Cannot call - service unavailable

Hi Guys, trying to make a lan to lan sip call, its an account issue as another account works fine

Any thoughts?

2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:24] GotoIf(“SIP/1005-0000000e”, “0?next3:continue”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx_builtins.c: Goto (macro-dial-one,s,26)
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:26] GotoIf(“SIP/1005-0000000e”, “0?nodial”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:27] GosubIf(“SIP/1005-0000000e”, “1?dstring,1():dlocal,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] Set(“SIP/1005-0000000e”, “DSTRING=”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:2] Set(“SIP/1005-0000000e”, “DEVICES=1001”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:3] ExecIf(“SIP/1005-0000000e”, “0?Return()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:4] ExecIf(“SIP/1005-0000000e”, “0?Set(DEVICES=001)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:5] Set(“SIP/1005-0000000e”, “LOOPCNT=1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:6] Set(“SIP/1005-0000000e”, “ITER=1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:7] Set(“SIP/1005-0000000e”, “THISDIAL=SIP/1001”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:8] GosubIf(“SIP/1005-0000000e”, “1?zap2dahdi,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] ExecIf(“SIP/1005-0000000e”, “0?Return()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:2] Set(“SIP/1005-0000000e”, “NEWDIAL=”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:3] Set(“SIP/1005-0000000e”, “LOOPCNT2=1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:4] Set(“SIP/1005-0000000e”, “ITER2=1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:5] Set(“SIP/1005-0000000e”, “THISPART2=SIP/1001”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:6] ExecIf(“SIP/1005-0000000e”, “0?Set(THISPART2=DAHDI/1001)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:7] Set(“SIP/1005-0000000e”, “NEWDIAL=SIP/1001&”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:8] Set(“SIP/1005-0000000e”, “ITER2=2”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:9] GotoIf(“SIP/1005-0000000e”, “0?begin2”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:10] Set(“SIP/1005-0000000e”, “THISDIAL=SIP/1001”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:11] Return(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:9] GotoIf(“SIP/1005-0000000e”, “1?docheck”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx_builtins.c: Goto (macro-dial-one,dstring,15)
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:15] GotoIf(“SIP/1005-0000000e”, “0?skipset”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:16] Set(“SIP/1005-0000000e”, “DSTRING=SIP/1001&”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:17] Set(“SIP/1005-0000000e”, “ITER=2”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:18] GotoIf(“SIP/1005-0000000e”, “0?begin”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:19] ExecIf(“SIP/1005-0000000e”, “0?Return()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:20] Set(“SIP/1005-0000000e”, “DSTRING=SIP/1001”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:21] Return(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:28] GotoIf(“SIP/1005-0000000e”, “0?nodial”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:29] GotoIf(“SIP/1005-0000000e”, “0?skiptrace”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:30] GosubIf(“SIP/1005-0000000e”, “1?ctset,1():ctclear,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] Set(“SIP/1005-0000000e”, “DB(CALLTRACE/1001)=1005”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:2] Return(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:31] Set(“SIP/1005-0000000e”, “D_OPTIONS=HhTtr”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:32] GosubIf(“SIP/1005-0000000e”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:33] NoOp(“SIP/1005-0000000e”, "Blind Transfer: , Attended Transfer: , User: 1005, Alert Info: “) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:34] ExecIf(“SIP/1005-0000000e”, “1?Set(ALERT_INFO=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:35] ExecIf(“SIP/1005-0000000e”, “0?Set(ALERT_INFO=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:36] ExecIf(“SIP/1005-0000000e”, “0?Set(ALERT_INFO=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:37] ExecIf(“SIP/1005-0000000e”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:38] ExecIf(“SIP/1005-0000000e”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:39] GosubIf(“SIP/1005-0000000e”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:40] ExecIf(“SIP/1005-0000000e”, “0?Set(CHANNEL(musicclass)=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:41] GosubIf(“SIP/1005-0000000e”, “0?qwait,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:42] Set(“SIP/1005-0000000e”, “__CWIGNORE=”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:43] Set(“SIP/1005-0000000e”, “__KEEPCID=TRUE”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:44] GotoIf(“SIP/1005-0000000e”, “0?usegoto,1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:45] GotoIf(“SIP/1005-0000000e”, “0?godial”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:46] Gosub(“SIP/1005-0000000e”, “sub-presencestate-display,s,1(1001)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] Goto(“SIP/1005-0000000e”, “state-not_set,1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx_builtins.c: Goto (sub-presencestate-display,state-not_set,1)
