@expire wrote:

Dear FreePBX Community,

I have been struggling with an issue for about 2 weeks now, I tried allot of settings buts until now I still don’t know what is wrong.

When 2 users call each other and a third user (USER_3 in the logs) tries to call user 2 (USER_2 in the logs), he doesn’t immediately get the busy tone and the connection is not directly ended.

We are using SSL and what I seem to be able to read from the logs is that the RTP connection between USER_2 and USER_3 is destroyed immediately but the SIP channel isn’t destroyed immediately and delays up to around 10 seconds…

We had this running in the past without SSL and the busy tone was given within a second.

Here you will find the logs here:
https://drive.google.com/file/d/1ZGywKGR8YKrUUmxTxbxsHACd0zLVrutf/view?usp=sharing

The long delay is from [2019-09-30 19:00:52] to [2019-09-30 19:01:02], I put some blocks in the logs with the word “INTERESTING”, where I thought that that could interesting.
And I also noted where I think the USER_2 disconnected and USER_3.

Best regards,
x

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@dan_ce wrote:

I’ve got a script that I call from a custom destination. I’m trying to debug why it isn’t being called by the custom destination anymore. I’ve tried turning agi debugging on (because the logs show the script being called, then nothing else at all relating to the script) but that doesn’t seem to show anything helpful related to the script. Any thoughts most welcome!

cheers

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@lmon wrote:

Hi,
Just tried to login to the admin console this morning and get the following error message :

“/var/www/html/admin/modules/framework/module.xml does not exist”

Any suggestions on why I get this ? I checked and indeed, there is no module.xml file in this directory.
I don’t think I made any changes to the system, so don’t see why this happens …
thanks !

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@GaryCameron wrote:

When trying to update the Freepbx system I get the following errors:
Transaction check error:
file /etc/asterisk/amd.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/ari.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/asterisk.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/ccss.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/cdr.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/cdr_adaptive_odbc.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86 _64
file /etc/asterisk/cel.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/cel_odbc.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/confbridge.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/dnsmgr.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/enum.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/extconfig.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/extensions.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/features.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/http.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/iax.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/indications.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/logger.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/manager.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/meetme.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/modules.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/musiconhold.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/pjsip.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/pjsip_notify.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/queuerules.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/queues.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/res_odbc.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/res_parking.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/rtp.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/sip.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/sip_notify.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/udptl.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64
file /etc/asterisk/voicemail.conf from install of asterisk13-configs-13.27.1-1.sng7.x86_64 conflicts with file from package freepbx-14.1-1.sng7.x86_64

What do I need to do to go about repairing this? I have had to switch to a backup server because this serve is down and out of commission

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@dotcomm wrote:

Hello,

When trying to record a new system recording via FreePBX, it fails and shows in the logs:

-- Executing [s@macro-user-callerid:34] Gosub("Local/445@from-internal-000001f3;2", "macro-user-callerid,lang-playback,1(hook_1)") in new stack
-- Executing [lang-playback@macro-user-callerid:1] GosubIf("Local/445@from-internal-000001f3;2", "1?macro-user-callerid,en,hook_1():macro-user-callerid,en,hook_1()") in new stack
-- Executing [en@macro-user-callerid:3] Playback("Local/445@from-internal-000001f3;2", "beep&im-sorry&your&simul-call-limit-reached&goodbye") in new stack

It’s the same notification as you see when you are trying to dial with a device where no user is linked to it. (We have a devices & users setup.)

Just before the issue showed up, I was playing with the commands “database del AMPUSER & database put DEVICE/xxx user none”… so I think I may have deleted too much.

I already did a force download & install of the system recordings module, without luck.

Any of you guys have an idea to fix it?

Thanks for your help!

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@gordon wrote:

So I have been looking at the Zulu UC app for android. It is complicated to have an always on vpn with Android and the Freepbx certificate manager let’s encrypt options expect http auth to work. Which means the FreePBX website needs to be accessible from the general web.

So my question… is it now recommended that FreePBX is exposed over the internet?

