Increase Ringing timeout

Dear all
I am using Asterisk 13.11.2 ( Sip Client : Eyebeam) and want to increase the Ringing timeout for no-answer call.
In my case, Asterisk always terminates out-going call after ~ 50 seconds if the callee does not answer.
Please help me how to increase this time.
Thanks so much

The value is controlled in the Dial() used to place the outgoing call. However some clients will automatically reject calls if they have been ringing for a period of time, which is outside the control of Asterisk and may or may not be configurable on the client.

Thanks Mr Jcolp.
I have change another client like Linphone / Eyebeam but not success.
i am new with Asterisk so can you tell me more about Dial function or where we can found this in source code ?
Thanks again.

It’s in the Asterisk dialplan, which is in extensions.conf. This is configured by you (or a tool). If you aren’t familiar with the dialplan then it’s something you should definitely read about[1] and if you are using a tool (such as FreePBX) then you should stick within it.

[1] http://www.asteriskdocs.org/

Thanks for your fast support,
According to your recommend, i think dialplan timeout is about 120 s.
====================Calling log==============================================
500 is my extension, 10123456 is callee number
– Executing [s@macro-dialout-trunk:23] Dial(“SIP/500-00000000”, “SIP/pstn1/10123456,120,T”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/pstn1/10123456

Asterisk always terminates out-going call after ~ 50 seconds if the callee does not answer.
Please help me how to increase this time.

You have specified 120 as the timeout. Please provide the output of “sip set debug on” to confirm that Asterisk itself is terminating. It may actually be the server you have called is rejecting it after that time.

This is the reject log

=================================
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:24] NoOp(“SIP/500-00000000”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18”) in new stack
– Executing [s@macro-dialout-trunk:25] GotoIf(“SIP/500-00000000”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
– Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“SIP/500-00000000”, “RC=18”) in new stack
– Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“SIP/500-00000000”, “18,1”) in new stack

Hope you can help me.
Thanks

I need to see the output of “sip set debug on” with a call attempt as I mentioned to be certain. First glance though makes it seem as though it’s not Asterisk terminating the call.

Sorry for my mistake.
This is debug log with core set debug 4.

