@cdsJerryw wrote:

I did several module updates this week as they became available so I'm not sure which one made the change. However since doing the updates incoming calls now result in multiple lines ringing with that incoming call to a number with Follow-me active.

Here are the details: If follow-me is active with RingallV2 and the extension itself is NOT listed in the follow-me group the extension rings once. The other numbers in the group ring as expected.

If follow-me is on and the extension IS listed in the follow-me group, TWO incoming calls come to the extension and BOTH continue to ring until answered. The other numbers in the group ring as expected. In the past, only one line would ring on the extension, not two.

FreePBX 13.0.192.16 All modules are running the most current version available.

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@Trezeguet17 wrote:

Hi people today im having this big big real problem with fop2 , im trying to access to it trough my public adress to the server of fop2 , but when i give click on it , it stays loading eternally by finally say : System unavailable , this is the output from my test i did

/usr/local/fop2/fop2_server --test
Flash Operator Panel 2 - Valid License (1)
Connection to manager OK!

but i stil can load anything

this is the access log
1xx.25.5x.9x -my enterprrise [23/Aug/2017:14:27:15 -0500] "POST /fop2/setvar.php HTTP/1.1" 200 6 "http://x8x.4x17x.125:51050/fop2/" "Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/62.0.3178.0 Safari/537.36"

seems that i can acces , but i cant does it have something to do with asterisk_manager?

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@Trezeguet17 wrote:

Hi im trying to call from my extensions to another city , where i have another server with free pbx , i call from xxx extension to xxx extension and i cant connect the call , i was just making a debug to my extension and i enabled Nat on the config of my extension and this is the output of the debug i put

• Executing [continue@macro-dialout-trunk:4] Set("SIP/311-b75812c0", "CALLERID(number)=311") in new stack
  -- Executing [1173@from-internal:6] Macro("SIP/311-b75812c0", "outisbusy|") in new stack
  -- Executing [s@macro-outisbusy:1] Progress("SIP/311-b75812c0", "") in new stack
  Audio is at 172.16.1.5 port 15618
  Adding codec 0x4 (ulaw) to SDP
  Adding codec 0x8 (alaw) to SDP
  Adding codec 0x2 (gsm) to SDP
  Adding non-codec 0x1 (telephone-event) to SDP
  SURENVIOS-NEIVA*CLI>
  <--- Transmitting (NAT) to 172.16.1.111:5060 --->
  SIP/2.0 183 Session Progress
  Via: SIP/2.0/UDP 172.16.1.111:5060;branch=z9hG4bKbe3949a09376695a;received=172.16.1.111
  From: "Aux Fact y Gestion2" ;tag=688dba7f557f9678
  To: ;tag=as421553a0
  Call-ID: 3a2a70dc368f2f7f@172.16.1.111
  CSeq: 58952 INVITE
  User-Agent: FPBX-2.9.0(1.4.26)
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  Supported: replaces
  Contact:
  Content-Type: application/sdp
  Content-Length: 281

v=0
o=root 2952 2952 IN IP4 172.16.1.5
s=session
c=IN IP4 172.16.1.5
t=0 0
m=audio 15618 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- Executing [s@macro-outisbusy:2] Playback("SIP/311-b75812c0", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'es')
SURENVIOS-NEIVA*CLI>
<--- SIP read from 172.16.1.111:5060 --->
CANCEL sip:1173@172.16.1.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.111:5060;branch=z9hG4bKbe3949a09376695a
From: "Aux Fact y Gestion2" ;tag=688dba7f557f9678
To:
Supported: path
Call-ID: 3a2a70dc368f2f7f@172.16.1.111
CSeq: 58952 CANCEL
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

can some one please help me ? the trunk its fine but i cant make any call to remote extensions, those extensions are sip

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@philipt wrote:

I work for a school using a FreePBX server. They had an old phone system that they could dial * when an outside line was ringing, and it would pick it up instantly if they were in the call group or not. This allowed the Administrator to be in her office and not distrubed by the phone when she shut her door, but if the phone was ringing and there was no one to answer, she could just dial * and pick it up.