I am looking at a lot of trouble to get android vpns working for Zulu and there isn’t an easy way to have auto on/off based on the phone having a local wifi connection or not. Letting clients access FreePBX via internet would certainly simplify things. But it’s always been my understanding that the pbx should only offer local access.

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@salvy12477 wrote:

After restarting the asterisk service in FreePBX command line the service now shows;

root@TOWER-NYSAU-PBX-02 ~]# service asterisk status
asterisk dead but subsys locked
[root@TOWER-NYSAU-PBX-02 ~]#

(The commands I used to restart was service asterisk restart)
I’m assuming that there’s a process thats stuck and or locked out but am unable to locate it. Also I am fairly new to linux and only recently have been put incharge of maintaining the phone system.

I’ve also started asterisk using; asterisk -vvvc with the below output;

Asterisk 13.18.2, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

[ Initializing Custom Configuration Options ]
== Parsing ‘/etc/asterisk/asterisk.conf’: Found
Couldn’t find manager DBGet in XML documentation
Couldn’t find manager DBGet in XML documentation
== Manager registered action DBGet
Couldn’t find manager DBPut in XML documentation
Couldn’t find manager DBPut in XML documentation
== Manager registered action DBPut
Couldn’t find manager DBDel in XML documentation
Couldn’t find manager DBDel in XML documentation
== Manager registered action DBDel
Couldn’t find manager DBDelTree in XML documentation
Couldn’t find manager DBDelTree in XML documentation
== Manager registered action DBDelTree
PBX UUID: f846665d-a4e5-4879-8ab7-5b06f24da8f3
== Registered ‘audio’ codec ‘g723’ at sample rate ‘8000’ with id ‘1’
== Created cached format with name ‘g723’
== Registered ‘audio’ codec ‘ulaw’ at sample rate ‘8000’ with id ‘2’
== Created cached format with name ‘ulaw’
== Registered ‘audio’ codec ‘alaw’ at sample rate ‘8000’ with id ‘3’
== Created cached format with name ‘alaw’
== Registered ‘audio’ codec ‘gsm’ at sample rate ‘8000’ with id ‘4’
== Created cached format with name ‘gsm’
== Registered ‘audio’ codec ‘g726’ at sample rate ‘8000’ with id ‘5’
== Created cached format with name ‘g726’
== Registered ‘audio’ codec ‘g726aal2’ at sample rate ‘8000’ with id ‘6’
== Created cached format with name ‘g726aal2’
== Registered ‘audio’ codec ‘adpcm’ at sample rate ‘8000’ with id ‘7’
== Created cached format with name ‘adpcm’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘8000’ with id ‘8’
== Created cached format with name ‘slin’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘12000’ with id ‘9’
== Created cached format with name ‘slin12’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘16000’ with id ‘10’
== Created cached format with name ‘slin16’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘24000’ with id ‘11’
== Created cached format with name ‘slin24’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘32000’ with id ‘12’
== Created cached format with name ‘slin32’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘44100’ with id ‘13’
== Created cached format with name ‘slin44’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘48000’ with id ‘14’
== Created cached format with name ‘slin48’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘96000’ with id ‘15’
== Created cached format with name ‘slin96’
== Registered ‘audio’ codec ‘slin’ at sample rate ‘192000’ with id ‘16’
== Created cached format with name ‘slin192’
== Registered ‘audio’ codec ‘lpc10’ at sample rate ‘8000’ with id ‘17’
== Created cached format with name ‘lpc10’
== Registered ‘audio’ codec ‘g729’ at sample rate ‘8000’ with id ‘18’
== Created cached format with name ‘g729’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘8000’ with id ‘19’
