@tharmon77 wrote:

I have a real estate client and what they want is this:

1. Dial into the system to an extension, log into that extension
2. Change destination based on time schedule to go to an extension
3. Change the FMFM destionation OR call forward all destinations to an external number.
4. When shift is over, revert back to another extension or change destional.

Reason: Agents will not know who is on call until that morning usually Saturday or Sunday due to either a showing service has booked someone, and when they are on schedule, they dial into the system, change the mode to forward to a “On call” extension, log into that extension, change the call forward all destionation matching their cell phone.

When shift is over, change the mode back and have it to go to IVR.

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@Scion wrote:

Greetings,

What would be the best way for a call center to remain compliant with FCC do not call registry using FreePBX?

Thanks in advance for helpful answers.

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@elmenyawy wrote:

We are using Bria 5 the licensed version with Freepbx 15. Calls working fine with no issues but presence doesn’t work. Whenever we change the status from Bria nothing happens and users in directory are still showns as “Available”.

Checked the pcap when changing the presence status and I can see Bria sending a PUBLISH request to Freepbx which is responding first with 401 Unauthorized. Then Bria sens again a PUBLISH request with the new presence state and Freepbx responds with 489 Bad Event.

Please let me know if there is something I can do to fix that or there is a module needs to be purchased in order to have that working. Also, if it is a compatibility issue, what other solutions are guaranteed to work?

Please find the pcap output below:
(Bria’s computer IP is 192.168.239.1 and Freepbx’s IP is 192.168.239.133)

No. Time Source Destination Protocol Length Info
13 5.959568 192.168.239.1 192.168.239.133 SIP/XML 1051 Request: PUBLISH sip:500@192.168.239.133 |

Frame 13: 1051 bytes on wire (8408 bits), 1051 bytes captured (8408 bits) on interface 0
Ethernet II, Src: Vmware_c0:00:08 (00:50:56:c0:00:08), Dst: Vmware_fd:27:46 (00:0c:29:fd:27:46)
Internet Protocol Version 4, Src: 192.168.239.1, Dst: 192.168.239.133
User Datagram Protocol, Src Port: 54369, Dst Port: 5060
Session Initiation Protocol (PUBLISH)
Request-Line: PUBLISH sip:500@192.168.239.133 SIP/2.0
Method: PUBLISH
Request-URI: sip:500@192.168.239.133
Request-URI User Part: 500
Request-URI Host Part: 192.168.239.133
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.239.1:54369;branch=z9hG4bK-524287-1—a0adb84387f05b00;rport
Transport: UDP
Sent-by Address: 192.168.239.1
Sent-by port: 54369
Branch: z9hG4bK-524287-1—a0adb84387f05b00
RPort: rport
Max-Forwards: 70
Contact: sip:500@192.168.239.1:54369
Contact URI: sip:500@192.168.239.1:54369
To: sip:500@192.168.239.133
SIP to address: sip:500@192.168.239.133
From: "test1"sip:500@192.168.239.133;tag=b9337e17
SIP Display info: “test1”
SIP from address: sip:500@192.168.239.133
SIP from tag: b9337e17
Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI
[Generated Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI]
CSeq: 1 PUBLISH
Sequence Number: 1
Method: PUBLISH
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/pidf+xml
User-Agent: Bria 5 release 5.6.2 stamp 99262
Event: presence
Content-Length: 445
Message Body
eXtensible Markup Language
<?xml
version=“1.0”
encoding=“UTF-8”
?>




open



<dm:person
id=“75e4ef07f933e640da”>
dm:note
On the phone
</dm:note>
rpid:activities
rpid:on-the-phone/
</rpid:activities>
</dm:person>

No. Time Source Destination Protocol Length Info
14 5.960511 192.168.239.133 192.168.239.1 SIP 587 Status: 401 Unauthorized |