Is there any way to make a feature code in which she could dial something like ## and pickup a call coming from the sip provider? I had her dial **100 followed by send and it worked some of the time, but sometimes put the call on park.

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@stevensedory wrote:

Running FreePBX 13.0.192.16 and Asterisk 13.17.0

Recently used the "warm spare" method to move to a new server (new VM on KVM/proxmox)

The server has about 120 remote extension, and had no real problems before.

I posted about this crash issue yesterday here, but my hypothesis was off: https://community.freepbx.org/t/media-index-c-failed-to-stat/43645

Today we had a ton of users call and say their phones weren't working. Funny thing is, they show OK with IP address in peers list when running "sip show peers" in cli.

So we did a fwconsole restart, and things started working again.

This crash has happened three times this week already. Sunday morning, yesterday morning, and today.

I starting digging through the logs, and these are the errors that may or may not be the cause. I'm hoping someone can give me some insight. Here are some error examples:

These ones show all over the logs, way before, way after, and right around the crash time:

[2017-08-22 09:30:59] ERROR[32499][C-00000028] pbx_functions.c: Function PJSIP_HEADER not registered

These ones yesterday were fairly close to before the crash, but there were none today before the crash:

[2017-08-22 09:38:08] ERROR[1488] netsock2.c: getaddrinfo("2605:e000:6045:3a00:20b:82ff:feac:c151:13312", "(null)", ...): Name or service not known
[2017-08-22 09:38:08] WARNING[1488] chan_sip.c: Could not resolve socket address for '2605:e000:6045:3a00:20b:82ff:feac:c151:13312'

These existing on all three instances:

Line 33144: [2017-08-20 10:34:38] ERROR[30555] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data

And finally, these ones look the like the most likely culprit, but didn't show on Yesterday's crash (just today and Sunday's):
*note that the difference between Today's and Sunday's, vs Yesterday's, is that the former showed all endpoints "OK", though they truly weren't, the latter showed only about half of them

[2017-08-23 13:42:49] ERROR[26004] astobj2.c: FRACK!, Failed assertion bad magic number 0x0 for object 0x3de7430 (0)

So all that to say, I hope someone can help us find the root cause of all this.

Again, this server was a fresh v13 FreePBX server that we just "warm spare" copied to from an existing server. The existing was running on an ESXi host, fully updated to ....66-21. We fully updated the fresh VM to 66-21 as well before running the backup/restore. The new server is a VM on KVM/proxmox.

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@jgiebler wrote:

I'm sure If I could figure out how to search for this question better, I would find an answer already out there. However I cannot.

I have an operator group that is responsible for answering the phones. Their phones need to ring 4 times (26 seconds) first. If they do not answer, their phones need to continue to ring but then a second operator group's phones will ring for an additional 3 rings (18 Seconds). If, after a total of 7 rings (42 seconds) the call is not answered, it is sent to an IVR.

I am able to do the initial ring group and IVR. However, I am having a challenge bringing in the second operator group after 4 rings.

I have tried setting up a queue and after 24 seconds it goes to another queue that includes both Operator groups and then goes to IVR after 18 seconds. I have also tried this with ring groups.

The issue is that when the first queue/ring group completes after 24 seconds the main operator phones stop ringing briefly. They then start ringing again. This is confusing because it appears that there is a second caller. However it is the first call just moving queues/groups.

Any thoughts/help would be grateful

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@lrgt2003 wrote:

My box is showing an unexplained surge in CPU usage. The system is now running at 70-99% CPU at idle, and with 1 call, it's always 99%, and the call quality has degraded immensely.

Worked fine, and then all of a sudden this happened.

Both of the following log files are over 1 GIGABYTE EACH, and growing quick. Is it ok to delete them, and start fresh?......
/var/log/asterisk/full
/var/log/asterisk/full.0

Thanks!

