@comtech wrote:

Anyone on this forum have positive experience using Asternic Call Center Stats with newer versions of FreePBX (14/15)? I am looking for something of this reporting detail for queues, with limited effort/maintenance to keep alive and working, as the majority of our cycles are dedicated to other initiatives. Going for the paid, work out of box approach.

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@edwinr wrote:

If I activate this one, the call will route to next trunk but in cdr no logging was happen. Please advise. By the way it only logged the last one trunk attempted. TY in advance!

FreePBX 12
Asterisk 11.23.0

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@TonyX wrote:

Hey guys,

Using a call file without parameters like WaitTime, MaxRetries oder RetryTime all is working perfectly fine. The waittime is like the default configurated and i dont get any other calls if i dont answer the first one because default retries is 0.

The problem is happening when i set e.g. MaxRetries to 3 and RetryTime to 30 or sth. Instead of 4 calls im getting multiple calls imcoming with less retrytime than 15 seconds. Sometimes its like getting simultaneous calls.

My phone rings for about 3 seconds and i can see 5 missed calls.
Waiting for about 10-20 seconds and i can see 20+ missed calls. Checking CDR i can see that a few of the calls are completed as noanswer and some of them ending with answer.
Different WaitTime, RetryTime or anything is making it even worse.

I really have no idea what causes the problem.

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@azidhaka wrote:

Hi,

The problem began after some updates were installed on my FreePBX 13.0.197.22, Asterisk 13.29.2 system last week. My extension names contain cyrillic, where the problem probably is.

First, all CEL insert queries started to fail:

[2020-02-09 07:42:51] WARNING[20290] res_odbc.c: SQL Execute returned an error: HY000: [MySQL][ODBC 5.1 Driver][mysqld-5.7.28-0ubuntu0.18.04.4]Data too long for column ‘appdata’ at row 1
[2020-02-09 07:42:51] WARNING[20290] res_odbc.c: SQL Execute error -1!
[2020-02-09 07:42:51] WARNING[20290] cel_odbc.c: Insert failed on ‘asteriskcdrdb:cel’. CEL failed: INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES (‘APP_START’,{ts ‘2020-02-09 07:42:51.471548’},‘0892296793’,‘0892296793’,‘0892296793’,’’,‘046994001’,‘recordcheck’,‘sub-record-check’,‘SIP/Bulgaria-00058f78’,‘MixMonitor’,’/data/recordings/2020/02/09/in-046994001-0892296793-20200209-074251-1581226971.598369.wav,abi(LOCAL_MIXMON_ID),’,3,’’,‘1581226971.598369’,‘1581226971.598369’,’’,’’,’’)

I’ve extended the column ‘appdata’ to VARCHAR(255) to alleviate that.

After that, all the CDR insert queries started failing:

[2020-02-14 15:24:50] WARNING[26840] res_odbc.c: SQL Execute returned an error: HY000: [MySQL][ODBC 5.1 Driver][mysqld-5.7.28-0ubuntu0.18.04.4]Incorrect string value: ‘\xD0’ for column ‘lastdata’ at row 1
[2020-02-14 15:24:50] WARNING[26840] res_odbc.c: SQL Execute error -1!
[2020-02-14 15:24:50] WARNING[26840] 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,accountcode,uniqueid,recordingfile,cnum,cnam,outbound_cnum) VALUES ({ ts ‘2020-02-14 15:23:59’ },’"" <0040316301139>’,‘0040316301139’,‘0040722727289’,‘qlog-queuedial’,‘SIP/229-00000462’,‘SIP/International-00000463’,‘Dial’,‘SIP/International/0040722727289,300,gM(queuedial-answer^1581686639.1487^229_С�’,51,38,‘ANSWERED’,3,‘Outbound’,‘1581686639.1487’,‘out-0040722727289-229-20200214-152359-1581686639.1487.wav’,‘229’,‘229 Склад Русе’,‘0040316301139’)

/etc/odbc.ini contains Charset=utf8
Both tables (cdr, cel) collation is utf8_unicode_ci

Please help me solve that.

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@rcpldirector wrote:

We have a ring group on 602 that is set to ring all phones in the building. This worked for a little while, but now we have two cordless extensions that do not ring when the ring group is called, but do ring if you call the extension direct.

