[2014-04-29 07:53:23] VERBOSE[5516][C-0000009e] pbx.c: -- Goto (macro-dial-one,s,43)
[2014-04-29 07:53:23] VERBOSE[5516][C-0000009e] pbx.c: -- Executing [s@macro-dial-one:43] Dial("SIP/nextivia-000000a5", "SIP/101,15,Ttr") in new stack
[2014-04-29 07:53:23] VERBOSE[5516][C-0000009e] netsock2.c: == Using SIP RTP TOS bits 184
[2014-04-29 07:53:23] VERBOSE[5516][C-0000009e] netsock2.c: == Using SIP RTP CoS mark 5
[2014-04-29 07:53:23] VERBOSE[5516][C-0000009e] app_dial.c: -- Called SIP/101
[2014-04-29 07:53:23] VERBOSE[5516][C-0000009e] app_dial.c: -- SIP/101-000000a6 is ringing
[2014-04-29 07:53:25] VERBOSE[5516][C-0000009e] app_dial.c: -- SIP/101-000000a6 answered SIP/nextivia-000000a5
[2014-04-29 07:53:25] VERBOSE[5516][C-0000009e] res_musiconhold.c: -- Started music on hold, class '80s', on channel 'SIP/nextivia-000000a5'
[2014-04-29 07:53:55] WARNING[1702] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[2014-04-29 07:53:55] WARNING[1702] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] res_musiconhold.c: -- Stopped music on hold on SIP/nextivia-000000a5
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: -- Executing [h@macro-dial-one:1] Macro("SIP/nextivia-000000a5", "hangupcall,") in new stack
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/nextivia-000000a5", "1?theend") in new stack
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: -- Goto (macro-hangupcall,s,3)
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/nextivia-000000a5", "0?Set(CDR(recordingfile)=)") in new stack
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/nextivia-000000a5", " ") in new stack
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/nextivia-000000a5' in macro 'hangupcall'
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/nextivia-000000a5'
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] app_macro.c: == Spawn extension (macro-dial-one, s, 43) exited non-zero on 'SIP/nextivia-000000a5' in macro 'dial-one'
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] app_macro.c: == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/nextivia-000000a5' in macro 'exten-vm'
[2014-04-29 07:53:55] VERBOSE[5516][C-0000009e] pbx.c: == Spawn extension (from-did-direct, 101, 2) exited non-zero on 'SIP/nextivia-000000a5'
[2014-04-29 07:53:55] VERBOSE[5521][C-0000009e] app_mixmonitor.c: == MixMonitor close filestream (mixed)
[2014-04-29 07:53:55] VERBOSE[5521][C-0000009e] app_mixmonitor.c: == End MixMonitor Recording SIP/nextivia-000000a5
My Calls going to direct extensions are getting cut off from 0-45 seconds. I have my normal calling going to a ring group and that works perfectly fine. Also calling out from those extensions have no problems either, just calling in. Anyone have any ideas?
Can you post your SIP trunk configuration please?
The trunk its self or the asterisk sip settings?
Your problem is that you need to setup your network/router and sip nat settings to agree with each other and correctly pass rtp traffic to the right destination.
The trunk itself.