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] Set(“SIP/1005-0000000e”, “PRESENCESTATE_DISPLAY=”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:2] Return(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:47] Set(“SIP/1005-0000000e”, “CONNECTEDLINE(name,i)=Doorbird”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:48] Set(“SIP/1005-0000000e”, “CONNECTEDLINE(num)=1001”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:49] Set(“SIP/1005-0000000e”, “D_OPTIONS=HhTtrI”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:50] Macro(“SIP/1005-0000000e”, “dialout-one-predial-hook,”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] MacroExit(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:51] ExecIf(“SIP/1005-0000000e”, “0?Set(D_OPTIONS=HhtrII)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:52] NoOp(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:53] Dial(“SIP/1005-0000000e”, “SIP/1001,HhTtrIb(func-apply-sipheaders^s^1)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5383][C-00000007] app_mixmonitor.c: Begin MixMonitor Recording SIP/1005-0000000e
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] netsock2.c: Using SIP VIDEO TOS bits 136
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] netsock2.c: Using SIP VIDEO CoS mark 6
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] netsock2.c: Using SIP RTP TOS bits 184
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] netsock2.c: Using SIP RTP CoS mark 5
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_stack.c: SIP/1001-0000000f Internal Gosub(func-apply-sipheaders,s,1) start
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] NoOp(“SIP/1001-0000000f”, “Applying SIP Headers to channel”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:2] Set(“SIP/1001-0000000f”, “SIPHEADERKEYS=”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:3] ExecIf(“SIP/1001-0000000f”, “0?Set(Rheader=1)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:4] While(“SIP/1001-0000000f”, “0”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_while.c: Jumping to priority 8
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:9] ExecIf(“SIP/1001-0000000f”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:10] ExecIf(“SIP/1001-0000000f”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:11] Return(“SIP/1001-0000000f”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_stack.c: Spawn extension (from-internal, 1001, 1) exited non-zero on ‘SIP/1001-0000000f’
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_stack.c: SIP/1001-0000000f Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_dial.c: Called SIP/1001
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_dial.c: Connected line update to SIP/1005-0000000e prevented.
[2019-10-01 17:28:39] NOTICE[3882][C-00000007] chan_sip.c: Failed to authenticate on INVITE to ‘“1005” <sip:[email protected]>;tag=as5b9275b7’
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_dial.c: SIP/1001-0000000f is circuit-busy
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:54] ExecIf(“SIP/1005-0000000e”, “0?MacroExit()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:55] ExecIf(“SIP/1005-0000000e”, “0?Set(DIALSTATUS=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:56] GosubIf(“SIP/1005-0000000e”, “0?s-CONGESTION,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:57] MacroExit(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:15] Set(“SIP/1005-0000000e”, “SV_DIALSTATUS=CONGESTION”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:16] GosubIf(“SIP/1005-0000000e”, “0?docfu,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:17] GosubIf(“SIP/1005-0000000e”, “0?docfb,1()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:18] Set(“SIP/1005-0000000e”, “DIALSTATUS=CONGESTION”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:19] ExecIf(“SIP/1005-0000000e”, “0?MacroExit()”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:20] GotoIf(“SIP/1005-0000000e”, “1?s-CONGESTION,1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx_builtins.c: Goto (macro-exten-vm,s-CONGESTION,1)
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] GotoIf(“SIP/1005-0000000e”, “0?exit,1”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:2] PlayTones(“SIP/1005-0000000e”, “congestion”) in new stack
[2019-10-01 17:28:39] WARNING[5382][C-00000007] translate.c: no samples for ulawtolin
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:3] Congestion(“SIP/1005-0000000e”, “10”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_macro.c: Spawn extension (macro-exten-vm, s-CONGESTION, 3) exited non-zero on ‘SIP/1005-0000000e’ in macro ‘exten-vm’
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Spawn extension (ext-local, 1001, 2) exited non-zero on ‘SIP/1005-0000000e’
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] Macro(“SIP/1005-0000000e”, “hangupcall,”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:1] GotoIf(“SIP/1005-0000000e”, “1?theend”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:3] ExecIf(“SIP/1005-0000000e”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:4] NoOp(“SIP/1005-0000000e”, " monior file= /var/spool/asterisk/monitor/2019/10/01/internal-1001-1005-20191001-172839-1569914919.14.wav”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:5] AGI(“SIP/1005-0000000e”, “attendedtransfer-rec-restart.php,/var/spool/asterisk/monitor/2019/10/01/internal-1001-1005-20191001-172839-1569914919.14.wav”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] res_agi.c: <SIP/1005-0000000e>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Executing [[email protected]:6] Hangup(“SIP/1005-0000000e”, “”) in new stack
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/1005-0000000e’ in macro ‘hangupcall’
[2019-10-01 17:28:39] VERBOSE[5382][C-00000007] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/1005-0000000e’
[2019-10-01 17:28:39] VERBOSE[5383][C-00000007] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2019-10-01 17:28:39] VERBOSE[5383][C-00000007] app_mixmonitor.c: End MixMonitor Recording SIP/1005-0000000e

