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DHCPv6 Client - SGN7 FreePBX 14

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@kolpinkb wrote:

Does anyone have FreePBX 14 Distro working with IPv6 as a DHCPv6 client?

The distro seems to get one during early steps of the installation process as seen in my router V6 leases. But then the the distro switches to an IPv6 autoconf script that doesn't obtain its own IP properly.

There is a UUID generated in:

/etc/sysconfig/network-scripts/ifcfg-eth0

But when I add the option:

DHCPV6C=yes

And restart the adapter or reboot nothing happens.

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Minor Module Upgrade Issue

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@johnjces wrote:

Just upgraded 30 of 31 modules tonight and all seems okay on my distro version 10.13.66-20. However, I do not use Zulu and it is disabled and I do not use Customer Relations module and that too is disabled. When completed I receive the following message:

Module Administration
Errors with selection:
Customer Relationship Management Link cannot be upgraded:
The Module Named "Zulu" is required, but yours is disabled.
Please try again after the dependencies have been installed.

It's nice to have everything upgraded with those that are not desired disabled so they don't come back on system upgrades.

Is there any way to update this module without enabling then disabling Zulu? I don't see that doing such is much of an issue or problem, just nice to have seamless module updates.

Thanks!

John

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Roadmap for asterisk 13.16?

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@tonyg wrote:

hi, i am trying to get a polycom to work with my freepbx 14 (asterisk 13.15.1) system. All my phones are configured for sip-tls and srtp. i can receive calls on the polycom with no issue, but i don't get any audio when i make a call. i get this error in asterisk

[2017-06-01 15:41:22] WARNING[25516][C-0000006c]: res_srtp.c:415 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110

I have found this on the asterisk site and I think it is the bug I am experiencing, looks like it is fixed in 13.16, so I was wondering when freepbx might include 13.16

https://issues.asterisk.org/jira/browse/ASTERISK-26979
thanks

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Enabling SNMP Monitoring

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@SysAdminKen wrote:

Hi Everyone,

I have been searching around on the web (and these forums) for the past few days on how to enable SNMP on the distro to do basic system monitoring (CPU/DISK/Mem/Network/etc). At this point I'm not even really worried/concerned about Asterisk SNMP monitoring (although that will be the next step). I get that the distro is based on Schmooze/Centos/RedHat so in theory, simply searching for info on how to configure SNMP in any of those 3 distros should have revealed some decent information however all of my tests so far have failed.

I went on a bit of a tangent yesterday looking to figure out if maybe the firewall was blocking the SNMP ports or something like that... but after sitting down with my first coffee I've decided to simply ask... are there any distro specific instructions out there that clearly explain the process of setting up/configuring SNMP within the distro?

Thx
Ken

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Network Card

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@bcampbell wrote:

I have a Sangoma FreePBX Appliance 500 and have a need to add an additional network card. I know that there are two PCI Express slots open on the system, but couldnt find a list of recommended Network Cards (found a list of expansion cards that were all telephony related T1 cards, etc..). In hoping to stick to something that is well supported I was thinking about this card (Intel EXPI9402PT) which utilizes the Intel 82571GB chipset. I am hopeful that the chipset itself is supported in the kernel, but apart from that I am not sure what the specs are for the pci express slots in the appliance. I cant find any documentation on the sangoma site as they seem to have removed any existence of it ever being sold. The best I could find was a youtube video from Chris at Crosstalk solutions doing an unboxing of the 500 appliance. The video shows that they exist but he doesnt talk about what speed they both support. Anyone know for sure? The NIC I picked out above is a x4.

Thanks,

Brian

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New install, no MOTD showing

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@cullenl wrote:

I recently installed a fresh system using the 10.13.66 ISO to a vultr.com instance. Since then, there's been no MOTD when logging in via ssh; no warnings, either, but this makes me wonder what else is wrong. Anyone else seen this or can explain what went wrong?