[2016-11-08 19:05:28] DEBUG[26430][C-00001382]: audiohook.c:276 audiohook_read_frame_both: Read factory 0x7f7ea00be798 and write factory 0x7f7ea00bf1d8 both fail to provide 160 samples
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:3424 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:3424 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:3424 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:5555 set_format: Channel SIP/600-00002702 setting write format path: ulaw -> ulaw
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: app_macro.c:439 _macro_exec: Executed application: Playback
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:2825 pbx_extension_helper: Launching 'Congestion'
    -- Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion("SIP/600-00002702", "20") in new stack
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: chan_sip.c:3744 __sip_xmit: Trying to put 'SIP/2.0 503' onto UDP socket destined for 192.168.0.198:10051
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: chan_sip.c:3400 sip_alreadygone: Setting SIP_ALREADYGONE on dialog e727df685f4c5802@TVNJLVBD
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:2534 ast_softhangup_nolock: Soft-Hanging (0x01) up channel 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[3703]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 600
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:5555 set_format: Channel SIP/600-00002702 setting write format path: slin -> ulaw
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:3424 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:4905 ast_prod: Prodding channel 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[3703]: chan_sip.c:30129 sip_devicestate: Checking device state for peer 600
[2016-11-08 19:05:28] WARNING[26424][C-00001382]: channel.c:4910 ast_prod: Prodding channel 'SIP/600-00002702' failed
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:7980 ast_channel_start_silence_generator: Started silence generator on 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[3703]: devicestate.c:474 do_state_change: Changing state for SIP/600 - state 2 (In use)
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:3424 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:7991 deactivate_silence_generator: Trying to stop silence generator when there is no generator on 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: app_macro.c:433 _macro_exec: Spawn extension (macro-dialout-trunk,s-NOANSWER,4) exited non-zero on 'SIP/600-00002702' in macro 'dialout-trunk'
  == Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on 'SIP/600-00002702' in macro 'dialout-trunk'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:4296 __ast_pbx_run: Spawn extension (from-internal,10123456789,5) exited non-zero on 'SIP/600-00002702'
  == Spawn extension (from-internal, 10123456789, 5) exited non-zero on 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:2534 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:2534 ast_softhangup_nolock: Soft-Hanging (0x80) up channel 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:2825 pbx_extension_helper: Launching 'Macro'
    -- Executing [h@from-internal:1] Macro("SIP/600-00002702", "hangupcall") in new stack
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx_variables.c:777 pbx_substitute_variables_helper_full: Expression result is '1'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:2825 pbx_extension_helper: Launching 'GotoIf'
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/600-00002702", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: app_macro.c:439 _macro_exec: Executed application: GotoIf
[2016-11-08 19:05:28] DEBUG[3706]: cdr.c:1277 cdr_object_finalize: Finalized CDR for SIP/600-00002702 - start 1478606727.242435 answer 0.000000 end 1478606728.903486 dispo FAILED
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx_variables.c:708 pbx_substitute_variables_helper_full: Function CDR(recordingfile) result is 'out-10123456789-600-20161108-190436-1478606676.9986.wav'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx_variables.c:777 pbx_substitute_variables_helper_full: Expression result is '0'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:2825 pbx_extension_helper: Launching 'ExecIf'
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/600-00002702", "0?Set(CDR(recordingfile)=)") in new stack
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: app_macro.c:439 _macro_exec: Executed application: ExecIf
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx_variables.c:508 ast_str_substitute_variables_full: Function CDR(recordingfile) result is 'out-10123456789-600-20161108-190436-1478606676.9986.wav'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx_variables.c:566 ast_str_substitute_variables_full: Expression result is '0'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:2825 pbx_extension_helper: Launching 'Hangup'
    -- Executing [s@macro-hangupcall:4] Hangup("SIP/600-00002702", "") in new stack
