[Asterisk-Users] dropped calls

[Ross Donaldson]

There is new firmware that may help http://www.grandstream.com/BETATEST/.
Grandstream acknowledges this problem. They say it is a codec issue with
asterisk. I don't know if this update addresses this problem but it may be
worth a try.

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Paulo Loureiro
> Sent: Friday, March 05, 2004 10:26 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] dropped calls
> 
> 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
> 
> 
>