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@regev wrote:

Hey
we have installed asterisk 13.16 with freepbx 12.0-
and i cant see the cdr it doenst find anything

here is the log :

[2017-08-24 09:35:09] NOTICE[7790] res_odbc.c: Registered ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]
[2017-08-24 09:35:09] NOTICE[7790] res_odbc.c: Registered ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]
[2017-08-24 09:35:19] WARNING[7790] cdr_adaptive_odbc.c: Unable to load cdr_adaptive_odbc.conf. No adaptive ODBC CDRs.
[2017-08-24 09:35:19] WARNING[7790] cdr_adaptive_odbc.c: Unable to load cdr_adaptive_odbc.conf. No adaptive ODBC CDRs.
[2017-08-24 09:35:50] ERROR[7799] cdr_custom.c: Unable to load cdr_custom.conf. Not logging custom CSV CDRs.
[2017-08-24 09:35:50] ERROR[7799] cdr_custom.c: Unable to load cdr_custom.conf. Not logging custom CSV CDRs.
[2017-08-24 09:35:50] WARNING[7799] cdr_adaptive_odbc.c: Unable to load cdr_adaptive_odbc.conf. No adaptive ODBC CDRs.
[2017-08-24 09:35:50] WARNING[7799] cdr_adaptive_odbc.c: Unable to load cdr_adaptive_odbc.conf. No adaptive ODBC CDRs.
[2017-08-24 09:35:50] WARNING[7799] cel_odbc.c: No such connection 'asteriskcdrdb' in the 'cel' section of cel_odbc.conf. Check res_odbc.conf.
[2017-08-24 09:35:50] WARNING[7799] cel_odbc.c: No such connection 'asteriskcdrdb' in the 'cel' section of cel_odbc.conf. Check res_odbc.conf.

when i added the file
/etc/asterisk/cdr_custom.conf
as suggested like this :

;
;
; Mappings for custom config file
;
; To get your CSV output in a format tailored to your liking, uncomment the
; following lines and look for the output in the cdr-custom directory (usually
; in /var/log/asterisk). Depending on which mapping you uncomment, you may see
; Master.csv, Simple.csv, or both.
;
[mappings]
Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)

; High Resolution Time for billsec and duration fields
Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)
Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}

then i get this error and still CDR is blank :
pbx_functions.c: Can't find trailing parenthesis for function 'CDR(dst'?

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@PitzKey wrote:

Hello,

We are running:
FreePBX 10.0.192.16
UCP: 13.0.42.2
User Management: 13.0.76.27
XMPP: 13.0.17.4

We can't remove XMPP in UCP, we already checked for updates, tried to disable and re-enable XMPP in module admin, and Turned on and Off in userman, won't go away.
It's set to NO in the Group and in the User settings,still shows up.

Anyone?

Thanks

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@jcroy wrote:

I'm trying to use a custom ringtone on a installation, I uploaded the wav file using the endpoint manager (custom ringtone section). That part works but it seems to be changing the file in the process. On my computer I edited to speed it up and double ring. It sounds just the way I wanted it on my computer, but when I upload it its like nothing changed. Ideas? Thoughts?

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@mitterhuemer wrote:

Hello,
on our SNG7 FPBX14 our CDR Module show the wrong time Zones.

We want to Show CET+1 but the Module shows all times +2 Hours too much.

The times in PHP and inside the Distro are correct.

How can i tell the CDR module to use the correct time?

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@sentinelace wrote:

I have a wav file I am trying to use in my IVR. The logs show it's playing but its dead air. I converted it to mono with same results. I would like to have the highest possible quality as well. What is the process of converting the file so I get the best quality and supported method. I have tried using audicity with no luck

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@JacobWoods wrote:

Setup FreePBX 14 on CentOS 7 following this guide:
https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+CentOS+7#InstallingFreePBX14onCentOS7

While in /usr/src/freepbx and attempting to run the last command, ./install -n, I received the error displayed in the provided screenshot. When I navigate to IP of FreePBX in a web browser, I get the Apache HTTP Server Test Page. Assuming that is because FreePBX did not install. I'm thinking I may have a typo in the asterisk database name or something like that. Any help would be greatly appreciated. I'm a Linux noob.

