Call waiting drops instead of going to VM

Asterisk Asterisk Support
1

I am new to asterisk, I’m running TrixBox CE 2.6 I have one extension that when they get a call while on the line, instead of going to VM, it drops the call. I have confirmed they are not pressing ignore and call waiting is enabled on the extension (X103) below are the debugs

--  dialparties.agi: dbset CALLTRACE/103 to 5705555555
--  dialparties.agi: Filtered ARG3: 103

== Manager ‘admin’ logged off from 127.0.0.1
– AGI Script dialparties.agi completed, returning 0
– Executing [s@macro-dial:7] Dial(“Zap/2-1”, “SIP/103|15|tr”) in new stack
– Called 103
Extension Changed 103[ext-local] new state InUse&Ringing for Notify User 103
– SIP/103-08f0d540 is ringing
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Connect attempt from ‘127.0.0.1’ unable to authenticate
– Nobody picked up in 15000 ms
– Executing [s@macro-dial:8] Set(“Zap/2-1”, “DIALSTATUS=BUSY”) in new stack
– Executing [s@macro-dial:9] GosubIf(“Zap/2-1”, “0?BUSY|1”) in new stack
– Executing [s@macro-exten-vm:10] GotoIf(“Zap/2-1”, “0?exit|return”) in new stack
– Executing [s@macro-exten-vm:11] Set(“Zap/2-1”, “SV_DIALSTATUS=BUSY”) in new stack
– Executing [s@macro-exten-vm:12] GosubIf(“Zap/2-1”, “0?docfu|1”) in new stack
– Executing [s@macro-exten-vm:13] GosubIf(“Zap/2-1”, “0?docfb|1”) in new stack
– Executing [s@macro-exten-vm:14] Set(“Zap/2-1”, “DIALSTATUS=BUSY”) in new stack
– Executing [s@macro-exten-vm:15] NoOp(“Zap/2-1”, “Voicemail is 103”) in new stack
– Executing [s@macro-exten-vm:16] GotoIf(“Zap/2-1”, “0?s-BUSY|1”) in new stack
– Executing [s@macro-exten-vm:17] NoOp(“Zap/2-1”, “Sending to Voicemail box 103”) in new stack
– Executing [s@macro-exten-vm:18] Macro(“Zap/2-1”, “vm|103|BUSY|”) in new stack
– Executing [s@macro-vm:1] Macro(“Zap/2-1”, “user-callerid|SKIPTTL”) in new stack
– Executing [s@macro-user-callerid:1] Set(“Zap/2-1”, “AMPUSER=5705555555”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“Zap/2-1”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“Zap/2-1”, “0|Set|REALCALLERIDNUM=5705555555”) in new stack
– Executing [s@macro-user-callerid:4] Set(“Zap/2-1”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“Zap/2-1”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“Zap/2-1”, “1?report”) in new stack
– Goto (macro-user-callerid,s,11)
– Executing [s@macro-user-callerid:11] GotoIf(“Zap/2-1”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,20)
– Executing [s@macro-user-callerid:20] NoOp(“Zap/2-1”, “Using CallerID “SCRANTON PA” <5705555555>”) in new stack
Extension Changed 103[ext-local] new state InUse for Notify User 103
– Executing [s@macro-vm:2] Set(“Zap/2-1”, “VMGAIN=”"") in new stack
– Executing [s@macro-vm:3] GotoIf(“Zap/2-1”, “1?vmx|1”) in new stack
– Goto (macro-vm,vmx,1)
– Executing [vmx@macro-vm:1] GotoIf(“Zap/2-1”, “0?s-BUSY|1”) in new stack
– Executing [vmx@macro-vm:2] Set(“Zap/2-1”, “MODE=busy”) in new stack
– Executing [vmx@macro-vm:3] GotoIf(“Zap/2-1”, “1?notdirect”) in new stack
– Goto (macro-vm,vmx,5)
– Executing [vmx@macro-vm:5] NoOp(“Zap/2-1”, "Checking if ext 103 is enabled: ") in new stack
– Executing [vmx@macro-vm:6] GotoIf(“Zap/2-1”, “1?s-BUSY|1”) in new stack
– Goto (macro-vm,s-BUSY,1)
– Executing [s-BUSY@macro-vm:1] NoOp(“Zap/2-1”, “BUSY voicemail”) in new stack
– Executing [s-BUSY@macro-vm:2] Macro(“Zap/2-1”, “get-vmcontext|103”) in new stack
– Executing [s@macro-get-vmcontext:1] Set(“Zap/2-1”, “VMCONTEXT=default”) in new stack
– Executing [s@macro-get-vmcontext:2] GotoIf(“Zap/2-1”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [s@macro-get-vmcontext:300] NoOp(“Zap/2-1”, “”) in new stack
– Executing [s-BUSY@macro-vm:3] VoiceMail(“Zap/2-1”, “103@default|sb”) in new stack
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on ‘Zap/2-1’ in macro ‘vm’
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on ‘Zap/2-1’ in macro ‘exten-vm’
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on ‘Zap/2-1’
– Hungup ‘Zap/2-1’
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Spawn extension (macro-vm, o, 13) exited non-zero on ‘Zap/1-1’ in macro ‘vm’
== Spawn extension (macro-vm, o, 13) exited non-zero on ‘Zap/1-1’ in macro ’