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Strange Behaviour of Outbound Routes

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@richarduk wrote:

So I've just installed a new Freepbx from the distro, and the install went well generally. Ran an yum update and a FreePBX module update after install and everything is now up to date. Now at FreePBX 14.0.1rc1.17 and Asterisk 14.5

Then spent the day setting up the trunks and extensions and now I'm doing the outbound routes. When I add an outbound route that's great and it seems all good, but if I edit that route then submit I'll get two with the same name.

If I delete both and then add two outbound routes with different trunks then the trunks will merge between them.

So, deleted all the outgoing routes, created a single route called national. Great.
Then selected that route, changed nothing and submit and now I've got two called national

from the table outbound_routes

+----------+----------+--------+-------------+----------+-----------------+--------------------+----------+---------------+------+-----------+----------+-------------+-------------------+
| route_id | name     | outcid | outcid_mode | password | emergency_route | intracompany_route | mohclass | time_group_id | dest | time_mode | timezone | calendar_id | calendar_group_id |
+----------+----------+--------+-------------+----------+-----------------+--------------------+----------+---------------+------+-----------+----------+-------------+-------------------+
|        0 | national |        |             |          |                 |                    | default  |             0 |      |           | default  | NULL        | NULL              |
|        0 | national |        |             |          |                 |                    | default  |             0 |      |           | default  | NULL        | NULL              |
+----------+----------+--------+-------------+----------+-----------------+--------------------+----------+---------------+------+-----------+----------+-------------+-------------------+

So now I'm stuck. I can't add any routes and really don't want to have to re-install and start from scratch again. Any ideas?

Just FYI I've uninstalled all the commercial modules so none of those are affecting anything.

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Delete IP address from firewall


How to enable SNMP?

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@MAA wrote:

Hello,

How to enable SNMP on FreePBX 13 Distro?
I try yum install snmpd, but no success:

[root@server ~]# yum install snmpd
Loaded plugins: fastestmirror, kmod
Setting up Install Process
Loading mirror speeds from cached hostfile
base | 2.0 kB 00:00
extras | 1.3 kB 00:00
pbx | 2.9 kB 00:00
schmooze-commercial | 2.9 kB 00:00
updates | 1.3 kB 00:00
No package snmpd available.
Error: Nothing to do

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Keeping removed Commercial modules removed, after Distro Upgrade

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@4allbusiness wrote:

Hello FreePBX,

I know that removed (not disabled) commercial modules will be reinstalled and enabled after a distro Upgrade.
This is not bad or so, but some modules are giving me extra overhead on CPU, such as PagingPro and QueueReports.

After removing these modules, my systems are getting very quiet and have low cpu resources.

Is there in any way a solution to keep these Commercial modules removed?
I understand the commercial importance for Sangoma, but it is strange, that after removing these commercial modules, they get reinstalled after each upgrade of the Distro.

I hope you can understand this from a users view. For now, I have to create a script to delete/remove these modules in a cron job. But I hope there is a better solution for the future?

It is strange, that the Distro decides to reinstall these commercial modules without asking this, after we have deleted them, isn't it? FreePBX stands for Freedom, but is feels a little not so 'Freedom', if Commercial modules keeps being pushed after each upgrade.

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FreePbx Backup and Restore DR planning help please

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@255 wrote:

Good Morning,

I'm currently working of a DR strategy for our FreePBX sever and have some questions please?

We use the native "Backup" module included with FreePBX.

