@anhdv1979 wrote:

Dear All,
I have successfully install the lastest freepbx distro (freepbx 14 & asterisk 15) and also updated to lastest for all modules. I have a problem with paging & intercom without playing full alert tone (play only a half & clients in a group stop). I have also check the quality of the file via music on hold with good quality. Pls help me check the log and give me some suggestion. Many thanks!
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:4] While(“PJSIP/207-0000004a”, “1”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:5] Set(“PJSIP/207-0000004a”, “sipheader=Ring Answer”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:6] SIPAddHeader(“PJSIP/207-0000004a”, “Alert-Info: Ring Answer”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:7] Set(“PJSIP/207-0000004a”, “PJSIP_HEADER(add,Alert-Info)=Ring Answer”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:8] EndWhile(“PJSIP/207-0000004a”, “”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:4] While(“PJSIP/207-0000004a”, “0”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@func-apply-sipheaders:9] Return(“PJSIP/207-0000004a”, “”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] pbx.c: Executing [s@autoanswer:4] Return(“PJSIP/207-0000004a”, “”) in new stack
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] app_stack.c: Spawn extension (from-internal, PAGE207, 1) exited non-zero on ‘PJSIP/207-0000004a’
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] app_stack.c: PJSIP/207-0000004a Internal Gosub(autoanswer,s,1(Ring Answer,<uri>;answer-after=0)) complete GOSUB_RETVAL=
[2018-08-23 02:44:03] VERBOSE[15992][C-0000007a] app_dial.c: Called PJSIP/207/sip:207@192.168.1.100:5060
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] res_agi.c: <PJSIP/205-00000049>AGI Script page.agi completed, returning 0
[2018-08-23 02:44:03] VERBOSE[10233] netsock2.c: Using SIP RTP Audio TOS bits 184
[2018-08-23 02:44:03] VERBOSE[10233] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2018-08-23 02:44:03] VERBOSE[10233] netsock2.c: Using SIP RTP Audio CoS mark 5
[2018-08-23 02:44:03] VERBOSE[15990] dial.c: Called s@app-page-stream
[2018-08-23 02:44:03] VERBOSE[15993][C-0000007b] pbx.c: Executing [s@app-page-stream:1] Wait(“Local/s@app-page-stream-0000006a;2”, “1”) in new stack
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] pbx.c: Executing [2057@app-pagegroups:11] Set(“PJSIP/205-00000049”, “CONFBRIDGE(user,template)=page_user_duplex”) in new stack
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] pbx.c: Executing [2057@app-pagegroups:12] Set(“PJSIP/205-00000049”, “CONFBRIDGE(user,admin)=yes”) in new stack
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] pbx.c: Executing [2057@app-pagegroups:13] Set(“PJSIP/205-00000049”, “CONFBRIDGE(user,marked)=yes”) in new stack
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] pbx.c: Executing [2057@app-pagegroups:14] Answer(“PJSIP/205-00000049”, “”) in new stack
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] pbx.c: Executing [2057@app-pagegroups:15] ConfBridge(“PJSIP/205-00000049”, “1534992242814,admin_menu”) in new stack
[2018-08-23 02:44:03] VERBOSE[15996][C-00000078] bridge_channel.c: Channel CBAnn/1534992242814-0000006b;2 joined ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:44:03] VERBOSE[15976][C-00000078] bridge_channel.c: Channel PJSIP/205-00000049 joined ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:44:04] VERBOSE[15987][C-00000079] pbx.c: Executing [s@app-page-stream:2] Answer(“Local/s@app-page-stream-00000068;2”, “”) in new stack