2

Trixbox is a dead product. If you are new to Asterisk, start again with either bare Asterisk or a packaged Asterisk for which support is available. (In fact, I think the product was killed sufficiently long ago that it should be impossible t be both be using it and be new to Asterisk.)

The voicemail application has been called, but there isn’t a high enough logging level to indicate why it dropped out.

3

Thank you for your help. This is a client that has a Trixbox and I am new to supporting more advanced issues. Here a debug with logging bumped to level 4. Any assistance is appreciated

-- Executing [s-BUSY@macro-vm:2] Macro("Zap/2-1", "get-vmcontext|103") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("Zap/2-1", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("Zap/2-1", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("Zap/2-1", "") in new stack
-- Executing [s-BUSY@macro-vm:3] VoiceMail("Zap/2-1", "103@default|sb") in new stack

== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on ‘Zap/2-1’
– Hungup 'Zap/2-1’
Extension Changed 103[ext-local] new state InUse for Notify User 103
Really destroying SIP dialog ‘4f5752a2311e3b5b6eb45c9058024345@172.30.4.254’ Method: INVITE
Really destroying SIP dialog ‘60c0aa1b27935fe04dd4a4b82027679a@172.30.4.254’ Method: OPTIONS
Really destroying SIP dialog ‘7be00d723f3e86264d001a8e10cfa2a2@172.30.4.254’ Method: OPTIONS
Really destroying SIP dialog ‘7fcea0c8319695765f2141c46a1beb71@172.30.4.254’ Method: OPTIONS
Really destroying SIP dialog ‘51f4d5db5aebd3aa2adc00aa71f9f374@172.30.4.254’ Method: OPTIONS
Really destroying SIP dialog ‘3e9bb9bd6c5127e176b041ee2d77ba51@172.30.4.254’ Method: OPTIONS
Really destroying SIP dialog ‘7268484e282946f5553990786070634b@172.30.4.254’ Method: OPTIONS
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Connect attempt from ‘127.0.0.1’ unable to authenticate
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘Zap/1-1’
– Executing [h@macro-dial:1] Macro(“Zap/1-1”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“Zap/1-1”, “vw”) in new stack
Extension Changed 103[ext-local] new state Idle for Notify User 103
– Executing [s@macro-hangupcall:2] NoCDR(“Zap/1-1”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“Zap/1-1”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“Zap/1-1”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“Zap/1-1”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“Zap/1-1”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘Zap/1-1’
– Hungup 'Zap/1-1’
Really destroying SIP dialog ‘0dc1946743cf88147bdec07e6c0812d3@172.30.4.254’ Method: BYE
== Connect attempt from ‘127.0.0.1’ unable to authenticate

4

Trixbox is essentially unsupportable.

