Asterisk Ready. *CLI> (dial "123") Dec 29 20:54:09 DEBUG[1090894768]: chan_sip.c:5397 check_user_full: Setting NAT on RTP to 0 Dec 29 20:54:09 DEBUG[1090894768]: chan_sip.c:828 __sip_ack: Stopping retransmission on 'ba2e3985-8c46e254@10.10.6.155' of Response 101: Found Dec 29 20:54:09 DEBUG[1090894768]: chan_sip.c:5397 check_user_full: Setting NAT on RTP to 0 Dec 29 20:54:09 DEBUG[1090894768]: chan_sip.c:7227 handle_request: Check for res for 119 Dec 29 20:54:09 DEBUG[1090894768]: chan_sip.c:1671 update_user_counter: Call from user '119' is 1 out of 0 Dec 29 20:54:09 DEBUG[1090894768]: chan_sip.c:4612 build_route: build_route: Contact hop: greg m -- Executing Dial("SIP/119-db82", "Zap/1/93860550||") in new stack Dec 29 20:54:09 DEBUG[1151105968]: app_dial.c:491 dial_exec: SIMPLE DIAL (NO URL) Dec 29 20:54:09 DEBUG[1151105968]: chan_zap.c:1639 zt_call: Dialing '93860550' Dec 29 20:54:09 DEBUG[1151105968]: chan_zap.c:1700 zt_call: Deferring dialing... -- Called 1/93860550 Dec 29 20:54:10 DEBUG[1151105968]: chan_zap.c:3764 __zt_exception: Exception on 27, channel 1 Dec 29 20:54:10 DEBUG[1151105968]: chan_zap.c:3076 zt_handle_event: Got event Hook Transition Complete(12) on channel 1 (index 0) Dec 29 20:54:12 DEBUG[1151105968]: chan_zap.c:3764 __zt_exception: Exception on 27, channel 1 Dec 29 20:54:12 DEBUG[1151105968]: chan_zap.c:3076 zt_handle_event: Got event Dial Complete(9) on channel 1 (index 0) Dec 29 20:54:12 DEBUG[1151105968]: chan_zap.c:1221 zt_enable_ec: No echocancellation requested Dec 29 20:54:12 DEBUG[1151105968]: channel.c:1363 ast_read: Dropping duplicate answer! -- Zap/1-1 answered SIP/119-db82 Dec 29 20:54:12 DEBUG[1151105968]: rtp.c:1175 ast_rtp_write: Ooh, format changed from UNKN to ULAW Dec 29 20:54:12 DEBUG[1090894768]: chan_sip.c:828 __sip_ack: Stopping retransmission on 'ba2e3985-8c46e254@10.10.6.155' of Response 102: Found Dec 29 20:54:12 DEBUG[1151105968]: chan_sip.c:2271 sip_rtp_read: Oooh, format changed to 8 Dec 29 20:54:12 DEBUG[1151105968]: rtp.c:1175 ast_rtp_write: Ooh, format changed from ULAW to ALAW Dec 29 20:55:25 NOTICE[1151105968]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum Dec 29 20:55:29 NOTICE[1151105968]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum Dec 29 20:55:30 NOTICE[1151105968]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum *CLI>zap show channel 1 Channel: 1 File Descriptor: 27 Span: 1 Extension: Dialing: no Context: test-in Caller ID string: "Channel 1" <6001> Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Owner: Zap/1-1 Real: Zap/1-1 Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No Actual Hookstate: Onhook *CLI> Dec 29 20:56:17 NOTICE[1151105968]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum (now I hangup) Dec 29 20:57:09 DEBUG[1151105968]: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/119-db82 Dec 29 20:57:09 DEBUG[1151105968]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/119-db82 and Zap/1-1 Dec 29 20:57:09 DEBUG[1151105968]: chan_zap.c:1975 zt_hangup: Hangup: channel: 1 index = 0, normal = 27, callwait = -1, thirdcall = -1 Dec 29 20:57:09 DEBUG[1151105968]: chan_zap.c:2377 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Dec 29 20:57:09 DEBUG[1151105968]: chan_zap.c:1193 update_conf: Updated conferencing on 1, with 0 conference users -- Hungup 'Zap/1-1' Dec 29 20:57:09 DEBUG[1151105968]: app_dial.c:1029 dial_exec: Exiting with DIALSTATUS=ANSWER. == Spawn extension (test-out, 123, 1) exited non-zero on 'SIP/119-db82' Dec 29 20:57:09 DEBUG[1149008816]: rate_engine.c:708 poster_worker: Rating Engine poster thread processing Dec 29 20:57:09 DEBUG[1149008816]: rate_engine.c:770 poster_worker: Attempting to write queue entry to database Dec 29 20:57:09 DEBUG[1151105968]: chan_sip.c:1770 sip_hangup: update_user_counter(119) - decrement inUse counter and just fyi: *CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo test-in 1 test-in 2 test-in 3 test-in 4 test-in *CLI> zap show channel 1 Channel: 1 File Descriptor: 27 Span: 1 Extension: Dialing: no Context: test-in Caller ID string: "Channel 1" <6001> Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Owner: Real: Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No Actual Hookstate: Onhook *CLI> show dialplan test-out [ Context 'test-out' created by 'pbx_config' ] '*42' => 1. SayAlpha(${CALLERIDNAME}) [pbx_config] 2. Playback(extension) [pbx_config] 3. SayDigits(${CALLERIDNUM}) [pbx_config] '123' => 1. Dial(Zap/1/93860550||) [pbx_config] '_9NXXXXXX' => 1. Dial(Zap/1/${EXTEN}|20|tT) [pbx_config]