The cordless unit we are using is the Yealink W65H.

Looking at the debug, these lines are showing as busy, but I don’t know why.
https://pastebin.freepbx.org/view/d42ddd53
Fun starts on line 339

What I have tried:

1. Rebooted the W60 base
2. Rebooted the handsets
3. Dialed *79 on both handsets to ensure no DND
4. Cursed at them
5. Set Skip Busy Agent to Yes

When Skip Busy Agent is on, the phones act as they should and ring, but I feel like that is a band-aid, not a fix. What might cause this?

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@victorcasale wrote:

Dear friends, I’m running FreePBX 15.0.16.20 in a local machine.
I have created 4 PJSIP extensions under setup page and i’m using a Grandstrem HT814 where all extensions are running. I can call any extension with no problems.

I’m just concerned with this behavor:
My extensions numbers are 201, 202, 203, 204

If extension number 201 (Robert) calls extension 202(Jhon) , both calling parties can listen and talk perfeclty. Robert was wishing to call jhon, but he wasn’t in the room and alex placed the call. Alex know that jhon is in the next room where the extension is 204. In my opinion, in a normal situation, when Alex press flash key, Robert would listen to hold music, when Alex would dialing to extension 204 and sayng to jhon that Robert wishes to talk to him, so when Alex puts his phone on hook, the connection between robert and jhoh, now on extension 204 would be stabilished. It’s not happening to me. When Alex presses flash key, both Robert and Alex begin listenin a dial tone, so when alex dial 204, both listen calling tone and when jhon picks up the extension 204 both listen the conversation, like in a conference.
Can someone please help me on how solving it ?

Many thanks

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@Quafi wrote:

Hey there,

I’m using a NUT server connected to a powerwalker UPS as my UPS-System. However, since it’s not APC, I can’t get the apcupsd from FreePBX to work with it. Is there a workaround to this? I’ve thought about manually installing nut on the server, but that doesn’t seem like that great of an idea.

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@Bufuair wrote:

Take a look at this, is a sample call I have made from extension 1001 to 1000 it was processed and returned to extension 1001 and after that routed through sohoo66 portal the other extensions .
everything works fine except the caller id. EVERY CALL that come out of my pbx shows up as 1000 .

there is no outbound caller ID

Screenshot_20200217-152418_Remote Desktop.jpg2220×1080 573 KB

All the calles routed through my pbx show up on my app under the 1000 extension … and the phone number shows as name but only for the last call.