== Created cached format with name ‘speex’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘16000’ with id ‘20’
== Created cached format with name ‘speex16’
== Registered ‘audio’ codec ‘speex’ at sample rate ‘32000’ with id ‘21’
== Created cached format with name ‘speex32’
== Registered ‘audio’ codec ‘ilbc’ at sample rate ‘8000’ with id ‘22’
== Created cached format with name ‘ilbc’
== Registered ‘audio’ codec ‘g722’ at sample rate ‘16000’ with id ‘23’
== Created cached format with name ‘g722’
== Registered ‘audio’ codec ‘siren7’ at sample rate ‘16000’ with id ‘24’
== Created cached format with name ‘siren7’
== Registered ‘audio’ codec ‘siren14’ at sample rate ‘32000’ with id ‘25’
== Created cached format with name ‘siren14’
== Registered ‘audio’ codec ‘testlaw’ at sample rate ‘8000’ with id ‘26’
== Created cached format with name ‘testlaw’
== Registered ‘audio’ codec ‘g719’ at sample rate ‘48000’ with id ‘27’
== Created cached format with name ‘g719’
== Registered ‘audio’ codec ‘opus’ at sample rate ‘48000’ with id ‘28’
== Created cached format with name ‘opus’
== Registered ‘image’ codec ‘jpeg’ at sample rate ‘0’ with id ‘29’
== Created cached format with name ‘jpeg’
== Registered ‘image’ codec ‘png’ at sample rate ‘0’ with id ‘30’
== Created cached format with name ‘png’
== Registered ‘video’ codec ‘h261’ at sample rate ‘1000’ with id ‘31’
== Created cached format with name ‘h261’
== Registered ‘video’ codec ‘h263’ at sample rate ‘1000’ with id ‘32’
== Created cached format with name ‘h263’
== Registered ‘video’ codec ‘h263p’ at sample rate ‘1000’ with id ‘33’
== Created cached format with name ‘h263p’
== Registered ‘video’ codec ‘h264’ at sample rate ‘1000’ with id ‘34’
== Created cached format with name ‘h264’
== Registered ‘video’ codec ‘mpeg4’ at sample rate ‘1000’ with id ‘35’
== Created cached format with name ‘mpeg4’
== Registered ‘video’ codec ‘vp8’ at sample rate ‘1000’ with id ‘36’
== Created cached format with name ‘vp8’
== Registered ‘video’ codec ‘vp9’ at sample rate ‘1000’ with id ‘37’
== Created cached format with name ‘vp9’
== Registered ‘text’ codec ‘red’ at sample rate ‘0’ with id ‘38’
== Created cached format with name ‘red’
== Registered ‘text’ codec ‘t140’ at sample rate ‘0’ with id ‘39’
== Created cached format with name ‘t140’
== Registered ‘audio’ codec ‘none’ at sample rate ‘8000’ with id ‘40’
== Created cached format with name ‘none’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘8000’ with id ‘41’
== Created cached format with name ‘silk8’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘12000’ with id ‘42’
== Created cached format with name ‘silk12’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘16000’ with id ‘43’
== Created cached format with name ‘silk16’
== Registered ‘audio’ codec ‘silk’ at sample rate ‘24000’ with id ‘44’
== Created cached format with name ‘silk24’
== Sorcery registered wizard ‘bucket’
== Sorcery registered wizard ‘bucket_file’
Cannot update type ‘bucket’ in module ‘core’ because it has no existing documentation!
Failed to register ‘bucket’ object type in Bucket sorcery
Bucket API initialization failed. ASTERISK EXITING!
== Manager unregistered action DBGet
== Manager unregistered action DBPut
== Manager unregistered action DBDel
== Manager unregistered action DBDelTree

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@sborne wrote:

Hey All,

We are curious if a hot reload (using “Apply Config” in Freepbx) of Dial Options, added Ww, to try and get on demand recording working, will cause calls to be dropped?

Asterisk 11.16.0
FreePBX 12

SIP trunk
IAX2 trunk
DAHDI with 2 PRI

Thanks in advance,

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@v70ff wrote:

Situation

Outside caller : calls my main line

My PBX: rings

Outside caller: Ends call

My PBX: Still ringing (4-5s) after call was ended by outside caller.