To be clear - what you are describing isn’t what’s happening.

Asterisk is a Back To Back User Agent, which means you don’t call “station to station”. You dial a number, connect to Asterisk. Asterisk connects to the phone you are trying to call in a new call, then bridges the two endpoints together.

From the logs, Asterisk appears to be having trouble contacting the second phone - you are getting “Busy Here” at the remote end. Options include local DND is set, phone is not provisioned correctly, and the phone isn’t registered.

Let us know what you find out with those.

Hi Dave

1001 is provisioned correctly as I’ve been using it for over a year. 1005 is a new extension just configured.

1003 can call 1005 just fine, 1003 can call 1001 just fine
1005 can’t call 1001 yet is configured at least to my eyes the same as 1003. This debug i believe shows 1005.

No extensions have DND

Can you elaborate a bit more on your setup?

Certainly.

All devices are on the LAN. The newly created extension 1005 cant call (call failed). Its calling a Doorbird intercom on 1001.

1003 can call it just fine.
1005 cannot.

Ive checked the freepbx configuration screens for the extension and they are identical.

FreePBX is running on Centos. There are no firewalls. Clients register to an internal DNS name

At the Asterisk console prompt, type
sip set debug on
make a failing test call and post a new log.

Try this one:

Asterisk*CLI>
Reliably Transmitting (NAT) to 192.168.0.57:51785:
OPTIONS sip:[email protected]:51785;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK3384ef67;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as768fa327
To: sip:[email protected]:51785;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.5.2(13.23.1)
Date: Tue, 01 Oct 2019 21:09:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:192.168.0.57:51785 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.US6rFs3ZY;rport
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected]
CSeq: 20 INVITE
Call-ID: HXycLgxpLf
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 670
Contact: sip:[email protected]:51785;transport=udp;expires=3600;+sip.instance=“urn:uuid:66e5c77b-6389-0000-ab3f-0f8e408b137f”;+org.linphone.specs=“lime”
User-Agent: CallBirdiOS/1.0 (kris’s iPad) LinphoneSDK/4.2 (belle-sip/1.6.3)

v=0
o=1005 152 2726 IN IP4 192.168.0.57
s=Talk
c=IN IP4 192.168.0.57
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7260 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
m=video 9256 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 ccm fir
<------------->
— (13 headers 25 lines) —
Sending to 192.168.0.57:51785 (NAT)
Sending to 192.168.0.57:51785 (NAT)
Using INVITE request as basis request - HXycLgxpLf
Found peer ‘1005’ for ‘1005’ from 192.168.0.57:51785

<— Reliably Transmitting (NAT) to 192.168.0.57:51785 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.US6rFs3ZY;received=192.168.0.57;rport=51785
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as02298f98
Call-ID: HXycLgxpLf
CSeq: 20 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“64520ef5”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘HXycLgxpLf’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.0.57:51785 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK3384ef67;rport
From: “Unknown” sip:[email protected];tag=as768fa327
To: sip:[email protected]:51785;transport=udp;tag=Ih0dN
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS

<------------->
— (6 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Retransmitting #1 (NAT) to 192.168.0.57:51785:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.US6rFs3ZY;received=192.168.0.57;rport=51785
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as02298f98
Call-ID: HXycLgxpLf
CSeq: 20 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“64520ef5”
Content-Length: 0


<— SIP read from UDP:192.168.0.57:51785 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.US6rFs3ZY;rport
Call-ID: HXycLgxpLf
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as02298f98
Contact: sip:[email protected]:51785;transport=udp;expires=3600;+sip.instance=“urn:uuid:66e5c77b-6389-0000-ab3f-0f8e408b137f”;+org.linphone.specs=“lime”
Max-Forwards: 70
CSeq: 20 ACK

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

<— SIP read from UDP:192.168.0.57:51785 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.zRQu7mkad;rport
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected]
CSeq: 21 INVITE
Call-ID: HXycLgxpLf
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 670
Contact: sip:[email protected]:51785;transport=udp;expires=3600;+sip.instance=“urn:uuid:66e5c77b-6389-0000-ab3f-0f8e408b137f”;+org.linphone.specs=“lime”
User-Agent: CallBirdiOS/1.0 (kris’s iPad) LinphoneSDK/4.2 (belle-sip/1.6.3)
Authorization: Digest realm=“asterisk”, nonce=“64520ef5”, algorithm=MD5, username=“1005”, uri=“sip:[email protected]”, response=“1a0878c49f2184ef89fc04b6417dc4ae”

v=0
o=1005 152 2726 IN IP4 192.168.0.57
s=Talk
c=IN IP4 192.168.0.57
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7260 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
m=video 9256 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 ccm fir
<------------->
— (14 headers 25 lines) —
Sending to 192.168.0.57:51785 (NAT)
Using INVITE request as basis request - HXycLgxpLf
Found peer ‘1005’ for ‘1005’ from 192.168.0.57:51785
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 99
Found RTP audio format 100
Found audio description format opus for ID 96
Found audio description format speex for ID 97
Found audio description format speex for ID 98
Found unknown media description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 99
Found audio description format telephone-event for ID 100
Found RTP video format 96
Found video description format H264 for ID 96
Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|alaw|opus|speex16|speex)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264|speex|speex|opus)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.57:7260
Peer video RTP is at port 192.168.0.57:9256
Looking for 1001 in from-internal (domain voice-sip.nsautomate.com.au)
sip_route_dump: route/path hop: sip:[email protected]:51785;transport=udp

<— Transmitting (NAT) to 192.168.0.57:51785 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.zRQu7mkad;received=192.168.0.57;rport=51785
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected]
Call-ID: HXycLgxpLf
CSeq: 21 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

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

<— SIP read from UDP:192.168.0.57:51785 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.US6rFs3ZY;rport
Call-ID: HXycLgxpLf
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as02298f98
Contact: sip:[email protected]:51785;transport=udp;expires=3600;+sip.instance=“urn:uuid:66e5c77b-6389-0000-ab3f-0f8e408b137f”;+org.linphone.specs=“lime”
Max-Forwards: 70
CSeq: 20 ACK

<------------->
— (8 headers 0 lines) —
[2019-10-02 07:09:48] ERROR[32182][C-00000010]: pbx_functions.c:701 ast_func_write: Function AUDIOHOOK_INHERIT not registered
We think we can do text
Audio is at 16386
Video is at 192.168.0.6:16390
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g722 to SDP
Adding codec g723 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding video codec h261 to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec mpeg4 to SDP
Adding video codec vp8 to SDP
Adding video codec vp9 to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.141:5060:
INVITE sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK24127d8a;rport
Max-Forwards: 70
From: “1005” sip:[email protected];tag=as2f7390d5
To: sip:[email protected]:5060;ob
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.5.2(13.23.1)
Date: Tue, 01 Oct 2019 21:09:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “1005” sip:[email protected]
Content-Type: application/sdp
Content-Length: 1183

v=0
o=root 1997092884 1997092884 IN IP4 192.168.0.6
s=Asterisk PBX 13.23.1
c=IN IP4 192.168.0.6
b=CT:384
t=0 0
m=audio 16386 RTP/AVP 0 8 3 111 9 4 112 5 10 118 7 18 110 117 119 97 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv
m=video 16390 RTP/AVP 99 31 34 103 104 100 108
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:108 VP9/90000
a=sendrecv


