[Asterisk-Users] Beginning of voicemail missed by sip phone

Mark Spencer markster at digium.com
Thu Mar 13 14:01:42 MST 2003


Can somebody look at the RTP packets with "ethereal" and tell me if they
notice any difference between what we send and what we receive?  Perhaps
we're starting out with values that are too high or something?

Mark

On 13 Mar 2003, Matteo Brancaleoni wrote:

> I can confirm that.
> With the snom, I get no delay.
> with a sip-fxs gw, I get 2 seconds delay.
>
> Matteo
>
> Il gio, 2003-03-13 alle 19:59, Lele Forzani ha scritto:
> > On Thursday 13 March 2003 18:00, Benjamin Miller wrote:
> >
> > > Actually I've seen this exact issue with my Cisco 7960.
> > > And it's any voice prompt I dial.  I loose just the very first .5
> > > seconds of the audio for whatever reason.
> > > So the sip users hear "eedian Mail" and "nk you for calling".
> > > Any one else dealing with this?
> > > Any ideas?
> >
> > Same here. But it seems to be somewhat related with the hardware: the 7960
> > looses about .5 seconds, and an old siemens optipoint 100 I have around
> > looses up to 2 seconds.
> >
> > But there's no delay with the ATA186 (sip) and with the Pingtel Expressa. With
> > the ATA I can hear the change in the background noise shortly before the
> > beginning of the recording, which is probably the connection of the rtp
> > stream.
> >
> > lele
> >
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Matteo Brancaleoni <mbrancaleoni at espia.it>
> Espia - Emmegi Srl
>
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>




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