@jnachbar wrote:

Security and the FreePBX vsftpd provisioning server

I am upgrading to FreePBX from an older Elastix server where I had configured my Polycom phones to pull their provisioning from an ftp server on the Elastix box.

Because it is possible to see the ftp username and password from the Polycom phones, I had set up the login that the phones used so that the username would not be able to login and so that ftp user was chroot’ed to the directory with the provisioning info only (and thus could not snoop around the server).

In FreePBX, I enabled the ftp server in System Admin / Provisioning Protocols, and after I created a username and password, the /etc/vsftpd/vsftpd.conf file was created. However, when I use ftp to connect using my username and password, I am not chroot’ed, and I can roam around the entire file system using the ftp client.

That seems way too obvious to have been overlooked. Is that really the way the ftp server is configured on FreePBX?

Also, the vsftpd.conf that is created by FreePBX includes userlist_deny=no, which would seem to allow, rather than deny, the users in user_list (which includes, e.g. root, bin, and daemon), to log in using the ftp server. Is that what is intended? Wouldn’t we want to exclude the root user from logging in using ftp?

I can manually configure the ftp server to work the way I want (which I had done on the Elastix box), but I doubt that would match the way the Polycom portion of the Endpoint Manager is configured. However, if I set up that user to chroot into /tftpboot, it might work.

Am I missing something? Is there a better way to configure the ftp server?

Thanks!

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@Bradbpw wrote:

I have some user complaining of calls going directly to VM and not ringing an extension. I’m trying to track down the route the call took in the logs, but it’s overwhelming looking through that info. I can kind of see where the call went in the CDR, but I don’t see why it may have skipped an extension. Is there a good way to determine why the call was routed in a particular direction? Maybe a specific log?

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@mvogel4949 wrote:

I’ve noticed a trend that after running a yum update on a fresh freepbx14 distro the system grinds to a halt. 5-10 minutes to shutdown and bootup. terrible audio quality. Prior to the yum update the system is super fast. Is there anyway to go backwards?

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@dlemmink wrote:

I have a custom agi-bin script I’m running in my dialplan.

When Asterisk boots normally the script will not run returning 0

If I shut down asterisk with 'asterisk -rx “core stop now” and restart it with asterisk -vvvvgp the scripts work.

Permissions on the agi file are 755 with asterisk as the owner.

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@MedicSlim wrote:

Hello, Need some help with an outbound trunk. Currently have ATT PRI with 15 channels. Trunk configured from what I can see correctly and OutBound route is set as a allow everything for testing but unable to make a call out. “all circuits are busy”

