@matthewljensen wrote:

I started messing around with the database commands from the CLI, and I wanted to try changing the followme number in an extension from there. It seems to work, meaning that calls get redirected to the new number, simply by running something like:

database put AMPUSER 201/followme/grplist 123456789#

But I read in this post: CLI commands, that this isn’t the best way to do this and that it may cause problems. And that it’s better to change it in the MySQL database, and then run a fwconsole reload.

Is this still the case, and if so, what are the repercussions of changing them directly in the asterisk database rather than in the MySQL database?

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@TrueTech wrote:

Hi guys,

I’m not very familiar with Asterisk coding and most things we use for our PBX is simply done via the GUI.
We have now come across an issue where a cellular provider no longer allows diverted calls if the diversion header is not in the correct format.
The diversion header that is being sent from my PBX is as follows (sign changed to > for posting)

Diversion: >tel:042xxxxxxx>;reason=no-answer;screen=no;privacy=off

The provider blocks these calls as they require the header to be in international format, so the 0 at the start of the number needs to be +27
I have seen posts on how to modify the header, etc. but I can’t figure out how to change that number to show +27 instead of the 0, as I can’t just add the +27.

Apologies if this has been covered elsewhere.

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@mvogel4949 wrote:

I have a relatively new server running FreePBX14. HD Space is only at 4% capacity and using the graph on the front screen memory consumption is very low. Yesterday the system froze up. all the phones came unregistered. A reboot of the system brought everything back to life. I’ve searched the Asterisk logfiles for possible reasons and haven’t found anything worthwhile. Is there a better logfile to search? Thanks

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@necto_random wrote:

I have freepbx 14, asterisk 13.
All my calls internal; internal to out; out to internal, falls after 31 seconds.
I use nat, do not use nat, situation same.
It not depeds of phone, becose when I came to queues it fail too.
this is my logs when i call.
Spawn extension (macro-dial, s, 22) exited non-zero on in macro ‘dial’
I use microtik and it have no udp limits.
I have no idea what is it.

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@allisone wrote:

Hello Everyone,

I have searched for this particular issue but I don’t see any topics that mention the specific problem that we are having. We recently moved our company from an on-premise FreePBX platform to a cloud-hosted FreePBX. In the transition, we moved from Chan_SIP to PJSIP on our trunks and extensions. We also are now running FreePBX 14.0.5.2 We are using Yealink T48S with 35.83.0.50 firmware.

Calls picked up from a parking lot are dropped after 24 seconds. This happens on all extensions with inbound and outbound calls that have been parked. I can listen to the recording of the call and all of the audio is captured until the call is dropped.

When I look in the CDR reports on the calls that are dropped, I see that the call is transferred to the lot and is picked back up but then a HANGUP, CHAN_END, PARK_END, and BRIDGE_EXIT all trigger.

RPID is set to Remote=Party-ID header and Trust RPID is set to Yes.

I have tried increasing the parking timeout, RTP Timeout, and RTP Hold Timeout but none of those setting affected the number of seconds before the call was dropped.

I have tried leaving the call in the parking lot until the timer sends the call back to its destination and the call is not dropped at that point. I can talk as long as I want without losing the call.

Are there any other settings that would impact this?
Could this be related to the change to PJSIP?

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@CAHOP240 wrote:

I’ve been wrestling with a problem and I can’t seem to pinpoint the issue. My setup:

AudioCodes VE SBC - 7.20A.204.222 - fqdn = audiocodes. mydomian .com (resolves internally and externally)

FreePBX - 14.0.5.2 - fqdn = pbx. mydomain. com (resolves internally)

Asterisk - 15.5.0

Using pjsip for UDP, TCP and TLS

I have a pjsip trunk setup between my PBX and the SBC. Trunk name is ‘AudioCodes’, context is set to from-ptsn, and I’m using the FQDN of my SBC in the trunk configuration. Local users register directly to the PBX. I’m trying to allow remote users to register to the PBX through the SBC. I point remote users to the external FQDN of the SBC. When a register packet comes in, the SBC will change the host portion of the SIP URI in the To and From headers from the FQDN of the SBC to the FQDN of the PBX and then send it off. When the PBX receives the register request it replies with a ‘404 Not Found’. Checking the logs on the PBX I get the error 'AOR not found for endpoint ‘AudioCodes’ ’

Looking at pcaps from a local register and one coming from the SBC, everything looks the same, except the Request-URI for an SBC registration will still contain the FQDN of the SBC while a local registration will have the FQDN of the PBX. I feel like there’s something simple that I’m missing but can’t seem to put my finger on it.

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@mst wrote:

Experts,

If I buy endpoint manager and activate under specific installation ID, then if I need to replace the server or move that endpoint to another freepbx server - is that possible to move the license to another FreePBX installation?

I dont want to believe its tight to only one installation and not possible to move the license to another server. So the solution is to paid another 150 for endpoint for new freepbx installation???

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@hardocp wrote:

I have an Asterisk v 16.1 FPBX v14 server with PJSIP and TLS SRTP setup

I have a bunch of phones on the same lan all setup the same as far as i can tell – the phones are all either yealink t5 or t4 models

One extension rings when dialed – the rest do not

Outbound calls work fine on all the extensions as does an echo test

x118 rings and looks like this:

