Was working now can't dial out but can dial in

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>
pbx
CLI>
pbxCLI>
pbx
CLI>
pbxCLI>
pbx
CLI>
pbxCLI>
pbx
CLI>
pbxCLI>
pbx
CLI>
pbxCLI>
pbx
CLI>
pbxCLI>
pbx
CLI>

<— SIP read from UDP:HIDDEN-PHONE-IP:1025 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-faa048dc
From: “Anonymous” sip:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected]
Call-ID: [email protected]
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 - [email protected]
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:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected];tag=as6ff1c700
Call-ID: [email protected]
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 ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:HIDDEN-PHONE-IP:1025 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-faa048dc
From: “Anonymous” sip:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected];tag=as6ff1c700
Call-ID: [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-de731894
From: “Anonymous” sip:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“500”,realm=“asterisk”,nonce=“5cb9fd1c”,uri=“sip:[email protected]”,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 - [email protected]
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:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected]
Call-ID: [email protected]
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:[email protected]:5060
Content-Length: 0

<------------>
– Executing [[email protected]:1] Macro(“SIP/500-00000005”, “user-callerid,LIMIT”) in new stack
– Executing [[email protected]:1] Set(“SIP/500-00000005”, “TOUCH_MONITOR=1428329210.5”) in new stack
– Executing [[email protected]:2] Set(“SIP/500-00000005”, “AMPUSER=500”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/500-00000005”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/500-00000005”, “1?Set(REALCALLERIDNUM=500)”) in new stack
– Executing [[email protected]:5] Set(“SIP/500-00000005”, “AMPUSER=500”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/500-00000005”, “0?limit”) in new stack
– Executing [[email protected]:7] Set(“SIP/500-00000005”, “AMPUSERCIDNAME=Musicroom-HomeLine”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/500-00000005”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/500-00000005”, “AMPUSERCID=500”) in new stack
– Executing [[email protected]:10] Set(“SIP/500-00000005”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:11] Set(“SIP/500-00000005”, “CALLERID(all)=“Musicroom-HomeLine” <500>”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/500-00000005”, “0?limit”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/500-00000005”, “1?Set(GROUP(concurrency_limit)=500)”) in new stack
– Executing [[email protected]:14] GosubIf(“SIP/500-00000005”, “7?sub-ccss,s,1(from-internal,HIDDEN_NUMBER_TO_CALL)”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/500-00000005”, “0?Return()”) in new stack
– Executing [[email protected]:2] Set(“SIP/500-00000005”, “CCSS_SETUP=TRUE”) in new stack
– Executing [[email protected]: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 [[email protected]:1] GotoIf(“SIP/500-00000005”, “0?is_exten”) in new stack
– Executing [[email protected]:2] StackPop(“SIP/500-00000005”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/500-00000005”, “FALSE”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/500-00000005”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/500-00000005”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,30)
– Executing [[email protected]:30] Set(“SIP/500-00000005”, “CALLERID(number)=500”) in new stack
– Executing [[email protected]:31] Set(“SIP/500-00000005”, “CALLERID(name)=Musicroom-HomeLine”) in new stack
– Executing [[email protected]:32] Set(“SIP/500-00000005”, “CDR(cnum)=500”) in new stack
– Executing [[email protected]:33] Set(“SIP/500-00000005”, “CDR(cnam)=Musicroom-HomeLine”) in new stack
– Executing [[email protected]:34] Set(“SIP/500-00000005”, “CHANNEL(language)=en”) in new stack
– Executing [[email protected]:2] Set(“SIP/500-00000005”, “ROUTEUSER=500”) in new stack
– Executing [[email protected]: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 ‘[email protected]’ 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:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected];tag=as31138788
Call-ID: [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.133:5060;branch=z9hG4bK-de731894
From: “Anonymous” sip:[email protected];tag=8d589749eb9d9554o3
To: “RICHARD DUVAL” sip:[email protected];tag=as31138788
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“500”,realm=“asterisk”,nonce=“5cb9fd1c”,uri=“sip:[email protected]”,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) —

You need to fix your routes :slight_smile:

How can it be broken? I install, it works. I dial out once, it works. I dial out again and I get this problem.

How can it be “breaking” itself? How can an extension that is valid and working for incoming calls and (once only for outgoing calls) become invalid by itself?

Also, I haven’t changed anything manually, everything I’ve done is through the GUI.

You misunderstand the extension bit, the phone extension is working, but you are sending calls through the context “restrictedroute-0” which has no “extension match” for HIDDEN_NUMBER_TO_CALL and that context has no “i” (invalid extension) so the call dies.

I appreciate your info but since I’m just using the GUI, doesn’t it mean the GUI is “breaking” this? Can you tell me what makes something go to a restricted route and what I might have done to create this problem?

Thanks

Not really, I believe you are using a commercial module from Schmooze perhaps “Outbound call limits” or “Class of Service”

COMMERCIAL MODULE REQUIRES A LICENSE FOR IT TO FUNCTION. Please visit www.schmoozecom.com/oss.php - Restrict Outbound Calls at an outbound route level

COMMERCIAL MODULE, REQUIRES A LICENSE.
Class of Service allows you to limit access, per extension, to feature codes, trunks, queues, ring groups, etc.

They are closed source so I can’t use them or help in any material way, try and disable them and see if that fixes it, if it doesn’t perhaps post in the Commercial modules forum.

I wasn’t actually, just the normal extension module but I figured it out, just a stupid mistake on my part.

It seems that in the “Allowed Routes” section on the extensions setup page, if you don’t actually select ANY extension it will work but unpredictably and intermittently.

As soon as I checked a route box and reloaded I got a bunch of error about files it couldn’t open, etc, but I rebooted the server and now it’s working fine.

Your earlier info on the “restricted-route” got me thinking and that’s how I found it.

Wouldn’t be a bad idea if a new extension in the GUI automatically selected a route or at least warned if none was selected on save.

I don’t believe that “Allowed Routes” exists without those modules being installed (used or otherwise).