Screenshot_20200217-153353_GS Wave.jpg1080×2220 391 KB

[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [4@ext-miscdests:1] NoOp(“SIP/Soho66_1000-00000000”, “MiscDest: 1001”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [4@ext-miscdests:2] Goto(“SIP/Soho66_1000-00000000”, “from-internal,1001,1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (from-internal,1001,1)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:1] Macro(“SIP/Soho66_1000-00000000”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/Soho66_1000-00000000”, “TOUCH_MONITOR=1581891416.0”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/Soho66_1000-00000000”, “AMPUSER=1001184314”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:3] Set(“SIP/Soho66_1000-00000000”, “HOTDESCKCHAN=Soho66_1000-00000000”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:4] Set(“SIP/Soho66_1000-00000000”, “HOTDESKEXTEN=Soho66_1000”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/Soho66_1000-00000000”, “HOTDESKCALL=0”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:6] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(HOTDESKCALL=1)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:7] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(CALLERID(name)=)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:8] GotoIf(“SIP/Soho66_1000-00000000”, “0?report”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:9] ExecIf(“SIP/Soho66_1000-00000000”, “1?Set(REALCALLERIDNUM=1001184314)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:10] Set(“SIP/Soho66_1000-00000000”, “AMPUSER=”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:11] GotoIf(“SIP/Soho66_1000-00000000”, “0?limit”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:12] Set(“SIP/Soho66_1000-00000000”, “AMPUSERCIDNAME=”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:13] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:14] GotoIf(“SIP/Soho66_1000-00000000”, “1?report”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (macro-user-callerid,s,22)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:22] NoOp(“SIP/Soho66_1000-00000000”, “Macro Depth is 1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:23] GotoIf(“SIP/Soho66_1000-00000000”, “1?report2:macroerror”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (macro-user-callerid,s,24)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:24] GotoIf(“SIP/Soho66_1000-00000000”, “1?continue”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (macro-user-callerid,s,43)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:43] Set(“SIP/Soho66_1000-00000000”, “CALLERID(number)=1001184314”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:44] Set(“SIP/Soho66_1000-00000000”, “CALLERID(name)=1001184314”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:45] GotoIf(“SIP/Soho66_1000-00000000”, “0?cnum”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:46] Set(“SIP/Soho66_1000-00000000”, “CDR(cnam)=1001184314”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:47] Set(“SIP/Soho66_1000-00000000”, “CDR(cnum)=1001184314”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-user-callerid:48] Set(“SIP/Soho66_1000-00000000”, “CHANNEL(language)=en_GB”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:2] Gosub(“SIP/Soho66_1000-00000000”, “sub-record-check,s,1(out,1001,dontcare)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:1] GotoIf(“SIP/Soho66_1000-00000000”, “10?initialized”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (sub-record-check,s,10)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:10] NoOp(“SIP/Soho66_1000-00000000”, “Recordings initialized”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:11] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(ARG3=dontcare)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:12] Set(“SIP/Soho66_1000-00000000”, “REC_POLICY_MODE_SAVE=”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:13] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(REC_STATUS=NO)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:14] GotoIf(“SIP/Soho66_1000-00000000”, “3?checkaction”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (sub-record-check,s,17)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@sub-record-check:17] GotoIf(“SIP/Soho66_1000-00000000”, “1?sub-record-check,out,1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (sub-record-check,out,1)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [out@sub-record-check:1] NoOp(“SIP/Soho66_1000-00000000”, “Outbound Recording Check from 1001184314 to 1001”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [out@sub-record-check:2] Set(“SIP/Soho66_1000-00000000”, “RECMODE=”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [out@sub-record-check:3] ExecIf(“SIP/Soho66_1000-00000000”, “1?Goto(routewins)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (sub-record-check,out,7)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [out@sub-record-check:7] Gosub(“SIP/Soho66_1000-00000000”, “recordcheck,1(dontcare,out,1001)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“SIP/Soho66_1000-00000000”, “Starting recording check against dontcare”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“SIP/Soho66_1000-00000000”, “dontcare”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:3] Return(“SIP/Soho66_1000-00000000”, “”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [out@sub-record-check:8] Return(“SIP/Soho66_1000-00000000”, “”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:3] ExecIf(“SIP/Soho66_1000-00000000”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:4] Set(“SIP/Soho66_1000-00000000”, “INTRACOMPANYROUTE=YES”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:5] Set(“SIP/Soho66_1000-00000000”, “MOHCLASS=none”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:6] Set(“SIP/Soho66_1000-00000000”, “_NODEST=”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [1001@from-internal:7] Macro(“SIP/Soho66_1000-00000000”, “dialout-trunk,1,1001,off”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:1] Set(“SIP/Soho66_1000-00000000”, “DIAL_TRUNK=1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf(“SIP/Soho66_1000-00000000”, “1?Set(DIAL_OPTIONS=Hhtr)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf(“SIP/Soho66_1000-00000000”, “0?sub-pincheck,s,1()”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(CALLERID(num)=)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf(“SIP/Soho66_1000-00000000”, “0?disabletrunk,1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:6] Set(“SIP/Soho66_1000-00000000”, “DIAL_NUMBER=1001”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:7] Set(“SIP/Soho66_1000-00000000”, “DIAL_TRUNK_OPTIONS=Hhtr”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:8] Set(“SIP/Soho66_1000-00000000”, “OUTBOUND_GROUP=OUT_1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:9] Set(“SIP/Soho66_1000-00000000”, “DIAL_TRUNK_OPTIONS=T”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/Soho66_1000-00000000”, “0?nomax”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf(“SIP/Soho66_1000-00000000”, “0?chanfull”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf(“SIP/Soho66_1000-00000000”, “1?skipoutcid”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (macro-dialout-trunk,s,14)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf(“SIP/Soho66_1000-00000000”, “0?sub-flp-1,s,1()”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:15] Set(“SIP/Soho66_1000-00000000”, “OUTNUM=1001”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:16] Set(“SIP/Soho66_1000-00000000”, “custom=SIP/Soho66_1000”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf(“SIP/Soho66_1000-00000000”, “1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^none)T)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^none)TM(confirm))”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:19] Macro(“SIP/Soho66_1000-00000000”, “dialout-trunk-predial-hook,”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/Soho66_1000-00000000”, “”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf(“SIP/Soho66_1000-00000000”, “1?skipcrm”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx_builtins.c: Goto (macro-dialout-trunk,s,26)
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp(“SIP/Soho66_1000-00000000”, “CRM Finished”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf(“SIP/Soho66_1000-00000000”, “0?bypass,1”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(CONNECTEDLINE(num,i)=1001)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(CONNECTEDLINE(name,i)=“1001184314)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf(“SIP/Soho66_1000-00000000”, “0?Set(CONNECTEDLINE(name,i)=”(Hidden)1001184314)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf(“SIP/Soho66_1000-00000000”, “0?customtrunk”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:32] ExecIf(“SIP/Soho66_1000-00000000”, “1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^none))”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@macro-dialout-trunk:33] Dial(“SIP/Soho66_1000-00000000”, “SIP/Soho66_1000/1001,300,M(setmusic^none)b(func-apply-sipheaders^s^1,(1))”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] netsock2.c: Using SIP RTP TOS bits 184
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] app_stack.c: SIP/Soho66_1000-00000001 Internal Gosub(func-apply-sipheaders,s,1(1)) start
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/Soho66_1000-00000001”, “1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/Soho66_1000-00000001”, “Applying SIP Headers to channel SIP/Soho66_1000-00000001”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/Soho66_1000-00000001”, “TECH=SIP”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@func-apply-sipheaders:4] Set(“SIP/Soho66_1000-00000001”, “SIPHEADERKEYS=”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/Soho66_1000-00000001”, “0”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] app_while.c: Jumping to priority 13
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] pbx.c: Executing [s@func-apply-sipheaders:14] Return(“SIP/Soho66_1000-00000001”, “”) in new stack
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] app_stack.c: Spawn extension (from-trunk, 1001, 1) exited non-zero on ‘SIP/Soho66_1000-00000001’
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] app_stack.c: SIP/Soho66_1000-00000001 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] app_dial.c: Called SIP/Soho66_1000/1001
[2020-02-16 22:17:28] VERBOSE[21259][C-00000001] app_dial.c: SIP/Soho66_1000-00000001 is ringing
[2020-02-16 22:17:29] VERBOSE[21259][C-00000001] app_dial.c: SIP/Soho66_1000-00000001 answered SIP/Soho66_1000-00000000!

