record individual callers in confbridge

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Asterisk Users 2 Comments

> I’d suggest try recording in ulaw first and then convert all to wav after.

> It may have something to do with timing since you’re using iax, what are you using as a timing source? Hardware or software?
>
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2 thoughts on - record individual callers in confbridge

  • I believe I am using software, since I don’t have any special hardware
    installed. I do have a Digium card somewhere in a box. I can try
    recording in ULAW, but I’m unsure how to convert it to WAV. I think
    Asterisk can do it, so I’ll give it a try and look it up.

    If I can get this sorted out, I think it’ll be a great podcast solution.

    Thanks.

  • On Mon, Mar 21, 2011 at 8:33 PM, Neeraj Chand
    wrote:
    installed.
    directory

    Hi again,

    I tried recording a podcast a couple days ago. I recorded in ulaw format,
    and converted to wav afterwards, as you said. Our session ran over double
    the length they usually do, over 3 hours. The sync is much better than my
    first tests, and that is with the jitter buffer enabled – or at least, I
    believe it is – I saw this output:

    *CLI> iax2 show channels
    Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx)
    Lag Jitter JitBuf Format FirstMsg LastMsg
    IAX2/tayler-129 70.xx.xx.169 tayler 00129/25402 00048/00048
    00075ms 0065ms 0119ms ulaw Rx:NEW Rx:ACK
    IAX2/josh-346 216.xx.xx.2 josh 00346/21060 00138/00175
    00088ms 0069ms 0114ms ulaw Rx:NEW Tx:ACK
    IAX2/dana-1985 66.xx.xx.7 dana 01985/21748 00013/00043
    00082ms 0005ms 0079ms ulaw Rx:NEW Tx:ACK
    3 active IAX channels
    *CLI> Resyncing the jb. last_delay 67, this delay 2079, threshold 1126, new
    offset -2079
    Resyncing the jb. last_delay -189, this delay 1097, threshold 1136, new
    offset -3176

    However, the sync goes out of sync slowly. I recorded both in and out
    channels for all three participants, and even the pairs don’t match up
    properly. System load was around 0-0.01 the whole time, according to top.
    Now, for the first hour or so, the sync was fairly OK (only minor adjustment
    needed), but it drifts after that and I need to adjust more frequently in
    Audacity.

    I am not sure which timing modules I am using. I tried to find the one you
    mentioned:

    $ sudo find / | grep -i dahdi_dummy | wc -l
    0

    I figure this is because I compiled Asterisk from source (because I had
    originally planned to use wideband codec but couldn’t find any
    cross-platform IAX client with wideband support). Is there a way to tell
    which timer module is in use, or which I compiled in? If I specify in
    modules.conf which to noload and which to load, would that do the trick (and
    does that even work when compiled from scratch?). I can always recompile,
    and I still have my source tree. I also found my old Digium card (it’s
    around 6 years old).

    Thanks for your help.

    Dana