asterisk users - Sep 2009 - Native bridging analog phones trouble DAHDI channels.

I own a TDM2400 board, with three FXO modules and one FXS.
I'am having trouble with analog sip phones, from two different 
equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), 
sometimes when I am calling someone, then I press flash, and then call 
someone else, both calls stay connected after I hang up.

[Sep 29 07:18:06] VERBOSE[3218] logger.c:     -- Called g2/16
[Sep 29 07:18:06] DEBUG[3218] chan_dahdi.c: Sent deferred digit string: T16w
[Sep 29 07:18:07] VERBOSE[3218] logger.c:     -- DAHDI/9-1 answered 
SIP/130-085c6958
[Sep 29 07:18:17] VERBOSE[3218] logger.c:     -- Started music on hold, 
class 'default', on DAHDI/9-1
[Sep 29 07:18:17] NOTICE[3218] rtp.c: Unknown RTP codec 126 received 
from '192.168.0.105'
[Sep 29 07:18:17] VERBOSE[3056] logger.c:     -- Stopped music on hold 
on DAHDI/8-1
[Sep 29 07:18:17] VERBOSE[3056] logger.c:     -- Stopped music on hold 
on DAHDI/9-1
[Sep 29 07:18:17] DEBUG[3056] chan_sip.c: SIP transfer: Succeeded to 
masquerade channels.
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: New owner for channel 8 is 
DAHDI/8-1
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: master: 8, slave: 9, 
nothingok: 0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Stopping tones on 8/0 
talking to 9/0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Stopping tones on 9/0 
talking to 8/0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Making 9 slave to master 8 at 0
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Added 20 to conference 9/8
[Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Added 19 to conference 9/9
[Sep 29 07:18:17] VERBOSE[3218] logger.c:     -- Native bridging 
DAHDI/8-1 and DAHDI/9-1

I am using Elastix, 1.5.2-2.3. I do not know if the trouble is caused by 
elastix or some trouble with the digium board configuration. I am using 
DAHDI modules.


Dahdi show status.
Description                              Alarms     IRQ        
bpviol     CRC4     
Wildcard TDM2400P Board 1                OK         2          
0          0        

Core show version

Asterisk 1.4.25.1 built by root @ rpmbuild32.elastix.palosanto.com on a i686
running Linux on 2009-06-14 11:49:25 UTC



I do not understand what it is happening.

Thank you in advance

Maurizio Faccio

Maurizio Faccio adinet wrote:
> I own a TDM2400 board, with three FXO modules and one FXS.
> I'am having trouble with analog sip phones, from two different 
> equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), 
> sometimes when I am calling someone, then I press flash, and then call 
> someone else, both calls stay connected after I hang up.
That's because you have just completed a flash-hook based transfer of
the first call to the second call. If you don't want this feature, set
'transfer=no' for the relevant channels in chan_dahdi.conf.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org