[Asterisk-Users] Softphone Audio problem

[Michael Van Donselaar]

On Thu, 20 May 2004 14:18:16 +0100, "Andy Farnsworth" <farnsaw at stonedoor.com>
wrote:

>As a test, I was trying to use Iaxcomm and Iaxphone to connect to
>Asterisk and dial out to my other line.  Using either of these soft
>phones, I can connect to Asterisk and listed to audio just fine.  I can
>even connect across the net to another asterisk server and hear audio
>just fine, however, when I dial out to my second land line the audio
>that is transmitted is horribly broken up.  It is as if the audio stream
>is broken into 8 parts every second and then every other part is
>dropped.  I then tried the asterisk echo test and got the same thing.  I
>am running Asterisk under RH9 on an AMD 2600+ 512 Mb RAM Desktop and the
>soft phones on my laptop running Windows XP (Laptop is Sony Vaio
>PCG-GRT815E, 2.8 Ghz processor, 512Mb Ram).
>
>Is this an asterisk problem or a soft phone problem?  If asterisk, any
>ideas on how to fix it?

What kind of PSTN interfaces are you using?  I'm not sure from the description:
are you seeing the problem of both lines, or only the second line?  Do you get
the same kind of results when using a SIP softphone?

I'll be posting new binaries to sourceforge this weekend, because there have
been some library changes related to jitter, but I haven't heard or seen
anything as drastic as you describe.

BTW what version of asterisk?

>
>Thanks,
>
>Andy Farnsworth
>
>
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