Well, after days chasing this ghost, the problem seems to be solved: - There was a problem with the NTP server (not responding to clients due to config restrictions) - upgrade gs firmware to latest available: 1.4.50.
If the NTP server is down gs phones keep trying to reach it and drop some of the packets received from * during a call. * thinks the phone is "dead" and drops the call. Anyone can confirm this? --- Paulo Loureiro. On Fri, 2004-03-05 at 19:24, Bartosz Jozwiak wrote: > I have couple of GS phone and CISCO 7960. > The funny thing is that two of that GS phone keep disconnecting and also > CISCO 7960 phone keeps disconnecting. > But the problem appear month ago! This is really strange! > > Bart > > > > > 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 > > > > _______________________________________________ > 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