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@mclusky wrote:

A have problems with SayNumber command on newly installed FreePBX 15.0.16.20. There is no sound for numbers played with SayNumber, but when I put eg. Playback(digits/90) in dialplan - it works fine. I tested this for en, en?_GB and fr languages using such simple dialplan (commenting out necessary lines):

[test]
exten => 100,1,NoOP(test)
 same => n,Set(CHANNEL(language)=en)
 same => n,SayNumber(90)
; same => n,Playback(digits/90)
 same => n,Hangup()

The log output during silent playing is:
[2020-02-17 15:26:38] VERBOSE[1524][C-0000000f] pbx_builtins.c: Goto (test,100,1)
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] pbx.c: Launching 'NoOp'
[2020-02-17 15:26:38] VERBOSE[1524][C-0000000f] pbx.c: Executing [100@test:1] NoOp("SIP/457-0000000e", "test") in new stack
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] pbx.c: Launching 'Set'
[2020-02-17 15:26:38] VERBOSE[1524][C-0000000f] pbx.c: Executing [100@test:2] Set("SIP/457-0000000e", "CHANNEL(language)=en") in new stack
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] pbx.c: Launching 'SayNumber'
[2020-02-17 15:26:38] VERBOSE[1524][C-0000000f] pbx.c: Executing [100@test:3] SayNumber("SIP/457-0000000e", "90") in new stack
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] media_cache.c: Failed to obtain media at 'digits/90'
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] channel.c: Channel SIP/457-0000000e setting write format path: ulaw -> ulaw
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] res_rtp_asterisk.c: Setting the marker bit due to a source update
[2020-02-17 15:26:38] DEBUG[1524][C-0000000f] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2020-02-17 15:26:38] VERBOSE[1524][C-0000000f] file.c: <SIP/457-0000000e> Playing 'digits/90.ulaw' (language 'en')

