@Mark_Twain007 wrote:

Hello, I’ve got an issue with out FreePBX setup that unfortunately I inherited with no training or documentation so I am hoping someone can help and I apologize if I have to ask for help with seemingly simple things like Log collection. I’ve never used FreePBX before, and I’m also pretty bad with command line Linux.

So we have an issue where some of our users calls drop after 30 seconds, or the calls drop if they put some on hold. The users with these issues are our remote uses that use a VOIP Softphone (Bria 5 from Counterpath) to use the system. All these users are remote. Inside our 4 walls everything works normally. Additionally if they are receiving the call, then the call works fine, except if they put the person on hold, then it drops.

All these users are remote, and they aren’t using VPN, and they are reporting that they never used the VPN in the past to get this to work.

I have seen some answers on the community about UDP timeout of the firewall, so I changed that and it did nothing. I’ve also seen some posts about proper port forwarding on the NAT being the issue but I am a little confused about where to where the port forward should be. Am I looking at the Free PBX box directly or at the Session Boarder Controller? I’m not sure the path in and out of the network the traffic should be traveling so I’m not sure where to look.

Any help would be appreciated. Thank you.

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@M_DigHip wrote:

Hello, thank you for reading my post!

Inherited a FreePBX system setup at a new client but I’m not at all familiar with it.

I am trying to figure out how to change the user’s Display Name that shows up on the phone (at idle). Model is Aastra 6757i.

I’ve logged into the GUI and updated the name under Applications, Extensions. Applied config and rebooted the phone, but the Display Name will not change.

Any help is appreciated!

Thank you

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@dtobal wrote:

Hi friends!

I am trying to deploy a setup where my extensions need to have a child TLS Certificate. If it does not have, it wont connect or can not do any call.

With Lets Encrypt, I didnt figure out how to generate child certificates. I have a wildcard valid, and I did import successfully to my freepbx. I am using it with WebRTC. Now, I dont know how to create, and how I can revoke these child certificates. Is it possible this cenario with Freepbx? Shell?

Thanks,
Denilson

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@fropa wrote:

Hi, I have a scheme like this: trunk -> inbound route -> Queue -> Extensions as operators. I use feature codes *72 to forward calls to external numbers. Everything works fine, but when I check CDR Records, there are three records.

Src (external) number - Dst Queue number.
Src (external) number - Dst External number where I forward call.
Src (external) number - Dst Extension number.

And I need to have a call record for all of them, but there is no record for the second CDR where a destination is an external number. I know that they are the same, but I need it for some reason. I turned on call recording in extension, inbound, outbound, queue menus. Also, I tried the “call recording” menu with a destination of that queue, but that did not help.

Can I have this call recording somehow?

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@GeekBoy wrote:

I have this FreePBX 15 sever to play around with. I have not touched it in over a month, and a visit the web interface today and notice the SSL certificate is not valid.

Checking Certificate Manager, I see the certificate expired back on January 6. I try to manually renew it, but fails.

It presents…

/.freepbx-known/3822eb3393279885ef039392cc669e39’ - 404 Not Found Not Found The requested URL /.freepbx-known/3822eb3393279885ef039392cc669e39 was not found on this server.

Looking in /var/www/html/.freepbx-known/ and sure enough not there.

I delete the expire certificated and try again, and getting similar error.

In addition, I also tried some systems updates, and reboot, in addition to fwconsole chown and still same.

It currently has certificate manager module version 15.0.37 on it.

I am just know nothing about what is going on behind the scenes of that .freepbx-known folder, and see nothing online. Delete everything in there?

Anyone know what’s going on?

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@DirectRaw wrote:

Hello guys! I have problem with FreePBX.

PBX Version:
15.0.16.81
PBX Distro:
12.7.8-2011-5.sng7
Asterisk Version:
16.13.0

Hello guys! I have problem with FreePBX.
I can’t make forward call on external number but i can tmake forward on internal phones.

== Setting global variable ‘SIPDOMAIN’ to ‘192.168.201.218’
– Executing [790512@from-pstn:1] Set(“PJSIP/sip_trunk_1-00000052”, “__DIRECTION=INBOUND”) in new stack
– Executing [790512@from-pstn:2] Gosub(“PJSIP/sip_trunk_1-00000052”, “sub-record-check,s,1(in,790512,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/sip_trunk_1-00000052”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/sip_trunk_1-00000052”, “NOW=1611829659”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/sip_trunk_1-00000052”, “__DAY=28”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/sip_trunk_1-00000052”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/sip_trunk_1-00000052”, “__YEAR=2021”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/sip_trunk_1-00000052”, “__TIMESTR=20210128-102739”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/sip_trunk_1-00000052”, “__FROMEXTEN=unknown”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/sip_trunk_1-00000052”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/sip_trunk_1-00000052”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/sip_trunk_1-00000052”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/sip_trunk_1-00000052”, “2?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?sub-record-check,in,1”) in new stack
– Goto (sub-record-check,in,1)
– Executing [in@sub-record-check:1] NoOp(“PJSIP/sip_trunk_1-00000052”, “Inbound Recording Check to 790512”) in new stack
– Executing [in@sub-record-check:2] Set(“PJSIP/sip_trunk_1-00000052”, “FROMEXTEN=unknown”) in new stack
– Executing [in@sub-record-check:3] ExecIf(“PJSIP/sip_trunk_1-00000052”, “11?Set(FROMEXTEN=87057452722)”) in new stack
– Executing [in@sub-record-check:4] Gosub(“PJSIP/sip_trunk_1-00000052”, “recordcheck,1(dontcare,in,790512)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/sip_trunk_1-00000052”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/sip_trunk_1-00000052”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [in@sub-record-check:5] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [790512@from-pstn:3] Set(“PJSIP/sip_trunk_1-00000052”, “CHANNEL(tonezone)=us”) in new stack
– Executing [790512@from-pstn:4] Set(“PJSIP/sip_trunk_1-00000052”, “__FROM_DID=790512”) in new stack
– Executing [790512@from-pstn:5] Set(“PJSIP/sip_trunk_1-00000052”, “returnhere=1”) in new stack
– Executing [790512@from-pstn:6] Gosub(“PJSIP/sip_trunk_1-00000052”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“PJSIP/sip_trunk_1-00000052”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [790512@from-pstn:7] Set(“PJSIP/sip_trunk_1-00000052”, “CDR(did)=790512”) in new stack
– Executing [790512@from-pstn:8] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?”) in new stack
– Executing [790512@from-pstn:9] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0 ?Set(CALLERID(name)=87057452722)”) in new stack
– Executing [790512@from-pstn:10] Set(“PJSIP/sip_trunk_1-00000052”, “__MOHCLASS=”) in new stack
– Executing [790512@from-pstn:11] Set(“PJSIP/sip_trunk_1-00000052”, “__REVERSAL_REJECT=FALSE”) in new stack
– Executing [790512@from-pstn:12] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?post-reverse-charge”) in new stack
– Goto (from-pstn,790512,14)
– Executing [790512@from-pstn:14] NoOp(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [790512@from-pstn:15] Set(“PJSIP/sip_trunk_1-00000052”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
– Executing [790512@from-pstn:16] Set(“PJSIP/sip_trunk_1-00000052”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
– Executing [790512@from-pstn:17] Set(“PJSIP/sip_trunk_1-00000052”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
– Executing [790512@from-pstn:18] Set(“PJSIP/sip_trunk_1-00000052”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
– Executing [790512@from-pstn:19] NoOp(“PJSIP/sip_trunk_1-00000052”, “CallerID Entry Point”) in new stack
– Executing [790512@from-pstn:20] Set(“PJSIP/sip_trunk_1-00000052”, “__CRM_DIRECTION=INBOUND”) in new stack
– Executing [790512@from-pstn:21] Set(“PJSIP/sip_trunk_1-00000052”, “__CRM_SOURCE=87057452722”) in new stack
– Executing [790512@from-pstn:22] Set(“PJSIP/sip_trunk_1-00000052”, “__CRM_LINKEDID=1611829659.92”) in new stack
– Executing [790512@from-pstn:23] AGI(“PJSIP/sip_trunk_1-00000052”, “agi://127.0.0.1/sangomacrm.agi,true”) in new stack
– <PJSIP/sip_trunk_1-00000052>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [790512@from-pstn:24] ExecIf(“PJSIP/sip_trunk_1-00000052”, “1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [790512@from-pstn:25] Goto(“PJSIP/sip_trunk_1-00000052”, “from-did-direct,222,1”) in new stack
– Goto (from-did-direct,222,1)
– Executing [222@from-did-direct:1] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?ext-local,222,1:followme-check,222,1”) in new stack
– Goto (ext-local,222,1)
– Executing [222@ext-local:1] Set(“PJSIP/sip_trunk_1-00000052”, “__RINGTIMER=15”) in new stack
– Executing [222@ext-local:2] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(__CWIGNORE=)”) in new stack
– Executing [222@ext-local:3] Macro(“PJSIP/sip_trunk_1-00000052”, “exten-vm,novm,222,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“PJSIP/sip_trunk_1-00000052”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/sip_trunk_1-00000052”, “TOUCH_MONITOR=1611829659.92”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/sip_trunk_1-00000052”, “AMPUSER=87057452722”) in new stack
– Executing [s@macro-user-callerid:3] Set(“PJSIP/sip_trunk_1-00000052”, “HOTDESCKCHAN=sip_trunk_1-00000052”) in new stack
– Executing [s@macro-user-callerid:4] Set(“PJSIP/sip_trunk_1-00000052”, “HOTDESKEXTEN=sip_trunk_1”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/sip_trunk_1-00000052”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-user-callerid:6] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-user-callerid:7] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“PJSIP/sip_trunk_1-00000052”, “1?Set(REALCALLERIDNUM=87057452722)”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/sip_trunk_1-00000052”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/sip_trunk_1-00000052”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?report”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [s@macro-user-callerid:23] NoOp(“PJSIP/sip_trunk_1-00000052”, “Macro Depth is 2”) in new stack
– Executing [s@macro-user-callerid:24] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,25)
– Executing [s@macro-user-callerid:25] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:26] ExecIf(“PJSIP/sip_trunk_1-00000052”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
– Executing [s@macro-user-callerid:27] Set(“PJSIP/sip_trunk_1-00000052”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:28] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,44)
– Executing [s@macro-user-callerid:44] Set(“PJSIP/sip_trunk_1-00000052”, “CALLERID(number)=87057452722”) in new stack
– Executing [s@macro-user-callerid:45] Set(“PJSIP/sip_trunk_1-00000052”, “CALLERID(name)=87057452722”) in new stack
– Executing [s@macro-user-callerid:46] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:47] Set(“PJSIP/sip_trunk_1-00000052”, “CDR(cnam)=87057452722”) in new stack
– Executing [s@macro-user-callerid:48] Set(“PJSIP/sip_trunk_1-00000052”, “CDR(cnum)=87057452722”) in new stack
– Executing [s@macro-user-callerid:49] Set(“PJSIP/sip_trunk_1-00000052”, “CHANNEL(language)=ru”) in new stack
– Executing [s@macro-exten-vm:2] Set(“PJSIP/sip_trunk_1-00000052”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“PJSIP/sip_trunk_1-00000052”, “__EXTTOCALL=222”) in new stack
– Executing [s@macro-exten-vm:4] Set(“PJSIP/sip_trunk_1-00000052”, “__PICKUPMARK=222”) in new stack
– Executing [s@macro-exten-vm:5] Set(“PJSIP/sip_trunk_1-00000052”, “RT=”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:6] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Macro(vm,novm,DIRECTDIAL,)”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:7] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?MacroExit()”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:8] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Gosub(ext-intercom,*80222,1())”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:9] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?MacroExit()”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:10] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?ChanSpy(PJSIP/222,q)”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:11] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?MacroExit()”) in new stack
[2021-01-28 10:27:40] WARNING[157092][C-00000030]: chan_sip.c:23279 func_header_read: This function can only be used on SIP channels.
– Executing [s@macro-exten-vm:12] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Macro(vm,novm,DIRECTDIAL,)”) in new stack
– Executing [s@macro-exten-vm:13] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:14] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Gosub(ext-intercom,*80222,1())”) in new stack
– Executing [s@macro-exten-vm:15] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:16] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?ChanSpy(PJSIP/222,q)”) in new stack
– Executing [s@macro-exten-vm:17] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:18] Gosub(“PJSIP/sip_trunk_1-00000052”, “sub-record-check,s,1(exten,222,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/sip_trunk_1-00000052”, “11?initialized”) in new stack
– Goto (sub-record-check,s,10)
– Executing [s@sub-record-check:10] NoOp(“PJSIP/sip_trunk_1-00000052”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/sip_trunk_1-00000052”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/sip_trunk_1-00000052”, “5?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?sub-record-check,exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] NoOp(“PJSIP/sip_trunk_1-00000052”, “Exten Recording Check between 87057452722 and 222”) in new stack
– Executing [exten@sub-record-check:2] Set(“PJSIP/sip_trunk_1-00000052”, “CALLTYPE=external”) in new stack
– Executing [exten@sub-record-check:3] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(CALLTYPE=)”) in new stack
– Executing [exten@sub-record-check:4] Set(“PJSIP/sip_trunk_1-00000052”, “CALLEE=dontcare”) in new stack
– Executing [exten@sub-record-check:5] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(CALLEE=dontcare)”) in new stack
– Executing [exten@sub-record-check:6] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?callee”) in new stack
– Goto (sub-record-check,exten,11)
– Executing [exten@sub-record-check:11] Gosub(“PJSIP/sip_trunk_1-00000052”, “recordcheck,1(dontcare,external,222)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/sip_trunk_1-00000052”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/sip_trunk_1-00000052”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [exten@sub-record-check:12] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [s@macro-exten-vm:19] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,25)
– Executing [s@macro-exten-vm:25] GosubIf(“PJSIP/sip_trunk_1-00000052”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:26] Macro(“PJSIP/sip_trunk_1-00000052”, “dial-one,HhTtr,222”) in new stack
– Executing [s@macro-dial-one:1] Set(“PJSIP/sip_trunk_1-00000052”, “DEXTEN=222”) in new stack
– Executing [s@macro-dial-one:2] Set(“PJSIP/sip_trunk_1-00000052”, “__CRM_SOURCE=87057452722”) in new stack
– Executing [s@macro-dial-one:3] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(__EXTTOCALL=222)”) in new stack
– Executing [s@macro-dial-one:4] Set(“PJSIP/sip_trunk_1-00000052”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:5] GosubIf(“PJSIP/sip_trunk_1-00000052”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:6] GosubIf(“PJSIP/sip_trunk_1-00000052”, “1?cf,1()”) in new stack
– Executing [cf@macro-dial-one:1] Set(“PJSIP/sip_trunk_1-00000052”, “CFAMPUSER=87057452722”) in new stack
– Executing [cf@macro-dial-one:2] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Return()”) in new stack
– Executing [cf@macro-dial-one:3] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ARG1=0)”) in new stack
– Executing [cf@macro-dial-one:4] ExecIf(“PJSIP/sip_trunk_1-00000052”, “1?Set(ARG1=)”) in new stack
– Executing [cf@macro-dial-one:5] Set(“PJSIP/sip_trunk_1-00000052”, “DEXTEN=9999972#”) in new stack
– Executing [cf@macro-dial-one:6] Set(“PJSIP/sip_trunk_1-00000052”, “__DIVERSION_REASON=unconditional”) in new stack
– Executing [cf@macro-dial-one:7] ExecIf(“PJSIP/sip_trunk_1-00000052”, “1?Return()”) in new stack
– Executing [s@macro-dial-one:7] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,10)
– Executing [s@macro-dial-one:10] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?continue”) in new stack
– Goto (macro-dial-one,s,27)
– Executing [s@macro-dial-one:27] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GosubIf(“PJSIP/sip_trunk_1-00000052”, “0?dstring,1():dlocal,1()”) in new stack
– Executing [dlocal@macro-dial-one:1] Set(“PJSIP/sip_trunk_1-00000052”, “DSTRING=9999972”) in new stack
– Executing [dlocal@macro-dial-one:2] Set(“PJSIP/sip_trunk_1-00000052”, “USEGOTO=1”) in new stack
– Executing [dlocal@macro-dial-one:3] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [s@macro-dial-one:29] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:30] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?skiptrace”) in new stack
– Goto (macro-dial-one,s,32)
– Executing [s@macro-dial-one:32] Set(“PJSIP/sip_trunk_1-00000052”, “D_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dial-one:33] GosubIf(“PJSIP/sip_trunk_1-00000052”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:34] NoOp(“PJSIP/sip_trunk_1-00000052”, "Blind Transfer: , Attended Transfer: , User: , Alert Info: ") in new stack
– Executing [s@macro-dial-one:35] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:36] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:37] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:38] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:39] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:40] GosubIf(“PJSIP/sip_trunk_1-00000052”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:41] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:42] GosubIf(“PJSIP/sip_trunk_1-00000052”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:43] Set(“PJSIP/sip_trunk_1-00000052”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:44] Set(“PJSIP/sip_trunk_1-00000052”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:45] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?usegoto,1”) in new stack
– Goto (macro-dial-one,usegoto,1)
– Executing [usegoto@macro-dial-one:1] Set(“PJSIP/sip_trunk_1-00000052”, “USEGOTO=”) in new stack
– Executing [usegoto@macro-dial-one:2] Goto(“PJSIP/sip_trunk_1-00000052”, “from-internal,9999972,1”) in new stack
– Goto (from-internal,9999972,1)
– Executing [9999972@from-internal:1] Macro(“PJSIP/sip_trunk_1-00000052”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/sip_trunk_1-00000052”, “TOUCH_MONITOR=1611829659.92”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/sip_trunk_1-00000052”, “AMPUSER=87057452722”) in new stack
– Executing [s@macro-user-callerid:3] Set(“PJSIP/sip_trunk_1-00000052”, “HOTDESCKCHAN=sip_trunk_1-00000052”) in new stack
– Executing [s@macro-user-callerid:4] Set(“PJSIP/sip_trunk_1-00000052”, “HOTDESKEXTEN=sip_trunk_1”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/sip_trunk_1-00000052”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-user-callerid:6] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-user-callerid:7] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(REALCALLERIDNUM=87057452722)”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/sip_trunk_1-00000052”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/sip_trunk_1-00000052”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?report”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [s@macro-user-callerid:23] NoOp(“PJSIP/sip_trunk_1-00000052”, “Macro Depth is 3”) in new stack
– Executing [s@macro-user-callerid:24] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,25)
– Executing [s@macro-user-callerid:25] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,44)
– Executing [s@macro-user-callerid:44] Set(“PJSIP/sip_trunk_1-00000052”, “CALLERID(number)=87057452722”) in new stack
– Executing [s@macro-user-callerid:45] Set(“PJSIP/sip_trunk_1-00000052”, “CALLERID(name)=87057452722”) in new stack
– Executing [s@macro-user-callerid:46] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:47] Set(“PJSIP/sip_trunk_1-00000052”, “CDR(cnam)=87057452722”) in new stack
– Executing [s@macro-user-callerid:48] Set(“PJSIP/sip_trunk_1-00000052”, “CDR(cnum)=87057452722”) in new stack
– Executing [s@macro-user-callerid:49] Set(“PJSIP/sip_trunk_1-00000052”, “CHANNEL(language)=ru”) in new stack
– Executing [9999972@from-internal:2] NoCDR(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [9999972@from-internal:3] Progress(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
– Executing [9999972@from-internal:4] Wait(“PJSIP/sip_trunk_1-00000052”, “1”) in new stack
– Executing [9999972@from-internal:5] Playback(“PJSIP/sip_trunk_1-00000052”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <PJSIP/sip_trunk_1-00000052> Playing ‘silence/1.ulaw’ (language ‘ru’)
– <PJSIP/sip_trunk_1-00000052> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘ru’)
– <PJSIP/sip_trunk_1-00000052> Playing ‘check-number-dial-again.ulaw’ (language ‘ru’)
== Channel ‘PJSIP/sip_trunk_1-00000052’ jumping out of macro ‘exten-vm’
– Executing [h@from-internal:1] Macro(“PJSIP/sip_trunk_1-00000052”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/sip_trunk_1-00000052”, " montior file= ") in new stack
– Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/sip_trunk_1-00000052”, “1?skipagi”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] Hangup(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/sip_trunk_1-00000052’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/sip_trunk_1-00000052’
– PJSIP/sip_trunk_1-00000052 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/sip_trunk_1-00000052”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/sip_trunk_1-00000052”, “HANGUP CAUSE: 16”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/sip_trunk_1-00000052”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/sip_trunk_1-00000052”, “MASTER CHANNEL: 1611829659.92 = 1611829659.92”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/sip_trunk_1-00000052”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/sip_trunk_1-00000052”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/sip_trunk_1-00000052”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
– <PJSIP/sip_trunk_1-00000052>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/sip_trunk_1-00000052”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/sip_trunk_1-00000052’
– PJSIP/sip_trunk_1-00000052 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

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@Linux_01 wrote:

How to Download CDR Report using Command in FreePBX v16

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@cyberco wrote:

Customer has raised the question again about connecting FreePBX to MSTeams

We can do it using followme and teams direct calling and that allow some sort of blended solution with Desk and Teams getting calls ok.

We have noticed a product called call2teams that does list asterisk as It seems to be just a SBC alloing you to connect sip extensions to it. This seems to fit the bill exactly as will allow call recording etc.

So my question is has anyone got call2teams working live in the field and is it any good? and if they have do they allow connection to 3rd party PBXs ?

Ian

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@benkohl wrote:

I’ve the exact same problem with the same config like in the thread named “Outboud Call drops voice after 333 seconds” (I’m not allowed to post a link or reopen this thread)

Calls with fritzbox as trunk have no audio after 5 and half minutes and drops after that on external side with a delay of a half minute. The internal side doesn’t drop the call.

The pjsip debug log doesn’t show anything related to the call (only when I start and hang up the call on the internal side).

I tried many of setting changes like “Rewrite Contact” or other rtp timeouts.

I also have OPNSense between fritzbox and FreePBX, but it only acts as an firewall and for test purposes it does not block anything.

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@robdgem wrote:

Hi All

I have a strange situation with my queues. I initially thought the whole PBX was going offline but its actually limited to queues. It appears that calls come through, the usual announcement “welcome to our business, please hold” plays perfectly and then silence, no on hold music and queue agents don’t receive calls but they are piling up on the system. This generally only happens during the morning and never after hours. I am completely stumped.
External calls with inbound routes to extensions work fine, its only if they go to a queue.

Does anyone have any suggestions?

FreePBX has always been one of those products rock solid products but i seem to have done something to mess it up

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@Paul_N wrote:

Hi,

My server has been up and running for the past year without any issues and it is configured to use both IPv4 and IPv6 but I’m struggling to set up several trunks over IPv6 and hoping someone here will help point me in the right direction…

First, a little info about my server:

• It’s a cloud-based VPS running Debian with the following FreePBX/Asterisk versions installed:

• • FreePBX 15.0.17.12
• • Asterisk Version 16.2.1

We use CHAN_PJSIP and we have the following extensions setup:

200, 201, 203, 204, 205, 206, 207, 208, 209, 210

Every extension except 203 utilises TLS over port 5061 with call encryption enabled. Extension 203 connects via IPv6 and utilises UDP over port 5060. Extensions work just fine, no issues there.

We have five trunks all setup, working and they register over IPv4, UDP and use port 5060.

Our pjsip.transports_custom file contains the following:

[ipv6-udp]
type=transport
protocol=udp
bind=[::]:5060
allow_reload=no
tos=cs3
cos=3

[ipv6-tls]
type=transport
protocol=tls
bind=[::]:5061
ca_list_file=/etc/ssl/certs/ca-certificates.crt
cert_file=/etc/asterisk/keys/domain.pem
priv_key_file=/etc/asterisk/keys/domain.key
method=tlsv1_2
verify_client=no
verify_server=yes
allow_reload=no
tos=cs3
cos=3

Our pjsip.endpoint_custom_post file contains the following:

[203](+)
transport=ipv6-udp

If we change the pjsip.endpoint_custom_post file to the following:

[203](+)
transport=ipv6-udp

[TRUNKNAME](+)
transport=ipv6-udp

The trunk TRUNKNAME fails to register. There is nothing in the log, it’s as if it’s not trying to register at all.

If we ping the SIP Server from our FreePBX server, we get a successful IPv6 result:
PING domain(domain (2001:8b0:0:30::5060:1)) 56 data bytes

Our pjsip show transports result is:

Transport: 0.0.0.0-tls tls 3 96 0.0.0.0:5061
Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5060
Transport: ipv6-tls tls 3 96 [::]:5061
Transport: ipv6-udp udp 3 96 [::]:5060

Extension 203 works just fine over IPv6 but no trunk will register if configured to use IPv6. I will try and grab some more data using tcpdump and pjsip set logger on but so far, am I missing something obvious?

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@dwsiemens wrote:

in he certificate manager, every day I have to add my certificates back. I have no idea why but Certs keep going away as expired every day.

I am on a open source only code base on Rhel 7 so I don’t have system admin.
latest version of freepbx.

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@joshzone90 wrote:

I see that is Asterisk 18.2 is out. Any idea when this will be pushed to asterisk-version-switch or do i need to manually update it?

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@dwsiemens wrote:

I see where in on-touch-recording it logs the filename which generaes a cel event. Is there any events for when a user stops the recording?

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@djdealer wrote:

Hi,

I’ve tried the whole day to get this working.
I have 2 variants that are confusing me. The first works fine.
First:
I (A) initiate an external call to some phone number (B) and then transfering the call to another external number ©.
All is fine C sees the whole number of A.
Second:
B calls A and then A transfer the call to C. C now sees only the extension number of A not the complete phone number.

Neither overriding the CID in the Outbound Route nor trunk work. C everytime see only the extension of A.

Am i missing some setting?

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@AxelLottel wrote:

I am trying to setup Zabbix monitor on my servers.
There are two computers, Zabbix Server and Zabbix Agent.
I add the Zabbix Server’s IP as a Trusted (Exclude from Firewall) in the firewall of the Agent.
I add the Zabbix Agent’s IP as a Trusted (Exclude from Firewall) in the firewall of the Servre.
I also added port 10050 TCP (Zabbix port) as an additional Custom Service for Internet and Local on both machines.
If I turn off the firewall I can talk to the Agent from the server.
If I enable the firewall on either I get a timeout.
I did a packet capture and I see re-transmits with firewall enabled.

Any clues?

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@sorvani wrote:

Brand new, clean install on FreePBXHosting.com
System & Modules updated to current. System rebooted.
Even after being online for an hour, it says Asterisk has been running for 7 seconds.
And the updated time is negative.

Going down. It is counting down from something.

Is there some setting that got set someplace before a timezone was changed or something?

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@GRIFFCOMM wrote:

Hi

How would i customize the dashboard, there are other items i need to see like which extensions are signed in, on what IP addresses, trunk status, conference room use, parking slot use etc…

While here i only seem to have a “modules” menu, is there a way to break these up to items like settings, extensions etc…so i dont have one huge menu with every option in that single menu click?

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@snaggy wrote:

I found that the official firmware for goip -1 (http://dbltek.com/update/GHSFVT-1.1-67-5.pkg) contains a link to a porn site
the link is in the file /usr/etc/syscfg.default (SIP_RC4_KEY=“etoall.net”)
in the web interface you can find this link in the section: Advance VoIP - Signaling Encryption - RC4 - RC4 Encryption Key
I do not know how this link is used by hardware and software
maybe this is just a joke from Chinese manufacturers

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@munozj wrote:

I have my public interface setup in the “internet” mode
I’ve enabled PJSIP in the responsive firewall section since it’s a non standard port and all my remote workers are able to connect using that. and left chan_sip disabled.

One of my Telco’s provides us service over PRI’s but they also have a SIP trunk for backup. I have their IP listed in the firewall in the networks tab setup as “Trusted, excluded from firewall” They use port 5060 to connect.

Under Connectivity - Trunks - SIP Settings - Outgoing
I have the trunk setup as:
host=8.45.245.21
type=peer
qualify=yes

This originally worked but I noticed a few months ago that it’s showing up as UNREACHABLE and calls are not going through the SIP trunk.

When attempting to place a call from that trunk I’m getting this error in the log:

[2021-01-14 16:05:46] WARNING[11211]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 115353446faac8385fe6e59031ed393e@xx.x.xx.xx:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

Calls placed directly into that trunk do not even hit the log.

What settings need to be set on the firewall to allow this connection?

Just making it a trusted network? Just it being listed as a trunk? Do I need to enable chan_sip (5060) on the firewall?

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