[2018-08-23 02:44:04] VERBOSE[15986] dial.c: Local/s@app-page-stream-00000068;1 answered
[2018-08-23 02:44:04] VERBOSE[15987][C-00000079] pbx.c: Executing [s@app-page-stream:3] Set(“Local/s@app-page-stream-00000068;2”, “CONFBRIDGE(user,template)=page_user_duplex”) in new stack
[2018-08-23 02:44:04] VERBOSE[15987][C-00000079] pbx.c: Executing [s@app-page-stream:4] Set(“Local/s@app-page-stream-00000068;2”, “CONFBRIDGE(user,marked)=yes”) in new stack
[2018-08-23 02:44:04] VERBOSE[15987][C-00000079] pbx.c: Executing [s@app-page-stream:5] ConfBridge(“Local/s@app-page-stream-00000068;2”, “1534992242814,”) in new stack
[2018-08-23 02:44:04] VERBOSE[15987][C-00000079] bridge_channel.c: Channel Local/s@app-page-stream-00000068;2 joined ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:44:04] VERBOSE[15993][C-0000007b] pbx.c: Executing [s@app-page-stream:2] Answer(“Local/s@app-page-stream-0000006a;2”, “”) in new stack
[2018-08-23 02:44:04] VERBOSE[15990] dial.c: Local/s@app-page-stream-0000006a;1 answered
[2018-08-23 02:44:04] VERBOSE[15990] file.c: <Local/s@app-page-stream-0000006a;1> Playing ‘beep.slin16’ (language ‘en’)
[2018-08-23 02:44:04] VERBOSE[15993][C-0000007b] pbx.c: Executing [s@app-page-stream:3] Set(“Local/s@app-page-stream-0000006a;2”, “CONFBRIDGE(user,template)=page_user_duplex”) in new stack
[2018-08-23 02:44:04] VERBOSE[15993][C-0000007b] pbx.c: Executing [s@app-page-stream:4] Set(“Local/s@app-page-stream-0000006a;2”, “CONFBRIDGE(user,marked)=yes”) in new stack
[2018-08-23 02:44:04] VERBOSE[15993][C-0000007b] pbx.c: Executing [s@app-page-stream:5] ConfBridge(“Local/s@app-page-stream-0000006a;2”, “1534992242814,”) in new stack
[2018-08-23 02:44:04] VERBOSE[15993][C-0000007b] bridge_channel.c: Channel Local/s@app-page-stream-0000006a;2 joined ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:44:04] VERBOSE[15993][C-0000007b] bridge_channel.c: Channel Local/s@app-page-stream-0000006a;2 left ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:44:04] VERBOSE[15992][C-0000007a] app_dial.c: PJSIP/207-0000004a is ringing
[2018-08-23 02:44:04] VERBOSE[15992][C-0000007a] app_dial.c: PJSIP/207-0000004a is ringing
[2018-08-23 02:44:04] VERBOSE[15991] dial.c: Local/PAGE207@app-paging-00000069;1 is ringing
[2018-08-23 02:44:04] VERBOSE[15992][C-0000007a] app_dial.c: PJSIP/207-0000004a answered Local/PAGE207@app-paging-00000069;2
[2018-08-23 02:44:04] VERBOSE[15992][C-0000007a] file.c: <PJSIP/207-0000004a> Playing ‘custom/07_Track.ulaw’ (language ‘en’)
[2018-08-23 02:44:09] VERBOSE[15987][C-00000079] bridge_channel.c: Channel Local/s@app-page-stream-00000068;2 left ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:44:41] VERBOSE[6143] res_pjsip/pjsip_configuration.c: Endpoint 205 is now Reachable
[2018-08-23 02:44:41] VERBOSE[6143] res_pjsip/pjsip_options.c: Contact 205/sip:205@192.168.1.57:5060 is now Reachable. RTT: 14.932 msec
[2018-08-23 02:46:03] VERBOSE[15992][C-0000007a] pbx.c: Spawn extension (app-paging, PAGE207, 7) exited non-zero on ‘Local/PAGE207@app-paging-00000069;2’
[2018-08-23 02:46:06] VERBOSE[15976][C-00000078] bridge_channel.c: Channel PJSIP/205-00000049 left ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:46:06] VERBOSE[15996][C-00000078] bridge_channel.c: Channel CBAnn/1534992242814-0000006b;2 left ‘softmix’ base-bridge <a775068a-d9b4-4cce-b467-d1914dfa7810>
[2018-08-23 02:46:06] VERBOSE[15976][C-00000078] pbx.c: Executing [h@app-pagegroups:1] ExecIf(“PJSIP/205-00000049”, “1?Set(DEVICE_STATE(Custom:PAGE2057)=NOT_INUSE)”) in new stack