Any relevant logging will appear between these lines:

-- Executing [s-BUSY@macro-vm:3] VoiceMail("Zap/2-1", "103@default|sb") in new stack == Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'Zap/2-1'

You may need to use the real log files (they have timestamps and they can be configured to include debug level output) and you may need to enable debug as well as verbose output. At least level 5 for both is normally advised.

You cannot assume that anyone here knows the details of Fonality’s dialplan.

5

thanks, here are the debugs

e[Ktrixbox1*CLI>
– Executing [s-BUSY@macro-vm:1] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40mBUSY voicemaile[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
– Executing [s-BUSY@macro-vm:2] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40mget-vmcontext|103e[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
– Executing [s@macro-get-vmcontext:1] e[1;36;40mSete[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40mVMCONTEXT=defaulte[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
– Executing [s@macro-get-vmcontext:2] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40m0?200:300e[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
– Goto (macro-get-vmcontext,s,300)

e[Ktrixbox1*CLI>
– Executing [s@macro-get-vmcontext:300] e[1;36;40mNoOpe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
– Executing [s-BUSY@macro-vm:3] e[1;36;40mVoiceMaile[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40m103@default|sbe[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on ‘Zap/2-1’

e[Ktrixbox1*CLI>
– Hungup ‘Zap/2-1’

6

wiki.asterisk.org/wiki/display/ … nformation

7

thank you, I am going to try to read and learn more about asterisk in general. Here is the output data

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

e[Ktrixbox1*CLI>
Transmitting (NAT) to 172.30.4.31:5060:
ACK sip:103@172.30.4.31:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK68001127;rport

From: “SCRANTON PA” sip:5705555555@172.30.4.254;tag=as1ab7b63d

To: sip:103@172.30.4.31:5060;ob;tag=vuagpb8hxriDdp66xjNajJ9eQYWP08ol

Contact: sip:5705555555@172.30.4.254

Call-ID: 1c2f0e7f2c6ba2f555ddbcd309488b10@172.30.4.254

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘1c2f0e7f2c6ba2f555ddbcd309488b10@172.30.4.254’ Method: INVITE

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘h8ASZZI.ambFycbYW53ZzJagw4jkUsfi’ Method: REGISTER

e[Ktrixbox1*CLI>
== Connect attempt from ‘127.0.0.1’ unable to authenticate

e[Ktrixbox1*CLI>
Reliably Transmitting (NAT) to 172.30.4.24:5060:
OPTIONS sip:105@172.30.4.24:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK6dc80707;rport

From: “Unknown” sip:Unknown@172.30.4.254;tag=as63aef432

To: sip:105@172.30.4.24:5060;ob

Contact: sip:Unknown@172.30.4.254

Call-ID: 16c8d71067560a0105ee322e107f78a2@172.30.4.254

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 23 Apr 2013 15:38:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY

Supported: replaces

Content-Length: 0

e[Ktrixbox1*CLI>

<— SIP read from 172.30.4.24:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK6dc80707

Call-ID: 16c8d71067560a0105ee322e107f78a2@172.30.4.254

From: “Unknown” sip:Unknown@172.30.4.254;tag=as63aef432

To: sip:105@172.30.4.24;ob;tag=z9hG4bK6dc80707

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/si mple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, timer, norefersub

Allow-Events: presence, message-summary, refer

User-Agent: Digium D50 1_1_2_0_51236

Content-Type: application/sdp

Content-Length: 417

v=0

o=- 62102805 62102805 IN IP4 172.30.4.24

s=digphn

c=IN IP4 172.30.4.24

t=0 0

m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96

a=rtcp:4001 IN IP4 172.30.4.24

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:58 L16/16000

a=rtpmap:118 L16/8000

a=rtpmap:58 L16-256/16000

a=sendrecv

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

<------------->
— (13 headers 18 lines) —

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘16c8d71067560a0105ee322e107f78a2@172.30.4.254’ Method: OPTIONS

e[Ktrixbox1*CLI>
Reliably Transmitting (NAT) to 172.30.4.32:5060:
OPTIONS sip:104@172.30.4.32:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK16a27135;rport