  1. If we create a "full" and include all audio, I take it we should be able to do a total restore?
  2. What happen if the "version" of freePBX differs from the version that the backup was waken from?
    3, What happens if we don't have the same modules enabled or installed when the restore is carried out? (Example extension dialing)

With thanks

Alan Scott

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Problem with fax to email

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@senbac wrote:

Hi all,
I am getting trouble with fax to email
WARNING[16886][C-00000081]: res_fax.c:1976 receivefax_t38_init: channel 'SIP/S2_SPT-000000a7' refused to negotiate T.38
I had try some advice on other related topic but it seem not working
any suggestions are welcome
thanks all

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BLF extension status online even if the extension is offline

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@grekasa wrote:

Hello,
I am using FreePBX Distro v13 with Asterisk v13 64bit latest stable.
My ipphones are Grandstream GXP1628, Fanvil C60 and Grandstream GSWave softphones.
At my mobile phone i enabled the blf function of GSWave but all the extensions are showed green online but this isn't correct because some of them are offline! If an (true) online extension is busy the blf show it correctly as busy with red light. The same thing happens with my GXP1628 & Fanvil c60.
It seems that IPPBX doesn't sent a Notify with Offline status!
Thank you very much in advance.

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Alias to Extension number

Source packages

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@dwsiemens wrote:

I am using the Freepbx SNG7 beta for version 14.

I have enabled the src repo. When I try to download source http://package1.sangoma.net/sangoma/src

The error is

Loaded plugins: fastestmirror
http://package1.sangoma.net/sangoma/src/repodata/repomd.xml: [Errno 14] HTTP Error 404 - Not Found
Trying other mirror.
To address this issue please refer to the below knowledge base article

https://access.redhat.com/articles/1320623

If above article doesn't help to resolve this issue please create a bug on https://issues.freepbx.org/

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FreePBX Upgrade from 10.13.x (SHMZ) to 14 RC (EL7) Issues

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@fender_bender5150 wrote:

I'm creating this topic for those of us performing the much anticipated upgrade from SHMZ based FreePBX to the new EL7 platform.

Has anyone noticed any problems, and if so, what are they? Let's help the devs!

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How many calls can I record similtaneously?

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@v4704 wrote:

Hi!
I set the "Call Recording" to "Force" on the Ring Group confirmation page.
Can you please tell me, how many calls will the system be able to record at the same time?

Thank you!

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CPU spikes on Asterisk

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@Loki wrote:

Hi all,

I am migrating the old elastix system to new FreePBX Distro (latest 10.13.66) and I am experiencing strange CPU spikes from time to time even though the system is under no load. Spikes sometimes exceed 100% CPU (more than one fully utilized core).
Could you please point me to a direction, how to figure out, why is this happening? I tried ps -LlFm and pstack with no luck so far. I attach top output.
System has currently configured with cca 250 PJSIP endpoints (cca 150 online).
Thank you.

top - 12:46:03 up  2:51,  2 users,  load average: 0.10, 0.14, 0.13
Tasks: 175 total,   1 running, 174 sleeping,   0 stopped,   0 zombie
Cpu0  : 32.7%us,  4.3%sy,  0.0%ni, 63.0%id,  0.0%wa,  0.0%hi,  0.0%si,
0.0%st
Cpu1  :  9.0%us,  3.3%sy,  0.0%ni, 87.7%id,  0.0%wa,  0.0%hi,  0.0%si,
0.0%st
Cpu2  : 17.9%us,  1.6%sy,  0.0%ni, 74.3%id,  0.0%wa,  0.0%hi,  0.0%si,
6.3%st
Cpu3  : 58.0%us,  2.3%sy,  0.0%ni, 39.7%id,  0.0%wa,  0.0%hi,  0.0%si,
0.0%st
Mem:   3922216k total,  2187576k used,  1734640k free,    93200k buffers
Swap:   786428k total,        0k used,   786428k free,   534736k cached

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
 2061 asterisk  20   0 3021m  88m  16m S 99.0  2.3  17:45.63 asterisk
20974 asterisk  20   0  331m  26m 9056 S  4.3  0.7   0:00.13 php
20950 root      20   0  110m 2444 1252 S  2.7  0.1   0:00.82 htop
 1492 mysql     20   0  947m  40m 6284 S  1.7  1.1   0:36.99 mysqld
 3114 mongodb   20   0  327m  43m  20m S  0.7  1.1   0:33.33 mongod
    4 root      20   0     0    0    0 S  0.3  0.0   0:00.16 ksoftirqd/0
  860 root      20   0     0    0    0 S  0.3  0.0   0:00.39 kauditd
 1746 asterisk  20   0 4607m 484m  18m S  0.3 12.6   2:59.63 java
 2140 root      20   0  323m  16m 9100 S  0.3  0.4   0:06.46 php
20951 root      20   0 15032 1308  948 R  0.3  0.0   0:00.08 top
    1 root      20   0 19360 1500 1196 S  0.0  0.0   0:00.81 init