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:2534 ast_softhangup_nolock: Soft-Hanging (0x20) up channel 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: app_macro.c:433 _macro_exec: Spawn extension (macro-hangupcall,s,4) exited non-zero on 'SIP/600-00002702' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/600-00002702' in macro 'hangupcall'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: pbx.c:4107 ast_pbx_h_exten_run: Spawn extension (from-internal,h,1) exited non-zero on 'SIP/600-00002702'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: channel.c:2683 ast_hangup: Hanging up channel 'SIP/600-00002702'
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: chan_sip.c:7129 sip_hangup: Hangup call SIP/600-00002702, SIP callid e727df685f4c5802@TVNJLVBD
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: chan_sip.c:6741 update_call_counter: Updating call counter for incoming call
[2016-11-08 19:05:28] DEBUG[26424][C-00001382]: res_rtp_asterisk.c:4917 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f7e640026e8'
[2016-11-08 19:05:28] DEBUG[3703]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 600
[2016-11-08 19:05:28] DEBUG[3703]: chan_sip.c:30129 sip_devicestate: Checking device state for peer 600
[2016-11-08 19:05:28] DEBUG[3703]: devicestate.c:474 do_state_change: Changing state for SIP/600 - state 1 (Not in use)
[2016-11-08 19:05:28] DEBUG[3751]: app_queue.c:2477 device_state_cb: Device 'SIP/600' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2016-11-08 19:05:28] DEBUG[3708]: app_queue.c:2552 extension_state_cb: Extension '600@ext-local' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2016-11-08 19:05:28] DEBUG[26430][C-00001382]: autochan.c:71 ast_autochan_destroy: Removed autochan 0x7f7ea00d1f40 from the list, about to free it
[2016-11-08 19:05:28] DEBUG[3710]: res_odbc.c:864 _ast_odbc_request_obj2: Reusing ODBC handle 0xd46f18 from class 'asteriskcdrdb'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/600-00002702
[2016-11-08 19:05:28] DEBUG[26430][C-00001382]: app_mixmonitor.c:776 mixmonitor_thread: No recipients to forward monitor to, moving on.
[2016-11-08 19:05:28] DEBUG[3710]: cel_odbc.c:765 odbc_log: Executing SQL statement: [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('HANGUP',{ts '2016-11-08 19:05:28.903920'},'600','600','600','','10123456789','h','from-internal','SIP/600-00002702','','',3,'','1478606676.9986','1478606676.9986','','','{"hangupcause":34,"hangupsource":"dialplan/builtin","dialstatus":"NOANSWER"}')]
[2016-11-08 19:05:28] DEBUG[3703]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 600
[2016-11-08 19:05:28] DEBUG[3703]: chan_sip.c:30129 sip_devicestate: Checking device state for peer 600
[2016-11-08 19:05:28] DEBUG[3703]: devicestate.c:474 do_state_change: Changing state for SIP/600 - state 1 (Not in use)
[2016-11-08 19:05:28] DEBUG[3743][C-00001382]: chan_sip.c:28605 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[2016-11-08 19:05:28] DEBUG[3743][C-00001382]: chan_sip.c:4522 __sip_ack: Stopping retransmission on 'e727df685f4c5802@TVNJLVBD' of Response 2: Match Found
[2016-11-08 19:05:28] DEBUG[3743]: chan_sip.c:6563 sip_pvt_dtor: Destroying SIP dialog e727df685f4c5802@TVNJLVBD
[2016-11-08 19:05:28] DEBUG[3743]: rtp_engine.c:378 instance_destructor: Destroyed RTP instance '0x7f7e640026e8'
[2016-11-08 19:05:29] DEBUG[3710]: res_odbc.c:713 ast_odbc_release_obj: Releasing ODBC handle 0xd46f18 into pool
[2016-11-08 19:05:29] DEBUG[3706]: res_odbc.c:864 _ast_odbc_request_obj2: Reusing ODBC handle 0xd46f18 from class 'asteriskcdrdb'
[2016-11-08 19:05:29] DEBUG[3706]: cdr_adaptive_odbc.c:740 odbc_log: Executing [INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid,recordingfile,cnum,cnam,outbound_cnum,outbound_cnam) VALUES ({ ts '2016-11-08 19:04:36' },'"600" <600>','600','10123456789','from-internal','SIP/600-00002702','SIP/pstn1-00002703','Dial','SIP/pstn1/10123456789,300,L(100000:90000)',51,0,'NO ANSWER',3,'1478606676.9986','out-10123456789-600-20161108-190436-1478606676.9986.wav','600','600','600','600')]
[2016-11-08 19:05:29] DEBUG[3706]: res_odbc.c:713 ast_odbc_release_obj: Releasing ODBC handle 0xd46f18 into pool
[2016-11-08 19:05:29] DEBUG[3706]: cdr.c:3262 post_cdr: Skipping CDR  for SIP/600-00002702 since we weren't answered
[2016-11-08 19:05:29] DEBUG[3710]: res_odbc.c:864 _ast_odbc_request_obj2: Reusing ODBC handle 0xd46f18 from class 'asteriskcdrdb'
[2016-11-08 19:05:29] DEBUG[3710]: cel_odbc.c:765 odbc_log: Executing SQL statement: [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('CHAN_END',{ts '2016-11-08 19:05:29.6490'},'600','600','600','','10123456789','h','from-internal','SIP/600-00002702','','',3,'','1478606676.9986','1478606676.9986','','','')]
[2016-11-08 19:05:29] DEBUG[3710]: res_odbc.c:713 ast_odbc_release_obj: Releasing ODBC handle 0xd46f18 into pool
[2016-11-08 19:05:29] DEBUG[3710]: res_odbc.c:864 _ast_odbc_request_obj2: Reusing ODBC handle 0xd46f18 from class 'asteriskcdrdb'
[2016-11-08 19:05:29] DEBUG[3710]: cel_odbc.c:765 odbc_log: Executing SQL statement: [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('LINKEDID_END',{ts '2016-11-08 19:05:29.39645'},'600','600','600','','10123456789','h','from-internal','SIP/600-00002702','','',3,'','1478606676.9986','1478606676.9986','','','')]
[2016-11-08 19:05:29] DEBUG[3710]: res_odbc.c:713 ast_odbc_release_obj: Releasing ODBC handle 0xd46f18 into pool