From: “Unknown” sip:Unknown@172.30.4.254;tag=as2ecb4462

To: sip:104@172.30.4.32:5060;ob

Contact: sip:Unknown@172.30.4.254

Call-ID: 5dd9ee9053f555c6517e6abb4bfc0bd6@172.30.4.254

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 23 Apr 2013 15:38:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY

Supported: replaces

Content-Length: 0

e[Ktrixbox1*CLI>

<— SIP read from 172.30.4.32:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK16a27135

Call-ID: 5dd9ee9053f555c6517e6abb4bfc0bd6@172.30.4.254

From: “Unknown” sip:Unknown@172.30.4.254;tag=as2ecb4462

To: sip:104@172.30.4.32;ob;tag=z9hG4bK16a27135

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/si mple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, timer, norefersub

Allow-Events: presence, message-summary, refer

User-Agent: Digium D50 1_1_2_0_51236

Content-Type: application/sdp

Content-Length: 417

v=0

o=- 62102805 62102805 IN IP4 172.30.4.32

s=digphn

c=IN IP4 172.30.4.32

t=0 0

m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96

a=rtcp:4001 IN IP4 172.30.4.32

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:58 L16/16000

a=rtpmap:118 L16/8000

a=rtpmap:58 L16-256/16000

a=sendrecv

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

<------------->
— (13 headers 18 lines) —

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘5dd9ee9053f555c6517e6abb4bfc0bd6@172.30.4.254’ Method: OPTIONS

e[Ktrixbox1*CLI>
Reliably Transmitting (NAT) to 172.30.4.31:5060:
OPTIONS sip:103@172.30.4.31:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK270e7750;rport

From: “Unknown” sip:Unknown@172.30.4.254;tag=as500a3b97

To: sip:103@172.30.4.31:5060;ob

Contact: sip:Unknown@172.30.4.254

Call-ID: 3428023d237377e01f6e3ba636605980@172.30.4.254

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 23 Apr 2013 15:38:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY

Supported: replaces

Content-Length: 0

e[Ktrixbox1*CLI>

<— SIP read from 172.30.4.31:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK270e7750

Call-ID: 3428023d237377e01f6e3ba636605980@172.30.4.254

From: “Unknown” sip:Unknown@172.30.4.254;tag=as500a3b97

To: sip:103@172.30.4.31;ob;tag=z9hG4bK270e7750

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/si mple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, timer, norefersub

Allow-Events: presence, message-summary, refer

User-Agent: Digium D50 1_1_2_0_51236

Content-Type: application/sdp

Content-Length: 417

v=0

o=- 62102805 62102805 IN IP4 172.30.4.31

s=digphn

c=IN IP4 172.30.4.31

t=0 0

m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96

a=rtcp:4001 IN IP4 172.30.4.31

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:58 L16/16000

a=rtpmap:118 L16/8000

a=rtpmap:58 L16-256/16000

a=sendrecv

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

<------------->
— (13 headers 18 lines) —

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘3428023d237377e01f6e3ba636605980@172.30.4.254’ Method: OPTIONS

e[Ktrixbox1*CLI>
Reliably Transmitting (NAT) to 172.30.4.29:5060:
OPTIONS sip:102@172.30.4.29:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK1dd5babe;rport

From: “Unknown” sip:Unknown@172.30.4.254;tag=as75595695

To: sip:102@172.30.4.29:5060;ob

Contact: sip:Unknown@172.30.4.254

Call-ID: 17087faa7338c75473026724321a8411@172.30.4.254

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 23 Apr 2013 15:38:35 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY

Supported: replaces

Content-Length: 0

e[Ktrixbox1*CLI>

<— SIP read from 172.30.4.29:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK1dd5babe

