[Asterisk-Users] dropped calls

[Paulo Loureiro]

Hello list,

I'm getting droped calls on an asterisk installation. When on GS phone
dials another one, the call is dropped after some (usually random) time
but most of the tome within 3 to 20 seconds.
I think the cause is stated on the logs, see bellow, and is related with
the message  "Didn't get a frame from channel: SIP/3805-df43", but I
can't figure why.


asterisk logs:
-------------------------------------
Mar  5 15:57:26 DEBUG[1116957488]: build_route: Contact hop:
<sip:192.168.60.106>
Mar  5 15:57:26 VERBOSE[1217669936]:     -- SIP/3805-df43 answered
SIP/-08122450
Mar  5 15:57:26 VERBOSE[1217669936]:     -- Attempting native bridge of
SIP/-08122450 and SIP/3805-df43
Mar  5 15:57:26 DEBUG[1116957488]: Stopping retransmission on
'c9f88915cb5c25fd at 192.168.60.107' of Response\ 25663: Found
Mar  5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524
Mar  5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW
Mar  5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel:
SIP/3805-df43
Mar  5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels
SIP/-08122450 and SIP/3805-df43
Mar  5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse
counter
Mar  5 15:57:38 DEBUG[1217669936]:  is not a local user
Mar  5 15:57:38 VERBOSE[1217669936]:   == Spawn extension (local, 3805,
1) exited non-zero on 'SIP/-0812245\0'
-----------------

The scenario:
1 server (redhat 9),  asterisk (stable) and a 2 x hisax ISDN BRI.
One of the BRI boards is used to dial out (ppp) on one channel and a
mgetty on the other channel. The other board is in ptp and used by *.
The phones are Grandstream BT101 and Handytone and are all on a switched
network (3 procurve switches, stacked).

The configs are ok, since the same files on another server work ok (no
dropped calls), but I can post them if needed.


Any help will be greatly appreciated.

Thanks in advance,



--- Paulo Loureiro