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [6523126@from-internal:1] Macro(“SIP/106-00000003”, “user-callerid,LIMIT”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/106-00000003”, “TOUCH_MONITOR=1548704927.19”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/106-00000003”, “AMPUSER=106”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/106-00000003”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/106-00000003”, “1?Set(REALCALLERIDNUM=106)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/106-00000003”, “AMPUSER=106”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/106-00000003”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/106-00000003”, "AMPUSERCIDNAME=Jeremy Barrow ") in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“SIP/106-00000003”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/106-00000003”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/106-00000003”, “AMPUSERCID=106”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/106-00000003”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/106-00000003”, "CALLERID(all)=“Jeremy Barrow " <106>”) in new stack
[2019-01-28 19:48:47] WARNING[27847][C-00000003]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘>’, expecting ‘-’ or ‘!’ or ‘(’ or ‘’; Input:
"LIMIT"=“LIMIT” & 3 & 1 & >0 & 0>=
** ^**
[2019-01-28 19:48:47] WARNING[27847][C-00000003]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/106-00000003”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/106-00000003”, “1?Set(GROUP(concurrency_limit)=106)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“SIP/106-00000003”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“SIP/106-00000003”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“SIP/106-00000003”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“SIP/106-00000003”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“SIP/106-00000003”, “CALLERID(number)=106”) in new stack
– Executing [s@macro-user-callerid:38] Set(“SIP/106-00000003”, "CALLERID(name)=Jeremy Barrow ") in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“SIP/106-00000003”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“SIP/106-00000003”, “CDR(cnam)=Jeremy Barrow”) in new stack
– Executing [s@macro-user-callerid:41] Set(“SIP/106-00000003”, “CDR(cnum)=106”) in new stack
– Executing [s@macro-user-callerid:42] Set(“SIP/106-00000003”, “CHANNEL(language)=en”) in new stack
– Executing [6523126@from-internal:2] Set(“SIP/106-00000003”, “ROUTEUSER=106”) in new stack
– Executing [6523126@from-internal:3] Set(“SIP/106-00000003”, “ROUTEUSER=106”) in new stack
– Executing [6523126@from-internal:4] GotoIf(“SIP/106-00000003”, “1?notblind”) in new stack
– Goto (from-internal,6523126,7)
– Executing [6523126@from-internal:7] GotoIf(“SIP/106-00000003”, “1?restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8,6523126,2:outbound-allroutes,6523126,2”) in new stack
– Goto (restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8,6523126,2)
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:2] Gosub(“SIP/106-00000003”, “sub-record-check,s,1(out,6523126,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/106-00000003”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/106-00000003”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/106-00000003”, “NOW=1548704927”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/106-00000003”, “__DAY=28”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/106-00000003”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/106-00000003”, “__YEAR=2019”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/106-00000003”, “__TIMESTR=20190128-194847”) in new stack
== Extension Changed 106[ext-local] new state InUse for Notify User 101
– Executing [s@sub-record-check:8] Set(“SIP/106-00000003”, “__FROMEXTEN=106”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/106-00000003”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/106-00000003”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/106-00000003”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/106-00000003”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/106-00000003”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/106-00000003”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/106-00000003”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“SIP/106-00000003”, “Outbound Recording Check from 106 to 6523126”) in new stack
– Executing [out@sub-record-check:2] Set(“SIP/106-00000003”, “RECMODE=dontcare”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“SIP/106-00000003”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“SIP/106-00000003”, “recordcheck,1(dontcare,out,6523126)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/106-00000003”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/106-00000003”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“SIP/106-00000003”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“SIP/106-00000003”, “”) in new stack
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:3] ExecIf(“SIP/106-00000003”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:4] Set(“SIP/106-00000003”, “MOHCLASS=default”) in new stack
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:5] ExecIf(“SIP/106-00000003”, “1?Set(TRUNKCIDOVERRIDE=“Medics First”<2175350100>)”) in new stack
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:6] Set(“SIP/106-00000003”, “_NODEST=”) in new stack
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:7] Macro(“SIP/106-00000003”, “dialout-trunk,1,6523126,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/106-00000003”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] UserEvent(“SIP/106-00000003”, “zulu-outbound-call,from:106,to:6523126”) in new stack
– Executing [s@macro-dialout-trunk:3] ExecIf(“SIP/106-00000003”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:4] GosubIf(“SIP/106-00000003”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:5] ExecIf(“SIP/106-00000003”, “0?Set(CALLERID(num)=106)”) in new stack
– Executing [s@macro-dialout-trunk:6] GotoIf(“SIP/106-00000003”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“SIP/106-00000003”, “DIAL_NUMBER=6523126”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“SIP/106-00000003”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“SIP/106-00000003”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/106-00000003”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:11] GotoIf(“SIP/106-00000003”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:12] GotoIf(“SIP/106-00000003”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:13] GotoIf(“SIP/106-00000003”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:14] Macro(“SIP/106-00000003”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“SIP/106-00000003”, “106”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“SIP/106-00000003”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“SIP/106-00000003”, “off”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“SIP/106-00000003”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“SIP/106-00000003”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“SIP/106-00000003”, “0?Set(REALCALLERIDNUM=106)”) in new stack
