Outbound calls drop after 30ish seconds

General Help
1

Outbound calls drop after 30 sec. i’m at a loss. I see there are other threads on this problem, and I have tried all the suggested solns. If someone with more experience could look at the log and tell me what is going on i would be greatful.

CR

[2018-08-22 20:41:00] VERBOSE[16400][C-00000016] app_dial.c: Called SIP/Flowrout_Trunk/16203883279
[2018-08-22 20:41:03] VERBOSE[16400][C-00000016] app_dial.c: SIP/Flowrout_Trunk-0000000f is ringing
[2018-08-22 20:41:03] VERBOSE[16400][C-00000016] app_dial.c: SIP/Flowrout_Trunk-0000000f is making progress passing it to PJSIP/207-0000001a
[2018-08-22 20:41:05] VERBOSE[16400][C-00000016] app_dial.c: SIP/Flowrout_Trunk-0000000f answered PJSIP/207-0000001a
[2018-08-22 20:41:05] VERBOSE[16493][C-00000016] bridge_channel.c: Channel SIP/Flowrout_Trunk-0000000f joined ‘simple_bridge’ basic-bridge <fb93ecd7-8d0b-46bd-b78d-3567a2274704>
[2018-08-22 20:41:05] VERBOSE[16400][C-00000016] bridge_channel.c: Channel PJSIP/207-0000001a joined ‘simple_bridge’ basic-bridge <fb93ecd7-8d0b-46bd-b78d-3567a2274704>
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] bridge_channel.c: Channel PJSIP/207-0000001a left ‘simple_bridge’ basic-bridge <fb93ecd7-8d0b-46bd-b78d-3567a2274704>
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] app_macro.c: Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on ‘PJSIP/207-0000001a’ in macro ‘dialout-trunk’
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Spawn extension (from-internal, 6203883279, 7) exited non-zero on ‘PJSIP/207-0000001a’
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/207-0000001a”, “hangupcall”) in new stack
[2018-08-22 20:41:37] VERBOSE[16493][C-00000016] bridge_channel.c: Channel SIP/Flowrout_Trunk-0000000f left ‘simple_bridge’ basic-bridge <fb93ecd7-8d0b-46bd-b78d-3567a2274704>
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/207-0000001a”, “1?theend”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/207-0000001a”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/207-0000001a”, "SIP/Flowrout_Trunk-0000000f monior file= ") in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/207-0000001a”, “attendedtransfer-rec-restart.php,SIP/Flowrout_Trunk-0000000f,”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] res_agi.c: <PJSIP/207-0000001a>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/207-0000001a”, “”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/207-0000001a’ in macro ‘hangupcall’
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/207-0000001a’
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] app_stack.c: PJSIP/207-0000001a Internal Gosub(crm-hangup,s,1) start
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/207-0000001a”, “Sending Hangup to CRM”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/207-0000001a”, “HANGUP CAUSE: 16”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/207-0000001a”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/207-0000001a”, “MASTER CHANNEL: 1534970460.41 = 1534970460.41”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/207-0000001a”, “0?return”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/207-0000001a”, “__CRM_HANGUP=1”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/207-0000001a”, “sangomacrm.agi”) in new stack
[2018-08-22 20:41:37] VERBOSE[16400][C-00000016] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-08-22 20:41:38] VERBOSE[16400][C-00000016] res_agi.c: <PJSIP/207-0000001a>AGI Script sangomacrm.agi completed, returning 0
[2018-08-22 20:41:38] VERBOSE[16400][C-00000016] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/207-0000001a”, “”) in new stack
[2018-08-22 20:41:38] VERBOSE[16400][C-00000016] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/207-0000001a’
[2018-08-22 20:41:38] VERBOSE[16400][C-00000016] app_stack.c: PJSIP/207-0000001a Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

2

This is always a NAT issue. Something is sending to the wrong IP Address, so a SIP request is timing out, and one or the other ends are dropping the call.

You will need to go through and verify your SIP settings (localnet, externip, etc) and make sure any routers you have in the way have SIP ‘Helpers’ turned off.

3

I guess that was the problem. I reset my edgerouter to factory and redid the config from scratch. Works now. Thanks for the help.

CR

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