Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem?
I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: > 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 -- Cumprimentos, --- Paulo Loureiro Netmania _______________________________________________ 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