– Executing [s@macro-outbound-callerid:7] ExecIf(“SIP/106-00000003”, “0?Set(AMPUSER=106)”) in new stack
– Executing [s@macro-outbound-callerid:8] GotoIf(“SIP/106-00000003”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] Set(“SIP/106-00000003”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“SIP/106-00000003”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:14] Set(“SIP/106-00000003”, “TRUNKOUTCID=“Medics First”<2175350100>”) in new stack
– Executing [s@macro-outbound-callerid:15] GotoIf(“SIP/106-00000003”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,20)
– Executing [s@macro-outbound-callerid:20] ExecIf(“SIP/106-00000003”, “1?Set(CALLERID(all)=“Medics First”<2175350100>)”) in new stack
– Executing [s@macro-outbound-callerid:21] ExecIf(“SIP/106-00000003”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“SIP/106-00000003”, “1?Set(CALLERID(all)=Medics First<2175350100>)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/106-00000003”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/106-00000003”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:25] Set(“SIP/106-00000003”, “CDR(outbound_cnum)=2175350100”) in new stack
– Executing [s@macro-outbound-callerid:26] Set(“SIP/106-00000003”, “CDR(outbound_cnam)=Medics First”) in new stack
– Executing [s@macro-dialout-trunk:15] GosubIf(“SIP/106-00000003”, “1?sub-flp-1,s,1()”) in new stack
– Executing [s@sub-flp-1:1] ExecIf(“SIP/106-00000003”, “1?Return()”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“SIP/106-00000003”, “OUTNUM=6523126”) in new stack
– Executing [s@macro-dialout-trunk:17] Set(“SIP/106-00000003”, “custom=DAHDI/g0”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/106-00000003”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/106-00000003”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:20] Macro(“SIP/106-00000003”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/106-00000003”, “”) in new stack
– Executing [s@macro-dialout-trunk:21] GotoIf(“SIP/106-00000003”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“SIP/106-00000003”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:23] Set(“SIP/106-00000003”, “__CRM_DESTINATION=6523126”) in new stack
– Executing [s@macro-dialout-trunk:24] Set(“SIP/106-00000003”, “__CRM_SOURCE=106”) in new stack
– Executing [s@macro-dialout-trunk:25] AGI(“SIP/106-00000003”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/106-00000003>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:26] Set(“SIP/106-00000003”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:27] NoOp(“SIP/106-00000003”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:28] GotoIf(“SIP/106-00000003”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“SIP/106-00000003”, “1?Set(CONNECTEDLINE(num,i)=6523126)”) in new stack
– Executing [s@macro-dialout-trunk:30] ExecIf(“SIP/106-00000003”, “1?Set(CONNECTEDLINE(name,i)=CID:2175350100)”) in new stack
– Executing [s@macro-dialout-trunk:31] ExecIf(“SIP/106-00000003”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)2175350100)”) in new stack
– Executing [s@macro-dialout-trunk:32] GotoIf(“SIP/106-00000003”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:33] Dial(“SIP/106-00000003”, “DAHDI/g0/6523126,300,Tb(func-apply-sipheaders^s^1,(1))”) in new stack
– DAHDI/i1/6523126-4 Internal Gosub(func-apply-sipheaders,s,1(1)) start
– Executing [s@func-apply-sipheaders:1] ExecIf(“DAHDI/i1/6523126-4”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@func-apply-sipheaders:2] UserEvent(“DAHDI/i1/6523126-4”, “zulu-call-b,type:func-apply-sipheaders,to:,from:106”) in new stack
– Executing [s@func-apply-sipheaders:3] NoOp(“DAHDI/i1/6523126-4”, “Applying SIP Headers to channel 1”) in new stack
– Executing [s@func-apply-sipheaders:4] Set(“DAHDI/i1/6523126-4”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:5] ExecIf(“DAHDI/i1/6523126-4”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:6] While(“DAHDI/i1/6523126-4”, “0”) in new stack
– Jumping to priority 10
– Executing [s@func-apply-sipheaders:11] ExecIf(“DAHDI/i1/6523126-4”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:12] ExecIf(“DAHDI/i1/6523126-4”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:13] Return(“DAHDI/i1/6523126-4”, “”) in new stack
== Spawn extension (from-digital, 6523126, 1) exited non-zero on ‘DAHDI/i1/6523126-4’
– DAHDI/i1/6523126-4 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
– Requested transfer capability: 0x00 - SPEECH
– Called DAHDI/g0/6523126
– Span 1: Channel 0/1 got hangup, cause 34
– DAHDI/i1/6523126-4 is circuit-busy
– Hungup ‘DAHDI/i1/6523126-4’
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:34] NoOp(“SIP/106-00000003”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34”) in new stack
– Executing [s@macro-dialout-trunk:35] GotoIf(“SIP/106-00000003”, “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/106-00000003”, “RC=34”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/106-00000003”, “34,1”) in new stack
– Goto (macro-dialout-trunk,34,1)
– Executing [34@macro-dialout-trunk:1] Goto(“SIP/106-00000003”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] NoOp(“SIP/106-00000003”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] ExecIf(“SIP/106-00000003”, “1?Set(CALLERID(number)=106)”) in new stack
– Executing [6523126@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:8] Macro(“SIP/106-00000003”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/106-00000003”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/106-00000003”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/106-00000003”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/106-00000003”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
– <SIP/106-00000003> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
== Extension Changed 106[ext-local] new state Idle for Notify User 101
– <SIP/106-00000003> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
– Executing [h@restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8:1] Hangup(“SIP/106-00000003”, “”) in new stack
== Spawn extension (restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8, h, 1) exited non-zero on ‘SIP/106-00000003’
– SIP/106-00000003 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/106-00000003”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“SIP/106-00000003”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/106-00000003”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“SIP/106-00000003”, “MASTER CHANNEL: 1548704927.19 = 1548704927.19”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/106-00000003”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“SIP/106-00000003”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“SIP/106-00000003”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <SIP/106-00000003>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“SIP/106-00000003”, “”) in new stack
== Spawn extension (restrictedroute-5aee0cc44da9dd93c7b7f7493389cea8, h, 1) exited non-zero on ‘SIP/106-00000003’
– SIP/106-00000003 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
freepbx*CLI>