Does anyone know why this is happening and giving a solution would be greatly appreciated

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@wassy83 wrote:

Hi to all, I have an ftth + voip connectivity provided by tiscali isp in italy. The plan comes with a free of charge locked/brand zyxel router, but the provider also releases the pppoe and sip parameters in an email in this way:

PPPoE static 
USER-VLAN 8/35 in 802.1Q
user ID PPP (Internet) userppp@ftth.tiscali.it
Password PPP (Internet) passwordppp
Username SIP (Voice) 0039XXXXXXXXXX@ims.tiscali.net
Password SIP (Voice) passwordsip
URI 0039XXXXXXXXXX
outbound proxy/proxy server:
srv:  srvmi.p.ims.tiscali.net
fqdn:  core1.p.ims.tiscali.net
IP:  213.205.21.8
Protocol UDP:  port 5060
Protocol Voip: SIP RFC 3261
Domain/Registrar: ims.tiscali.net
Codec list: g711alaw; g729
DTMF rfc2833 payload type 97 symmetric implementation
Fax g 711 pass-through (T38 disabled)
Session refresh Update method
PRACK Supported 100rel|
MWI notify unsolicited ( subscribe disable)
CLIP PAI-FROM

now I want to migrate from this zyxel router to a pfsense box for the pppoe interface and to freepbx for the sip trunk. no problems with pfsense, but I cannot figure out a way for configuring these parameters in my freepbx box (only chan_pjsip).
in my pjsip trunk I have added username, password, outbound proxy and sip server as above but I’m always receiving something like error 171005 missing route set.
Hope someone can give to me a starting point.
many thanks

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@johnjces wrote:

Many months back I wrote about issues with FreePBX DDNS not updating and I have to admit that it is sort of a pain.

My brother had Cox Cable go down for awhile and when it came back, having a dynamic IP from Cox, got a new IP. Cox in the Phx area seem to drop a lot and Internet IP’s change a lot. In any event, for several days his PBX was not reporting the new IP, BUT the correct Internet IP was reported in the box. As a result, “things” just weren’t working well especially IAX. Forcing the new IP will eventually update the DNS servers in an hour or two.

In checking System Admin, DDNS, it showed the correct Internet IP but for some unknown reason it hadn’t updated any DNS servers. This is similar behavior that I wrote about and I think I posted a bug report. It’s been so long I just can’t remember.

Can anyone on the FreePBX team provide some insight into the built in DDNS and what might be happening all to frequently. Seems to take many hours before a DNS update occurs. In this case days and finally had to force it.

Thanks!

John

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@kerencom wrote:

i am looking for a CTI solution for Free PBX so i can integrate my call center

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@LigerXT5 wrote:

We’ve recently, in the last year, installed a FreePBX system with a client. We’ve been informed the *80XXX feature code does not seem to work. Only rings the extension. Should auto answer and be placed on speaker.

In the phone’s Account > Intercom Settings > Allow Auto Answer by Call-Info/Alert-Info is set to Yes, the remainder is set to no, and no Alert-Info for the Auto Answer field.

This is set the same on my desk phone, same model.

I’ve read, and confirmed in the FreePBX feature codes, *54 is to allow intercom, however when used on their phones, it redials the last number called out from that phone. I don’t see that mentioned in any search result. On my phone, I hear the Intercom Enabled.

*80, *54, and *55 are default and enabled on the server.

We’d rather not setup paging groups, just for one to one (bi-directional) extension paging.

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@Davedog wrote:

message that appears approx 10 minutes after hitting Apply Config:

Error: Did not receive valid response from server

XHR response code: 0 XHR responseText: undefined jQuery status: error

searching forums for solutions did not have any suggestions that corrected problem

amportal a r from the CLI clears this until you go back to the GUI and press Apply Config, then same result

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@bmartindcs wrote:

I manage a few dozen system and am aware of the usual suspects that cause poor voice quality. Typical scenario has SIP endpoints at one place, and PBX elsewhere in a major datacenter. Real problems are rare, and some people exaggerate more than others, so it’s hard to tell if there really is a problem or not in some cases with certain people. Obviously it will never be perfect if you’re traversing public internet, and I do set that expectation.