That is core set debug, not “sip set debug on”. It does not include the SIP traffic.

Sorry, my mistake again :slight_smile:
My topology : 192.168.11.99 is Asterisk server, 192.168.11.11 is my Gateway.
I found the SIP “SIP/2.0 408 Request Timeout” from sip:600@192.168.11.99 to my Gateway.

-- Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/600-00002710", "2016/11/08/out-10123456789-600-20161108-191322-1478607202.10000.wav,ai(LOCAL_MIXMON_ID),") in new stack
  == Begin MixMonitor Recording SIP/600-00002710
-- Executing [recordcheck@sub-record-check:20] Set("SIP/600-00002710", "__MIXMON_ID=0x7f7ea007cca0") in new stack
-- Executing [recordcheck@sub-record-check:21] Set("SIP/600-00002710", "__RECORD_ID=SIP/600-00002710") in new stack
-- Executing [recordcheck@sub-record-check:22] Set("SIP/600-00002710", "__REC_STATUS=RECORDING") in new stack
-- Executing [recordcheck@sub-record-check:23] Set("SIP/600-00002710", "CDR(recordingfile)=out-10123456789-600-20161108-191322-1478607202.10000.wav") in new stack
-- Executing [recordcheck@sub-record-check:24] Return("SIP/600-00002710", "") in new stack
-- Executing [out@sub-record-check:6] Return("SIP/600-00002710", "") in new stack
-- Executing [10123456789@from-internal:3] Set("SIP/600-00002710", "MOHCLASS=default") in new stack
-- Executing [10123456789@from-internal:4] Set("SIP/600-00002710", "_NODEST=") in new stack
-- Executing [10123456789@from-internal:5] Macro("SIP/600-00002710", "dialout-trunk,3,10123456789,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/600-00002710", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/600-00002710", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/600-00002710", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/600-00002710", "DIAL_NUMBER=10123456789") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/600-00002710", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/600-00002710", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/600-00002710", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/600-00002710", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/600-00002710", "DIAL_TRUNK_OPTIONS=L(100000:90000)") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/600-00002710", "outbound-callerid,3") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/600-00002710", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/600-00002710", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/600-00002710", "0?Set(REALCALLERIDNUM=600)") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/600-00002710", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,7)
-- Executing [s@macro-outbound-callerid:7] Set("SIP/600-00002710", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/600-00002710", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] Set("SIP/600-00002710", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/600-00002710", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/600-00002710", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/600-00002710", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/600-00002710", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/600-00002710", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/600-00002710", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:20] Set("SIP/600-00002710", "CDR(outbound_cnum)=600") in new stack
-- Executing [s@macro-outbound-callerid:21] Set("SIP/600-00002710", "CDR(outbound_cnam)=600") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/600-00002710", "1?sub-flp-3,s,1()") in new stack
-- Executing [s@sub-flp-3:1] ExecIf("SIP/600-00002710", "1?Return()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/600-00002710", "OUTNUM=10123456789") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/600-00002710", "custom=SIP/pstn1") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/600-00002710", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)L(100000:90000))") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/600-00002710", "0?Set(DIAL_TRUNK_OPTIONS=L(100000:90000)M(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/600-00002710", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/600-00002710", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/600-00002710", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/600-00002710", "1?Set(CONNECTEDLINE(num,i)=10123456789)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/600-00002710", "1?Set(CONNECTEDLINE(name,i)=CID:600)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/600-00002710", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)600)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/600-00002710", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("SIP/600-00002710", "SIP/pstn1/10123456789,300,L(100000:90000)") in new stack
   > Limit Data for this call:
   > timelimit      = 100000 ms (100.000 s)
   > play_warning   = 90000 ms (90.000 s)
   > play_to_caller = yes
   > play_to_callee = no
   > warning_freq   = 0 ms (0.000 s)
   > start_sound    = 
   > warning_sound  = timeleft
   > end_sound      = 
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 11338
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.11.11:5060:
INVITE sip:10123456789@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK3a4f2207;rport
Max-Forwards: 70
From: "600" <sip:600@192.168.11.99>;tag=as2a247484
To: <sip:10123456789@192.168.11.11>
Contact: <sip:600@192.168.11.99:5060>
Call-ID: 35ebe9b635c6400b43a5a6bf2b6ff122@192.168.11.99:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.188.9(13.11.2)
Date: Tue, 08 Nov 2016 12:13:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1038894438 1038894438 IN IP4 192.168.11.99
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.11.99
t=0 0
m=audio 11338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
-- Called SIP/pstn1/10123456789