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@dataspeed wrote:

Employees are in an area where there is bad cell service which is resulting them to used Wifi calling feature on their phone. When they call in, the line is dead silent and eventually hangs up. In the call logs I can see their number hitting the server. I am able to dial from my freepbx landline to a cellphone that is using Wifi calling. Anyone know what may cause this? All other regular calls from landlines and mobile networksto the freepbx work fine

WIFI CALL.JPG2804x399 123 KB

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@adtopkek wrote:

Converted to PJSIP and we are now getting this error for the parking lot extensions:

[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145293 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘2496’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145295 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘2645’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145297 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘3047’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145299 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘3195’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145301 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘31’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145303 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘33’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145305 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘345’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145307 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘365’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145309 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘3665’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145311 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘3696’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145313 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘386’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145315 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘3927’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145317 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘3946’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145319 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘4027’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145321 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘4125’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145323 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘4246’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145325 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘4296’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145327 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘4597’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145329 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘4790’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145331 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘494’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145333 of /etc/asterisk/extensions_additional.conf
[2018-08-23 03:15:17] WARNING[19149] pbx.c: Unable to register extension ‘5190’ priority -1 in ‘park-hints’, already in use
[2018-08-23 03:15:17] WARNING[19149] pbx_config.c: Unable to register extension at line 145335 of /etc/asterisk/extensions_additional.conf

We are having problems but we don’t know if this is causing it. It is a new error after converting a server PJSIP. Is this a “normal” error or is it unusual?

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@jimmycrack215 wrote:

Hey, whatup! I am new here so if i am providing the wrong information or asking it wrong i apologize. So ive been able to work though a good amount of issues and feel comfortable setting up a basic setup. (no paid modules, CLI update, manually setting up trunks in/outbound routes and extensions, Nat and Firewall issues, remote client, CLI debugging…)

So i bring it up at work the other day and my boss asks me to fix a clients setup. I would appreciate some wisdom from someone more experienced before i make any decisions. The tiny differences in versions combined with the flood of ‘answers’ online point to the only solution… Join the FreePBX forums and get some help from someone knowledgeable instead of trying to plug in answers from other peoples questions … no matter how similar they may look.

Their setup! So the client runs an ‘organic’ pizza place. They have two locations with two PBX systems (sadly different versions!) Both are using pfSense as a firewall. The system was running fine until one location switched ISP’s. Now when calling the 1800 number it gives you two options. Location A and Location B. They also have their own local area code numbers configured too. Calling the 1800 number and selecting Option TWO will drop the call.

I think it has something to do with the caller ID being passed between the two PBX systems.

Verbosity was 0 and is now 6
  == CDR updated on SIP/dt-[COUCHTOMWC]-0000000e
    -- Executing [2@ivr-7:1] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "ext-miscdests,6,1") in new stack
    -- Goto (ext-miscdests,6,1)
    -- Executing [6@ext-miscdests:1] NoOp("SIP/dt-[COUCHTOMWC]-0000000e", "MiscDest: Mobile-Web-Number-2-West-Chester") in new stack
    -- Executing [6@ext-miscdests:2] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "from-internal,8001,1") in new stack
    -- Goto (from-internal,8001,1)
    -- Executing [8001@from-internal:1] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "TOUCH_MONITOR=1534913595.123") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/dt-[COUCHTOMWC]-0000000e", "AMPUSER=[MYCELLPHON]") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?Set(REALCALLERIDNUM=[MYCELLPHON])") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/dt-[COUCHTOMWC]-0000000e", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/dt-[COUCHTOMWC]-0000000e", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?report") in new stack
    -- Goto (macro-user-callerid,s,16)
    -- Executing [s@macro-user-callerid:16] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s@macro-user-callerid:29] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CALLERID(number)=[MYCELLPHON]") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CALLERID(name)=[MYCELLPHON]") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CDR(cnum)=[MYCELLPHON]") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CDR(cnam)=[MYCELLPHON]") in new stack
    -- Executing [s@macro-user-callerid:33] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CHANNEL(language)=en") in new stack
    -- Executing [8001@from-internal:2] Set("SIP/dt-[COUCHTOMWC]-0000000e", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing [8001@from-internal:3] Set("SIP/dt-[COUCHTOMWC]-0000000e", "MOHCLASS=default") in new stack
    -- Executing [8001@from-internal:4] Set("SIP/dt-[COUCHTOMWC]-0000000e", "_NODEST=") in new stack
    -- Executing [8001@from-internal:5] Gosub("SIP/dt-[COUCHTOMWC]-0000000e", "sub-record-check,s,1(out,8001,)") in new stack
    -- Executing [s@sub-record-check:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [s@sub-record-check:11] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Return()") in new stack
    -- Executing [s@sub-record-check:12] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [s@sub-record-check:13] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?out,1") in new stack
    -- Executing [s@sub-record-check:14] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:15] Set("SIP/dt-[COUCHTOMWC]-0000000e", "NOW=1534913610") in new stack
    -- Executing [s@sub-record-check:16] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__DAY=22") in new stack
    -- Executing [s@sub-record-check:17] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__MONTH=08") in new stack
    -- Executing [s@sub-record-check:18] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__YEAR=2018") in new stack
    -- Executing [s@sub-record-check:19] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__TIMESTR=20180822-005330") in new stack
    -- Executing [s@sub-record-check:20] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__FROMEXTEN=[MYCELLPHON]") in new stack
    -- Executing [s@sub-record-check:21] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__CALLFILENAME=out-8001-[MYCELLPHON]-20180822-005330-1534913595.123") in new stack
    -- Executing [s@sub-record-check:22] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [out@sub-record-check:2] GosubIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?record,1(exten,8001,[MYCELLPHON])") in new stack
    -- Executing [out@sub-record-check:3] Return("SIP/dt-[COUCHTOMWC]-0000000e", "") in new stack
    -- Executing [8001@from-internal:6] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "dialout-trunk,4,8001,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "DIAL_TRUNK=4") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/dt-[COUCHTOMWC]-0000000e", "DIAL_NUMBER=8001") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/dt-[COUCHTOMWC]-0000000e", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/dt-[COUCHTOMWC]-0000000e", "OUTBOUND_GROUP=OUT_4") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?sub-flp-4,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/dt-[COUCHTOMWC]-0000000e", "OUTNUM=8001") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/dt-[COUCHTOMWC]-0000000e", "custom=IAX2/inter-company-iax") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Ttr)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=TtrM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/dt-[COUCHTOMWC]-0000000e", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(CONNECTEDLINE(num,i)=8001)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(CONNECTEDLINE(name,i)=CID:[MYCELLPHON])") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/dt-[COUCHTOMWC]-0000000e", "IAX2/inter-company-iax/8001,300,Ttr") in new stack