I’m looking for a way to track/report on call quality. Is there such a way/tool at the asterisk level or within fpbx? Ideally a simple report/graph like the other server resource utilization on the main dashboard screen within fpbx, so I can get an overview of what a day looked like, vs sitting there staring at a trace for potentially hours on end.

FPBX distro 14.0.13.4 and Asterisk 13.22 with pjsip and usually g.711 end to end.

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@cursor wrote:

Today the pbx is simply refusing to connect calls to any phones. Everything seems correctly configured and the pbx can send and receive calls via an E1 but calls simply do not flow to phones and cpu usage is above 100%. I have seen load averages over 30 with all the stuck calls. DNS is working and all phones report as available. A core show channels shows all the stuck calls:

Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgeID
DAHDI/5-1 macro-dial-one s 59 Up Dial PJSIP/1528/sip:1528@192.1 5510834437 00:04:40 E1
DAHDI/10-1 macro-dialout-trunk s 26 Up Dial DAHDI/G0/5552654848,300,T 5551302800 00:03:51 E1 E1 73796caa-beff-43fe-a
PJSIP/1423-00000040 func-apply-sipheader s 10 Down Return (Empty) 1423 00:00:37 E1 E1
Local/901128@zulu-mo zulu-mobile-phone-pu 901128 1 Ring Stasis zulu-mobile-call-push-pro 9611807869 00:00:00 E1 E1
Local/901128@zulu-mo zulu-mobile-phone-pu 901128 1 Down AppDial (Outgoing Line) 901128 00:00:00 E1 E1
PJSIP/2813-0000003b func-apply-sipheader s 10 Down Return (Empty) 2813 00:01:01 E1 E1
PJSIP/2813-0000004c func-apply-sipheader s 10 Down Return (Empty) 2813 00:00:00 E1 E1
Local/901635@zulu-mo zulu-mobile-phone-pu 901635 1 Down AppDial (Outgoing Line) 901635 00:00:02 E1 E1
Local/901635@zulu-mo zulu-mobile-phone-pu 901635 1 Ring Stasis zulu-mobile-call-push-pro 6671380230 00:00:02 E1 E1
Local/901635@zulu-ca zulu-call 901635 5 Ring Dial Local/901635@zulu-mobile- 6671380230 00:00:02 E1 E1
Local/901635@zulu-ca zulu-call 112 1 Ringing AppDial (Outgoing Line) 112 00:00:02 E1 E1
Local/901128@zulu-mo zulu-mobile-phone-re 901128 1 Ring Stasis zulu-mobile-call-register 9611807869 00:00:00 E1 E1
DAHDI/2-1 macro-dial s 65 Up Dial PJSIP/1128/sip:1128@192.1 9611807869 00:01:05 E1
Local/901128@zulu-mo zulu-mobile-phone-re 901128 1 Down AppDial (Outgoing Line) 901128 00:00:00 E1 E1
DAHDI/13-1 macro-dial s 65 Up Dial PJSIP/2813/sip:2813@192.1 6144294000 00:00:47 E1
Local/902813@zulu-mo zulu-mobile-phone-pu 902813 1 Ring Stasis zulu-mobile-call-push-pro 6144294000 00:00:00 E1 E1
Local/902813@zulu-mo zulu-mobile-phone-pu 902813 1 Down AppDial (Outgoing Line) 902813 00:00:00 E1 E1
PJSIP/1528-0000003a func-apply-sipheader s 10 Down Return (Empty) 1528 00:01:04 E1 E1
DAHDI/61-1 from-digital 1 Up AppDial (Outgoing Line) 05552654848 00:03:39 E1 E1 73796caa-beff-43fe-a
DAHDI/9-1 macro-dial-one s 59 Up Dial PJSIP/1000/sip:1000@192.1 9848033447 00:02:22 E1
Local/901128@zulu-mo zulu-mobile-phone-wa 901128 1 Down AppDial (Outgoing Line) 901128 00:00:00 E1 E1
Local/901128@zulu-mo zulu-mobile-phone-wa 901128 1 Ring Stasis zulu-mobile-call-wait-pro 9611807869 00:00:00 E1 E1
PJSIP/1403-00000045 func-apply-sipheader s 10 Down Return (Empty) 1403 00:00:21 E1 E1
Local/901128@zulu-de zulu-desktop-phone 901128 1 Ring Stasis zulu-desktop-call-process 9611807869 00:00:00 E1 E1
Local/901128@zulu-de zulu-desktop-phone 901128 1 Down AppDial (Outgoing Line) 901128 00:00:00 E1 E1
PJSIP/1049-0000003d func-apply-sipheader s 10 Down Return (Empty) 1049 00:00:49 E1 E1
Local/902813@zulu-mo zulu-mobile-phone-wa 902813 1 Ring Stasis zulu-mobile-call-wait-pro 6144294000 00:00:00 E1 E1
Local/902813@zulu-mo zulu-mobile-phone-wa 902813 1 Down AppDial (Outgoing Line) 902813 00:00:00 E1 E1
-------- snip --------------