<--- SIP read from UDP:192.168.11.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK3a4f2207;rport=5060
From: "600" <sip:600@192.168.11.99>;tag=as2a247484
To: <sip:10123456789@192.168.11.11>
Call-ID: 35ebe9b635c6400b43a5a6bf2b6ff122@192.168.11.99:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.11.11:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK3a4f2207;rport=5060
From: "600" <sip:600@192.168.11.99>;tag=as2a247484
To: <sip:10123456789@192.168.11.11>;tag=3bb901a617dd5d7606ce6a1eae50a4bc
Call-ID: 35ebe9b635c6400b43a5a6bf2b6ff122@192.168.11.99:5060
CSeq: 102 INVITE
Contact: <sip:10123456789@192.168.11.11>
Supported: replaces
Content-Type: application/sdp
Content-Length: 207

v=0
o=DinStar 254034811 254034812 IN IP4 192.168.11.11
s=-
c=IN IP4 192.168.11.11
t=0 0
m=audio 8004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<------------->
--- (10 headers 10 lines) ---
sip_route_dump: route/path hop: <sip:10123456789@192.168.11.11>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.11.11:8004
-- SIP/pstn1-00002711 is making progress passing it to SIP/600-00002710
Audio is at 14900
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.0.198:10051 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.198:10051;branch=z9hG4bK-d87543-e230837d7504b13f-1--d87543-;received=192.168.0.198;rport=10051
From: "600"<sip:600@192.168.0.36>;tag=ef43c026
To: <sip:10123456789@192.168.0.36>;tag=as6072574b
Call-ID: f27e173670741911@TVNJLVBD
CSeq: 2 INVITE
Server: FPBX-13.0.188.9(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10123456789@192.168.0.36:5060>
Content-Type: application/sdp
Content-Length: 299

v=0
o=root 1158855147 1158855147 IN IP4 192.168.0.36
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.0.36
t=0 0
m=audio 14900 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
   > 0x7f7e64018550 -- Probation passed - setting RTP source address to 192.168.0.198:10052
   > 0x7f7ea00c2b30 -- Probation passed - setting RTP source address to 192.168.11.11:8004
Reliably Transmitting (no NAT) to 127.0.0.1:5065:
OPTIONS sip:500@127.0.0.1:5065;ob SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK78e705b3
Max-Forwards: 70
From: "Unknown" <sip:Unknown@127.0.0.1>;tag=as645ad8fc
To: <sip:500@127.0.0.1:5065;ob>
Contact: <sip:Unknown@127.0.0.1:5060>
Call-ID: 232d55856fb576193cc13f9b3b0a670d@127.0.0.1:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.188.9(13.11.2)
Date: Tue, 08 Nov 2016 12:13:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (no NAT) to 127.0.0.1:5065:
OPTIONS sip:500@127.0.0.1:5065;ob SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK78e705b3
Max-Forwards: 70
From: "Unknown" <sip:Unknown@127.0.0.1>;tag=as645ad8fc
To: <sip:500@127.0.0.1:5065;ob>
Contact: <sip:Unknown@127.0.0.1:5060>
Call-ID: 232d55856fb576193cc13f9b3b0a670d@127.0.0.1:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.188.9(13.11.2)
Date: Tue, 08 Nov 2016 12:13:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0



Really destroying SIP dialog '232d55856fb576193cc13f9b3b0a670d@127.0.0.1:5060' Method: OPTIONS

<--- SIP read from UDP:127.0.0.1:5065 --->
REGISTER sip:localhost SIP/2.0
Via: SIP/2.0/UDP 192.168.0.36:5065;rport;branch=z9hG4bKPj78b5b68f-fb01-4076-b6a4-405341deeaab
Max-Forwards: 70
From: <sip:500@localhost>;tag=1b0eae24-3e60-4c9d-a8ad-acb58263c8a9
To: <sip:500@localhost>
Call-ID: 07ce2961-1ada-401b-8e68-6b307de6cff3
CSeq: 29904 REGISTER
Contact: <sip:500@192.168.0.36:5065;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 127.0.0.1:5065 (NAT)
Sending to 127.0.0.1:5065 (NAT)