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@ozarktech wrote:

PBXact Version 10.13.66-21 When I click apply config, I get a red notification in the right hand side of the screen that says:
Symfony\Component\Process\Exception\ProcessFailedException The command “cd /var/www/html/admin/modules/pm2/node && mkdir -p /home/asterisk/.pm2 && mkdir -p /var/www/html/admin/modules/pm2/node/logs && export HOME=/home/asterisk && export PM2_HOME=/home/asterisk/.pm2 && export ASTLOGDIR=/var/log/asterisk && export ASTVARLIBDIR=/var/lib/asterisk && export PATH=$HOME/.node/bin:$PATH && export NODE_PATH=$HOME/.node/lib/node_modules:$NODE_PATH && export MANPATH=$HOME/.node/share/man:$MANPATH && /var/www/html/admin/modules/pm2/node/node_modules/pm2/bin/pm2 jlist” failed. Exit Code: 127(Command not found) Working directory: /var/www/html/admin Output: ================ Error Output: ================ /usr/bin/env: node: No such file or directory File:/var/www/html/admin/libraries/Composer/vendor/symfony/process/Process.php:221
Then a window pop up Error: Did not receive valid response from the server XHR response code: 500 XHR responseText: undefined jQuery status: error

Ive run all the updates. What next?

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@JimFromRamJack wrote:

My PolyCom SP IP 550 phones don’t register or allow me to provision them from FreePBX & I’d like to know why.

They worked at first, and still work with the configurations they’ve always had; but I can’t make any changes. Apparently one of my predecessors made some change to something that keeps the phones from successfully taking the configurations from the PBX. I’ll need to fix this to add/replace phones.

(I’d like to be able to push out the Line Key configuration to all endpoints without having to touch each one individually. I’d also like to set a wallpaper image on all phones as it was before said predecessor made his change(s).)

It appears there’s some problem in logging in to the Provisioning Server, but I’m having a hard time finding exactly what was changed to cause this. I’ve tried changing SSL, TLS, etc. and don’t really know what to enter on the PolyCom phones to get the handshake to ‘take’.

I think the predecessor must have changed something in the PBX since when he did this, he was on his way out & didn’t have time to touch each phone.

Can anyone point me in the right direction to get my SP IP 550s “talking” to the Provisioning Server again? (It’s the same server as the PBX, just on a different IP address mapped to the same NIC.)

Thanks,

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@Matthew99 wrote:

I created a queue with call strategy set to fewest calls answered. My expectation was after calling the first agent with fewest calls it would then bounce over to the next person with fewest calls and so on before going to the failover destination.