    -- Called IAX2/inter-company-iax/8001
    -- Call accepted by 10.0.0.2 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/inter-company-iax-19368 is making progress passing it to SIP/dt-[COUCHTOMWC]-0000000e
    -- IAX2/inter-company-iax-19368 is making progress passing it to SIP/dt-[COUCHTOMWC]-0000000e
    -- IAX2/inter-company-iax-19368 is circuit-busy
    -- Hungup 'IAX2/inter-company-iax-19368'
  
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/dt-[COUCHTOMWC]-0000000e", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?continue,1:s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "RC=34") in new stack
    -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "34,1") in new stack
    -- Goto (macro-dialout-trunk,34,1)
    -- Executing [34@macro-dialout-trunk:1] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/dt-[COUCHTOMWC]-0000000e", "TRUNK Dial failed due to CONGESTiON HANGUPCAUSE: 34 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CALLERID(number)=") in new stack
    -- Executing [8001@from-internal:7] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/dt-[COUCHTOMWC]-0000000e", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?intracompany,1") in new stack
    -- Goto (macro-outisbusy,intracompany,1)
    -- Executing [intracompany@macro-outisbusy:1] Playback("SIP/dt-[COUCHTOMWC]-0000000e", "all-circuits-busy-now&pls-tRy-call-later, noanswer") in new stack
    -- <SIP/dt-[COUCHTOMWC]-0000000e> Playing 'all-circuits-busy-now.ulaw' (language 'en')
    -- <SIP/dt-[COUCHTOMWC]-0000000e> Playing 'pls-try-call-later.ulaw' (language 'en')
    -- Executing [intracompany@macro-outisbusy:2] Congestion("SIP/dt-[COUCHTOMWC]-0000000e", "20") in new stack

localhost*CLI> core set verbose 0

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@adtopkek wrote:

We switched all extensions to PJSIP and now we have these errors flooding asterisk -r. I’m not sure if it bug or is it another “normal” warning/error.

