search for: trunk503out

...but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail. Here is the printout of the log file for both boxes. Free PBX: [2015-02-10 12:12:34] VERBOSE[10502] pbx.c: -- Executing [XXXXXXXXXX at subMachine:4] Playback("SIP/trunk503out-00009728", "temp/0250002") in new stack [2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XXXXXXXXXX @subMachine:5] Wait("SIP/trunk503out-00009728", "1") in new stack [2015-02-10 12:13:30] VERBOSE[10502] pbx.c: -- Executing [XXXXXXXXXX @subMachine:...

...(ulaw|alaw|g722), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 172.17.184.93:28196 Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com) list_route: hop: <sip:172.17.184.46;transport=tcp;lr> <--- Transmitting (NAT) to 172.17.184.46:31285 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 Via: SIP/2.0/TCP 172.18.7...