What is happening is the first person is called then it goes to my failover destination (voicemail)

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@etoosamoe wrote:

Good day everyone!

I stuckd with a trouble. First of all, if the Russian language is installed in the Sound Languages module, then in the settings you set Global Language - Ru - the following happens in the console:

<SIP/1371-00000000> Playing 'digits/12.slin' (language 'ru')
       > 0x7f01f8025f30 -- Strict RTP learning complete - Locking on source address 192.168.5.105:8000
    -- <SIP/1371-00000000> Playing 'digits/18.slin' (language 'ru')
    -- <SIP/1371-00000000> Playing 'vm-and.slin' (language 'ru')
    -- <SIP/1371-00000000> Playing 'digits/40.slin' (language 'ru')
    -- <SIP/1371-00000000> Playing 'seconds.slin' (language 'ru')

We see that the language is used Russian, but the phrases are pronounced in English and in the format .slin. I tried to add a custom language manually, but the problem is the same.

Then i tried to deletesoundlang module and its dependencies: fwconsole module delete ivr conferences findmefollow recordings soundlang

Then i execute: fwconsole module downloadinstall soundlang

root@111:/var/lib/asterisk/sounds# fwconsole ma downloadinstall soundlang
No repos specified, using: [standard,extended,unsupported] from last GUI settings

Downloading module 'soundlang'
Processing soundlang
Verifying local module download...Verified
Extracting...Done
Download completed in 5 seconds
Updating tables soundlang_packages, soundlang_settings, soundlang_customlangs, soundlang_prompts...Завершено
New install, downloading default english language set...
Installing core-sounds-g722...Завершено
Installing core-sounds-ulaw...Завершено
Installing extra-sounds-g722...Завершено
Installing extra-sounds-ulaw...Завершено
Finished installing default sounds
Installing additional package core-sounds-alaw...Завершено
Installing additional package core-sounds-alaw...
In Soundlang.class.php line 1123:

  SQLSTATE[22001]: String data, right truncated: 1406 Data too long for column 'filename' at row 1

I began to look for column named filename, and found it in table soundlang_prompts with varchar(80). There is about 4200 rows of data. I tried to

ALTER TABLE soundlang_prompts MODIFY COLUMN filename VARCHAR(100); (or 1000 or 2000)

but after i run downloadinstall soundlang the column still varchar(80).

I can’t find a command in install files that make this column exactly varchar(80).
I tried to rename or delete soundlang tables, after that the script said “new install downloading “en” package”, but… the same error

My build is Ubuntu 18.04.1 (not live), Current PBX Version: 14.0.5.25.

upd: just found, i had to do:

rm -rf /var/lib/asterisk/moh
ln -s /usr/share/asterisk/moh /var/lib/asterisk/moh
rm -rf /usr/share/asterisk/sounds
ln -s /var/lib/asterisk/sounds /usr/share/asterisk/sounds
chown -R asterisk.asterisk /usr/share/asterisk

on another system it make the asterisk speaks russian. now i have to install soundlang module on actual system, but it tells me an error

upd:
just commented the lines which installing additional modules in install.php, and it installed succesfully. then i install recordings, ivr etc, refreshsignatures and apply config correctly in webgui.
but i think it is not correct way to fix it…

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@andybengel wrote:

Hi all,

User & Devices Mode with FreePBX 14 + Asterisk 13 behaves somehow strange together with queues…

• I’m having a user 82 connected from 2 fixed devices 8201 and 8211 (and other users and devices as well, but that is not relevant right now…)
• Queue (9910, ringall) has dynamic agents 82 and others
• Device 8211 logs into the queue using *45 and it shows up as ‘(Invalid)’ in Asterisk Info/Queues

Calling the queue does not ring the phone…

• The device 8211 can login to the queue as device using 4582119910 (without defining it as dynamic agent!?), and shows up in Asterisk Info/Queues

Calling the queue does not ring the phone…

• Defining 8211 as static agent

Calling the queue now rings both devices: 8211 (as expected earlier) as well as 8201, which did neither explititely login to the queue nor is defined as agent for this queue

• Logging off the queue using *45 again results in an obviously assigned (static) member 8211, but it is not ringing on calling the queue

While the following handling of queue members seems to be a workaround, I’m not sure, if the workaround is really valid or will work in newer versions of Asterisk and/or FreePBX:

In Users&Devices-Mode you have to define one of the devices, which connected to a user who should log in to a queue, as a static agent of the queue and then you can assign this user as a dynamic agent. The user can now login using *45 as expected.