[2018-08-27 14:45:02] ERROR[11129][C-0000130e]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:45:03] WARNING[11129][C-0000130e]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:45:41] WARNING[11578][C-0000130f]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:45:41] ERROR[11578][C-0000130f]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:45:41] ERROR[11578][C-0000130f]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:45:41] WARNING[11840][C-0000130f]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:45:44] WARNING[11874][C-00001313]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:45:51] ERROR[12044][C-00001315]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:46:04] WARNING[12271][C-00001316]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
    --     -- LazyMembers debugging - Numbusies: 0, Nummems: 3
[2018-08-27 14:46:38] ERROR[12939][C-0000131c]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:46:38] WARNING[12939][C-0000131c]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:46:59] WARNING[13280][C-00001322]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
    --     -- LazyMembers debugging - Numbusies: 0, Nummems: 3
[2018-08-27 14:48:06] WARNING[14686][C-0000132a]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:48:24] WARNING[14957][C-0000132d]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:48:31] WARNING[15141][C-00001330]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:48:41] WARNING[15088][C-0000132f]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:48:45] ERROR[15435][C-00001332]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:48:52] WARNING[15141][C-00001330]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:49:01] WARNING[15711][C-00001336]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
    --     -- LazyMembers debugging - Numbusies: 0, Nummems: 6
[2018-08-27 14:49:11] ERROR[15986][C-00001338]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:11] ERROR[15985][C-00001338]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:11] ERROR[15987][C-00001338]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:11] ERROR[15988][C-00001338]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:11] ERROR[15989][C-00001338]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:11] ERROR[15990][C-00001338]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:11] WARNING[15962][C-00001337]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:49:11] ERROR[15962][C-00001337]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:49:11] ERROR[15962][C-00001337]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:49:11] WARNING[16000][C-00001337]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:49:12] ERROR[16000][C-00001337]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:49:12] ERROR[16000][C-00001337]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:49:27] WARNING[16412][C-0000133e]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:49:27] ERROR[16423][C-0000133f]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
[2018-08-27 14:49:27] WARNING[16423][C-0000133f]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
[2018-08-27 14:49:27] ERROR[16423][C-0000133f]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:49:27] ERROR[16423][C-0000133f]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
[2018-08-27 14:49:27] ERROR[16428][C-0000133f]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered

I think it’s related to dialparties.agi

 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'William' number is 'XXXXXXXXXX'
 dialparties.agi: CW Ignore is: 
 dialparties.agi: CF Ignore is: 
 dialparties.agi: CW IN_USE/BUSY is: 0
    -- AGI Script Executing Application: (SIPAddHeader) Options: (Alert-Info:Info=Inbound)
[2018-08-27 15:16:32] WARNING[21470][C-00001583]: res_agi.c:3148 handle_exec: Could not find application (SIPAddHeader)
 dialparties.agi: Methodology of ring is  'ringall'

The Audiohook is:

-- Executing [recordcheck@sub-record-check:16] NoOp("PJSIP/5402-00003efc", "Starting recording: out, 0004321") in new stack
-- Executing [recordcheck@sub-record-check:17] Set("PJSIP/5402-00003efc", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
[2018-08-27 15:16:37] ERROR[21557][C-00001584]: pbx_functions.c:699 ast_func_write: Function AUDIOHOOK_INHERIT not registered
-- Executing [recordcheck@sub-record-check:18] Set("PJSIP/5402-00003efc", "__CALLFILENAME=out-longnumber") in new stack

PJSIP is:

-- Local/5009431337@from-internal-0000100d;1 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] NoOp("Local/5009431337@from-internal-0000100d;1", "Applying SIP Headers to channel") in new stack
-- Executing [s@func-apply-sipheaders:2] Set("Local/5009431337@from-internal-0000100d;1", "SIPHEADERKEYS=Alert-Info") in new stack
-- Executing [s@func-apply-sipheaders:3] ExecIf("Local/5009431337@from-internal-0000100d;1", "0?Set(Rheader=1)") in new stack
-- Executing [s@func-apply-sipheaders:4] While("Local/5009431337@from-internal-0000100d;1", "1") in new stack
-- Executing [s@func-apply-sipheaders:5] Set("Local/5009431337@from-internal-0000100d;1", "sipheader=Info=Inbound") in new stack
-- Executing [s@func-apply-sipheaders:6] Set("Local/5009431337@from-internal-0000100d;1", "PJSIP_HEADER(add,Alert-Info)=Info=Inbound") in new stack
[2018-08-27 15:18:59] ERROR[24608][C-0000158d]: res_pjsip_header_funcs.c:513 func_write_header: This function requires a PJSIP channel.
-- Executing [s@func-apply-sipheaders:7] EndWhile("Local/5009431337@from-internal-0000100d;1", "") in new stack
-- Executing [s@func-apply-sipheaders:4] While("Local/5009431337@from-internal-0000100d;1", "0") in new stack
-- Executing [s@func-apply-sipheaders:8] ExecIf("Local/5009431337@from-internal-0000100d;1", "0?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
-- Executing [s@func-apply-sipheaders:9] Return("Local/5009431337@from-internal-0000100d;1", "") in new stack

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@josephchrz wrote:

Hello it has been a long time sense i have tried to setup tftp server. So i was wondering how hard would it be to setup the TFTP server in freepbx 14 compare to the older one i was using the freepbx 12?