--- (11 headers 0 lines) ---
[2016-11-08 19:13:32] NOTICE[3743]: chan_sip.c:24422 handle_response_peerpoke: Peer '500' is now Reachable. (1ms / 2000ms)
Really destroying SIP dialog '311365fd2b4d2d1c579093c97e993dcf@127.0.0.1:5060' Method: OPTIONS

<--- SIP read from UDP:127.0.0.1:5065 --->
INVITE sip:118001090@localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5065;rport;branch=z9hG4bKPj0a0e0395-1ca8-41be-9c94-233e1c3b782c
Max-Forwards: 70
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost
Contact: <sip:500@127.0.0.1:5065;ob>
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14956 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 473

v=0
o=- 3687596014 3687596014 IN IP4 192.168.0.36
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.0.36
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.0.36
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (14 headers 22 lines) ---
Sending to 127.0.0.1:5065 (NAT)
Sending to 127.0.0.1:5065 (NAT)
Using INVITE request as basis request - 95775f37-539d-464d-8914-85a5a847e80a
Found peer '500' for '500' from 127.0.0.1:5065

<--- Reliably Transmitting (no NAT) to 127.0.0.1:5065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5065;branch=z9hG4bKPj0a0e0395-1ca8-41be-9c94-233e1c3b782c;received=127.0.0.1;rport=5065
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost;tag=as1d2e0abf
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14956 INVITE
Server: FPBX-13.0.188.9(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06eafabb"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '95775f37-539d-464d-8914-85a5a847e80a' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:127.0.0.1:5065 --->
ACK sip:118001090@localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5065;rport;branch=z9hG4bKPj0a0e0395-1ca8-41be-9c94-233e1c3b782c
Max-Forwards: 70
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost;tag=as1d2e0abf
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14956 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:127.0.0.1:5065 --->
INVITE sip:118001090@localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5065;rport;branch=z9hG4bKPj27081f5f-f744-4579-ab99-dfb1976f85ab
Max-Forwards: 70
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost
Contact: <sip:500@127.0.0.1:5065;ob>
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14957 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: Digest username="500", realm="asterisk", nonce="06eafabb", uri="sip:118001090@localhost", response="979c58b4719ec536922b014114373cbe", algorithm=MD5
Content-Type: application/sdp
Content-Length: 473

v=0
o=- 3687596014 3687596014 IN IP4 192.168.0.36
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.0.36
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.0.36
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
<------------->
--- (15 headers 22 lines) ---
Sending to 127.0.0.1:5065 (no NAT)
Using INVITE request as basis request - 95775f37-539d-464d-8914-85a5a847e80a
Found peer '500' for '500' from 127.0.0.1:5065
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 99
Found RTP audio format 104
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 96
Found audio description format speex for ID 98
Found audio description format speex for ID 97
Found audio description format speex for ID 99
Found audio description format iLBC for ID 104
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 96
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|speex|speex16|speex32|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.36:4000
Looking for 118001090 in from-internal (domain localhost)
sip_route_dump: route/path hop: <sip:500@127.0.0.1:5065;ob>

<--- Transmitting (no NAT) to 127.0.0.1:5065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5065;branch=z9hG4bKPj27081f5f-f744-4579-ab99-dfb1976f85ab;received=127.0.0.1;rport=5065
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14957 INVITE
Server: FPBX-13.0.188.9(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:118001090@127.0.0.1:5060>
Content-Length: 0

-- Executing [s@macro-dialout-trunk:23] Dial("SIP/500-00002712", "SIP/pstn1/118001090,300,L(100000:90000)") in new stack
   > Limit Data for this call:
   > timelimit      = 100000 ms (100.000 s)
   > play_warning   = 90000 ms (90.000 s)
   > play_to_caller = yes
   > play_to_callee = no
   > warning_freq   = 0 ms (0.000 s)
   > start_sound    = 
   > warning_sound  = timeleft
   > end_sound      = 
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 17978
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.11.11:5060:
INVITE sip:118001090@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK60b68955;rport
Max-Forwards: 70
From: "500" <sip:500@192.168.11.99>;tag=as1f55d140
To: <sip:118001090@192.168.11.11>
Contact: <sip:500@192.168.11.99:5060>
Call-ID: 36079f6b5fc77c4f15f801240c203b71@192.168.11.99:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.188.9(13.11.2)
Date: Tue, 08 Nov 2016 12:13:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1762018350 1762018350 IN IP4 192.168.11.99
s=Asterisk PBX 13.11.2
c=IN IP4 192.168.11.99
t=0 0
m=audio 17978 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
-- Called SIP/pstn1/118001090

