asteriskRecorder*CLI> core set debug 20 Core debug was 10 and is now 20 -- Executing [123456@default:1] Set("SIP/279-081dd848", "MONITOR_EXEC=/etc/asterisk/soxmixwav.sh") in new stack -- Executing [123456@default:2] Playback("SIP/279-081dd848", "sera_grabada") in new stack -- Playing 'sera_grabada' (language 'en') Really destroying SIP dialog 'bc864c7f024f1fb7@192.168.1.100' Method: REGISTER -- Executing [123456@default:3] Answer("SIP/279-081dd848", "") in new stack -- Executing [123456@default:4] Monitor("SIP/279-081dd848", "wav|123456|mb") in new stack -- Executing [123456@default:5] Dial("SIP/279-081dd848", "SIP/123456") in new stack -- Called 123456 -- SIP/123456-081cbd00 is ringing -- SIP/279-081dd848 requested special control 17, passing it to SIP/123456-081cbd00 -- Call on SIP/123456-081cbd00 left from hold -- SIP/123456-081cbd00 answered SIP/279-081dd848 Really destroying SIP dialog '1E0820E80665DED21A795BD1AB96021B@192.168.1.27' Method: REGISTER Really destroying SIP dialog 'bc864c7f024f1fb7@192.168.1.100' Method: REGISTER asteriskRecorder*CLI> sip show channel channels channel asteriskRecorder*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 192.168.1.27 (None) 1E0820E8066 00101/25363 unkn No Rx: REGISTER 192.168.1.27 123456 7aecbcff123 00102/00000 ulaw No Tx: ACK 192.168.1.100 279 22545ee1d28 00103/04769 ulaw No Tx: ACK 3 active SIP channels asteriskRecorder*CLI> sip s set show asteriskRecorder*CLI> sip show chan channels channel asteriskRecorder*CLI> sip show channel 7aecbcff12350d1c78dbfa7a20a77f2c@192.168.1.27 asteriskRecorder*CLI> * SIP Call Curr. trans. direction: Outgoing Call-ID: 7aecbcff12350d1c78dbfa7a20a77f2c@192.168.1.27 Owner channel ID: SIP/123456-081cbd00 Our Codec Capability: 1548 Non-Codec Capability (DTMF): 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format: 0x4 (ulaw) MaxCallBR: 384 kbps Theoretical Address: 192.168.1.27:5060 Received Address: 192.168.1.27:5060 SIP Transfer mode: open NAT Support: RFC3581 Audio IP: 192.168.1.27 (local) Our Tag: as04c567d7 Their Tag: 1361079526 SIP User agent: Username: 123456 Peername: 123456 Original uri: sip:123456@192.168.1.27:5060 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:123456@192.168.1.27:5060 DTMF Mode: rfc2833 SIP Options: (none) asteriskRecorder*CLI> sip show channel 22545ee1d28e4273@192.168.1.100 asteriskRecorder*CLI> * SIP Call Curr. trans. direction: Outgoing Call-ID: 22545ee1d28e4273@192.168.1.100 Owner channel ID: SIP/279-081dd848 Our Codec Capability: 1548 Non-Codec Capability (DTMF): 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format: 0x4 (ulaw) MaxCallBR: 384 kbps Theoretical Address: 192.168.1.100:5060 Received Address: 192.168.1.100:5060 SIP Transfer mode: open NAT Support: RFC3581 Audio IP: 192.168.1.27 (Outside bridge) Our Tag: as4412a9f8 Their Tag: beb39dcf0df8e9df SIP User agent: Grandstream HT286 1.0.7.19 Username: 279 Peername: 279 Original uri: sip:279@192.168.1.100 Caller-ID: 279 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:279@192.168.1.100 DTMF Mode: inband SIP Options: replaces replace Really destroying SIP dialog '1E0820E80665DED21A795BD1AB96021B@192.168.1.27' Method: REGISTER == Spawn extension (default, 123456, 5) exited non-zero on 'SIP/279-081dd848' Really destroying SIP dialog '7aecbcff12350d1c78dbfa7a20a77f2c@192.168.1.27' Method: INVITE Really destroying SIP dialog '22545ee1d28e4273@192.168.1.100' Method: BYE asteriskRecorder*CLI>