Has anyone using User & Devices Mode as well as Queues experienced something similar (or other things)?

Or has any of the experts of this forum a hint of what could be done to get the expected behaviour (without the workaround above: only define the user as dynamic agent and being able to login to the queue and receive calls from the queue)?

TIA,
Andy

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@sentinelace wrote:

I have two trunks by the same carrier. They require info for incoming and rfc2833 for outgoing for DTMF. How do I tell the PBX to use the correct trunk for incoming? Its working for outgoing fine but incoming calls are going to the outgoing trunk

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@dave69s wrote:

Sorry if my terminology is wrong on this, but I could use some input.

My agent is talking to a client. They want to bridge in a 3rd party, make an introduction, and then drop of the call, leaving the client and the 3rd party on the phone. I’m using Yealink T41p phones and can do Conference, then split and transfer, but that seems a bit clumsy. I’m happy to do this through the AGI if necessary.

Any help would be greatly appreciated!

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@faisalkhan wrote:

HI guys,

I am into a strange issue. when I checked the CDRs it’s reflecting the CDRs properly but when I check in the database asteriskcdrdb it’s not showing any results of the current date.

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@afhit_andrew wrote:

Hello,

Has anyone ever attempted to provision a polycom phone with Free PBX to work from home? I’m in a rut on how to get this done.

If someone could provide feedback or instructions on how to do this, that would be great!

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@sorvani wrote:

New system. Extension users are NOT going to be in User Manager.

But I need to setup a user to be the Fax Destination.

How do I do that?

image.png1088x313 42.5 KB

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@mike366 wrote:

Phones were working fine, now just disconnected cannot HOST UNREACHABLE, vpn dead. How to fix? I rebooted the server, it did not START vpn

[root@pbx mike]# systemctl status openvpn-server@pbx.XXX.biz
● openvpn-server@pbx.XXX.biz.service - OpenVPN service for pbx.XXX.biz
Loaded: loaded (/usr/lib/systemd/system/openvpn-server@.service; disabled; vendor preset: disabled)
Active: inactive (dead)
Docs: man:openvpn(8)
https://community.openvpn.net/openvpn/wiki/Openvpn24ManPage
https://community.openvpn.net/openvpn/wiki/HOWTO

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@seancharters1 wrote:

When using Zulu (this is a fresh install) I can make calls but incoming calls dont work. Other softphones that i have tested work find but for some reason Zulu cannot receive calls.

Is there a setting that i have missed?

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@bajramia wrote:

I have a client who have two Pbxact UC 100 with HA, he have sonicwall with two internet service providers xxx.xxx.xxx.xxx and yyy.yyy.yyy.yyy i have tried to use ddns under advance sip settings no audio. When I failover i have to login to Pbxact—>advance sip setting detect external address for the audio to work my question is how i can do this automatically if ti can be done or if is there any solution to make this work

Thank you.

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@Zirwo01 wrote:

I configured and installed mizudroid on my tables I am able to register a phone internaly but when I try to register externally using my external IP address or my hostname it does not register any Ideas? I have tried using my external ip address X.X.X.X:5060 and also my dns.net name and is the same I cannot register I know it can be access from the out side because I constantly see faulire from intruders [2019-01-29 22:05:55] NOTICE[2401] chan_sip.c: Registration from ‘112 sip:112@71.223.111.125’ failed for ‘80.93.217.133:39569’ - Wrong password
[2019-01-29 22:05:55] NOTICE[2401] chan_sip.c: Registration from ‘112 sip:112@71.223.111.125’ failed for ‘80.93.217.133:45233’ - Wrong password

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@Azer wrote:

I installed sangoma freePBX and then it’s failed on vmware vsphera ESXI host, the license also was removed. how to restore that license to new a FreePBX?

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