<--- SIP read from UDP:192.168.11.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK60b68955;rport=5060
From: "500" <sip:500@192.168.11.99>;tag=as1f55d140
To: <sip:118001090@192.168.11.11>
Call-ID: 36079f6b5fc77c4f15f801240c203b71@192.168.11.99:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---


-- Executing [s-INVALIDNMBR@macro-dialout-trunk:3] Playback("SIP/500-00002712", "ss-noservice,noanswer") in new stack

---

<--- SIP read from UDP:192.168.0.198:10051 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.36:5060;branch=z9hG4bK788d12ba
Contact: <sip:192.168.0.198:10051>
To: <sip:600@192.168.0.198:10051;rinstance=0f6fce0ddbcfe657>;tag=9c04d675
From: "Unknown"<sip:Unknown@192.168.0.36>;tag=as03c0ad28
Call-ID: 0dc5eeae61eab76a6eaedf686ae55c22@192.168.0.36:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
User-Agent: eyeBeam release 3015c stamp 27107
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '0dc5eeae61eab76a6eaedf686ae55c22@192.168.0.36:5060' Method: OPTIONS
-- Executing [s-INVALIDNMBR@macro-dialout-trunk:4] Busy("SIP/500-00002712", "20") in new stack

<--- Reliably Transmitting (no NAT) to 127.0.0.1:5065 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 127.0.0.1:5065;branch=z9hG4bKPj27081f5f-f744-4579-ab99-dfb1976f85ab;received=127.0.0.1;rport=5065
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost;tag=as04f4a05c
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14957 INVITE
Server: FPBX-13.0.188.9(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>

<--- SIP read from UDP:127.0.0.1:5065 --->
ACK sip:118001090@localhost SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5065;rport;branch=z9hG4bKPj27081f5f-f744-4579-ab99-dfb1976f85ab
Max-Forwards: 70
From: sip:500@localhost;tag=24d85253-9be0-420d-9215-7fa436b14f9f
To: sip:118001090@localhost;tag=as04f4a05c
Call-ID: 95775f37-539d-464d-8914-85a5a847e80a
CSeq: 14957 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[2016-11-08 19:13:39] WARNING[2313][C-0000138a]: channel.c:4910 ast_prod: Prodding channel 'SIP/500-00002712' failed
  == Spawn extension (macro-dialout-trunk, s-INVALIDNMBR, 4) exited non-zero on 'SIP/500-00002712' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 118001090, 5) exited non-zero on 'SIP/500-00002712'
-- Executing [h@from-internal:1] Macro("SIP/500-00002712", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/500-00002712", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/500-00002712", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/500-00002712", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/500-00002712' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/500-00002712'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/500-00002712
Really destroying SIP dialog '95775f37-539d-464d-8914-85a5a847e80a' Method: ACK
Really destroying SIP dialog '36079f6b5fc77c4f15f801240c203b71@192.168.11.99:5060' Method: INVITE
Reliably Transmitting (NAT) to 192.168.11.11:5060:
OPTIONS sip:192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK18caedc2;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.11.99>;tag=as322d8d3b
To: <sip:192.168.11.11>
Contact: <sip:Unknown@192.168.11.99:5060>
Call-ID: 02390b5e1d6c55e41cfb49541cedeec2@192.168.11.99:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.188.9(13.11.2)
Date: Tue, 08 Nov 2016 12:13:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 192.168.11.11:5060:
OPTIONS sip:192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK68b95c29;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.11.99>;tag=as77139cb0
To: <sip:192.168.11.11>
Contact: <sip:Unknown@192.168.11.99:5060>
Call-ID: 2feeadea32d1c66113e7c6c7440f2024@192.168.11.99:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.188.9(13.11.2)
Date: Tue, 08 Nov 2016 12:13:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---