Digit files in EN variant are available in 4 formats: ulaw, alaw, g722 and sln16, path is: /var/lib/asterisk/sounds/en/digits - all seems to be correct (this is newly installed system).
The only difference that comes to me analyzing logs is that log:
res_rtp_asterisk.c: Setting the marker bit due to a source update
occurs only when SayNumber command was issued.

Any ideas?

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@pstevens wrote:

FreePBX 14.0.13.17
Asterisk 13.22.0

We have a small dialplan script to change alert-info headers and trigger custom ringtones on endpoints. It’s been working for years, and had been migrated from FPBX 10 and up as we’ve upgraded, but recently added a couple of additional lines to the script but they won’t load when we perform the usual dialplan reload.

Custom dialplan:

[from-internal]

include => alert-info

[from-internal-original]
include => from-internal-noxfer
include => from-internal-xfer
include => bad-number ; auto-generated

[alert-info]
exten => _8XXXXX,1,Goto(from-internal-original,${EXTEN},1)

exten => _XXXXX,1,ExecIf($["${CALLERID(name):-4}" = “High”]?SIPAddHeader(Alert-Info: HIGH))
exten => _XXXXX,n,GotoIf($["${CALLERID(name):-4}" = “High”]?from-internal-original,${EXTEN},1)
exten => _XXXXX,1,ExecIf($["${CALLERID(name):-8}" = “Security”]?SIPAddHeader(Alert-Info: SECURITY)) ; New line not being loaded to dialplan via reload
exten => _XXXXX,n,GotoIf($["${CALLERID(name):-8}" = “Security”]?from-internal-original,${EXTEN},1) ; New line not being loaded to dialplan via reload
exten => _XXXXX,n,ExecIf($["${CALLERID(name):-6}" = “Urgent”]?SIPAddHeader(Alert-Info: URGENT))
exten => _XXXXX,n,GotoIf($["${CALLERID(name):-6}" = “Urgent”]?from-internal-original,${EXTEN},1)
exten => _XXXXX,n,ExecIf($["${CALLERID(name):-9}" = “Emergency”]?SIPAddHeader(Alert-Info: EMERGENCY))
exten => _XXXXX,n,GotoIf($["${CALLERID(name):-9}" = “Emergency”]?from-internal-original,${EXTEN},1)
exten => _XXXXX,n,ExecIf($["${CALLERID(name):-4}" = “Fire”]?SIPAddHeader(Alert-Info: FIRE))
exten => _XXXXX,n,GotoIf($["${CALLERID(name):-4}" = “Fire”]?from-internal-original,${EXTEN},1)
exten => _XXXXX,n,SIPAddHeader(Alert-Info: NORMAL)
exten => _XXXXX,n,Goto(from-internal-original,${EXTEN},1)

Result of dialplan show alert-info after adding new lines:

[ Context 'alert-info' created by 'pbx_config' ]

‘_8XXXXX’ => 1. Goto(from-internal-original,${EXTEN},1) [pbx_config]
‘_XXXXX’ => 1. ExecIf($["${CALLERID(name):-4}" = “High”]?SIPAddHeader(Alert-Info: HIGH)) [pbx_config]
2. n,GotoIf($["${CALLERID(name):-4}" = “HIgh”]?from-internal-original,${EXTEN},1) [pbx_config]
3. ExecIf($["${CALLERID(name):-6}" = “Urgent”]?SIPAddHeader(Alert-Info: URGENT)) [pbx_config]
4. GotoIf($["${CALLERID(name):-6}" = “Urgent”]?from-internal-original,${EXTEN},1) [pbx_config]
5. ExecIf($["${CALLERID(name):-9}" = “Emergency”]?SIPAddHeader(Alert-Info: EMERGENCY)) [pbx_config]
6. GotoIf($["${CALLERID(name):-9}" = “Emergency”]?from-internal-original,${EXTEN},1) [pbx_config]
7. ExecIf($["${CALLERID(name):-4}" = “Fire”]?SIPAddHeader(Alert-Info: FIRE)) [pbx_config]
8. GotoIf($["${CALLERID(name):-4}" = “Fire”]?from-internal-original,${EXTEN},1) [pbx_config]
9. SIPAddHeader(Alert-Info: NORMAL) [pbx_config]
10. Goto(from-internal-original,${EXTEN},1) [pbx_config]

-= 2 extensions (11 priorities) in 1 context. =-

We’ve tried adding (+) next to the context name i.e. [alert-info] (+) with no success.