Any ideas? Everything is up to date on the server and it had been working fine till today.

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@trixie_no5 wrote:

Each PJSIP phone connection is duplicated in ‘pjsip list contacts’ from CLI, two entries per phone. Voice all working OK, web admin Asterisk -> Info -> Chan_PJSip Endpoints shows no duplicates, is this normal?

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@clyde277 wrote:

Hi there,

I was wondering if there was anyone who can help me figure this problem out.

On the client side I have a mikrotik CC1009, with SIP ALG turned off. Extensions are configured to use NAT on the freePBX side and the NAT mode is set to (yes, force_rport, comedia) for all extensions. The freePBX server has only one Public IP assigned to it, and I am happy with that setup.

On the client side, we have two ISP’s, both with static public IP’s. I have configured the fail-over redundancy on the mikrotik and everything works perfectly; when the primary ISP goes down the secondary ISP picks up within seconds.

However, when the primary ISP goes down, phones won’t register using the secondary ISP. I’ve seen the sip debug logs on asterisk and the server keeps sending NAT requests to the primary ISP, and doesn’t switch over to the secondary ISP. Here is the example of what I am talking about:

[2019-10-04 11:06:47] VERBOSE[2578] chan_sip.c: Retransmitting #1 (NAT) to xx.xx.xxx.xxx:1250:
OPTIONS sip:410@10.10.32.195:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xx.xxx:5060;branch=z9hG4bK03f04c57;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@xxx.xxx.xx.xxx;tag=as5078d144
To: sip:410@10.10.32.195:5060
Contact: sip:Unknown@xxx.xxx.xx.xxx:5060
Call-ID: 2f62f28912172b437e37eb8e4f3e90bc@xxx.xxx.xx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.195.22(11.25.3)
Date: Fri, 04 Oct 2019 16:06:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

What I’ve tried so far and no luck:
Used SIP ALG and disabled NAT mode on the extension.
Used TCP/TLS for SIP as opposed to UDP.
Turned on keep alive on the phone side.

I am out of ideas, any help would be much appreciated!

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@ncaridi wrote:

Hi ,
I’m trying to upgrade my Fbpx 13 to 15.
when running the Restore using the backup restore and choose restore legacy CDR I get the following error :
In Process.php line 1335:
The process “zcat /tmp/backup/b3355818-6911-44d9-a57f-a7a4a22a0ee8/mysql-4.
sql.gz | mysql -u freepbxuser -pPassword asteriskcdrdb” exceeded the ti
meout of 300 seconds.

is this an issue of a CDR db being too big ?
can I adjust the timeout ?

Thank you .

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@Bradbpw wrote:

Some of my users are reporting poor audio quality (jumbled or broken audio) or calls dropping altogether. It’s pretty rare, but it is happening. When this happens, is there a best-practices procedure to find the culprit? Is there a log that may show me the issue or a test that I can run?

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