== Setting global variable ‘SIPDOMAIN’ to ‘xxx.xxx.com’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
– Executing [118@from-internal:1] GotoIf(“PJSIP/121-0000000a”, “1?ext-local,118,1:followme-check,118,1”) in new stack
– Goto (ext-local,118,1)
– Executing [118@ext-local:1] Set(“PJSIP/121-0000000a”, “__RINGTIMER=15”) in new stack
– Executing [118@ext-local:2] Macro(“PJSIP/121-0000000a”, “exten-vm,118,118,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“PJSIP/121-0000000a”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/121-0000000a”, “TOUCH_MONITOR=1544977594.10”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/121-0000000a”, “AMPUSER=121”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/121-0000000a”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/121-0000000a”, “1?Set(REALCALLERIDNUM=121)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/121-0000000a”, “AMPUSER=121”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/121-0000000a”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/121-0000000a”, “AMPUSERCIDNAME=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/121-0000000a”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/121-0000000a”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/121-0000000a”, “AMPUSERCID=121”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/121-0000000a”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/121-0000000a”, “CALLERID(all)=“Asher Toporovsky” <121>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/121-0000000a”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/121-0000000a”, “0?Set(GROUP(concurrency_limit)=121)”) in new stack
– Executing [s@macro-user-callerid:15] NoOp(“PJSIP/121-0000000a”, “Macro Depth is 2”) in new stack
– Executing [s@macro-user-callerid:16] GotoIf(“PJSIP/121-0000000a”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,17)
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/121-0000000a”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:18] Set(“PJSIP/121-0000000a”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/121-0000000a”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,35)
– Executing [s@macro-user-callerid:35] Set(“PJSIP/121-0000000a”, “CALLERID(number)=121”) in new stack
– Executing [s@macro-user-callerid:36] Set(“PJSIP/121-0000000a”, “CALLERID(name)=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:37] GotoIf(“PJSIP/121-0000000a”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/121-0000000a”, “CDR(cnam)=Asher Toporovsky”) in new stack
– Executing [s@macro-user-callerid:39] Set(“PJSIP/121-0000000a”, “CDR(cnum)=121”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/121-0000000a”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-exten-vm:2] Set(“PJSIP/121-0000000a”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“PJSIP/121-0000000a”, “__EXTTOCALL=118”) in new stack
– Executing [s@macro-exten-vm:4] Set(“PJSIP/121-0000000a”, “__PICKUPMARK=118”) in new stack
– Executing [s@macro-exten-vm:5] Set(“PJSIP/121-0000000a”, “RT=15”) in new stack
– Executing [s@macro-exten-vm:6] Gosub(“PJSIP/121-0000000a”, “sub-record-check,s,1(exten,118,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/121-0000000a”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/121-0000000a”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/121-0000000a”, “NOW=1544977594”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/121-0000000a”, “__DAY=16”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/121-0000000a”, “__MONTH=12”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/121-0000000a”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/121-0000000a”, “__TIMESTR=20181216-112634”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/121-0000000a”, “__FROMEXTEN=121”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/121-0000000a”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/121-0000000a”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/121-0000000a”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/121-0000000a”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/121-0000000a”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/121-0000000a”, “5?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/121-0000000a”, “1?sub-record-check,exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] NoOp(“PJSIP/121-0000000a”, “Exten Recording Check between 121 and 118”) in new stack
– Executing [exten@sub-record-check:2] Set(“PJSIP/121-0000000a”, “CALLTYPE=internal”) in new stack
– Executing [exten@sub-record-check:3] ExecIf(“PJSIP/121-0000000a”, “0?Set(CALLTYPE=)”) in new stack
– Executing [exten@sub-record-check:4] Set(“PJSIP/121-0000000a”, “CALLEE=dontcare”) in new stack
– Executing [exten@sub-record-check:5] ExecIf(“PJSIP/121-0000000a”, “0?Set(CALLEE=dontcare)”) in new stack
– Executing [exten@sub-record-check:6] GotoIf(“PJSIP/121-0000000a”, “0?callee”) in new stack
– Executing [exten@sub-record-check:7] GotoIf(“PJSIP/121-0000000a”, “1?caller”) in new stack
– Goto (sub-record-check,exten,13)
– Executing [exten@sub-record-check:13] Set(“PJSIP/121-0000000a”, “RECMODE=dontcare”) in new stack
– Executing [exten@sub-record-check:14] ExecIf(“PJSIP/121-0000000a”, “0?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:15] ExecIf(“PJSIP/121-0000000a”, “1?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:16] Gosub(“PJSIP/121-0000000a”, “recordcheck,1(dontcare,internal,118)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/121-0000000a”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/121-0000000a”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/121-0000000a”, “”) in new stack
– Executing [exten@sub-record-check:17] Return(“PJSIP/121-0000000a”, “”) in new stack
– Executing [s@macro-exten-vm:7] GotoIf(“PJSIP/121-0000000a”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,13)
– Executing [s@macro-exten-vm:13] GosubIf(“PJSIP/121-0000000a”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:14] Macro(“PJSIP/121-0000000a”, “dial-one,15,HhTtr,118”) in new stack
– Executing [s@macro-dial-one:1] Set(“PJSIP/121-0000000a”, “DEXTEN=118”) in new stack
– Executing [s@macro-dial-one:2] ExecIf(“PJSIP/121-0000000a”, “0?Set(__EXTTOCALL=118)”) in new stack
– Executing [s@macro-dial-one:3] Set(“PJSIP/121-0000000a”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“PJSIP/121-0000000a”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:5] GosubIf(“PJSIP/121-0000000a”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:6] GotoIf(“PJSIP/121-0000000a”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,9)
– Executing [s@macro-dial-one:9] GotoIf(“PJSIP/121-0000000a”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:10] GotoIf(“PJSIP/121-0000000a”, “0?continue”) in new stack
– Executing [s@macro-dial-one:11] Set(“PJSIP/121-0000000a”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:12] GotoIf(“PJSIP/121-0000000a”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [s@macro-dial-one:24] GotoIf(“PJSIP/121-0000000a”, “0?next3:continue”) in new stack
– Goto (macro-dial-one,s,26)