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@rmapes wrote:

Good Day Everyone,
I would like to know if connecting the FreePBX to Active Directory would have a negative impact on the operation of the application? The goal is to have all servers authenticate to AD so IT can manage users.
I worked on a couple Linux (Ubuntu, CentOS 6.9) test servers and successfully have them connected to AD. These are bare bones servers with Apache2 running, seems to be happy.

Has anyone configured their FreePBX servers to authenticate to AD? No issues to report?

Regards,
Ron

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@stom wrote:

I’m getting lots of those
res_odbc.c: SQL Execute returned an error: 23000: [MySQL][ODBC 5.2(w) Driver][mysqld-5.5.56-MariaDB]Duplicate entry ‘1537285440.4841’ for key ‘PRIMARY’

and
res_odbc.c: SQL Execute error -1!
followed by
cdr_adaptive_odbc.c: cdr_adaptive_odbc: Insert failed on ‘asteriskcdrdb:cdr’. CDR failed: INSERT INTO cdr (calldate, clid, src, dst, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, uniqueid, recordingfile, cnum, cnam, linkedid, sequence) VALUES ({ ts ‘2018-09-18 18:44:07’ }, ‘“M0:Sales:” <3410721395>’, ‘3410721395’, ‘414’, ‘ext-local’, ‘Local/414@from-queue-00000797;2’, ‘SIP/413-000003b6’, ‘Dial’, ‘SIP/414,HhtrM(auto-blkvm)Ib(func-apply-sipheaders^s^1)’, 350, 350, ‘ANSWERED’, 3, ‘1537285440.4841’, ‘in-3106981831-3410721395-20180918-184336-1537285416.4838.gsm’, ‘3410721395’, ‘Sales:’, ‘1537285416.4838’, 7040)

I have also tried mysqlcheck with no issues.
Searching via cdr reports for the unique id I do get entries, so SOME are accepted and some are not.
Any ideas on what to look for appreciated

This is a default sng7 install, with nothing special on cdr configuration
OS: sangoma Kernel: x86_64 Linux 3.10.0-862.9.1.el7.x86_64
FreePBX 14.0.3.13
odbc show

ODBC DSN Settings

Name: asteriskcdrdb
DSN: MySQL-asteriskcdrdb
Number of active connections: 1 (out of 5)

cdr show status

Call Detail Record (CDR) settings

Logging: Enabled
Mode: Simple
Log unanswered calls: No
Log congestion: No

• Registered Backends

  Adaptive ODBC

contents of /etc/asterisk/cdr_adaptive_odbc.conf
[asteriskcdrdb]
connection=asteriskcdrdb
loguniqueid=yes
table=cdr
alias start => calldate

cel | CREATE TABLE cel (
id int(11) NOT NULL AUTO_INCREMENT,
eventtype varchar(30) COLLATE utf8mb4_unicode_ci NOT NULL,
eventtime datetime NOT NULL,
cid_name varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
cid_num varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
cid_ani varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
cid_rdnis varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
cid_dnid varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
exten varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
context varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
channame varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
appname varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
appdata varchar(255) COLLATE utf8mb4_unicode_ci NOT NULL,
amaflags int(11) NOT NULL,
accountcode varchar(20) COLLATE utf8mb4_unicode_ci NOT NULL,
uniqueid varchar(32) COLLATE utf8mb4_unicode_ci NOT NULL,
linkedid varchar(32) COLLATE utf8mb4_unicode_ci NOT NULL,
peer varchar(80) COLLATE utf8mb4_unicode_ci NOT NULL,
userdeftype varchar(255) COLLATE utf8mb4_unicode_ci NOT NULL,
extra varchar(512) COLLATE utf8mb4_unicode_ci NOT NULL,
PRIMARY KEY (id),
KEY uniqueid_index (uniqueid),
KEY linkedid_index (linkedid),
KEY context_index (context)
) ENGINE=InnoDB AUTO_INCREMENT=2202933 DEFAULT CHARSET=utf8mb4 COLLATE=utf8mb4_unicode_ci |

Regards

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@POLLA wrote:

I have a few extensions that when I forward a call to their voicemail they get a message that says “Your call cannot be completed as dialed. Please check the number and dial again.”

I’m using a * Sangoma FreePBX Phone System 60

The voicemail worked for these extensions before.
I do not recall making any major changes but I did do all updates.
The Destination if no answer is set to Follow ME and Normal Extension behavior.
Voicemail is enabled.

