, or 02070437777 from a
    different line. If I dial 50000 from the sipgate phone the display shows
    'Connected' but there's no audio. If I also dial 02070437777 from my
    sipgate phone I get the same - connection but just silence.
    What's strange is if I call the voicemail number 02070437777 number from
    a different phone I hear audio so the problem has to be with my router.
    I've forwarded the port in the router, that didn't solve it. I know I
    need to run a debug on this but my question is: which debug should I
    run?
    I tried debug ccsip messages but I couldn't see anything in the log
    relevant to this problem. I also ran debug dialpeer, same result. I also
    ran debug voice call trap fsm but I couldn't see anything.
    I know the problem is in the router config because if I use xlite on my
    pc I can get audio on the voicemail no problem.
    thanks for any pointers.
    tg, May 19, 2009
    #1
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  2. tg

    alexd Guest

    tg wrote:

    > cisco 2651XM router
    > IOS: c2600-adventerprisek9-mz.124-15.T8.bin
    > sip trunk working and registered to sipgate.co.uk
    > 7940G ip phone
    >
    > I have a problem with my ip phone connection in that I can't get any
    > audio when calling my voicemail number, yet all other numbers work fine.


    Does the 10000 test number work? Ivor's speaking clock on 020 7043 1320 is
    another Sipgate number you could try.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    20:02:52 up 12 days, 22:36, 2 users, load average: 0.19, 0.15, 0.10
    A few flakes working together can unleash an avalanche of destruction
    alexd, May 19, 2009
    #2
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  3. tg

    tg Guest

    "alexd" <> wrote in message
    news:...
    > tg wrote:
    >
    > Does the 10000 test number work? Ivor's speaking clock on 020 7043
    > 1320 is
    > another Sipgate number you could try.


    thanks for your response, yes both 10000 and 020 7043 1320 work fine, I
    can hear audio when I ring them.
    tg, May 19, 2009
    #3
  4. tg

    tg Guest

    "tg" <> wrote in message
    news:4a12f469$0$24011$...

    anybody?
    please?
    this problem is driving me insane.
    tg, May 20, 2009
    #4
  5. tg

    alexd Guest

    tg wrote:

    > thanks for your response, yes both 10000 and 020 7043 1320 work fine, I
    > can hear audio when I ring them.


    Bit of a tough one to narrow down. Seems a bit odd that 10000 works but
    50000 doesn't. If the debug output doesn't show anything of any interest,
    then I would take a packet capture of each call and compare the
    differences.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    16:16:44 up 13 days, 18:52, 2 users, load average: 0.16, 0.11, 0.09
    A few flakes working together can unleash an avalanche of destruction
    alexd, May 20, 2009
    #5
  6. tg

    alexd Guest

    tg wrote:

    > thanks for your response, yes both 10000 and 020 7043 1320 work fine, I
    > can hear audio when I ring them.


    Are you seeing anything from 217.10.79.35 being dropped, perchance? I
    compared calls to each number, RTP comes from 217.10.79.30 when using the
    test number and 217.10.79.35 when dialling voicemail.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    17:42:58 up 13 days, 20:19, 2 users, load average: 0.06, 0.10, 0.09
    A few flakes working together can unleash an avalanche of destruction
    alexd, May 20, 2009
    #6
  7. tg

    tg Guest

    "alexd" <> wrote in message
    news:...
    > tg wrote:
    >
    >> thanks for your response, yes both 10000 and 020 7043 1320 work fine,
    >> I
    >> can hear audio when I ring them.

    >
    > Are you seeing anything from 217.10.79.35 being dropped, perchance? I
    > compared calls to each number, RTP comes from 217.10.79.30 when using
    > the
    > test number and 217.10.79.35 when dialling voicemail.


    ok thanks but which debug do you think I should run to catch the info
    you speak of?
    tg, May 20, 2009
    #7
  8. tg

    Barry OGrady Guest

    On Wed, 20 May 2009 13:17:30 +0100, "tg" <> wrote:

    >
    >"tg" <> wrote in message
    >news:4a12f469$0$24011$...
    >
    >anybody?
    >please?
    >this problem is driving me insane.


    What is the problem?

    Barry
    =====
    Home page
    http://members.iinet.net.au/~barry.og

    The notion that faith in Christ is to be rewarded by
    an eternity of bliss, while a dependence upon reason,
    observation, and experience merits everlasting pain,
    is too absurd for refutation, and can be relieved
    only by that unhappy mixture of insanity and
    ignorance called 'faith.'
    [Robert Green Ingersoll]
    Barry OGrady, May 21, 2009
    #8
  9. tg

    tg Guest

    "alexd" <> wrote in message
    news:...
    > tg wrote:


    ps: here is a small section of my config which shows the settings I have
    for voicemail with sipgate.co.uk. Does anything look wrong to you?

    voice class codec 1
    codec preference 1 g711ulaw
    <snip>
    dial-peer voice 3 voip
    description **voicemail**
    destination-pattern 50000
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    no vad

    seems to me there must be a wrong setting in there somewhere....

    in fact here's the whole router config is interested:
    http://www.zen73857.zen.co.uk/cisco/runconfig.pdf
    tg, May 21, 2009
    #9
  10. tg

    alexd Guest

    tg wrote:

    > "alexd" <> wrote in message
    > news:...


