On 12/28/09 01:14, Joseph wrote: >I solved the problem with calls out via FXO but internal call to to phone >connected to FXS on AudioCodes is not working: > >app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause >20 - Unknown) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [...@internal:2] Hangup("IAX2/iaxy-322-2569", "") in new > stack > == Spawn extension (internal, 369, 2) exited non-zero on > 'IAX2/iaxy-322-2569' > >I was under impression that as soon as the extension is registered with >AudioCodes MP-114 it should accept the call but it is not going through: > >Name/username Host Dyn Nat ACL Port Status >pstn-5665/pstn-5665 (Unspecified) D 0 Unmonitored >pstn-1270/pstn-1270 10.0.0.102 D 5065 Unmonitored >369/369 (Unspecified) D 0 Unmonitored >321/act1 10.0.0.107 D 5061 OK (28 ms) > >ext. 369 and pstn-5665 are registered with AudioCodes MP-114
I don't follow the logic. When I disconnect the FXO line /pstn-5665 the internal extension is ringing OK 369 When I connect the PSTN line the call goes out via FXO -- Joseph _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users