[Asterisk-Users] AArgh, * and the 7960

brian k. west brian at bkw.org
Tue May 18 18:22:44 MST 2004


Lets look at this and FIX the problem instead of hacking it.  What you need
to do is install etherreal and capture a call and parse the timestamp info
to see if they are slipping.  Because they are perfect here.

bkw

----- Original Message ----- 
From: "Brian Cuthie" <brian at systemix.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, May 18, 2004 5:07 PM
Subject: Re: [Asterisk-Users] AArgh, * and the 7960


>
> Iain,
>
> This is a known issue with the Cisco phone and Asterisk having to do
> with a change made later in the cvs tree. Try 1.0 stable, or modify
> rtp.c to comment out the two lines as follows:
>
>                 /* Re-calculate last TS */
>                 rtp->lastts = rtp->lastts + ms * 8;
> //              if (!f->delivery.tv_sec && !f->delivery.tv_usec) {
>                         /* If this isn't an absolute delivery time,
> Check if it is close to our prediction,
>                            and if so, go with our prediction */
>                         if (abs(rtp->lastts - pred) < 640)
>                                 rtp->lastts = pred;
>                         else {
>                                 ast_log(LOG_DEBUG, "Difference is %d, ms
> is %d\n", abs(rtp->lastts - pred), ms);
>                                 mark = 1;
>                         }
> //              }
>         } else {
>
> This seems to work for me. Others may have more insight.
>
> -brian
>
>
> Nik Martin wrote:
>
> >Out of context, this isn't much information.  Is your network connection
OK?
> >Is your broadband provider having troubles?  Has some upstream hardware
> >changed that you may not be aware of?
> >
> >
> >
> >
> >>-----Original Message-----
> >>From: asterisk-users-admin at lists.digium.com
> >>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> >>Iain Stevenson
> >>Sent: Tuesday, May 18, 2004 1:29 PM
> >>To: asterisk-users at lists.digium.com
> >>Subject: [Asterisk-Users] AArgh, * and the 7960
> >>
> >>
> >>
> >>I've just had the most appalling performance from * ever.  Dialling:
> >>
> >> Cisco 7960 => asterisk => IAX
> >>
> >>produces sound drop outs so extreme that the call is useless.
> >> I noted this
> >>in an earlier post. Dialling:
> >>
> >> Cisco ATA186 => asterisk => IAX
> >>
> >>is fine.
> >>
> >>Frankly, I think this is such a bad problem that it should be
> >>sorted in
> >>advance of any of the new features that seem to be getting
> >>such prominence
> >>nowadays.  It was not present earlier in the year and I
> >>haven't upgraded my
> >>7960.  So I don't think you can point the finger entirely in Cisco's
> >>direction.
> >>
> >>  Iain
> >>_______________________________________________
> >>Asterisk-Users mailing list
> >>Asterisk-Users at lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >>
> >>
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>





More information about the asterisk-users mailing list