Let me know if I need to post any other info!

error.PNG1035x437 7.73 KB

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@minkhant wrote:

Hi there,
I have a little problem fro Freepbx Dashboard error.I want to normal configuration in the dashboard.How can I solve this problem.

dashboard.PNG1259x542 42.3 KB

Freepbx version 13
Asterisk 13

Kind Regards.

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@chacaman wrote:

Hi, thanks everyone for your time, I need to create a custom context which will record the calls of the other context would be something like sub-record-check context, I need this context to be executed in all the context that pbx has, where I can Get an example?

for example
fallowme
conference
out call
in call
internal calls
queue

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@ashleydrees wrote:

i have FPBX 14 + Asterisk 13 (built with the patch for call manager) and i am trying to get the DTMF Signalling working on Comedian mail, when calling out to any IVR though my IAX2 trunk to voiptalk.org - or via my PSTN line - i can use the RFC2833 DTMF Signalling without any issue - but when i call the Comedian Mail - i cannot use the buttons to manage it - so no listening to voicemails / managing the Comedian using the handset. (I had this issue before applying the call manager patch).

Not so much of an issue as it does email me the messages, and i can get them via the UCP, but it would be nice to have it working. The handset seems to be working normally overall and can make and receive both video and voice calls.

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@adminsti wrote:

FreePBX 13.0.192.8 and A101D - IVR not recognizing DTMF tones on some inbound calls.

I have tried enabling/disabling DTMF detection and removal both with no noticeable effect.

I have tried to monitor in log, but I don't see any DTMF entries. I did verify that it is enabled under Asterisk Logfile Settings.

Sometimes it acts like it doesn't receive anything and other times it will send the call to a different extension or IVR menu.

Thanks,
Steve

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@bigmillz wrote:

Finally trying to switch us to PJSIP, because I don't like everyone needing a desk as well as mobile extension. I'd rather they just have a single one that they can use on multiple devices simultaneously.

Thing is, I've switched us to PJSIP, and upon testing the mobile extension part of my plan, I realized there are a couple of issues:

1. We use DND at night on our desk phones - but when the mobile extension logs on, and somebody calls it, Asterisk reports back that the extension is on DND.

2. When both devices (desk and mobile) are logged into the same extension, only the last one registered rings (would prefer both if possible), and upon disconnecting the last one, the original extension has to re-register to be reachable again.

Suggestions on these, or do I go back to having 2 extensions for everyone?

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@dcitelecom wrote:

I want to downgrade our Vultr VPS but they don't allow downgrades so I need to make a backup, destroy the VPS and get a new one on the same server. We can keep the same IP but the machine ID or hardware ID might change ao I am afraid we'll lose the FreePBX deployment activation. Is there any way to avoid that. I have already used up most of my Zend resets when we had to rebuild the server after it crashed.

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@SkydiverFL wrote:

Not sure how valuable this is here, but I've whipped up a simple app to receive SMS text messages from Flowroute numbers and forward them via email. Assuming you're using Docker on a public server, the command is:

docker run --name flowroute-proxy -p 3000:3000 \
    -e TO_EMAIL=bruce@batmail.com \
    -e SMTP_PASS=robin4ever \
    -e SMTP_USER=bruce@batcave.com \
    -e SMTP_HOST=smtp.batcave.com
    fredlackey/flowroute-proxy

Full information is on the Docker Hub at: https://hub.docker.com/r/fredlackey/flowroute-proxy/

The source code is on GitHub at: https://github.com/FredLackey/flowroute-sms-email-proxy

And, there's an even simpler blog entry at: http://www.fredlackey.com/flowroute-proxy/

I hope this helps someone.

-- Fred

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