I’ve poked around to see if i can figure out what’s wrong, and I’m ready to give up .

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@tonyg wrote:

hi, i am setting up a new FreePBX 14.0.3.17 install. i have set up my sip trunk and it is registering, however, when i put my did on an extension and test, i get “all circuits busy” and the logs show

[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [s@from-pstn:1] NoOp(“PJSIP/1VOIP-00000026”, “No DID or CID Match”) in new stack

I am sure i am doing something wrong but i don’t see it. I have a packet capture and confirmed that the provider is sending 10 digits, that is what i have as the DID on the extension.

any help would be appreciated!

full log:

[2018-09-20 10:54:34] VERBOSE[14396] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.168.18’
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [s@from-pstn:1] NoOp(“PJSIP/1VOIP-00000026”, “No DID or CID Match”) in new stack
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [s@from-pstn:2] Answer(“PJSIP/1VOIP-00000026”, “”) in new stack
[2018-09-20 10:54:34] WARNING[30330][C-00000017] chan_sip.c: This function can only be used on SIP channels.
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [s@from-pstn:3] Log(“PJSIP/1VOIP-00000026”, "WARNING,Friendly Scanner from ") in new stack
[2018-09-20 10:54:34] WARNING[30330][C-00000017] Ext. s: Friendly Scanner from
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [s@from-pstn:4] Wait(“PJSIP/1VOIP-00000026”, “2”) in new stack
[2018-09-20 10:54:36] VERBOSE[30330][C-00000017] pbx.c: Executing [s@from-pstn:5] Playback(“PJSIP/1VOIP-00000026”, “ss-noservice”) in new stack
[2018-09-20 10:54:36] VERBOSE[30330][C-00000017] file.c: <PJSIP/1VOIP-00000026> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [h@from-pstn:1] Macro(“PJSIP/1VOIP-00000026”, “hangupcall,”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/1VOIP-00000026”, “1?theend”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/1VOIP-00000026”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/1VOIP-00000026”, " monior file= ") in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [s@macro-hangupcall:5] AGI(“PJSIP/1VOIP-00000026”, “attendedtransfer-rec-restart.php,”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] res_agi.c: <PJSIP/1VOIP-00000026>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“PJSIP/1VOIP-00000026”, “”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/1VOIP-00000026’ in macro ‘hangupcall’
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘PJSIP/1VOIP-00000026’

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@Krunal wrote:

Hi,

I have recently update the modules for Freepbx Distro running asterisk 13.22.0.
Suddenly the GUI stops responding with the error " Exception (2002) SQLSTATE[HY000] [2002] Connection refused::SQLSTATE[HY000] [2002] Connection refused".

While login from SSH I got some information as below.
Exception: SQLSTATE[HY000] [2002] Connection refused::SQLSTATE[HY000] [2002] Connection refused in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:

1. Exception->() /var/www/html/admin/libraries/utility.functions.php:204
2. die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:142
3. PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:137
4. PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:137
5. FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/FreePBX.class.php:71
6. FreePBX->__construct() /var/www/html/admin/bootstrap.php:153
7. require_once() /etc/freepbx.conf:11
8. include() /var/www/html/admin/modules/backup/bin/restore.php:7

Even I can not restore the old backup from GUI.
Please help me regarding the same. asterisk is working fine I guess.

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@adtopkek wrote:

We did the server module update from Zulu 2 -> Zulu 3 14.0.4.6 on a Freepbx 14.0.3.18 box.

I tried running the Zulu 3 client on my Linux Mint 18.3 and at the top is a red bar saying “Error Fetching Contacts”. The contacts module should work since we use it in the UCP all of the time and it is setup for this user. Has anyone else had this problem and found a solution?

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@strooph wrote:

Hello,
I am using freepbx14
I create a inbound route that answer the call and perform cid superfecta task that take a wild. Then call is send to a ring groupe that use the same MOH config.
I expected that music on hold will be played while superfecta is running but unfortunately the music start when the call is transferred to the ring group.
is someone experienced that , is it normal or should be set as a issue.

best regards
Philippe

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@fastdrw wrote:

Freepbx Distro SNG7-PBX-64bit-1805-1
System-Admin Timezone was set to UTC/UTC (Clock next to it show correct time)
Advanced Settings/PHP Timezone = America/Vancouver
CDR records were out by 8 hours.

Changed System-Admin Timezone to America/Vancouver (Clock next to it shows correct time)

Now, no new CDR’s are being added to CDR Reports.

