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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
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
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