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