<— Transmitting (NAT) to 192.168.0.57:51785 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.zRQu7mkad;received=192.168.0.57;rport=51785
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as1993b719
Call-ID: HXycLgxpLf
CSeq: 21 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
P-Asserted-Identity: “Doorbird” sip:[email protected]
Content-Length: 0

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

<— SIP read from UDP:192.168.0.141:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.6:5060;rport=5060;received=192.168.0.6;branch=z9hG4bK24127d8a
Call-ID: [email protected]:5060
From: “1005” sip:[email protected];tag=as2f7390d5
To: sip:[email protected];ob
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:192.168.0.141:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.6:5060;rport=5060;received=192.168.0.6;branch=z9hG4bK24127d8a
Call-ID: [email protected]:5060
From: “1005” sip:[email protected];tag=as2f7390d5
To: sip:[email protected];ob;tag=181aff50-218b-4671-ad8a-86f9e29c19cc
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Transmitting (NAT) to 192.168.0.141:5060:
ACK sip:[email protected]:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK24127d8a;rport
Max-Forwards: 70
From: “1005” sip:[email protected];tag=as2f7390d5
To: sip:[email protected]:5060;ob;tag=181aff50-218b-4671-ad8a-86f9e29c19cc
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-14.0.5.2(13.23.1)
Content-Length: 0


[2019-10-02 07:09:48] NOTICE[3882][C-00000010]: chan_sip.c:24041 handle_response_invite: Failed to authenticate on INVITE to ‘“1005” sip:[email protected];tag=as2f7390d5’
[2019-10-02 07:09:48] WARNING[32182][C-00000010]: translate.c:407 framein: no samples for ulawtolin

<— Reliably Transmitting (NAT) to 192.168.0.57:51785 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.zRQu7mkad;received=192.168.0.57;rport=51785
From: “IPad” sip:[email protected]m.au;tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as1993b719
Call-ID: HXycLgxpLf
CSeq: 21 INVITE
Server: FPBX-14.0.5.2(13.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

<------------>
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE

<— SIP read from UDP:192.168.0.57:51785 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:51785;branch=z9hG4bK.zRQu7mkad;rport
Call-ID: HXycLgxpLf
From: “IPad” sip:[email protected];tag=SGBxIYfdI
To: “1001” sip:[email protected];tag=as1993b719
Contact: sip:[email protected]:51785;transport=udp;expires=3600;+sip.instance=“urn:uuid:66e5c77b-6389-0000-ab3f-0f8e408b137f”;+org.linphone.specs=“lime”
Max-Forwards: 70
CSeq: 21 ACK

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘HXycLgxpLf’ Method: ACK

<— SIP read from UDP:192.168.0.141:5060 —>

I am guessing that the DoorBird is rejecting the call because of a codec issue.

Possibly, it’s getting confused by the the huge number of codecs that the iPad is presenting; restricting the codec set in both extensions (1005 and 1001) may help. For example:
Disallowed Codecs: all
Allowed Codecs: ulaw&h264

Otherwise, it’s possible that the devices don’t have any codecs with compatible settings in common, in which case you’ll have to use a different app on the iPad.

What app or hardware is extension 1003?

Hi Stewart thanks for the reply

That doesnt make sense to me though,because from the IPad on 1004, it works fine. As does 1003

Only from 1005 does it fail. I will restrict the settings anyway and test

1003 is Note 8 phone
1005 is IPad Air 2

Tried the restriction, same issue :frowning:

Post for comparison, the SIP traces from a failed call from 1005 and from a successful call from 1004.

Otherwise, does the DoorBird have any settings that would restrict what numbers or what devices can call it?

I’m an idiot! Nice find Stewart1, the doorbird does indeed restrict it! Silly me! WORKS now :smiley: thanks so much

Glad that you got it working. IMO, the developers at DoorBird are partly to blame – the correct response for this situation is
403 Forbidden
rather than
401 UnAuthorized

Not that I would have worked that out :smiley: but I’ve certainly learnt something! Thanks again

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