[Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU

[luis.kibe at kddi.com.br]

Additional information : I use brazilian E1 variant "br"

Luis M. Kibe

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luis M. Kibe
Sent: Tuesday, June 14, 2005 6:00 PM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU

Hi Steve
I have a similiar problem with noise.
Asterisk SIP to SIP calls works without problems. During outbound and
inbound PSTN calls, if there is only single call, the system works perfectly
as well - voice is crystal clear.
However, 10 - 60 seconds after a 2nd simultaneous call in the E1 starts, the
voice becomes garbled and delay starts to increase to a point where the
quality is too bad for the call to continue. Any idea ?
Versions :
Hardware : Wildcard TE405P
asterisk-1.0.7
zaptel-1.0.7
libunicall-0.0.3pre3

Best Regards,
Luis M. Kibe

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Steve Underwood
Sent: Friday, June 10, 2005 10:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU


Hi,

We have now solved this problem. There was a bug in selecting codecs 
when chan_unicall generates DTMF or supervisory tones. If anyone else is 
having a similar problem with high CPU usage when running chan_unicall 
try stable version unicall-0.0.2b, or test version unicall-0.0.3pre3. 
They contain the fix.

Regards,
Steve


Andres Maduro wrote:

>  
> Hi,
>  
> I have recently found a bug when using Steve Underwood chan_unicall 
> with Asterisk 1.0.x (including 1.0.8RC)
>  
> When you place a call from a SIP phone with dtmfmode=rfc2833 or 
> dtmfmode=inband through MFCR2 via chan_unicall all goes well until you 
> press a dtmf key.  When you do this, the other end hears a garbage 
> sound (not the dtmf tone) and cpu goes to 99.9% rendering almost 
> unusable the PBX.  If there are more than 2 calls, audio start to get 
> choppy, more calls renders unusable the pbx.
>  
> If you hangup the calling extension, almost all the time it returns to 
> normality, if there is a moderate load on the * server, the only way 
> of shutting down * is by killing -9 it.
>  
> I have been working this with Steve and have reported this finding today.
>  
> If you have any suggestion in which things could be tweaked in 
> chan_sip.c, chan_zap.c or chan_unicall.c in order to see if this bug 
> could be solved, I will be happy to test it.
>  
> Any additional info you may require please let me know.
>  
> Regards.   
>            AM.


_______________________________________________
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