Any suggestions welcome! (Including tips to streamline this code…)

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@josephchrz wrote:

Hello i have a small home business. trying to keep up with freepbx and modules updating as well as security. I was wondering if it would be better to put it on one of the freepbx cloud hosting compnaies that offer this type of service as well as support they would do all the backups, update sa well if the system fails. Because 2 times my system has die on me. And i really don’t have the time to replace it or fix it as well as keeping up with everthing.

What does the community think, I know the monthly cost will be there but it would save me a lot of headaches. I’m just trying to get a second opinion.

joseph

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@mudasar321 wrote:

Hi, I’m using Freepbx version 15. When I edit any extension and make changes after I lost “parent_user_id” in table cxpanel_users. Please advise.

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@haylocc wrote:

I was able to work with this when we were chan_sip but since moving to pjsip I am not sure the best way to accomplish this. I have DIDs that have e911 service for 2 locations and was able to use outbound routes to map the extension to dial 911 with the corresponding DID. Now that pjsip allows multiple extension registrations the only constant variable I can use is the static IP we have from our ISP. I saw this in the forum and am having trouble trying to adapt it for our usage to comply with e911. We have a combination of chan_sip and pjsip extensions but have a static IP for each location. How could I get the below to work so that eg. Static IP of 123.12.123.12 would use DID 1234567890 and Static IP of 123.12.123.13 would use DID 1234567891 when dialing 911.

[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp()
exten => s,n,GotoIf($[$["${CALLERID(dnid)}" = "911"] | $["${EMERGENCYROUTE}" = "YES"]]?s_CHANNLE_TYPE,1:s_exit,1)

exten => s_CHANNLE_TYPE,1,NoOp()
exten => s_CHANNLE_TYPE,n,GotoIF($["${CHANNEL(channeltype)}" = "PJSIP"]?s_PJSIP_IP,1)
exten => s_CHANNLE_TYPE,n,GotoIF($["${CHANNEL(channeltype)}" = "SIP"]?s_SIP_IP,1)
exten => s_CHANNLE_TYPE,n,Log(NOTICE, <Emergency CID OVERRIDE> Unable to get channel type. CID-IP Mapping not used.)
exten => s_CHANNLE_TYPE,n,Goto(s_exit,1)

exten => s_PJSIP_IP,1,NoOp()
exten => s_PJSIP_IP,n,Set(endpoint_addr=${CHANNEL(pjsip,remote_addr)})
exten => s_PJSIP_IP,n,Set(ip_addr=${CUT(endpoint_addr,:,1)})
exten => s_PJSIP_IP,n,GotoIF($ ["${ip_addr}" != ""]?s_CID_IP_MAP,1)
exten => s_PJSIP_IP,n,Log(NOTICE, <Emergency CID OVERRIDE> Unable to get endpoint ip from PJSIP channel. CID-IP Mapping not used.)
exten => s_PJSIP_IP,n,Goto(s_exit,1)

exten => s_SIP_IP,1,NoOp()
exten => s_SIP_IP,n,Set(ip_addr=${CHANNEL(sip,peerip)})
exten => s_SIP_IP,n,GotoIF($ ["${ip_addr}" != ""]?s_CID_IP_MAP,1)
exten => s_SIP_IP,n,Log(NOTICE, <Emergency CID OVERRIDE> Unable to get endpoint ip from SIP channel. CID-IP Mapping not used.)
exten => s_SIP_IP,n,Goto(s_exit,1)