<--- SIP read from UDP:192.168.11.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK68689383;rport=5060
From: "Unknown" <sip:Unknown@192.168.11.99>;tag=as605db0e5
To: <sip:192.168.11.11>;tag=eb69b9ea8223d3c04e5bde63361049b4
Call-ID: 7c975b49415a920e052793286420ee8f@192.168.11.99:5060
CSeq: 102 OPTIONS
Contact: <sip:192.168.11.11>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, NOTIFY, REFER
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7c975b49415a920e052793286420ee8f@192.168.11.99:5060' Method: OPTIONS
Really destroying SIP dialog '07ce2961-1ada-401b-8e68-6b307de6cff3' Method: REGISTER

<--- SIP read from UDP:192.168.0.198:10051 --->


<------------->

<--- SIP read from UDP:192.168.11.11:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK3a4f2207;rport=5060
From: "600" <sip:600@192.168.11.99>;tag=as2a247484
To: <sip:10123456789@192.168.11.11>;tag=3bb901a617dd5d7606ce6a1eae50a4bc
Call-ID: 35ebe9b635c6400b43a5a6bf2b6ff122@192.168.11.99:5060
CSeq: 102 INVITE
Contact: <sip:10123456789@192.168.11.11>
Supported: replaces
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 192.168.11.11:5060:
ACK sip:10123456789@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.99:5060;branch=z9hG4bK3a4f2207;rport
Max-Forwards: 70
From: "600" <sip:600@192.168.11.99>;tag=as2a247484
To: <sip:10123456789@192.168.11.11>;tag=3bb901a617dd5d7606ce6a1eae50a4bc
Contact: <sip:600@192.168.11.99:5060>
Call-ID: 35ebe9b635c6400b43a5a6bf2b6ff122@192.168.11.99:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.188.9(13.11.2)
Content-Length: 0


---
Scheduling destruction of SIP dialog '35ebe9b635c6400b43a5a6bf2b6ff122@192.168.11.99:5060' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/600-00002710", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18") in new stack
-- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/600-00002710", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/600-00002710", "RC=18") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/600-00002710", "18,1") in new stack
-- Goto (macro-dialout-trunk,18,1)
-- Executing [18@macro-dialout-trunk:1] Goto("SIP/600-00002710", "s-NOANSWER,1") in new stack
-- Goto (macro-dialout-trunk,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp("SIP/600-00002710", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:2] Progress("SIP/600-00002710", "") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:3] Playback("SIP/600-00002710", "number-not-answering,noanswer") in new stack
-- <SIP/600-00002710> Playing 'number-not-answering.gsm' (language 'en')
-- Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion("SIP/600-00002710", "20") in new stack

<--- Reliably Transmitting (no NAT) to 192.168.0.198:10051 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.198:10051;branch=z9hG4bK-d87543-e230837d7504b13f-1--d87543-;received=192.168.0.198;rport=10051
From: "600"<sip:600@192.168.0.36>;tag=ef43c026
To: <sip:10123456789@192.168.0.36>;tag=as6072574b
Call-ID: f27e173670741911@TVNJLVBD
CSeq: 2 INVITE
Server: FPBX-13.0.188.9(13.11.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


<------------>
[2016-11-08 19:14:14] WARNING[2129][C-00001389]: channel.c:4910 ast_prod: Prodding channel 'SIP/600-00002710' failed
  == Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on 'SIP/600-00002710' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 10123456789, 5) exited non-zero on 'SIP/600-00002710'
-- Executing [h@from-internal:1] Macro("SIP/600-00002710", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/600-00002710", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/600-00002710", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/600-00002710", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/600-00002710' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/600-00002710'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording SIP/600-00002710

<--- SIP read from UDP:192.168.0.198:10051 --->
ACK sip:10123456789@192.168.0.36 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.198:10051;branch=z9hG4bK-d87543-e230837d7504b13f-1--d87543-;rport
To: <sip:10123456789@192.168.0.36>;tag=as6072574b
From: "600"<sip:600@192.168.0.36>;tag=ef43c026
Call-ID: f27e173670741911@TVNJLVBD
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'f27e173670741911@TVNJLVBD' Method: ACK

Your gateway is terminating the call after that period of time:

<--- SIP read from UDP:192.168.11.11:5060 --->
SIP/2.0 408 Request Timeout

It’s not Asterisk doing it.

@jcolp You are right .
Please close this topic.
Thanks for your support :slight_smile: