@synergy wrote:

Hello,
Found that Call drop every 15 minutes.
It only happens with specific Sip trunk provider.
Check with them why it’s happens and his answer is:
“Please note calls are disconnecting after 15 minutes as your end is not responding to our re-invites. Our switch will send a re-invite to your roughly 15 minutes into the call to prevent fraudulent behavior, if your end fails to respond as you did here, calls will drop”
My asterisk version is - Asterisk 11.6-cert7
And Sip trunk peer details:
type=peer
qualify=no
nat=no
insecure=very
host=X.X.X.X
dtmfmode=auto
context=from-pstn
canreinvite=no

Please advise, need help.

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@comtech wrote:

SNG7, FreePBX14/Asterisk 13

Hello,

We have built out a system to add Asterisk services to our mainly Avaya shop. One of these services is a callback assist.

Call comes in to Asterisk**>Callfile generated based on caller inputs>Asterisk calls back>**Caller presses 1 and is sent to a 10-digit number(go to misc destination), to our Avaya queue.

The trunk between our Avaya Communication Manager (Avaya Switch/PBX) and Asterisk is Session Manager (SIP Router, like Kamailio). This has been working great for over a year.

We upgraded our Avaya Communication and Session Managers last weekend and something strange started happening. After the call routes to the Avaya queue (via misc destination), an agent on an Avaya endpoint picks up.

When the agent on the far (Avaya) end puts the caller on hold, instead of hearing Avaya music on hold, the caller is hearing the Asterisk music on hold.

After confirming this behavior myself, when I look at the full log, I see it happening, the Avaya agent is placing the call on hold on his end, but Asterisk is invoking hold music:
[2018-07-02 07:15:47] VERBOSE[25884][C-00033fbb] res_musiconhold.c: Started music on hold, class ‘default’, on channel ‘SIP/AVAYA-2-0003df3c’
[2018-07-02 07:15:55] VERBOSE[25884][C-00033fbb] res_musiconhold.c: Stopped music on hold on SIP/AVAYA-2-0003df3c

The problem with this is that sometimes the agent comes back (from hold) and there is no one there, the caller is still hearing asterisk hold music.

Questions:

• Any ideas on why asterisk is controlling the caller side when the far side is taking action?

• I have our Avaya guys looking at this, but other than what I shared, I do not see a lot of information on the full log. Is there another log that might be more beneficial as to understanding the exchange that is happening between the systems?

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@VoIPTek wrote:

Hello,

I know at one point with older versions of FreePBX we were able to hotdesk, but am I correct in understanding that today in FreePBX 13/14 we would need to use phoneapps?
Lastly, is there an updated support list for phoneapps https://wiki.freepbx.org/display/FPG/Phone+Apps-Supported+Devices ?

I would prefer to do this without phoneapps, but if that is my only choice, that will have to work.

Laslty, Sangoma phones provide hotdesking right?

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@handleric wrote:

Hello,

I just did a clean install of the latest FreePBX distro (1805) and notices that ever after making changes to the feature codes or disabling any in the GUI they all still respond as if the changes were never made. When I return to Feature Code Admin is shows the info I entered so I know it’s saving, it just isn’t taking effect. has anyone else run into issues like this?

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@etechsol wrote:

Hi I am looking for a simple solution to create a bill at the ned of each month for each user exstention on our freebpx system. I would like to set call costs based on the area code they dial but not include internal calls

Is there a simple module or solution to achieve this does any one recommend
Thanks
Rob

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@Gyomed wrote:

Hi all,

This is my first post on this Wiki (which helped me a lot for other common problems).
I now have a strange issue, and I can’t find the solution.
For one of our customer, we installed a FreePBX with framework version 14.0.3.6.
He’s got 2 differents DID and for this, I created 2 differents inbound routes (I created 2 “Time conditions” depending on the number called…)

So, here is the call flow:

1. 1st DID called --> Inbound route for 1st DID with a Time Condition as Destination
   –> Matched : go to Ring group 301 (with extensions 201 & 202) and if no answer --> go to extension 203 --> if no answer then go to Voicemail of this extension.

This is working perfectly for the 1st DID.

2. 2nd DID called --> Inbound route for 2nd DID with a Time Condition as Destination
   –> Matched : go to Ring group 301 (with same extensions as above : 201 & 202) and if no answer --> go to extension 203 --> if no answer then go to Voicemail of this extension.

I don’t know why, but with the second DID, the call just hang up after around 30 seconds on the last extension 203, without playing the voicemail message…

Remark:
There is also another DID assigned to the extension 203 (last extension with a private number), with another inbound route (call flow : Other DID Called --> Extension 203 --> Voicemail if no answer).
This one is also working perfectly, I’m reaching the extension voicemail if I call this particular DID.

So, I compare 2 calls from the 2 scenario explained above, and here are the logs I could retrieve.
I display only the end of the call flow below -

1. the working one -

…
[2018-07-03 12:17:01] VERBOSE[27313][C-0000024c] app_dial.c: Called SIP/203
[2018-07-03 12:17:01] VERBOSE[27313][C-0000024c] app_dial.c: Connected line update to SIP/3starsnet-00000510 prevented.
[2018-07-03 12:17:01] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Ringing for Notify User 201
[2018-07-03 12:17:01] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Ringing for Notify User 202
[2018-07-03 12:17:01] VERBOSE[27313][C-0000024c] app_dial.c: SIP/203-00000513 is ringing
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] app_dial.c: Nobody picked up in 30000 ms
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] app_stack.c: SIP/203-00000513 Internal Gosub(crm-hangup,s,1) start
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/203-00000513”, “Sending Hangup to CRM”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/203-00000513”, “HANGUP CAUSE: 16”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/203-00000513”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/203-00000513”, “MASTER CHANNEL: 1530613021.1353 = 1530612991.1350”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/203-00000513”, “1?return”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (crm-hangup,s,8)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/203-00000513”, “”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] app_stack.c: Spawn extension (from-internal, 203, 1) exited non-zero on ‘SIP/203-00000513’
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] app_stack.c: SIP/203-00000513 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-dial-one:56] ExecIf(“SIP/3starsnet-00000510”, “0?MacroExit()”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-dial-one:57] ExecIf(“SIP/3starsnet-00000510”, “0?Set(DIALSTATUS=)”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-dial-one:58] GosubIf(“SIP/3starsnet-00000510”, “0?s-NOANSWER,1()”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-dial-one:59] MacroExit(“SIP/3starsnet-00000510”, “”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-exten-vm:27] Set(“SIP/3starsnet-00000510”, “SV_DIALSTATUS=NOANSWER”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-exten-vm:28] GosubIf(“SIP/3starsnet-00000510”, “0?docfu,1()”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-exten-vm:29] GosubIf(“SIP/3starsnet-00000510”, “0?docfb,1()”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-exten-vm:30] Set(“SIP/3starsnet-00000510”, “DIALSTATUS=NOANSWER”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-exten-vm:31] ExecIf(“SIP/3starsnet-00000510”, “1?MacroExit()”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [203@ext-local:3] Set(“SIP/3starsnet-00000510”, “__PICKUPMARK=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [203@ext-local:4] GotoIf(“SIP/3starsnet-00000510”, “1?ext-local,vmu203,1”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (ext-local,vmu203,1)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmu203@ext-local:1] Macro(“SIP/3starsnet-00000510”, “vm,203,NOANSWER,”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-vm:1] Macro(“SIP/3starsnet-00000510”, “user-callerid,SKIPTTL”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/3starsnet-00000510”, “TOUCH_MONITOR=1530612991.1350”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/3starsnet-00000510”, “AMPUSER=0485830996”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/3starsnet-00000510”, “0?report”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/3starsnet-00000510”, “0?Set(REALCALLERIDNUM=)") in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/3starsnet-00000510”, “AMPUSER=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/3starsnet-00000510”, “0?limit”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/3starsnet-00000510”, “AMPUSERCIDNAME=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“SIP/3starsnet-00000510”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“SIP/3starsnet-00000510”, “1?report”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-user-callerid,s,16)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:16] NoOp(“SIP/3starsnet-00000510”, “Macro Depth is 2”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“SIP/3starsnet-00000510”, “1?report2:macroerror”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“SIP/3starsnet-00000510”, “1?continue”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/3starsnet-00000510”, "CALLERID(number)=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/3starsnet-00000510”, “CALLERID(name)=Manneh:") in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“SIP/3starsnet-00000510”, “0?cnum”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/3starsnet-00000510”, "CDR(cnam)=Manneh:”) in new stack
[2018-07-03 12:17:31] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Idle for Notify User 201
[2018-07-03 12:17:31] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Idle for Notify User 202
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/3starsnet-00000510”, “CDR(cnum)=**********”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-user-callerid:42] Set(“SIP/3starsnet-00000510”, “CHANNEL(language)=fr”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-vm:2] Set(“SIP/3starsnet-00000510”, “VMGAIN=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-vm:3] Macro(“SIP/3starsnet-00000510”, “blkvm-check,”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-blkvm-check:1] Set(“SIP/3starsnet-00000510”, “GOSUB_RETVAL=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-blkvm-check:2] ExecIf(“SIP/3starsnet-00000510”, “0?Set(GOSUB_RETVAL=TRUE)”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-blkvm-check:3] MacroExit(“SIP/3starsnet-00000510”, “”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-vm:4] GotoIf(“SIP/3starsnet-00000510”, “1?vmx,1”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-vm,vmx,1)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:1] Set(“SIP/3starsnet-00000510”, “__EXTTOCALL=203”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:2] Set(“SIP/3starsnet-00000510”, “__CRM_VOICEMAIL=203”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:3] Set(“SIP/3starsnet-00000510”, “MEXTEN=203”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:4] Set(“SIP/3starsnet-00000510”, “MMODE=NOANSWER”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:5] Set(“SIP/3starsnet-00000510”, “RETVM=”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:6] Set(“SIP/3starsnet-00000510”, “MODE=unavail”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:7] Macro(“SIP/3starsnet-00000510”, “get-vmcontext,203”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-get-vmcontext:1] Set(“SIP/3starsnet-00000510”, “VMCONTEXT=default”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/3starsnet-00000510”, “0?200:300”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-get-vmcontext:300] NoOp(“SIP/3starsnet-00000510”, “”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:8] Set(“SIP/3starsnet-00000510”, “MODE=unavail”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:9] NoOp(“SIP/3starsnet-00000510”, “MODE IS: unavail”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:10] GotoIf(“SIP/3starsnet-00000510”, “1?chknomsg”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-vm,vmx,12)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:12] GotoIf(“SIP/3starsnet-00000510”, “0?s-NOANSWER,1”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:13] GotoIf(“SIP/3starsnet-00000510”, “1?notdirect”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-vm,vmx,15)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:15] NoOp(“SIP/3starsnet-00000510”, "Checking if ext 203 is enabled: ") in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [vmx@macro-vm:16] GotoIf(“SIP/3starsnet-00000510”, “1?s-NOANSWER,1”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-vm,s-NOANSWER,1)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s-NOANSWER@macro-vm:1] Macro(“SIP/3starsnet-00000510”, “get-vmcontext,203”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-get-vmcontext:1] Set(“SIP/3starsnet-00000510”, “VMCONTEXT=default”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/3starsnet-00000510”, “0?200:300”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-get-vmcontext,s,300)
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-get-vmcontext:300] NoOp(“SIP/3starsnet-00000510”, “”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] pbx.c: Executing [s-NOANSWER@macro-vm:2] VoiceMail(“SIP/3starsnet-00000510”, “203@default,u”) in new stack
[2018-07-03 12:17:31] VERBOSE[27313][C-0000024c] file.c: <SIP/3starsnet-00000510> Playing ‘/var/spool/asterisk/voicemail/default/203/unavail.slin’ (language ‘fr’)
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] app_macro.c: Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on ‘SIP/3starsnet-00000510’ in macro ‘vm’
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Spawn extension (ext-local, vmu203, 1) exited non-zero on ‘SIP/3starsnet-00000510’
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [h@ext-local:1] Macro(“SIP/3starsnet-00000510”, “hangupcall,”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/3starsnet-00000510”, “1?theend”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/3starsnet-00000510”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/3starsnet-00000510”, " monior file= ") in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-hangupcall:5] AGI(“SIP/3starsnet-00000510”, “attendedtransfer-rec-restart.php,”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] res_agi.c: <SIP/3starsnet-00000510>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“SIP/3starsnet-00000510”, “”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/3starsnet-00000510’ in macro ‘hangupcall’
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/3starsnet-00000510’
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] app_stack.c: SIP/3starsnet-00000510 Internal Gosub(crm-hangup,s,1) start
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/3starsnet-00000510”, “Sending Hangup to CRM”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/3starsnet-00000510”, “HANGUP CAUSE: 16”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/3starsnet-00000510”, “1?Set(__CRM_VOICEMAIL=FAILED)”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/3starsnet-00000510”, “MASTER CHANNEL: 1530612991.1350 = 1530612991.1350”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/3starsnet-00000510”, “0?return”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/3starsnet-00000510”, “__CRM_HANGUP=1”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/3starsnet-00000510”, “sangomacrm.agi”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] res_agi.c: <SIP/3starsnet-00000510>AGI Script sangomacrm.agi completed, returning 0
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/3starsnet-00000510”, “”) in new stack
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] app_stack.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/3starsnet-00000510’
[2018-07-03 12:17:41] VERBOSE[27313][C-0000024c] app_stack.c: SIP/3starsnet-00000510 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

…
[2018-07-03 12:18:40] VERBOSE[27628][C-0000024d] app_dial.c: Called SIP/203
[2018-07-03 12:18:40] VERBOSE[27628][C-0000024d] app_dial.c: Connected line update to SIP/3starsnet-00000514 prevented.
[2018-07-03 12:18:40] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Ringing for Notify User 201
[2018-07-03 12:18:40] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Ringing for Notify User 202
[2018-07-03 12:18:40] VERBOSE[27628][C-0000024d] app_dial.c: SIP/203-00000517 is ringing
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] app_stack.c: SIP/203-00000517 Internal Gosub(crm-hangup,s,1) start
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/203-00000517”, “Sending Hangup to CRM”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/203-00000517”, “HANGUP CAUSE: 16”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/203-00000517”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/203-00000517”, “MASTER CHANNEL: 1530613120.1357 = 1530613089.1354”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/203-00000517”, “1?return”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx_builtins.c: Goto (crm-hangup,s,8)
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/203-00000517”, “”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] app_stack.c: Spawn extension (from-internal, 203, 1) exited non-zero on ‘SIP/203-00000517’
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] app_stack.c: SIP/203-00000517 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] app_macro.c: Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘SIP/3starsnet-00000514’ in macro ‘dial-one’
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘SIP/3starsnet-00000514’ in macro ‘exten-vm’
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Spawn extension (ext-local, 203, 2) exited non-zero on ‘SIP/3starsnet-00000514’
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [h@ext-local:1] Macro(“SIP/3starsnet-00000514”, “hangupcall,”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/3starsnet-00000514”, “1?theend”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-07-03 12:19:08] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Idle for Notify User 201
[2018-07-03 12:19:08] VERBOSE[11314] chan_sip.c: Extension Changed 203[ext-local] new state Idle for Notify User 202
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/3starsnet-00000514”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/3starsnet-00000514”, "SIP/203-00000517 monior file= ") in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@macro-hangupcall:5] AGI(“SIP/3starsnet-00000514”, “attendedtransfer-rec-restart.php,SIP/203-00000517,”) in new stack
[2018-07-03 12:19:08] VERBOSE[27628][C-0000024d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] res_agi.c: <SIP/3starsnet-00000514>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“SIP/3starsnet-00000514”, “”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/3starsnet-00000514’ in macro ‘hangupcall’
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/3starsnet-00000514’
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] app_stack.c: SIP/3starsnet-00000514 Internal Gosub(crm-hangup,s,1) start
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/3starsnet-00000514”, “Sending Hangup to CRM”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/3starsnet-00000514”, “HANGUP CAUSE: 16”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/3starsnet-00000514”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/3starsnet-00000514”, “MASTER CHANNEL: 1530613089.1354 = 1530613089.1354”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/3starsnet-00000514”, “0?return”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/3starsnet-00000514”, “__CRM_HANGUP=1”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/3starsnet-00000514”, “sangomacrm.agi”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] res_agi.c: <SIP/3starsnet-00000514>AGI Script sangomacrm.agi completed, returning 0
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/3starsnet-00000514”, “”) in new stack
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] app_stack.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/3starsnet-00000514’
[2018-07-03 12:19:09] VERBOSE[27628][C-0000024d] app_stack.c: SIP/3starsnet-00000514 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
…

Any help about this would be really appreciated.
Have a nice day all !!

Guillaume.

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@wzkds wrote:

FreePBX 13.0.195.4
Asterisk 13.19.1

I’m running into an issue in devices and user’s mode in which fop2 extensions will randomly get stuck ‘busy’ after a user enters the *11 or *12 code to login or logout of a phone. The extension will stay orange until I restart the fop2 deamon. I’m posting here to see if anyone has any thoughts on what I can check to make sure it’s not showing this on the asterisk end to make sure it’s not just an issue with fop2, or if it is something odd with fop2 that is getting hung up (would seem so, as fop2 is restarted, not asterisk). I haven’t had any luck with the dev getting back to me.

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@ghurty wrote:

I used to use a EasyOpenVPN script that I found in order to install VPN on the systems so I can have the yealink phones use their VPN connection. In the newest version of FreePBX using the freepbx distro, I cant install pkcs-helper-devel because the package is unavailable. Does anyone have a newer way of doing it.

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@Russix wrote:

So following these commands from post titled “How i reset freepbx GUI admin password”:

amportal admin auth_none
Changing “AUTHTYPE” from [usermanager] to [none]
amportal admin auth_database
Changing “AUTHTYPE” from [none] to [database]

How do I change the authtype back to usermanager???

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@pcdocmd wrote:

New to the forum and this will be the first time setting up FreePBX. I have a few questions before I replace their dying Panasonic PBX system. I’ve purchased an i7 desktop with 4GB of memory that I want to install FreePBX on a TDM410P Asterisk Card 4 module for the AT&T POTS to connect to and then 10 POE powered Cisco 4971. These will be connecting through existing network that spreads between 5 buildings. Each building has a Cisco 3750 and the buildings are connected through Mikrotik SXT. Each building has 1-3 Cisco Access points and all connections are wireless. They have satellite internet that is 12 down 5 up and data capped at 15 GB per month so VOIP over internet isn’t an option.

Will the hardware mentioned support this
Do I need 2 FXS modules or 2 FXO modules for the Asterisk card?
Will the existing Cisco 3750’s need to be configured for a VLAN and QoS ?
Will FreePBX walk me through POTS or is it setup more for SIP\Trunks?

I’ve been watching YouTube videos and scrolling through forums to try to absorb as much information on FreePBX prior to asking these questions. Thanks in advance.

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@ibenomar wrote:

Hello
Every once in awhile we get random calls, upwards of 50 at a time over a span of a few minutes. They all comes from the some caller id or different ones
is it possible to stop any new call when we have an active call with some caller id to make sure that we don’t have active calls with the some caller id .
Thanks

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@nabberuk wrote:

I have a cloud based pbx install with various remote extensions at two sites. I’ve opened up SIP on the freepbx firewall for just the external IPs at those two sites.

All extensions register fine but at site B i get one way audio. This is usually down to NAT or a firewall issue. yet at both sites im allowing all traffic from internally to the pbx. I’ve looked at the firewall settings on both and they are the same.

I have NAT set to yes on all extensions too.

Handsets at site A are Sangoma and at site B are Yealink. I’m a little puzzled on where to look next?

EDIT: after further investigation it appears only external calls are not working. When i try to call out externally i get a “all circuits are busy message”. This can’t be the case as it’s the only external call going on.

Internal calls work fine, which in my mind would rule out the firewall or nat issues.

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@fearx wrote:

I searched for links and images but found nothing that would answer that. I wanted to understand: What is the “architecture” to make a call?
E.G: Softphone initiates call -> sends to asterisk -> asterisk unzips data …

I think I may be searching for the wrong term. It would be of great help to my understanding if you could explain!

Thank you all and good day

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@Jmarc wrote:

Hello,

I’ve been running FreePBX fine for some years and everything was running smoothly.
I had to update my PfSense to V2.4 for some IPSEC VPN issues, which where resolved with the update.
Ever since the update, all my phones (mostly polycom) become unavailable every 60 seconds or so and then come back as reachable.

Here’s part of the log;

[2018-07-04 08:15:17] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3100’ is now UNREACHABLE! Last qualify: 1018
[2018-07-04 08:15:18] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3033’ is now UNREACHABLE! Last qualify: 3023
[2018-07-04 08:15:19] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3003’ is now UNREACHABLE! Last qualify: 17
[2018-07-04 08:15:25] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3026’ is now UNREACHABLE! Last qualify: 17
[2018-07-04 08:15:25] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3016’ is now UNREACHABLE! Last qualify: 17
[2018-07-04 08:15:26] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3012’ is now UNREACHABLE! Last qualify: 65
[2018-07-04 08:15:26] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3022’ is now UNREACHABLE! Last qualify: 16
[2018-07-04 08:15:28] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3015’ is now UNREACHABLE! Last qualify: 1034
[2018-07-04 08:15:29] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3006’ is now UNREACHABLE! Last qualify: 17
[2018-07-04 08:15:29] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3032’ is now UNREACHABLE! Last qualify: 18
[2018-07-04 08:15:29] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3031’ is now UNREACHABLE! Last qualify: 17
[2018-07-04 08:15:36] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3012’ is now Reachable. (16ms / 2000ms)
[2018-07-04 08:15:36] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3026’ is now Reachable. (1014ms / 2000ms)
[2018-07-04 08:15:36] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3024’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:15:36] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3006’ is now Reachable. (16ms / 2000ms)
[2018-07-04 08:15:38] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3003’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:15:41] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3016’ is now Reachable. (16ms / 2000ms)
Contact 3028/sip:3028@192.168.175.85:51946;rinstance=9ae0176e13a4e62b is now Reachable. RTT: 14.768 msec
Endpoint 3028 is now Reachable
[2018-07-04 08:15:44] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3031’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:15:56] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3033’ is now Reachable. (32ms / 2000ms)
[2018-07-04 08:15:58] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3021’ is now UNREACHABLE! Last qualify: 3037
[2018-07-04 08:16:04] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3022’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:16:06] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3015’ is now Reachable. (51ms / 2000ms)
[2018-07-04 08:16:07] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3032’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:16:08] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3021’ is now Reachable. (58ms / 2000ms)
[2018-07-04 08:16:09] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3100’ is now Reachable. (26ms / 2000ms)
[2018-07-04 08:16:12] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3000’ is now Reachable. (50ms / 2000ms)
[2018-07-04 08:16:39] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3024’ is now Lagged. (3013ms / 2000ms)
[2018-07-04 08:16:39] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3024’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:17:08] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3015’ is now Lagged. (2035ms / 2000ms)
[2018-07-04 08:17:34] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3015’ is now UNREACHABLE! Last qualify: 2036
[2018-07-04 08:17:57] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3015’ is now Reachable. (1035ms / 2000ms)
[2018-07-04 08:18:11] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3021’ is now Lagged. (3034ms / 2000ms)
[2018-07-04 08:18:11] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3009’ is now Lagged. (2012ms / 2000ms)
[2018-07-04 08:18:13] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3009’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:18:15] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3100’ is now UNREACHABLE! Last qualify: 1019
[2018-07-04 08:18:16] NOTICE[1716]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘3000’ is now UNREACHABLE! Last qualify: 49
[2018-07-04 08:18:22] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3021’ is now Reachable. (1033ms / 2000ms)
[2018-07-04 08:18:38] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3010’ is now Lagged. (2032ms / 2000ms)
[2018-07-04 08:18:38] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3010’ is now Reachable. (36ms / 2000ms)
[2018-07-04 08:18:40] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3000’ is now Reachable. (51ms / 2000ms)
[2018-07-04 08:18:40] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3026’ is now Lagged. (3013ms / 2000ms)
[2018-07-04 08:18:41] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3026’ is now Reachable. (17ms / 2000ms)
[2018-07-04 08:18:43] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3029’ is now Lagged. (2013ms / 2000ms)
[2018-07-04 08:18:44] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3029’ is now Reachable. (16ms / 2000ms)
[2018-07-04 08:18:53] NOTICE[1716]: chan_sip.c:24403 handle_response_peerpoke: Peer ‘3100’ is now Reachable. (1021ms / 2000ms)

Tried disabling hardware checksum offloading on the firewall, didn’t change anything.

FreePBX 13.0.190.12
Current Asterisk Version: 13.9.1
All on a VM

Any help would be appreciated.

Thanks

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@Hosker wrote:

Afternoon,

Is the Interface dropping to 0% normal? or should it be constant. I dont have problems of calls dropping or anything but it seems to me that it shoudnt do that and perhaps I have a problem to address.

Basic setup:
Freepbx -> Switch -> WAN

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@handleric wrote:

Hello,

i’m trying to setup an external voicemail server to meet my retirements of expansion and future growth. I seem to have everything working except one part, I need to figure out how to modify the default context to send busy or unanswered calls to the external voicemail server. I am using an IAX trunk to connect the two boxes and can dial across into it as well as transfer calls over. I also already have the MWI publishing across, so once I get this figured out I should be set.

I’ve found and read some really old articles on this which seem to no longer pertain. Does anyone around have experience with this?

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@fearx wrote:

When I give Firefox permission to access the microphone, it automatically drops and displays this:
== Setting global variable ‘SIPDOMAIN’ to ‘xxxx.xxxx.xxx.xx’

It authenticates but can not make calls. Any idea?

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@SilkBC wrote:

Hello,

I have a similar issue to this one from 2008: https://community.freepbx.org/t/cid-name-prefix-lost-for-anonymous-callers/3657/5

I have several queues setup and calls with a caller ID of anonymous get assigned one specific prefix, regardless of which queue it really come sin on. What is sort of odd is that the CDR shows the correct queue prefix; it is just on the phone’s screen that the incorrect prefix shows up.

Phone in question is an Aastra (Mitel) 6869i.

I don’t know if this is just something stupid in the way Aastra handles anonymous CIDs, or if something isn’t being passed to the phone properly by FreePBX.

Versions I have are:
FreePBX: 10.13.66-22
Core: 13.0.122.33 (just recently updated; not sure if this was an issue before the update)

One thing I will do is to see if I can replicate this on a test system with the same phone my client has. Once i do, I will swap my phone with a Sangoma and see if the issue persists and report back. That should at least identify if it is an Aastra/Mitel issue or something to do with FreePBX.

(Obviously if it is an Aastra/Mitel issue, I strongly suspect there isn’t much, if anything, that could be done on the FreePBX side; it’s something I would have to take up with them?)

I will report back shortly.

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@fox_hu wrote:

Hello , everyone , My server is running FreePBX 13.0.192.19 , and installed Outbound CNAM module. The module is written by @lgaetz, and is great.

Outbound CNAM module works ,but I just facing an issue, the Chinese characters were garbled on userfield.

Hers is the example:

wrong characters.jpg1414x655 209 KB

Please help me to fix it, thank you !

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@fearx wrote:

Hello. I’m learning about FreePBX and Asterisk, and to test a SIP call, I’ve used Zoiper. But it return this to me:

Just created it on “extensions” as a Chan_SIP extension. What can be?
PS: the user and password are correct…

PS2: My CLI return this to me:
ERROR[4662]: pjproject:0 <?>: sip_transport.c Error processing 42 bytes packet from UDP 192.168.1.8:45625 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 1:

Sorry, I did not understand my error yet

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