exten => s_CID_IP_MAP,1,NoOp()
exten => s_CID_IP_MAP,n,GoSubIF(${REGEX("(192\.168\.7\.[0-9])" ${ip_addr})}?s_SetCID,1(5551117777)) ;location1
exten => s_CID_IP_MAP,n,GoSubIF(${REGEX("(192\.168\.8\.[0-9])" ${ip_addr})}?s_SetCID,1(5551118888)) ;location2
exten => s_CID_IP_MAP,n,GoSubIF(${REGEX("(192\.168\.9\.[0-9])" ${ip_addr})}?s_SetCID,1(5551119999)) ;location3
exten => s_CID_IP_MAP,n,GoSubIF(${REGEX("(192\.168\.0\.[0-9])" ${ip_addr})}?s_SetCID,1(5551110000)) ;location4
exten => s_CID_IP_MAP,n,Log(NOTICE, <Emergency CID OVERRIDE> Unable to find CID mapping for ${ip_addr} . CID-IP Mapping not used.)
exten => s_CID_IP_MAP,n,Goto(s_exit,1)

exten => s_SetCID,1,NoOp()
exten => s_SetCID,n,Set(CALLERID(num)=${ARG1})   
exten => s_SetCID,n,Goto(s_exit,1)

exten => s_exit,1,NoOP()
exten => s_exit,n,MacroExit()

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@gw22 wrote:

With the new caller id regulations coming into play, starting to have FollowMe calls failing if the calling party has their number blocked, or anonymous. The call will ring on a local phone and end up in voicemail, but wont externally send the call out on a sip trunk.

Executing [s@macro-user-callerid:9] ExecIf(“SIP/SIP1-0000024d”, “0?Set(REALCALLERIDNUM=anonymous)”) in new stack

Got SIP response 603 “Declined” back from 184.75.215.114:5060

I have a Route CID set in the outbound route and override extension set to yes, but still fails the call

Have tried setting sendrpid to pai and yes but they didnt seem to make a difference.

Also tried setting a failover trunk, but not sure if thats the right way to go

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@josephchrz wrote:

Hello i post asking about endpoint manger and tftp server. But now the topic is closed i can not reply. I had to recreate this one and ask which commercial modules would i need for the oss endpoint manger and tftp server please?

Joseph

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@carlyle705 wrote:

Really need help on this.

Call forwarding setup:

1. Misc Destination to my cellphone
2. In my extension advanced tab, scroll all the way down, set Optional Destinations to Misc Destination
3. I have a inbound call route for my phone number to directly call my extension.

When I call my number, the call will be transferred to my cellphone, but no audio.

Any help would be appreciated.

Thanks!!

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@AmidouFlorian92 wrote:

Hi all
I’m searching a way to configure an SMS gateway on my freepbx server.
I want to be able to send and receive automatic SMS on my server.
Is there a solution to do It using the CLI or what solution can I install to have an SMS module working with freepbx ?
Thanks

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@Cooltronics wrote:

I am having an issue where it tells me UCP, RESTful API, and LetsEncrypt under Port Management are all disabled. When I try to enable them to their defaults and click “update now” nothing at all happens. Tried 3 different web browsers, both http & https and different Pc’s just in case.

Any ideas?

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@Randee wrote:

Our system is working fine, can make inbound and outbound calls just fine. However when we call a number that’s been disconnected or no longer in service we don’t get any kind of message, just silence.

An employee was explaining that the phones don’t work because some numbers are just silence, well when I call from my cell phone I get the “we’re sorry your call did not go through” type message. Is there anyway to change outbound routes / trunk settings to let callers know the number they are calling is disconnected? is this a VOIP provider issue? I am using flowroute as our provider.

BTW: 309-755-7700 is currently disconnected…

Thanks for any info,
Randy

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@Philipp971 wrote:

Hi there!

so, we are sending a lot of faxes per day and it seems that we can just send two faxes simultaneously and then the message: “Outgoing Channels Are Busy. 0 Channels Available Now” orrcurs in the UCP-Fax-Dashboard.

I’am very happy with the Fax-Pro-Module, it was a worthwhile purchase, but i really need to set up this max-outgoing-fax counter to like… 4 simultaneous faxes or maybe higher.

We got two virtual extensions going out via the same CID (CID and the outgoing faxes are working perfect and without problems) but is there any way to set the “counter” for max-outgoing faxes higher than two simultaneous faxes?

Thanks a lot!
Philipp

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