Call-ID: 17087faa7338c75473026724321a8411@172.30.4.254

From: “Unknown” sip:Unknown@172.30.4.254;tag=as75595695

To: sip:102@172.30.4.29;ob;tag=z9hG4bK1dd5babe

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/si mple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, timer, norefersub

Allow-Events: presence, message-summary, refer

User-Agent: Digium D50 1_1_2_0_51236

Content-Type: application/sdp

Content-Length: 417

v=0

o=- 62102805 62102805 IN IP4 172.30.4.29

s=digphn

c=IN IP4 172.30.4.29

t=0 0

m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96

a=rtcp:4001 IN IP4 172.30.4.29

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:58 L16/16000

a=rtpmap:118 L16/8000

a=rtpmap:58 L16-256/16000

a=sendrecv

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

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

e[Ktrixbox1*CLI>
— (13 headers 18 lines) —

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘17087faa7338c75473026724321a8411@172.30.4.254’ Method: OPTIONS

e[Ktrixbox1*CLI>
Reliably Transmitting (NAT) to 172.30.4.30:5060:
OPTIONS sip:101@172.30.4.30:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK0c690722;rport

From: “Unknown” sip:Unknown@172.30.4.254;tag=as20cb0772

To: sip:101@172.30.4.30:5060;ob

Contact: sip:Unknown@172.30.4.254

Call-ID: 52a84422685a835c7c5a9ee576155ebb@172.30.4.254

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 23 Apr 2013 15:38:36 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY

Supported: replaces

Content-Length: 0

e[Ktrixbox1*CLI>

<— SIP read from 172.30.4.30:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK0c690722

Call-ID: 52a84422685a835c7c5a9ee576155ebb@172.30.4.254

From: “Unknown” sip:Unknown@172.30.4.254;tag=as20cb0772

To: sip:101@172.30.4.30;ob;tag=z9hG4bK0c690722

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/si mple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, timer, norefersub

Allow-Events: presence, message-summary, refer

User-Agent: Digium D50 1_1_2_0_51236

Content-Type: application/sdp

Content-Length: 417

v=0

o=- 62102805 62102805 IN IP4 172.30.4.30

s=digphn

c=IN IP4 172.30.4.30

t=0 0

m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96

a=rtcp:4001 IN IP4 172.30.4.30

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:58 L16/16000

a=rtpmap:118 L16/8000

a=rtpmap:58 L16-256/16000

a=sendrecv

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

<------------->
— (13 headers 18 lines) —

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘52a84422685a835c7c5a9ee576155ebb@172.30.4.254’ Method: OPTIONS

e[Ktrixbox1*CLI>
Reliably Transmitting (NAT) to 172.30.4.26:5060:
OPTIONS sip:100@172.30.4.26:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK7af6370e;rport

From: “Unknown” sip:Unknown@172.30.4.254;tag=as0fb95ef2

To: sip:100@172.30.4.26:5060;ob

Contact: sip:Unknown@172.30.4.254

Call-ID: 14e8e7b67d91e19e155b9c46790a9079@172.30.4.254

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 23 Apr 2013 15:38:36 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY

Supported: replaces

Content-Length: 0

e[Ktrixbox1*CLI>

<— SIP read from 172.30.4.26:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK7af6370e

Call-ID: 14e8e7b67d91e19e155b9c46790a9079@172.30.4.254

From: “Unknown” sip:Unknown@172.30.4.254;tag=as0fb95ef2

To: sip:100@172.30.4.26;ob;tag=z9hG4bK7af6370e

CSeq: 102 OPTIONS

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/si mple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain

