My system was working fine a couple of days ago now I can’t dial out but I can dial in no problem.
Here’s a capture from and SSH into asterisk debug (sensitive information hidden with appropriate text)… It is one complete attempt to dial out. Hoping some guru can take a look and tell me what’s going on.
BTW, I’m using 12.0.51 64 bit
Thanks much in advance, Rick.
EDITED A FEW MINUTES LATER: I restored a backup from 2 days ago that is now working. The only difference (I’m aware of) is that I updated the voicemail module “voicemail 12.0.34 (current: 12.0.33)” this morning…
bxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
pbxCLI>
<— SIP read from UDP:HIDDEN-PHONE-IP:1025 —>
INVITE sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-faa048dc
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “Anonymous” sip:[email protected]:5060
Expires: 240
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 401
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 50444682 50444682 IN IP4 192.168.1.133
s=-
c=IN IP4 192.168.1.133
t=0 0
m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (14 headers 18 lines) —
Sending to HIDDEN-PHONE-IP:1025 (NAT)
Sending to HIDDEN-PHONE-IP:1025 (NAT)
Using INVITE request as basis request - eb34122d-3abbb7c0@localhost
Found peer ‘500’ for ‘500’ from HIDDEN-PHONE-IP:1025
<— Reliably Transmitting (NAT) to HIDDEN-PHONE-IP:1025 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-faa048dc;received=HIDDEN-PHONE-IP;rport=1025
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP;tag=as6ff1c700
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 101 INVITE
Server: FPBX-12.0.51(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5cb9fd1c"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘eb34122d-3abbb7c0@localhost’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:HIDDEN-PHONE-IP:1025 —>
ACK sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-faa048dc
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP;tag=as6ff1c700
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 101 ACK
Max-Forwards: 70
Contact: “Anonymous” sip:[email protected]:5060
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0
<------------->
— (10 headers 0 lines) —
<— SIP read from UDP:HIDDEN-PHONE-IP:1025 —>
INVITE sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-de731894
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“500”,realm=“asterisk”,nonce=“5cb9fd1c”,uri=“sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP”,algorithm=MD5,response="c99a900343833ceb05fed578aaf98991"
Contact: “Anonymous” sip:[email protected]:5060
Expires: 240
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 401
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 50444682 50444682 IN IP4 192.168.1.133
s=-
c=IN IP4 192.168.1.133
t=0 0
m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (15 headers 18 lines) —
Sending to HIDDEN-PHONE-IP:1025 (NAT)
Using INVITE request as basis request - eb34122d-3abbb7c0@localhost
Found peer ‘500’ for ‘500’ from HIDDEN-PHONE-IP:1025
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.133:16428
Looking for HIDDEN_NUMBER_TO_CALL in from-internal (domain HIDDEN-FREEPBX-SERVER-IP)
list_route: hop: sip:[email protected]:5060
<— Transmitting (NAT) to HIDDEN-PHONE-IP:1025 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-de731894;received=HIDDEN-PHONE-IP;rport=1025
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 102 INVITE
Server: FPBX-12.0.51(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP:5060
Content-Length: 0
<------------>
– Executing [HIDDEN_NUMBER_TO_CALL@from-internal:1] Macro(“SIP/500-00000005”, “user-callerid,LIMIT”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/500-00000005”, “TOUCH_MONITOR=1428329210.5”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/500-00000005”, “AMPUSER=500”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/500-00000005”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/500-00000005”, “1?Set(REALCALLERIDNUM=500)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/500-00000005”, “AMPUSER=500”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/500-00000005”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/500-00000005”, “AMPUSERCIDNAME=Musicroom-HomeLine”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/500-00000005”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/500-00000005”, “AMPUSERCID=500”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/500-00000005”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/500-00000005”, “CALLERID(all)=“Musicroom-HomeLine” <500>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/500-00000005”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/500-00000005”, “1?Set(GROUP(concurrency_limit)=500)”) in new stack
– Executing [s@macro-user-callerid:14] GosubIf(“SIP/500-00000005”, “7?sub-ccss,s,1(from-internal,HIDDEN_NUMBER_TO_CALL)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/500-00000005”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/500-00000005”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/500-00000005”, “0?monitor_config,1(from-internal,HIDDEN_NUMBER_TO_CALL):monitor_default,1(from-internal,HIDDEN_NUMBER_TO_CALL)”) in new stack
– Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/500-00000005”, “0?is_exten”) in new stack
– Executing [monitor_default@sub-ccss:2] StackPop(“SIP/500-00000005”, “”) in new stack
– Executing [monitor_default@sub-ccss:3] Return(“SIP/500-00000005”, “FALSE”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“SIP/500-00000005”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] GotoIf(“SIP/500-00000005”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,30)
– Executing [s@macro-user-callerid:30] Set(“SIP/500-00000005”, “CALLERID(number)=500”) in new stack
– Executing [s@macro-user-callerid:31] Set(“SIP/500-00000005”, “CALLERID(name)=Musicroom-HomeLine”) in new stack
– Executing [s@macro-user-callerid:32] Set(“SIP/500-00000005”, “CDR(cnum)=500”) in new stack
– Executing [s@macro-user-callerid:33] Set(“SIP/500-00000005”, “CDR(cnam)=Musicroom-HomeLine”) in new stack
– Executing [s@macro-user-callerid:34] Set(“SIP/500-00000005”, “CHANNEL(language)=en”) in new stack
– Executing [HIDDEN_NUMBER_TO_CALL@from-internal:2] Set(“SIP/500-00000005”, “ROUTEUSER=500”) in new stack
– Executing [HIDDEN_NUMBER_TO_CALL@from-internal:3] GotoIf(“SIP/500-00000005”, “1?restrictedroute-0,HIDDEN_NUMBER_TO_CALL,2:outbound-allroutes,HIDDEN_NUMBER_TO_CALL,2”) in new stack
– Goto (restrictedroute-0,HIDDEN_NUMBER_TO_CALL,2)
[2015-04-06 10:06:50] WARNING[2401][C-00000005]: pbx.c:6646 __ast_pbx_run: Channel ‘SIP/500-00000005’ sent to invalid extension but no invalid handler: context,exten,priority=restrictedroute-0,HIDDEN_NUMBER_TO_CALL,2
Scheduling destruction of SIP dialog ‘eb34122d-3abbb7c0@localhost’ in 6400 ms (Method: INVITE)
<— Reliably Transmitting (NAT) to HIDDEN-PHONE-IP:1025 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-de731894;received=HIDDEN-PHONE-IP;rport=1025
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP;tag=as31138788
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 102 INVITE
Server: FPBX-12.0.51(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<— SIP read from UDP:HIDDEN-PHONE-IP:1025 —>
ACK sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-de731894
From: “Anonymous” sip:500@HIDDEN-FREEPBX-SERVER-IP;tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP;tag=as31138788
Call-ID: eb34122d-3abbb7c0@localhost
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“500”,realm=“asterisk”,nonce=“5cb9fd1c”,uri=“sip:HIDDEN_NUMBER_TO_CALL@HIDDEN-FREEPBX-SERVER-IP”,algorithm=MD5,response="c99a900343833ceb05fed578aaf98991"
Contact: “Anonymous” sip:[email protected]:5060
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0
<------------->
— (11 headers 0 lines) —