[Asterisk-Users] AArgh, * and the 7960
Brian Cuthie
brian at systemix.com
Tue May 18 16:07:20 MST 2004
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
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>>
>
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