– Executing [s@macro-dial-one:26] GotoIf(“PJSIP/121-0000000a”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:27] GosubIf(“PJSIP/121-0000000a”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“PJSIP/121-0000000a”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“PJSIP/121-0000000a”, “DEVICES=118”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“PJSIP/121-0000000a”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“PJSIP/121-0000000a”, “0?Set(DEVICES=18)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“PJSIP/121-0000000a”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“PJSIP/121-0000000a”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“PJSIP/121-0000000a”, “THISDIAL=PJSIP/118”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“PJSIP/121-0000000a”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“PJSIP/121-0000000a”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“PJSIP/121-0000000a”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“PJSIP/121-0000000a”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“PJSIP/121-0000000a”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“PJSIP/121-0000000a”, “THISPART2=PJSIP/118”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“PJSIP/121-0000000a”, “0?Set(THISPART2=DAHDIIP/118)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“PJSIP/121-0000000a”, “NEWDIAL=PJSIP/118&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“PJSIP/121-0000000a”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“PJSIP/121-0000000a”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“PJSIP/121-0000000a”, “THISDIAL=PJSIP/118”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“PJSIP/121-0000000a”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“PJSIP/121-0000000a”, “0?docheck”) in new stack
– Executing [dstring@macro-dial-one:10] NoOp(“PJSIP/121-0000000a”, “Debug: Found PJSIP Destination PJSIP/118”) in new stack
– Executing [dstring@macro-dial-one:11] GotoIf(“PJSIP/121-0000000a”, “0?doset”) in new stack
– Executing [dstring@macro-dial-one:12] NoOp(“PJSIP/121-0000000a”, “Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS”) in new stack
– Executing [dstring@macro-dial-one:13] Set(“PJSIP/121-0000000a”, “THISDIAL=PJSIP/118/sip:118@68.195.13.5:49396;transport=TLS”) in new stack
– Executing [dstring@macro-dial-one:14] ExecIf(“PJSIP/121-0000000a”, “0?Set(DIALSTATUS=CHANUNAVAIL)”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“PJSIP/121-0000000a”, “0?skipset”) in new stack
– Executing [dstring@macro-dial-one:16] Set(“PJSIP/121-0000000a”, “DSTRING=PJSIP/118/sip:118@68.195.13.5:49396;transport=TLS&”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“PJSIP/121-0000000a”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:18] GotoIf(“PJSIP/121-0000000a”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:19] ExecIf(“PJSIP/121-0000000a”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:20] Set(“PJSIP/121-0000000a”, “DSTRING=PJSIP/118/sip:118@68.195.13.5:49396;transport=TLS”) in new stack
– Executing [dstring@macro-dial-one:21] Return(“PJSIP/121-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“PJSIP/121-0000000a”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:29] GotoIf(“PJSIP/121-0000000a”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:30] GosubIf(“PJSIP/121-0000000a”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“PJSIP/121-0000000a”, “DB(CALLTRACE/118)=121”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“PJSIP/121-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:31] Set(“PJSIP/121-0000000a”, “D_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dial-one:32] GosubIf(“PJSIP/121-0000000a”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:33] NoOp(“PJSIP/121-0000000a”, "Blind Transfer: , Attended Transfer: , User: 121, Alert Info: ") in new stack
– Executing [s@macro-dial-one:34] ExecIf(“PJSIP/121-0000000a”, “1?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:35] ExecIf(“PJSIP/121-0000000a”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:36] ExecIf(“PJSIP/121-0000000a”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:37] ExecIf(“PJSIP/121-0000000a”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:38] ExecIf(“PJSIP/121-0000000a”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:39] GosubIf(“PJSIP/121-0000000a”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:40] ExecIf(“PJSIP/121-0000000a”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:41] GosubIf(“PJSIP/121-0000000a”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:42] Set(“PJSIP/121-0000000a”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:43] Set(“PJSIP/121-0000000a”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:44] GotoIf(“PJSIP/121-0000000a”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:45] GotoIf(“PJSIP/121-0000000a”, “0?godial”) in new stack
– Executing [s@macro-dial-one:46] Gosub(“PJSIP/121-0000000a”, “sub-presencestate-display,s,1(118)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“PJSIP/121-0000000a”, “state-not_set,1”) in new stack
– Goto (sub-presencestate-display,state-not_set,1)
– Executing [state-not_set@sub-presencestate-display:1] Set(“PJSIP/121-0000000a”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“PJSIP/121-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:47] Set(“PJSIP/121-0000000a”, “CONNECTEDLINE(name,i)=Rinaldo Toporovsky”) in new stack
– Executing [s@macro-dial-one:48] Set(“PJSIP/121-0000000a”, “CONNECTEDLINE(num)=118”) in new stack
– Executing [s@macro-dial-one:49] Set(“PJSIP/121-0000000a”, “D_OPTIONS=HhTtrI”) in new stack
– Executing [s@macro-dial-one:50] Macro(“PJSIP/121-0000000a”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“PJSIP/121-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:51] ExecIf(“PJSIP/121-0000000a”, “0?Set(D_OPTIONS=HhtrII)”) in new stack
– Executing [s@macro-dial-one:52] NoOp(“PJSIP/121-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:53] ExecIf(“PJSIP/121-0000000a”, “0?Set(D_OPTIONS=HhTtrIg)”) in new stack
– Executing [s@macro-dial-one:54] Dial(“PJSIP/121-0000000a”, “PJSIP/118/sip:118@68.195.13.5:49396;transport=TLS,15,HhTtrIb(func-apply-sipheaders^s^1)”) in new stack
– PJSIP/118-0000000b Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/118-0000000b”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/118-0000000b”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:3] ExecIf(“PJSIP/118-0000000b”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/118-0000000b”, “0”) in new stack
– Jumping to priority 8
– Executing [s@func-apply-sipheaders:9] ExecIf(“PJSIP/118-0000000b”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/118-0000000b”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:11] Return(“PJSIP/118-0000000b”, “”) in new stack
== Spawn extension (from-internal, 118, 1) exited non-zero on ‘PJSIP/118-0000000b’
– PJSIP/118-0000000b Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/118/sip:118@68.195.13.5:49396;transport=TLS
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
– Connected line update to PJSIP/121-0000000a prevented.
– PJSIP/118-0000000b is ringing
– PJSIP/118-0000000b is ringing