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@TheRussian1968 wrote:

Asterisk 13.19.1 built by mockbuild @ jenkins7 on a x86_64 running Linux
PBX Firmware: 12.7.4-1804-2.sng7

Hello,

I am having issues with a “received incoming sip connection from unknown peer” message when routing a call into my Asterisk server. I just downloaded and installed a FreePBX (latest distro) onto a virtual machine.

Have researched and can’t seem to figure out the issue – I am a bit new to Asterisk configuration.

Below is a working configuration:

GATE ASTERISK – connects to my upstream SIP Providers and connects to internal ASTERISK servers.

Outgoing Settings

Trunk Name: MY_TRUNK

username=GATE2_LINK

type=friend

secret=<secret>

qualify=yes

insecure=very

host=voip4.<domain.tld> … this is the domain pointing to the internal Asterisk Server

dtmfmode=auto

context=from-internal

trustrpid=yes

Incoming Settings

Blank

VOIP Server – internal and sits “behind” the Gate Asterisk Server

Outgoing Settings

Trunk Name: GATE2_LINK

username=MY_TRUNK

type=friend

secret=<secret>

qualify=yes

insecure=very

host=gate2.<domain.tld> … this is the domain pointing to the gate server (upstream)

dtmfmode=auto

context=from-trunk

Incoming Settings

Blank

The Firewall is disabled and both servers sit behind a firewall. Calls flow in and out of VOIP4 as expected. All extensions are CHAN_SIP and connected to VOIP4 via CISCO SPA504g phone.

I setup in a different location a new instance using the latest FreePBX distro and installed using default settings (Asterisk 13). This time I used the included Firewall and setup using the default settings.

Created CHAN_SIP and PJSIP Extensions and connected to a CISCO SPA504g phone.

PBX Server – connected to Gate2 Asterisk server

Outgoing Settings

Trunk Name: GATE2_LINK

username=MY_TRUNK

type=friend

secret=<secret>

qualify=yes

insecure=very

host=gate2.thecomputerzone.ca

dtmfmode=auto

context=from-trunk

Incoming Settings

Blank

On the Extension connected to PBX Server I can make an outbound call. However, when I route a called to the same extension get the message “received incoming sip connection from unknown peer to …”

I can set “Allow Anonymous Inbound SIP Calls” on the PBX Server to “YES” and the calls come in but obviously this is not a permanent solution – not one I would deploy.

Any help or insight would be appreciated.

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@travis_farmer wrote:

Ok, i re-installed from the official distro (latest ISO download), as i changed the system to a different server, though i changed back when i realized the different server did not have enough memory.
so anyway, long story longer…
after a semi-fresh install (after setting up a few things), i may have somehow botched the configuration somewhere as i no longer have SSH access to the server. i have checked the firewall config, and can’t see where it would be denying access. this computer is listed as trusted.

second problem, iSymphony is installed (and re-installed), but appears to not be running. i have downloaded the RPM from getisymphony.com, but without SSH, i my hands are rather tied.

thanks in advance for any help.

~Travis

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@tonyg wrote:

hi, I thought I ran across this a while ago, but now that I need it I cannot find it…does anyone know if something like this exists? and if so, where?

thanks

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@PCaddict wrote:

Hi i try to put distinctive ring on inbound route but i face a problem. I choose my DID and when i go to the alert-info box i choose custom but i cannot edit the line… is show [CUSTOM] but cannot edit it… i try two different browser and got same result… is there a place i have to put my custom alert-info or is just a bug that i cannot edit the line…

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@tonyg wrote:

Hi, I am using FreePBX 14.0.3.25 and I noticed this on the dash about 2 weeks ago and it will not go away.

2018-10-22 08_47_53-FreePBX Administration.png901x186 17 KB

It says that ws and wss are enabled but they are not…

2018-10-22 08_49_18-FreePBX Administration.png1031x406 22.9 KB

how do i get rid of this error?

thanks in advance

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@RalphGraham wrote:

I am unable to get voicemail to email to work. Emails are sent by the system but are not received at my email address. My email provider is Hover.com who tell me that the Internal SMTP details are blocked because of the following reason
" I have taken a look and can see that the sending IP is listed on SpamHaus, which is a spam blacklist that we use."
I therefore tried to setup the external server but I am unable to obtain the reverse DNS of my system using hostname.
Hover further recommend setting an SPF record on my DNS but where in the settings can I do that?
I have checked several other sites for help but no-one else seems to have this difficulty.
A search for email here produces no results.
Any advise please.

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