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Pjsip over TCP and security

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@v4704 wrote:

Hi!
I would like to enable TCP support in order to have a native "background activity" with iPhone users (over UDP, the iOS client needs to workaround and keep the app alive without the iOS support, which eats the user's battery, and this solution is less reliable).

Now, with TCP enabled on the client side, the registration failed.
I see that the Firewall module only specify 5160 over UDP and not TCP (Advanced > Port/Service Maps).

Other than creating a custom service, do I need to do anything else to enable TCP support? maybe something else not related to the security-side of it?
Is it even recommended or known as problematic?

Asterisk: 13.15.0
FreePBX: 13.0.192.8

Many thanks!

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PJSIP Ext 2 ext hangup on answer and no VM

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@bobster9999 wrote:

Sangoma Appliance, PBX Firmware: 10.13.66-20, PBX Service Pack: 1.0.0.0

Hi,
My distro has been working perfectly until this morning. My SIP trunk was down and prior to reboot I updated modules.

My system does not now allow calls between extensions, users dial each other and the call is dropped when answered - as soon as the simple bridge is connected it gets disconnected.

Spawn extension (from-internal, 2002, 1) exited non-zero on 'PJSIP/2002-00000026'
-- PJSIP/2002-00000026 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called PJSIP/2002/sip:2002@192.168.10.197:5060
-- Connected line update to PJSIP/1999-00000025 prevented.
-- PJSIP/2002-00000026 is ringing
-- Nobody picked up in 20000 ms
-- Executing [s@macro-dial-one:53] ExecIf("PJSIP/1999-00000025", "0?MacroExit()") in new stack
-- Executing [s@macro-dial-one:54] ExecIf("PJSIP/1999-00000025", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-dial-one:55] GosubIf("PJSIP/1999-00000025", "0?s-NOANSWER,1()") in new stack
-- Executing [s@macro-dial-one:56] MacroExit("PJSIP/1999-00000025", "") in new stack
-- Executing [s@macro-exten-vm:17] Set("PJSIP/1999-00000025", "SV_DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:18] GosubIf("PJSIP/1999-00000025", "0?docfu,1()") in new stack
-- Executing [s@macro-exten-vm:19] GosubIf("PJSIP/1999-00000025", "0?docfb,1()") in new stack
-- Executing [s@macro-exten-vm:20] Set("PJSIP/1999-00000025", "DIALSTATUS=NOANSWER") in new stack
-- Executing [s@macro-exten-vm:21] ExecIf("PJSIP/1999-00000025", "0?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:22] GotoIf("PJSIP/1999-00000025", "0?s-NOANSWER,1") in new stack
-- Executing [s@macro-exten-vm:23] Macro("PJSIP/1999-00000025", "vm,2002,NOANSWER,") in new stack
-- Executing [s@macro-vm:1] Macro("PJSIP/1999-00000025", "user-callerid,SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/1999-00000025", "TOUCH_MONITOR=1497963392.37") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/1999-00000025", "AMPUSER=1999") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/1999-00000025", "10?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/1999-00000025", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("PJSIP/1999-00000025", "CALLERID(number)=1999") in new stack
-- Executing [s@macro-user-callerid:30] Set("PJSIP/1999-00000025", "CALLERID(name)=IT Support") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/1999-00000025", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("PJSIP/1999-00000025", "CDR(cnam)=IT Support") in new stack
-- Executing [s@macro-user-callerid:33] Set("PJSIP/1999-00000025", "CDR(cnum)=1999") in new stack