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Participants: 3

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@chaptech wrote:

Howdy,

I am trying to create a emergency evacuation page/intercom that pages all phones ( and a couple SNOM-PA1s ) and broadcasts a emergency evac announcement, trying to play for around 2 minutes. After exactly 16 seconds i see this in the console

== Spawn extension (app-paging, PAGE100, 7) exited non-zero on ‘Local/PAGE100@app-paging-000005ad;2’

and the page call is placed on hold, if i press resume it will pick up where it left off.

Any assistance appreciated on getting this not to go on hold or better way to do achieve this

cheers

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Participants: 1

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@DIYA wrote:

Hi

I recently upgraded our freepbx from version 2 to 14. We seem to have a few calls per day dropped when users try to transfer external calls to internal extensions. When I got to test I can never replicate and this does not occur on the vast majority of our calls. We use Snom 300/360/370 phones

pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/167-000016be”, “CALLERID(number)=167”) in new stack
pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/167-000016be”, “CALLERID(name)=Karen D”) in new stack
pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“SIP/167-000016be”, “0?cnum”) in new stack
pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/167-000016be”, “CDR(cnam)=Karen D”) in new stack
pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/167-000016be”, “CDR(cnum)=167”) in new stack
pbx.c: Executing [s@macro-user-callerid:42] Set(“SIP/167-000016be”, “CHANNEL(language)=en”) in new stack
pbx.c: Executing [s@macro-exten-vm:2] Set(“SIP/167-000016be”, “RingGroupMethod=none”) in new stack
pbx.c: Executing [s@macro-exten-vm:3] Set(“SIP/167-000016be”, “__EXTTOCALL=155”) in new stack
pbx.c: Executing [s@macro-exten-vm:4] Set(“SIP/167-000016be”, “__PICKUPMARK=155”) in new stack
pbx.c: Executing [s@macro-exten-vm:5] Set(“SIP/167-000016be”, “RT=15”) in new stack
pbx.c: Executing [s@macro-exten-vm:6] ExecIf(“SIP/167-000016be”, “0?Macro(vm,155,DIRECTDIAL,)”) in new stack
pbx.c: Executing [s@macro-exten-vm:7] ExecIf(“SIP/167-000016be”, “0?MacroExit()”) in new stack
pbx.c: Executing [s@macro-exten-vm:8] ExecIf(“SIP/167-000016be”, “0?Gosub(ext-intercom,*80155,1())”) in new stack
pbx.c: Executing [s@macro-exten-vm:9] ExecIf(“SIP/167-000016be”, “0?MacroExit()”) in new stack
pbx.c: Executing [s@macro-exten-vm:10] ExecIf(“SIP/167-000016be”, “0?ChanSpy(SIP/155,q)”) in new stack
pbx.c: Executing [s@macro-exten-vm:11] ExecIf(“SIP/167-000016be”, “0?MacroExit()”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx.c: Executing [s@macro-exten-vm:12] ExecIf(“SIP/167-000016be”, “0?Macro(vm,155,DIRECTDIAL,)”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx_functions.c: Function PJSIP_HEADER not registered
Pbx.c: Executing [s@macro-exten-vm:13] ExecIf(“SIP/167-000016be”, “0?MacroExit()”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx_functions.c: Function PJSIP_HEADER not registered
pbx.c: Executing [s@macro-exten-vm:14] ExecIf(“SIP/167-000016be”, “0?Gosub(ext-intercom,*80155,1())”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx_functions.c: Function PJSIP_HEADER not registered
pbx.c: Executing [s@macro-exten-vm:15] ExecIf(“SIP/167-000016be”, “0?MacroExit()”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx_functions.c: Function PJSIP_HEADER not registered
pbx.c: Executing [s@macro-exten-vm:16] ExecIf(“SIP/167-000016be”, “0?ChanSpy(SIP/155,q)”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx_functions.c: Function PJSIP_HEADER not registered
pbx.c: Executing [s@macro-exten-vm:17] ExecIf(“SIP/167-000016be”, “0?MacroExit()”) in new stack
pbx_functions.c: Function PJSIP_HEADER not registered
pbx.c: Executing [s@macro-exten-vm:18] Gosub(“SIP/167-000016be”, “sub-record-check,s,1(exten,155,dontcare)”) in new stack
pbx.c: Executing [s@sub-record-check:1] GotoIf(“SIP/167-000016be”, “0?initialized”) in new stack
pbx.c: Executing [s@sub-record-check:2] Set(“SIP/167-000016be”, “__REC_STATUS=INITIALIZED”) in new stack
pbx.c: Executing [s@sub-record-check:3] Set(“SIP/167-000016be”, “NOW=1545042941”) in new stack
pbx.c: Executing [s@sub-record-check:4] Set(“SIP/167-000016be”, “__DAY=17”) in new stack
pbx.c: Executing [s@sub-record-check:5] Set(“SIP/167-000016be”, “__MONTH=12”) in new stack
pbx.c: Executing [s@sub-record-check:6] Set(“SIP/167-000016be”, “__YEAR=2018”) in new stack
pbx.c: Executing [s@sub-record-check:7] Set(“SIP/167-000016be”, “__TIMESTR=20181217-103541”) in new stack
pbx.c: Executing [s@sub-record-check:8] Set(“SIP/167-000016be”, “__FROMEXTEN=167”) in new stack
pbx.c: Executing [s@sub-record-check:9] Set(“SIP/167-000016be”, “__MON_FMT=wav”) in new stack
pbx.c: Executing [s@sub-record-check:10] NoOp(“SIP/167-000016be”, “Recordings initialized”) in new stack
pbx.c: Executing [s@sub-record-check:11] ExecIf(“SIP/167-000016be”, “0?Set(ARG3=dontcare)”) in new stack
pbx.c: Executing [s@sub-record-check:12] Set(“SIP/167-000016be”, “REC_POLICY_MODE_SAVE=”) in new stack
pbx.c: Executing [s@sub-record-check:13] ExecIf(“SIP/167-000016be”, “0?Set(REC_STATUS=NO)”) in new stack
pbx.c: Executing [s@sub-record-check:14] GotoIf(“SIP/167-000016be”, “5?checkaction”) in new stack
pbx_builtins.c: Goto (sub-record-check,s,17)
pbx.c: Executing [s@sub-record-check:17] GotoIf(“SIP/167-000016be”, “1?sub-record-check,exten,1”) in new stack
pbx_builtins.c: Goto (sub-record-check,exten,1)
pbx.c: Executing [exten@sub-record-check:1] NoOp(“SIP/167-000016be”, “Exten Recording Check between 167 and 155”) in new stack
pbx.c: Executing [exten@sub-record-check:2] Set(“SIP/167-000016be”, “CALLTYPE=internal”) in new stack
pbx.c: Executing [exten@sub-record-check:3] ExecIf(“SIP/167-000016be”, “0?Set(CALLTYPE=)”) in new stack
pbx.c: Executing [exten@sub-record-check:4] Set(“SIP/167-000016be”, “CALLEE=dontcare”) in new stack
pbx.c: Executing [exten@sub-record-check:5] ExecIf(“SIP/167-000016be”, “0?Set(CALLEE=dontcare)”) in new stack
pbx.c: Executing [exten@sub-record-check:6] GotoIf(“SIP/167-000016be”, “0?callee”) in new stack
pbx.c: Executing [exten@sub-record-check:7] GotoIf(“SIP/167-000016be”, “1?caller”) in new stack
pbx_builtins.c: Goto (sub-record-check,exten,13)
pbx.c: Executing [exten@sub-record-check:13] Set(“SIP/167-000016be”, “RECMODE=dontcare”) in new stack
pbx.c: Executing [exten@sub-record-check:14] ExecIf(“SIP/167-000016be”, “0?Set(RECMODE=dontcare)”) in new stack
pbx.c: Executing [exten@sub-record-check:15] ExecIf(“SIP/167-000016be”, “1?Set(RECMODE=dontcare)”) in new stack
pbx.c: Executing [exten@sub-record-check:16] Gosub(“SIP/167-000016be”, “recordcheck,1(dontcare,internal,155)”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“SIP/167-000016be”, “Starting recording check against dontcare”) in new stack
pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“SIP/167-000016be”, “dontcare”) in new stack
pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
pbx.c: Executing [recordcheck@sub-record-check:3] Return(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [exten@sub-record-check:17] Return(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [s@macro-exten-vm:19] GotoIf(“SIP/167-000016be”, “1?macrodial”) in new stack
pbx_builtins.c: Goto (macro-exten-vm,s,25)
pbx.c: Executing [s@macro-exten-vm:25] GosubIf(“SIP/167-000016be”, “0?clrheader,1()”) in new stack
pbx.c: Executing [s@macro-exten-vm:26] Macro(“SIP/167-000016be”, “dial-one,15,HhTtr,155”) in new stack
pbx.c: Executing [s@macro-dial-one:1] Set(“SIP/167-000016be”, “DEXTEN=155”) in new stack
Pbx.c: Executing [s@macro-dial-one:2] Set(“SIP/167-000016be”, “__CRM_SOURCE=167”) in new stack
pbx.c: Executing [s@macro-dial-one:3] ExecIf(“SIP/167-000016be”, “0?Set(__EXTTOCALL=155)”) in new stack
pbx.c: Executing [s@macro-dial-one:4] Set(“SIP/167-000016be”, “DIALSTATUS_CW=”) in new stack
pbx.c: Executing [s@macro-dial-one:5] GosubIf(“SIP/167-000016be”, “0?screen,1()”) in new stack
pbx.c: Executing [s@macro-dial-one:6] GosubIf(“SIP/167-000016be”, “0?cf,1()”) in new stack
pbx.c: Executing [s@macro-dial-one:7] GotoIf(“SIP/167-000016be”, “1?skip1”) in new stack
pbx_builtins.c: Goto (macro-dial-one,s,10)
pbx.c: Executing [s@macro-dial-one:10] GotoIf(“SIP/167-000016be”, “0?nodial”) in new stack
pbx.c: Executing [s@macro-dial-one:11] GotoIf(“SIP/167-000016be”, “0?continue”) in new stack
pbx.c: Executing [s@macro-dial-one:12] Set(“SIP/167-000016be”, “EXTHASCW=ENABLED”) in new stack
pbx.c: Executing [s@macro-dial-one:13] GotoIf(“SIP/167-000016be”, “0?next1:cwinusebusy”) in new stack
pbx_builtins.c: Goto (macro-dial-one,s,25)
pbx.c: Executing [s@macro-dial-one:25] GotoIf(“SIP/167-000016be”, “0?next3:continue”) in new stack
pbx_builtins.c: Goto (macro-dial-one,s,27)
pbx.c: Executing [s@macro-dial-one:27] GotoIf(“SIP/167-000016be”, “0?nodial”) in new stack
pbx.c: Executing [s@macro-dial-one:28] GosubIf(“SIP/167-000016be”, “1?dstring,1():dlocal,1()”) in new stack
pbx.c: Executing [dstring@macro-dial-one:1] Set(“SIP/167-000016be”, “DSTRING=”) in new stack
pbx.c: Executing [dstring@macro-dial-one:2] Set(“SIP/167-000016be”, “DEVICES=155”) in new stack
pbx.c: Executing [dstring@macro-dial-one:3] ExecIf(“SIP/167-000016be”, “0?Return()”) in new stack
pbx.c: Executing [dstring@macro-dial-one:4] ExecIf(“SIP/167-000016be”, “0?Set(DEVICES=55)”) in new stack
pbx.c: Executing [dstring@macro-dial-one:5] Set(“SIP/167-000016be”, “LOOPCNT=1”) in new stack
pbx.c: Executing [dstring@macro-dial-one:6] Set(“SIP/167-000016be”, “ITER=1”) in new stack
pbx.c: Executing [dstring@macro-dial-one:7] Set(“SIP/167-000016be”, “THISDIAL=SIP/155”) in new stack
pbx.c: Executing [dstring@macro-dial-one:8] GosubIf(“SIP/167-000016be”, “1?zap2dahdi,1()”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/167-000016be”, “0?Return()”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/167-000016be”, “NEWDIAL=”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/167-000016be”, “LOOPCNT2=1”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/167-000016be”, “ITER2=1”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/167-000016be”, “THISPART2=SIP/155”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/167-000016be”, “0?Set(THISPART2=DAHDI/155)”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/167-000016be”, “NEWDIAL=SIP/155&”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/167-000016be”, “ITER2=2”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/167-000016be”, “0?begin2”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/167-000016be”, “THISDIAL=SIP/155”) in new stack
pbx.c: Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [dstring@macro-dial-one:9] GotoIf(“SIP/167-000016be”, “1?docheck”) in new stack
pbx_builtins.c: Goto (macro-dial-one,dstring,15)
pbx.c: Executing [dstring@macro-dial-one:15] GotoIf(“SIP/167-000016be”, “0?skipset”) in new stack
pbx.c: Executing [dstring@macro-dial-one:16] Set(“SIP/167-000016be”, “DSTRING=SIP/155&”) in new stack
pbx.c: Executing [dstring@macro-dial-one:17] Set(“SIP/167-000016be”, “ITER=2”) in new stack
pbx.c: Executing [dstring@macro-dial-one:18] GotoIf(“SIP/167-000016be”, “0?begin”) in new stack
pbx.c: Executing [dstring@macro-dial-one:19] ExecIf(“SIP/167-000016be”, “0?Return()”) in new stack
pbx.c: Executing [dstring@macro-dial-one:20] Set(“SIP/167-000016be”, “DSTRING=SIP/155”) in new stack
pbx.c: Executing [dstring@macro-dial-one:21] Return(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [s@macro-dial-one:29] GotoIf(“SIP/167-000016be”, “0?nodial”) in new stack
pbx.c: Executing [s@macro-dial-one:30] GotoIf(“SIP/167-000016be”, “0?skiptrace”) in new stack
pbx.c: Executing [s@macro-dial-one:31] GosubIf(“SIP/167-000016be”, “1?ctset,1():ctclear,1()”) in new stack
pbx.c: Executing [ctset@macro-dial-one:1] Set(“SIP/167-000016be”, “DB(CALLTRACE/155)=167”) in new stack
pbx.c: Executing [ctset@macro-dial-one:2] Return(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [s@macro-dial-one:32] Set(“SIP/167-000016be”, “D_OPTIONS=HhTtr”) in new stack
pbx.c: Executing [s@macro-dial-one:33] GosubIf(“SIP/167-000016be”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
pbx.c: Executing [s@macro-dial-one:34] NoOp(“SIP/167-000016be”, "Blind Transfer: , Attended Transfer: , User: 167, Alert Info: ") in new stack
pbx.c: Executing [s@macro-dial-one:35] ExecIf(“SIP/167-000016be”, “1?Set(ALERT_INFO=)”) in new stack
pbx.c: Executing [s@macro-dial-one:36] ExecIf(“SIP/167-000016be”, “0?Set(ALERT_INFO=)”) in new stack
pbx.c: Executing [s@macro-dial-one:37] ExecIf(“SIP/167-000016be”, “0?Set(ALERT_INFO=)”) in new stack
pbx.c: Executing [s@macro-dial-one:38] ExecIf(“SIP/167-000016be”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
pbx.c: Executing [s@macro-dial-one:39] ExecIf(“SIP/167-000016be”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
pbx.c: Executing [s@macro-dial-one:40] GosubIf(“SIP/167-000016be”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
pbx.c: Executing [s@macro-dial-one:41] ExecIf(“SIP/167-000016be”, “0?Set(CHANNEL(musicclass)=)”) in new stack
pbx.c: Executing [s@macro-dial-one:42] GosubIf(“SIP/167-000016be”, “0?qwait,1()”) in new stack
pbx.c: Executing [s@macro-dial-one:43] Set(“SIP/167-000016be”, “__CWIGNORE=”) in new stack
pbx.c: Executing [s@macro-dial-one:44] Set(“SIP/167-000016be”, “__KEEPCID=TRUE”) in new stack
pbx.c: Executing [s@macro-dial-one:45] GotoIf(“SIP/167-000016be”, “0?usegoto,1”) in new stack
pbx.c: Executing [s@macro-dial-one:46] GotoIf(“SIP/167-000016be”, “0?godial”) in new stack
pbx.c: Executing [s@macro-dial-one:47] Gosub(“SIP/167-000016be”, “sub-presencestate-display,s,1(155)”) in new stack
pbx.c: Executing [s@sub-presencestate-display:1] Goto(“SIP/167-000016be”, “state-not_set,1”) in new stack
pbx_builtins.c: Goto (sub-presencestate-display,state-not_set,1)
pbx.c: Executing [state-not_set@sub-presencestate-display:1] Set(“SIP/167-000016be”, “PRESENCESTATE_DISPLAY=”) in new stack
pbx.c: Executing [state-not_set@sub-presencestate-display:2] Return(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [s@macro-dial-one:48] Set(“SIP/167-000016be”, “CONNECTEDLINE(name,i)=Mike”) in new stack
pbx.c: Executing [s@macro-dial-one:49] Set(“SIP/167-000016be”, “CONNECTEDLINE(num)=155”) in new stack
pbx.c: Executing [s@macro-dial-one:50] Set(“SIP/167-000016be”, “D_OPTIONS=HhTtrI”) in new stack
pbx.c: Executing [s@macro-dial-one:51] Macro(“SIP/167-000016be”, “dialout-one-predial-hook,”) in new stack
pbx.c: Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [s@macro-dial-one:52] ExecIf(“SIP/167-000016be”, “0?Set(D_OPTIONS=HhtrII)”) in new stack
pbx.c: Executing [s@macro-dial-one:53] NoOp(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [s@macro-dial-one:54] ExecIf(“SIP/167-000016be”, “0?Set(D_OPTIONS=HhTtrIg)”) in new stack
pbx.c: Executing [s@macro-dial-one:55] Dial(“SIP/167-000016be”, “SIP/155,15,HhTtrIb(func-apply-sipheaders^s^1)”) in new stack
