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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Paulo Loureiro > Sent: Friday, March 05, 2004 10:26 AM > To: [EMAIL PROTECTED] > 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 > '[EMAIL PROTECTED]' 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 > > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