    >> Are you seeing anything from 217.10.79.35 being dropped, perchance? I
    >> compared calls to each number, RTP comes from 217.10.79.30 when using
    >> the
    >> test number and 217.10.79.35 when dialling voicemail.

    >
    > ok thanks but which debug do you think I should run to catch the info
    > you speak of?


    I meant check ACL logging in case RTP is being dropped unexpectedly,
    although you'd think having the SIP UA running on the edge would eliminate
    this kind of nonsense...

    My only experience with Cisco voice kit is having a UC500 demo kit for a few
    weeks and using a 7941G with Asterisk. A quick google came up with this:

    http://www.cisco.com/en/US/products...s_configuration_example09186a00808f9666.shtml

    There are several interesting looking debug commands, but you'll have to get
    someone else to interpret the output :) If the debug output is no use,
    there's always packet capture.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    17:14:01 up 14 days, 19:53, 2 users, load average: 0.17, 0.14, 0.10
    A few flakes working together can unleash an avalanche of destruction
    alexd, May 21, 2009
    #10
  11. tg

    tg Guest

    "alexd" <> wrote in message
    news:...
    > tg wrote:
    >

    I did the packet trace thing and the data is streaming in 'as though'
    the phone is receiving audio. I compared the packet stream with
    a call containing audio to a call (to voicemail) containing silence and
    they were practically identical. Even though I couldn't hear anything on
    the voicemail number the data was pouring into the phone so it's a case
    of the phone (or the router) can't turn it back into audio. There must
    be something about the voicemail stream that is different to a normal
    PSTN call. This made me think of codec but I've tested every codec
    available in the dial peer and nothing will bring on the audio. It's so
    wierd.
    tg, May 21, 2009
    #11
  12. tg

    tg Guest

    "alexd" <> wrote in message
    news:...
    > tg wrote:


    anyway it looks as as though I might have to throw in the towel on this.
    As a final test I bought some credit with draytel.org and gave them a
    try. I used exactly the same settings on my equipment as I did with
    sipgate and whaddya know...their voicemail worked perfectly first time.
    The voicemail of both draytel and tescointernetphone work fine on my
    setup but sipgate doesn't. There has got to be something wrong or
    slightly different about sipgate the voicemail system but multiple
    emails to sipgate support have drawn a blank.
    tg, May 21, 2009
    #12
  13. tg

    alexd Guest

    tg wrote:

    > Even though I couldn't hear anything on the voicemail number the data was
    > pouring into the phone so it's a case of the phone (or the router) can't
    > turn it back into audio.


    If you open up a packet capture in wireshark, you'll be able to play the
    captured audio stream back [if it's SIP/RTP, not sure about SCCP]. One
    thing I noticed with the voicemail call, the call setup packets coming from
    Sipgate didn't specify a packetisation time ['ptime']. May or may not be
    significant.

    > There must be something about the voicemail stream that is different to a
    > normal PSTN call.


    The only difference I've been able to discern is what peer the RTP goes
    to/from, and the lack of a ptime in the SDP part of the call.

    This looks useful:

    http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_example09186a0080672b8b.shtml

    > This made me think of codec but I've tested every codec
    > available in the dial peer and nothing will bring on the audio. It's so
    > wierd.


    The debug output I got from Asterisk when calling Sipgate voicemail and
    other numbers indicated the accepted range of codecs were identical. Also
    the call setup would fail if the codecs don't overlap - an essential
    feature of SIP.

    --
    <http://ale.cx/> (AIM:troffasky) ()
    20:15:09 up 14 days, 22:54, 2 users, load average: 0.16, 0.07, 0.07
    A few flakes working together can unleash an avalanche of destruction
    alexd, May 21, 2009
    #13
  14. tg

    alexd Guest

    tg wrote:

    > There has got to be something wrong or
    > slightly different about sipgate the voicemail system but multiple
    > emails to sipgate support have drawn a blank.


    Sipgate support is pretty much non-existent, and such support that does
    exist will be unlikely to include Call Manager. I'm not knocking Sipgate
    however; the service is excellent for the price!

    --
    <http://ale.cx/> (AIM:troffasky) ()
    12:33:04 up 15 days, 15:16, 2 users, load average: 0.07, 0.09, 0.09
    A few flakes working together can unleash an avalanche of destruction
    alexd, May 22, 2009
    #14
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