netsock2.c: Using SIP RTP TOS bits 184
netsock2.c: Using SIP RTP CoS mark 5
app_stack.c: SIP/155-000016bf Internal Gosub(func-apply-sipheaders,s,1) start
pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/155-000016bf”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/155-000016bf”, “Applying SIP Headers to channel”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/155-000016bf”, “SIPHEADERKEYS=”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:4] ExecIf(“SIP/155-000016bf”, “0?Set(Rheader=1)”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/155-000016bf”, “0”) in new stack
app_while.c: Jumping to priority 8
pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf(“SIP/155-000016bf”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:10] Return(“SIP/155-000016bf”, “”) in new stack
app_stack.c: Spawn extension (from-internal, 155, 1) exited non-zero on ‘SIP/155-000016bf’
app_stack.c: SIP/155-000016bf Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
app_dial.c: Called SIP/155
app_dial.c: Connected line update to SIP/167-000016be prevented.
chan_sip.c: Extension Changed 155[ext-local] new state Ringing for Notify User 170
chan_sip.c: Extension Changed 155[ext-local] new state Ringing for Notify User 142
app_dial.c: SIP/155-000016bf is ringing
chan_sip.c: Extension Changed 155[ext-local] new state Ringing for Notify User 170
chan_sip.c: Extension Changed 155[ext-local] new state Ringing for Notify User 142
app_dial.c: SIP/155-000016bf is ringing
app_macro.c: Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘SIP/167-000016be’ in macro ‘dial-one’
app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘SIP/167-000016be’ in macro ‘exten-vm’
pbx.c: Spawn extension (ext-local, 155, 2) exited non-zero on ‘SIP/167-000016be’
pbx.c: Executing [h@ext-local:1] Macro(“SIP/167-000016be”, “hangupcall,”) in new stack
pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/167-000016be”, “1?theend”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,3)
chan_sip.c: Extension Changed 155[ext-local] new state Idle for Notify User 170
chan_sip.c: Extension Changed 155[ext-local] new state Idle for Notify User 142
pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/167-000016be”, “0?Set(CDR(recordingfile)=)”) in new stack
pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/167-000016be”, “SIP/155-000016bf monior file= “) in new stack
pbx.c: Executing [s@macro-hangupcall:5] AGI(“SIP/167-000016be”, “attendedtransfer-rec-restart.php,SIP/155-000016bf,”) in new stack
res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
res_musiconhold.c: Stopped music on hold on SIP/sip_os-000016b9
bridge_channel.c: Channel SIP/167-000016bd left ‘simple_bridge’ basic-bridge <1366e421-e9b9-44b7-9d69-2f00bc62cda6>
bridge_channel.c: Channel SIP/sip_os-000016b9 left ‘simple_bridge’ basic-bridge <1366e421-e9b9-44b7-9d69-2f00bc62cda6>
app_macro.c: Channel ‘SIP/sip_os-000016b9’ jumping out of macro ‘dial’
chan_sip.c: Extension Changed 167[ext-local] new state Idle for Notify User 170
pbx.c: Executing [155@from-internal-xfer:1] GotoIf(“SIP/sip_os-000016b9”, “1?ext-local,155,1:followme-check,155,1”) in new stack
pbx_builtins.c: Goto (ext-local,155,1)
pbx.c: Executing [155@ext-local:1] Set(“SIP/sip_os-000016b9”, “__RINGTIMER=15”) in new stack
pbx.c: Executing [155@ext-local:2] Macro(“SIP/sip_os-000016b9”, “exten-vm,155,155,0,0,0”) in new stack
pbx.c: Executing [s@macro-exten-vm:1] Macro(“SIP/sip_os-000016b9”, “user-callerid,”) in new stack
pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/sip_os-000016b9”, “TOUCH_MONITOR=1545042915.5925”) in new stack
pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/sip_os-000016b9”, “AMPUSER=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/sip_os-000016b9”, “0?report”) in new stack
pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/sip_os-000016b9”, “0?Set(REALCALLERIDNUM=02085088892)”) in new stack
pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/sip_os-000016b9”, “AMPUSER=”) in new stack
pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/sip_os-000016b9”, “0?limit”) in new stack
VERBOSE[2185] chan_sip.c: Extension Changed 167[ext-local] new state Idle for Notify User 142
pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/sip_os-000016b9”, “AMPUSERCIDNAME=”) in new stack
pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“SIP/sip_os-000016b9”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“SIP/sip_os-000016b9”, “1?report”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,16)
pbx.c: Executing [s@macro-user-callerid:16] NoOp(“SIP/sip_os-000016b9”, “Macro Depth is 2”) in new stack
pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“SIP/sip_os-000016b9”, “1?report2:macroerror”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,18)
pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“SIP/sip_os-000016b9”, “0?continue”) in new stack
pbx.c: Executing [s@macro-user-callerid:19] ExecIf(“SIP/sip_os-000016b9”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
pbx.c: Executing [s@macro-user-callerid:20] Set(“SIP/sip_os-000016b9”, “__TTL=63”) in new stack
pbx.c: Executing [s@macro-user-callerid:21] GotoIf(“SIP/sip_os-000016b9”, “1?continue”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,37)
pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/sip_os-000016b9”, “CALLERID(number)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/sip_os-000016b9”, “CALLERID(name)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“SIP/sip_os-000016b9”, “0?cnum”) in new stack
pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/sip_os-000016b9”, “CDR(cnam)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/sip_os-000016b9”, “CDR(cnum)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:42] Set(“SIP/sip_os-000016b9”, “CHANNEL(language)=en”) in new stack
pbx.c: Executing [s@macro-exten-vm:2] Set(“SIP/sip_os-000016b9”, “RingGroupMethod=none”) in new stack
pbx.c: Executing [s@macro-exten-vm:3] Set(“SIP/sip_os-000016b9”, “__EXTTOCALL=155”) in new stack
pbx.c: Executing [s@macro-exten-vm:4] Set(“SIP/sip_os-000016b9”, “__PICKUPMARK=155”) in new stack
pbx.c: Executing [s@macro-exten-vm:5] Set(“SIP/sip_os-000016b9”, “RT=15”) in new stack
WARNING[7453][C-00000b4b] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected ‘’, expecting $end; Input:
““Do Not Disturb”” = “send_to_vm” | “” = “feature_send_to_vm”
^
WARNING[7453][C-00000b4b] ast_expr2.fl: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
pbx.c: Executing [s@macro-exten-vm:6] ExecIf(“SIP/sip_os-000016b9”, “””?Macro(vm,155,DIRECTDIAL,)”) in new stack
pbx.c: Executing [s@macro-vm:1] Macro(“SIP/sip_os-000016b9”, “user-callerid,SKIPTTL”) in new stack
pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/sip_os-000016b9”, “TOUCH_MONITOR=1545042915.5925”) in new stack
pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/sip_os-000016b9”, “AMPUSER=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/sip_os-000016b9”, “0?report”) in new stack
pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/sip_os-000016b9”, “0?Set(REALCALLERIDNUM=02085088892)”) in new stack
pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/sip_os-000016b9”, “AMPUSER=”) in new stack
pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/sip_os-000016b9”, “0?limit”) in new stack
pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/sip_os-000016b9”, “AMPUSERCIDNAME=”) in new stack
pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“SIP/sip_os-000016b9”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“SIP/sip_os-000016b9”, “1?report”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,16)
pbx.c: Executing [s@macro-user-callerid:16] NoOp(“SIP/sip_os-000016b9”, “Macro Depth is 3”) in new stack
pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“SIP/sip_os-000016b9”, “1?report2:macroerror”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,18)
pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“SIP/sip_os-000016b9”, “1?continue”) in new stack
pbx_builtins.c: Goto (macro-user-callerid,s,37)
pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/sip_os-000016b9”, “CALLERID(number)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/sip_os-000016b9”, “CALLERID(name)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“SIP/sip_os-000016b9”, “0?cnum”) in new stack
pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/sip_os-000016b9”, “CDR(cnam)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/sip_os-000016b9”, “CDR(cnum)=02085088892”) in new stack
pbx.c: Executing [s@macro-user-callerid:42] Set(“SIP/sip_os-000016b9”, “CHANNEL(language)=en”) in new stack
pbx.c: Executing [s@macro-vm:2] Set(“SIP/sip_os-000016b9”, “VMGAIN=”) in new stack
pbx.c: Executing [s@macro-vm:3] Macro(“SIP/sip_os-000016b9”, “blkvm-check,”) in new stack
pbx.c: Executing [s@macro-blkvm-check:1] Set(“SIP/sip_os-000016b9”, “GOSUB_RETVAL=”) in new stack
pbx.c: Executing [s@macro-blkvm-check:2] ExecIf(“SIP/sip_os-000016b9”, “0?Set(GOSUB_RETVAL=TRUE)”) in new stack
pbx.c: Executing [s@macro-blkvm-check:3] MacroExit(“SIP/sip_os-000016b9”, “”) in new stack
pbx.c: Executing [s@macro-vm:4] GotoIf(“SIP/sip_os-000016b9”, “1?vmx,1”) in new stack
pbx_builtins.c: Goto (macro-vm,vmx,1)
pbx.c: Executing [vmx@macro-vm:1] Set(“SIP/sip_os-000016b9”, “__EXTTOCALL=155”) in new stack
pbx.c: Executing [vmx@macro-vm:2] Set(“SIP/sip_os-000016b9”, “__CRM_VOICEMAIL=155”) in new stack
pbx.c: Executing [vmx@macro-vm:3] Set(“SIP/sip_os-000016b9”, “MEXTEN=155”) in new stack
pbx.c: Executing [vmx@macro-vm:4] Set(“SIP/sip_os-000016b9”, “MMODE=DIRECTDIAL”) in new stack
pbx.c: Executing [vmx@macro-vm:5] Set(“SIP/sip_os-000016b9”, “RETVM=”) in new stack
pbx.c: Executing [vmx@macro-vm:6] Set(“SIP/sip_os-000016b9”, “MODE=unavail”) in new stack
pbx.c: Executing [vmx@macro-vm:7] Macro(“SIP/sip_os-000016b9”, “get-vmcontext,155”) in new stack
pbx.c: Executing [s@macro-get-vmcontext:1] Set(“SIP/sip_os-000016b9”, “VMCONTEXT=default”) in new stack
pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/sip_os-000016b9”, “0?200:300”) in new stack
pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
pbx.c: Executing [s@macro-get-vmcontext:300] NoOp(“SIP/sip_os-000016b9”, “”) in new stack
res_agi.c: <SIP/167-000016be>AGI Script attendedtransfer-rec-restart.php completed, returning 0
pbx.c: Executing [s@macro-hangupcall:6] Hangup(“SIP/167-000016be”, “”) in new stack
pbx.c: Executing [vmx@macro-vm:8] Set(“SIP/sip_os-000016b9”, “MODE=unavail”) in new stack
app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/167-000016be’ in macro ‘hangupcall’
pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/167-000016be’
pbx.c: Executing [vmx@macro-vm:9] NoOp(“SIP/sip_os-000016b9”, “MODE IS: unavail”) in new stack
pbx.c: Executing [vmx@macro-vm:10] GotoIf(“SIP/sip_os-000016b9”, “1?chknomsg”) in new stack
pbx_builtins.c: Goto (macro-vm,vmx,12)
pbx.c: Executing [vmx@macro-vm:12] GotoIf(“SIP/sip_os-000016b9”, “0?s-DIRECTDIAL,1”) in new stack
pbx.c: Executing [vmx@macro-vm:13] GotoIf(“SIP/sip_os-000016b9”, “0?notdirect”) in new stack
pbx.c: Executing [vmx@macro-vm:14] Set(“SIP/sip_os-000016b9”, “MODE=unavail”) in new stack
pbx.c: Executing [vmx@macro-vm:15] NoOp(“SIP/sip_os-000016b9”, "Checking if ext 155 is enabled: ") in new stack
pbx.c: Executing [vmx@macro-vm:16] GotoIf(“SIP/sip_os-000016b9”, “1?s-DIRECTDIAL,1”) in new stack
pbx_builtins.c: Goto (macro-vm,s-DIRECTDIAL,1)
pbx.c: Executing [s-DIRECTDIAL@macro-vm:1] NoOp(“SIP/sip_os-000016b9”, “DIRECTDIAL voicemail”) in new stack
pbx.c: Executing [s-DIRECTDIAL@macro-vm:2] Macro(“SIP/sip_os-000016b9”, “get-vmcontext,155”) in new stack
pbx.c: Executing [s@macro-get-vmcontext:1] Set(“SIP/sip_os-000016b9”, “VMCONTEXT=default”) in new stack
pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/sip_os-000016b9”, “0?200:300”) in new stack
pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
pbx.c: Executing [s@macro-get-vmcontext:300] NoOp(“SIP/sip_os-000016b9”, “”) in new stack
pbx.c: Executing [s-DIRECTDIAL@macro-vm:3] VoiceMail(“SIP/sip_os-000016b9”, “155@default,u”) in new stack
file.c: <SIP/sip_os-000016b9> Playing ‘/var/spool/asterisk/voicemail/default/155/unavail.slin’ (language ‘en’)
file.c: <SIP/sip_os-000016b9> Playing ‘vm-intro.alaw’ (language ‘en’)
file.c: <SIP/sip_os-000016b9> Playing ‘beep.alaw’ (language ‘en’)
app_voicemail.c: Recording the message
app.c: x=0, open writing: /var/spool/asterisk/voicemail/default/155/tmp/eBJj4F format: wav, 0x7f4ed80a0920
app.c: User hung up
app_macro.c: Spawn extension (macro-vm, s-DIRECTDIAL, 3) exited non-zero on ‘SIP/sip_os-000016b9’ in macro ‘vm’
app_macro.c: Spawn extension (macro-exten-vm, s, 6) exited non-zero on ‘SIP/sip_os-000016b9’ in macro ‘exten-vm’
pbx.c: Spawn extension (ext-local, 155, 2) exited non-zero on ‘SIP/sip_os-000016b9’
pbx.c: Executing [h@ext-local:1] Macro(“SIP/sip_os-000016b9”, “hangupcall,”) in new stack
pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/sip_os-000016b9”, “1?theend”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,3)
pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/sip_os-000016b9”, “0?Set(CDR(recordingfile)=)”) in new stack
pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/sip_os-000016b9”, " monior file= ") in new stack
pbx.c: Executing [s@macro-hangupcall:5] AGI(“SIP/sip_os-000016b9”, “attendedtransfer-rec-restart.php,”) in new stack
res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
res_agi.c: <SIP/sip_os-000016b9>AGI Script attendedtransfer-rec-restart.php completed, returning 0
pbx.c: Executing [s@macro-hangupcall:6] Hangup(“SIP/sip_os-000016b9”, “”) in new stack
app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/sip_os-000016b9’ in macro ‘hangupcall’
pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/sip_os-000016b9’
app_stack.c: SIP/sip_os-000016b9 Internal Gosub(crm-hangup,s,1) start
pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/sip_os-000016b9”, “Sending Hangup to CRM”) in new stack
pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/sip_os-000016b9”, “HANGUP CAUSE: 16”) in new stack
pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/sip_os-000016b9”, “1?Set(__CRM_VOICEMAIL=SUCCESS)”) in new stack
pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/sip_os-000016b9”, “MASTER CHANNEL: 1545042915.5925 = 1545042915.5925”) in new stack
pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/sip_os-000016b9”, “0?return”) in new stack
pbx.c: Executing [s@crm-hangup:6] Set(“SIP/sip_os-000016b9”, “__CRM_HANGUP=1”) in new stack
pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/sip_os-000016b9”, “sangomacrm.agi”) in new stack
res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
res_agi.c: <SIP/sip_os-000016b9>AGI Script sangomacrm.agi completed, returning 0
pbx.c: Executing [s@crm-hangup:8] Return(“SIP/sip_os-000016b9”, “”) in new stack
app_stack.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/sip_os-000016b9’
app_stack.c: SIP/sip_os-000016b9 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

