Changed ISP, new WAN IP address, now calls drop at 30 seconds

We changed our ISP, so we got a new WAN IP address.

The new WAN IP was entered in Settings -> Asterisk SIP Settings -> General SIP Settings tab -> External Address. Actually the "Detect Network Settings button pulled the correct new WAN IP. I hit Submit and Apply Config. No other settings were changed. No issues existed on the old WAN IP.

The new WAN IP was entered in the SIP provider’s website (Twilio). We can make and receive calls just fine, but all incoming calls (Originating) drop right at 32 seconds, every time.

What are all the places I needed to update with a new WAN IP address? Maybe I missed somewhere.

The Twilio call properties for this call says: “Who Hung Up: callee”

Thank you.

Here are the Asterisk Log Files for the beginning of the call:

[2020-08-07 15:26:32] WARNING[45157] iax2/firmware.c: Error opening firmware directory '/var/lib/asterisk/firmware/iax': No such file or directory
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'chan_dahdi.so' (DAHDI Telephony w/PRI & SS7 & MFC/R2)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'chan_motif.so' (Motif Jingle Channel Driver)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_adsi.so' (ADSI Resource)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_fax.so' (Generic FAX Applications)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_ari.so' (Asterisk RESTful Interface)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound Registration Support)
[2020-08-07 15:26:32] ERROR[45157] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf'
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_confbridge.so' (Conference Bridge Application)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_parking.so' (Call Parking Resource)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/71/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/72/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/73/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/74/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/75/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/76/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/77/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/78/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/70/1, registrar=res_parking; con=<nil>((nil)); con->root=(nil)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_meetme.so' (MeetMe conference bridge)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'cel_odbc.so' (ODBC CEL backend)
[2020-08-07 15:26:32] VERBOSE[2502] chan_sip.c: Reloading SIP
[2020-08-07 15:26:32] VERBOSE[2502] netsock2.c: Using SIP TOS bits 96
[2020-08-07 15:26:32] VERBOSE[2502] netsock2.c: Using SIP CoS mark 4
[2020-08-07 15:26:32] VERBOSE[45157] cel_odbc.c: Found CEL table [email protected]
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_amd.so' (Answering Machine Detection Application)
[2020-08-07 15:26:32] VERBOSE[45157] app_amd.c: AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] maximumWordLength [5000]
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_playback.so' (Sound File Playback Application)
[2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_digium_phone.so' (Digium Phone Module for Asterisk)
[2020-08-07 15:26:33] VERBOSE[45157] loader.c: Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
[2020-08-07 15:26:33] WARNING[45157] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages
[2020-08-07 15:26:33] VERBOSE[45157] loader.c: Reloading module 'codec_dahdi.so' (Generic DAHDI Transcoder Codec Translator)
[2020-08-07 15:26:33] VERBOSE[45157] loader.c: Reloading module 'app_queue.so' (True Call Queueing)
[2020-08-07 15:26:33] VERBOSE[45157] asterisk.c: Remote UNIX connection disconnected
[2020-08-07 15:26:34] VERBOSE[6241] pbx_variables.c: Setting global variable 'SIPDOMAIN' to '136.228.122.109'
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:1] Set("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "CALLERID(number)=4054010812") in new stack
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:2] Goto("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "from-pstn,4055737881,1") in new stack
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (from-pstn,4055737881,1)
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:1] NoOp("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "Catch-All DID Match - Found 4055737881 - You probably want a DID for this.") in new stack
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:2] Set("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "__FROM_DID=4055737881") in new stack
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:3] Goto("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "ext-did,s,1") in new stack
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (ext-did,s,1)
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:1] Set("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "__DIRECTION=INBOUND") in new stack
[2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:2] Gosub("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "sub-record-check,s,1(in,s,no)") in new stack

Here are the Asterisk Log Files for the very end of the call:

[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] bridge_channel.c: Channel PJSIP/Twilio_PJSip_US2_Oregon-00000030 left 'simple_bridge' basic-bridge <bca56c76-ab91-4c39-93f4-0e0cd354c94a>
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] app_macro.c: Spawn extension (macro-dial, s, 53) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030' in macro 'dial'
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Spawn extension (ext-group, 602, 22) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030'
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:1] Macro("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "hangupcall,") in new stack
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:1] GotoIf("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "1?theend") in new stack
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-08-07 15:27:15] VERBOSE[45364][C-00000010] bridge_channel.c: Channel PJSIP/202-00000032 left 'simple_bridge' basic-bridge <bca56c76-ab91-4c39-93f4-0e0cd354c94a>
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:3] ExecIf("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "0?Set(CDR(recordingfile)=)") in new stack
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:4] NoOp("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "PJSIP/202-00000032 montior file= ") in new stack
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:5] GotoIf("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "1?skipagi") in new stack
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [[email protected]:7] Hangup("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "") in new stack
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030' in macro 'hangupcall'
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030'

Please confirm that you restarted (not just reloaded) Asterisk after updating the WAN IP.

I had not restarted, only Apply Config. Now I just did the following:

Admin -> System Admin -> Power Options -> pushed Reboot

After the reboot, an incoming call did not drop after a minute. I’ll call this resolved and will post again if the issue persists on Monday. Thank you @Stewart1

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