Supported: replaces, 100rel, timer, norefersub

Allow-Events: presence, message-summary, refer

User-Agent: Digium D70 1_1_2_0_51236

Content-Type: application/sdp

Content-Length: 417

v=0

o=- 62102806 62102806 IN IP4 172.30.4.26

s=digphn

c=IN IP4 172.30.4.26

t=0 0

m=audio 4000 RTP/AVP 0 8 9 111 18 58 118 58 96

a=rtcp:4001 IN IP4 172.30.4.26

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:111 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:58 L16/16000

a=rtpmap:118 L16/8000

a=rtpmap:58 L16-256/16000

a=sendrecv

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

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

e[Ktrixbox1*CLI>
— (13 headers 18 lines) —

e[Ktrixbox1*CLI>
Really destroying SIP dialog ‘14e8e7b67d91e19e155b9c46790a9079@172.30.4.254’ Method: OPTIONS

e[Ktrixbox1*CLI>
== Connect attempt from ‘127.0.0.1’ unable to authenticate

e[Ktrixbox1*CLI>
– Channel 0/2, span 1 got hangup request, cause 16

e[Ktrixbox1*CLI>
Scheduling destruction of SIP dialog ‘0b348b367f2682346f867ebd738a5ecd@172.30.4.254’ in 6400 ms (Method: INVITE)

e[Ktrixbox1*CLI>
set_destination: Parsing sip:103@172.30.4.31:5060;ob for address/port to send to
set_destination: set destination to 172.30.4.31, port 5060
Reliably Transmitting (NAT) to 172.30.4.31:5060:
BYE sip:103@172.30.4.31:5060;ob SIP/2.0

Via: SIP/2.0/UDP 172.30.4.254:5060;branch=z9hG4bK2f627af8;rport

From: “Private” sip:4045555555@172.30.4.254;tag=as2fcd2f94

To: sip:103@172.30.4.31:5060;ob;tag=hHlp6t31X3gZhAJ2eJ4FAePTiZfrIwzt

Call-ID: 0b348b367f2682346f867ebd738a5ecd@172.30.4.254

CSeq: 103 BYE

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

e[Ktrixbox1*CLI>
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘Zap/2-1’
– Executing [h@macro-dial:1] e[1;36;40mMacroe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40mhangupcalle[0;37;40m”) in new stack
– Executing [s@macro-hangupcall:1] e[1;36;40mResetCDRe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40mvwe[0;37;40m”) in new stack

<— SIP read from 172.30.4.31:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.4.254:5060;rport=5060;received=172.30.4.254;branch=z9hG4bK2f627af8

Call-ID: 0b348b367f2682346f867ebd738a5ecd@172.30.4.254

From: “Private” sip:4045555555@172.30.4.254;tag=as2fcd2f94

To: sip:103@172.30.4.31;ob;tag=hHlp6t31X3gZhAJ2eJ4FAePTiZfrIwzt

CSeq: 103 BYE

Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Executing [s@macro-hangupcall:2] e[1;36;40mNoCDRe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack

e[Ktrixbox1*CLI>
– Executing [s@macro-hangupcall:3] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40m1?skiprge[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40m1?skipblkvme[0;37;40m”) in new stack
Really destroying SIP dialog ‘0b348b367f2682346f867ebd738a5ecd@172.30.4.254’ Method: INVITE

e[Ktrixbox1*CLI>
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] e[1;36;40mGotoIfe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40m1?theende[0;37;40m”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] e[1;36;40mHangupe[0;37;40m(“e[1;35;40mZap/2-1e[0;37;40m”, “e[1;35;40me[0;37;40m”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘Zap/2-1’
– Hungup ‘Zap/2-1’

8

It is very difficult to understand the Fonality dial plan

You are working on a dead product without any support or updated documentation

running and old version of Asterisk…

If t you are new on Asterisk your best start it is migrate that Trixbox to a New Asterisk installation.

if you are runningng out time for learning try with products Like Freepbx Distro or Elastix

9

I’m willing to do that, I really like the HUD capability and don’t want to lose that is there an alternative? As for right now, I would like to get this box running. What is a good way to troubleshoot this issue and fix the existing box?

10

try to delete the extension 103 and create the extension back. and delete all messages for that extension.

11

that resolved it, thank you

12

How did you resolve the issue? Any information might come in handy when another person with a similar issue searches the forums …