Above is the “snippet” from the full log file. I have put in anything from when user1 attempts the transfer to user 2.

Any help would be appreciated. Thanks

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@wzkds wrote:

Using FreePBX 13/Asterisk 13

At issue is getting voicemail transcription into emails using the IBM Watson speech to text engine utilizing the apikey. I’ve noted that @kristiandg had the same question, but that thread is closed.

I’ve had the IBM Watson script working in the past when watson allowed the creation of a username and password following a freepbx blog post.

I’m trying to get the apikey method working using the script that Ward Mundy adjusted for the change by ibm, but it is not working.

Do I need to alter extensions_custom.conf as noted by Jerson? I don’t believe I’ve had to do this in the past, and the nerdvittles post with the altered ascript using apikey doesn’t mention this.

Has anyone had any luck getting the new script working? The only other thought I had is that ward’s script perhaps requires sendmail, whereas my server has postfix smtp server installed.

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@echo501 wrote:

Hello,

I have an issues where the inbound call is dropped after the caller chooses one specific extension. All my other extensions are operating normally meaning they receive inbound calls fine.

Historically we have numbered our extensions 200-299. However this phone is in a common area and it was requested that we give it an extension 300. I set up the phone and extension just like the others but now the calls drop as soon as the extension 300 is dialed.

The following is a snippet from the asterisk output upon receiving the call.

-- Goto (maingreeting,7036513067,1)
    -- Executing [7036513067@maingreeting:1] Answer("SIP/vitelity-sbc-in-00000126", "") in new stack
    -- Executing [7036513067@maingreeting:2] Ringing("SIP/vitelity-sbc-in-00000126", "") in new stack
    -- Executing [7036513067@maingreeting:3] Wait("SIP/vitelity-sbc-in-00000126", "1") in new stack
       > 0x7fdb9c2087c0 -- Probation passed - setting RTP source address to 64.2.142.239:17076
    -- Executing [7036513067@maingreeting:4] BackGround("SIP/vitelity-sbc-in-00000126", "custom/4dv-main") in new stack
    -- <SIP/vitelity-sbc-in-00000126> Playing 'custom/4dv-main.slin' (language 'en')
[2018-12-17 07:52:14] WARNING[29142][C-000000b2]: pbx.c:6846 __ast_pbx_run: Invalid extension '30', but no rule 'i' or 'e' in context 'maingreeting'

The item that stands out is the invalid extension 30. I don’t have 2 digit extension numbers never have. I have reviewed the extension config in FreePBX and there is no 30 only 300.

Why is the last zero being dropped? Is there a config that limits the number of extensions? If I recreate the extension with a 2xx number I’m sure it’ll work. But I want to know why the 3xx extension number isn’t working.

Any help is most appreciated!!

Thanks
Kenny

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@jamesnurse wrote:

Hi all,
Hopefully someone can just shed a bit of light on what these emails I have been getting mean. Mainly if I need to be concerned and if not how do I stop them being sent? They are sending about once an hour at the moment.

The email is coming from Cron, asterisk in regards to /usr/sbin/fwconsole userman --syncall -q
The contents of the message is just ‘Fork 1’ continuously
I’ve attached a screenshot the message that gets sent on repeat.

Asterisk email.png610x671 5.05 KB

Cheers

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@ghurty wrote:

Is there a way to hook up an Analog Valcom system to Asterisk? I tried to use an FXS adapter but a busy signal was coming over the speakers.

Thank you

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@wassy83 wrote:

Hi to all,
I need some help to fix this for my company…
FIRST AF ALL:
I have the latest freepbx updated and fully working in a production call center environment with around 60 extension. In a few words, when a call comes from outside it will be routed to a simple ring group with around 40 of 60 extensions then, after 15 seconds, a queue will play a message to the client and will ring on all the 40 exentensions again.
NOW THE PROBLEM
we use a really customized crm (VTECRM from vtevillage) this is a port of vtiger… I have passed a full permission credential to access my AMI interface to the crm’s developer but the developper tells me that the AMI interface is not always reporting to him who answered the call but only the group ID, this varied if the call was answered directly by the incoming caller group, by a blind transfered call or attended.

with this problem what happens is that we see a lot of incoming call popups in our CRM instead of viewing only the popup referred to the answered call.

any Idea or a starting point to fix this?
many many thanks

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@PitzKey wrote:

Hello all,

Disclaimer; I’m new to the calendar module.

We want to have followme ring an external number only after hours, so we created a local calendar with a single event that has the business hours which is reoccurring daily, we enabled followme, linked that local calendar and set Calendar Match Inverse to “No”.
It works fine.

But we want to allow that user during business hours to enable followme (Previously they would dial *21EXT.)

But apparently when using the calendar feature, followme is always enabled and if you call the feature code it’ll disable followme completely.

Any ideas?

Thanks

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@JasperE wrote:

In my organization I have some extensions which can be called directly by external parties (customers/suppliers/etc).

For these extensions I have set up an inbound route with a configured DID number and Extension destination.

When the inbound call from an external party remains unanswered at the configured extension, I’d like to have the call forwarded to an operator queue.

That is; the call should only be forwarded to the operator queue if an external party is calling using a DID inbound route. Internal extension->extension calls should remain unanswered or go to voicemail.

This way I aim to reduce workload for the operator while maximizing human response factor for whatever number is called.

Is this possible with FreePBX?

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@ccts wrote:

Hi,

I’ve got a fpbx 13 box hosted in the cloud and I’ve got about 4-5 clients running off the same server. All the client devices are connecting fine, except for one client. From the obihai phone it says: "Register Failed: No Response From Server (server=x.x.x.x:5060; retrying).

I’ve tried:
-checking responsive firewall and don’t see any client’s blocked there
-stopping responsive firewall from the gui, then seeing if client will register-no
-restarting firewall from cli using “fwconsole firewall stop
fwconsole firewall start”
-put in the client wan ip as filter in the log tab on the gui, but it returned nothing (in the last 500 rows at least).
-added the client wan ip to trusted sites on RF

I’m not sure where or what is blocking it, but I’m stuck…Please help.

I suspect perhaps they are running a dynamic public ip, not static & perhaps their wan ip changed and their extensions were already registered as the original ip, so when the ip changed, they were blocked, but I’m not 100% sure, not sure where to check this out.

All other end points at other sites are working fine, just this one site isn’t.

Any advise?

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@sentinelace wrote:

I got a Yealink cp860 that I need to put in recovery mode. It says online you can use usb or tftp. TFTP server doesn’t ever see it connect. I can do any other model with my TFTP but this one. It just says start updating and hangs

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@hardocp wrote:

I am currently running FPBX 14 and Asterisk 16 on a cloud VM

Would like to provide my users access over HTTPS to their VMs using UCP – however most of the solutions that are noted on the board refer to using the Sys Admin module which i do not have

How can this be done without that module?

thanks,

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