[SOLVED] I broke something

I have this single outgoing route containing one rule:

‘.’

I have a single DAHDI TDM 8xFXO card configured as Trunk 0 ascending with 1 test PSTN jacked into port 1 of 8.

I have several extensions installed and working.

I have one ring-group to which all extensions belong.

I can receive incoming calls and all extensions ring.

When I place an outgoing call I get this error:

== Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 [2013-06-11 12:00:35] DEBUG[16676][C-00000043]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1 [2013-06-11 12:00:35] DEBUG[16676][C-00000043]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d2fPhzEeVfuYlBauNh/1psiPYb4su8RGO2OgOE/k [2013-06-11 12:00:35] WARNING[16676][C-00000043]: chan_sip.c:10064 process_sdp: Declining non-primary audio stream: audio 18968 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101 [2013-06-11 12:00:35] WARNING[16676][C-00000043]: chan_sip.c:10064 process_sdp: Declining non-primary audio stream: audio 18968 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101 == Extension Changed 41712[ext-local] new state InUse for Notify User 41711 -- Executing [[email protected]:1] Macro("SIP/41712-00000067", "user-callerid,LIMIT") in new stack -- Executing [[email protected]:1] Set("SIP/41712-00000067", "TOUCH_MONITOR=1370966435.108") in new stack -- Executing [[email protected]:2] Set("SIP/41712-00000067", "AMPUSER=41712") in new stack -- Executing [[email protected]:3] GotoIf("SIP/41712-00000067", "0?report") in new stack -- Executing [[email protected]:4] ExecIf("SIP/41712-00000067", "1?Set(REALCALLERIDNUM=41712)") in new stack -- Executing [[email protected]:5] Set("SIP/41712-00000067", "AMPUSER=41712") in new stack -- Executing [[email protected]:6] Set("SIP/41712-00000067", "AMPUSERCIDNAME=James B Byrne") in new stack -- Executing [[email protected]:7] GotoIf("SIP/41712-00000067", "0?report") in new stack -- Executing [[email protected]:8] Set("SIP/41712-00000067", "AMPUSERCID=41712") in new stack -- Executing [[email protected]:9] Set("SIP/41712-00000067", "__DIAL_OPTIONS=Ttr") in new stack -- Executing [[email protected]:10] Set("SIP/41712-00000067", "CALLERID(all)="James B Byrne" <41712>") in new stack -- Executing [[email protected]:11] GotoIf("SIP/41712-00000067", "0?limit") in new stack -- Executing [[email protected]:12] ExecIf("SIP/41712-00000067", "1?Set(GROUP(concurrency_limit)=41712)") in new stack -- Executing [[email protected]:13] ExecIf("SIP/41712-00000067", "1?Set(CHANNEL(language)=en)") in new stack -- Executing [[email protected]:14] GosubIf("SIP/41712-00000067", "7?sub-ccss,s,1(from-internal,9055267697)") in new stack -- Executing [[email protected]:1] ExecIf("SIP/41712-00000067", "0?Return()") in new stack -- Executing [[email protected]:2] Set("SIP/41712-00000067", "CCSS_SETUP=TRUE") in new stack -- Executing [[email protected]:3] GosubIf("SIP/41712-00000067", "0?monitor_config,1(from-internal,9055267697):monitor_default,1(from-internal,9055267697)") in new stack -- Executing [[email protected]:1] GotoIf("SIP/41712-00000067", "0?is_exten") in new stack -- Executing [[email protected]:2] StackPop("SIP/41712-00000067", "") in new stack -- Executing [[email protected]:3] Return("SIP/41712-00000067", "FALSE") in new stack -- Executing [[email protected]:15] GotoIf("SIP/41712-00000067", "1?continue") in new stack -- Goto (macro-user-callerid,s,28) -- Executing [[email protected]:28] Set("SIP/41712-00000067", "CALLERID(number)=41712") in new stack -- Executing [[email protected]:29] Set("SIP/41712-00000067", "CALLERID(name)=James B Byrne") in new stack -- Executing [[email protected]:30] Set("SIP/41712-00000067", "CDR(cnum)=41712") in new stack -- Executing [[email protected]:31] Set("SIP/41712-00000067", "CDR(cnam)=James B Byrne") in new stack -- Executing [[email protected]:32] Set("SIP/41712-00000067", "CHANNEL(language)=en") in new stack -- Executing [[email protected]:2] Set("SIP/41712-00000067", "ROUTEUSER=41712") in new stack -- Executing [[email protected]:3] GotoIf("SIP/41712-00000067", "1?outbound-1-2,9055267697,2:outbound-allroutes,9055267697,2") in new stack -- Goto (outbound-1-2,9055267697,2) [2013-06-11 12:00:35] WARNING[6529][C-00000043]: pbx.c:6390 __ast_pbx_run: Channel 'SIP/41712-00000067' sent to invalid extension but no invalid handler: context,exten,priority=outbound-1-2,9055267697,2 [2013-06-11 12:00:35] WARNING[6529][C-00000043]: pbx.c:6390 __ast_pbx_run: Channel 'SIP/41712-00000067' sent to invalid extension but no invalid handler: context,exten,priority=outbound-1-2,9055267697,2 == Extension Changed 41712[ext-local] new state Idle for Notify User 41711

Now, I had this setup working earlier and other than applying yum updates and updating modules as notified from FreePBX the only configuration changes I have made have been with respect to adding new extensions and testing their connectivity.

Evidently I upset something else and I have no clue as to what it might be. I have checked the permissions and ownership to the best of my abilities. I have run

amportal chown
amportal a reload

without effect.

Any clues as to where I start get outside calls to work again?

You didn’t tell us the version of FreePBX, Asterisk, OS and how the system was installed.

True.

Arch = x86_64
OS = CentOS-6.4 (freepbx)
FreePBX = 2.11.0.2
Asterisk = 11.3.0

In any case, I restored a backup from a point when I definitely had things working and am moving forward from there. Everything seems to be fine at the moment.