-- Executing [s@macro-user-callerid:34] Set("PJSIP/1999-00000025", "CHANNEL(language)=en_GB") in new stack
-- Executing [s@macro-vm:2] Set("PJSIP/1999-00000025", "VMGAIN=") in new stack
-- Executing [s@macro-vm:3] Macro("PJSIP/1999-00000025", "blkvm-check,") in new stack
-- Executing [s@macro-blkvm-check:1] Set("PJSIP/1999-00000025", "GOSUB_RETVAL=") in new stack
-- Executing [s@macro-blkvm-check:2] ExecIf("PJSIP/1999-00000025", "0?Set(GOSUB_RETVAL=TRUE)") in new stack
-- Executing [s@macro-blkvm-check:3] MacroExit("PJSIP/1999-00000025", "") in new stack
-- Executing [s@macro-vm:4] GotoIf("PJSIP/1999-00000025", "1?vmx,1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] Set("PJSIP/1999-00000025", "__EXTTOCALL=2002") in new stack
-- Executing [vmx@macro-vm:2] Set("PJSIP/1999-00000025", "_CRMVOICEMAIL=2002") in new stack
-- Executing [vmx@macro-vm:3] Set("PJSIP/1999-00000025", "MEXTEN=2002") in new stack
-- Executing [vmx@macro-vm:4] Set("PJSIP/1999-00000025", "MMODE=NOANSWER") in new stack
-- Executing [vmx@macro-vm:5] Set("PJSIP/1999-00000025", "RETVM=") in new stack
-- Executing [vmx@macro-vm:6] Set("PJSIP/1999-00000025", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:7] Macro("PJSIP/1999-00000025", "get-vmcontext,2002") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("PJSIP/1999-00000025", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("PJSIP/1999-00000025", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("PJSIP/1999-00000025", "") in new stack
-- Executing [vmx@macro-vm:8] Set("PJSIP/1999-00000025", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:9] NoOp("PJSIP/1999-00000025", "MODE IS: unavail") in new stack
-- Executing [vmx@macro-vm:10] GotoIf("PJSIP/1999-00000025", "1?chknomsg") in new stack
-- Goto (macro-vm,vmx,12)
-- Executing [vmx@macro-vm:12] GotoIf("PJSIP/1999-00000025", "0?s-NOANSWER,1") in new stack
-- Executing [vmx@macro-vm:13] GotoIf("PJSIP/1999-00000025", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,15)
-- Executing [vmx@macro-vm:15] NoOp("PJSIP/1999-00000025", "Checking if ext 2002 is enabled: ") in new stack
-- Executing [vmx@macro-vm:16] GotoIf("PJSIP/1999-00000025", "1?s-NOANSWER,1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-vm:1] Macro("PJSIP/1999-00000025", "get-vmcontext,2002") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("PJSIP/1999-00000025", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("PJSIP/1999-00000025", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("PJSIP/1999-00000025", "") in new stack
-- Executing [s-NOANSWER@macro-vm:2] VoiceMail("PJSIP/1999-00000025", "2002@default,u") in new stack
-- Playing 'vm-theperson.ulaw' (language 'en_GB')
== Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'PJSIP/1999-00000025' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 23) exited non-zero on 'PJSIP/1999-00000025' in macro 'exten-vm'
== Spawn extension (ext-local, 2002, 2) exited non-zero on 'PJSIP/1999-00000025'
-- Executing [h@ext-local:1] Macro("PJSIP/1999-00000025", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/1999-00000025", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/1999-00000025", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/1999-00000025", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/1999-00000025' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/1999-00000025'

This has also prevented access to voicemail using *97 or *98.

Please help, system unusable.....

At the time of writing I came across this post which explains removing 2 lines of config for PJSIP - will there be a patch?
https://issues.freepbx.org/browse/FREEPBX-15009

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