Frame 14: 587 bytes on wire (4696 bits), 587 bytes captured (4696 bits) on interface 0
Ethernet II, Src: Vmware_fd:27:46 (00:0c:29:fd:27:46), Dst: Vmware_c0:00:08 (00:50:56:c0:00:08)
Internet Protocol Version 4, Src: 192.168.239.133, Dst: 192.168.239.1
User Datagram Protocol, Src Port: 5060, Dst Port: 54369
Session Initiation Protocol (401)
Status-Line: SIP/2.0 401 Unauthorized
Status-Code: 401
[Resent Packet: False]
[Request Frame: 13]
[Response Time (ms): 0]
Message Header
Via: SIP/2.0/UDP 192.168.239.1:54369;rport=54369;received=192.168.239.1;branch=z9hG4bK-524287-1—a0adb84387f05b00
Transport: UDP
Sent-by Address: 192.168.239.1
Sent-by port: 54369
RPort: 54369
Received: 192.168.239.1
Branch: z9hG4bK-524287-1—a0adb84387f05b00
Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI
[Generated Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI]
From: “test1” sip:500@192.168.239.133;tag=b9337e17
SIP Display info: “test1”
SIP from address: sip:500@192.168.239.133
SIP from tag: b9337e17
To: sip:500@192.168.239.133;tag=z9hG4bK-524287-1—a0adb84387f05b00
SIP to address: sip:500@192.168.239.133
SIP to tag: z9hG4bK-524287-1—a0adb84387f05b00
CSeq: 1 PUBLISH
Sequence Number: 1
Method: PUBLISH
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1600228298/e8883b2b900407f1da8239dd4fce19db”,opaque=“686986073a877d23”,algorithm=md5,qop=“auth”
Authentication Scheme: Digest
Realm: “asterisk”
Nonce Value: “1600228298/e8883b2b900407f1da8239dd4fce19db”
Opaque Value: “686986073a877d23”
Algorithm: md5
QOP: “auth”
Server: FPBX-15.0.16.72(16.11.1)
Content-Length: 0

No. Time Source Destination Protocol Length Info
15 5.970575 192.168.239.1 192.168.239.133 SIP/XML 1335 Request: PUBLISH sip:500@192.168.239.133 |

Frame 15: 1335 bytes on wire (10680 bits), 1335 bytes captured (10680 bits) on interface 0
Ethernet II, Src: Vmware_c0:00:08 (00:50:56:c0:00:08), Dst: Vmware_fd:27:46 (00:0c:29:fd:27:46)
Internet Protocol Version 4, Src: 192.168.239.1, Dst: 192.168.239.133
User Datagram Protocol, Src Port: 54369, Dst Port: 5060
Session Initiation Protocol (PUBLISH)
Request-Line: PUBLISH sip:500@192.168.239.133 SIP/2.0
Method: PUBLISH
Request-URI: sip:500@192.168.239.133
Request-URI User Part: 500
Request-URI Host Part: 192.168.239.133
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.239.1:54369;branch=z9hG4bK-524287-1—013cee25d8138077;rport
Transport: UDP
Sent-by Address: 192.168.239.1
Sent-by port: 54369
Branch: z9hG4bK-524287-1—013cee25d8138077
RPort: rport
Max-Forwards: 70
Contact: sip:500@192.168.239.1:54369
Contact URI: sip:500@192.168.239.1:54369
To: sip:500@192.168.239.133
SIP to address: sip:500@192.168.239.133
From: “test1"sip:500@192.168.239.133;tag=b9337e17
SIP Display info: “test1”
SIP from address: sip:500@192.168.239.133
SIP from tag: b9337e17
Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI
[Generated Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI]
CSeq: 2 PUBLISH
Sequence Number: 2
Method: PUBLISH
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/pidf+xml
User-Agent: Bria 5 release 5.6.2 stamp 99262
[truncated]Authorization: Digest username=“500”,realm=“asterisk”,nonce=“1600228298/e8883b2b900407f1da8239dd4fce19db”,uri=“sip:500@192.168.239.133”,response=“22e5038dd9d46c5eafee6afa6571b625”,cnonce=“a83ea6555f32ba6f718d40741715ee53”,nc=0
Authentication Scheme: Digest
Username: “500”
Realm: “asterisk”
Nonce Value: “1600228298/e8883b2b900407f1da8239dd4fce19db”
Authentication URI: "sip:500@192.168.239.133”
Digest Authentication Response: “22e5038dd9d46c5eafee6afa6571b625”
CNonce Value: “a83ea6555f32ba6f718d40741715ee53”
Nonce Count: 00000001
QOP: auth
Algorithm: md5
Opaque Value: “686986073a877d23”
Event: presence
Content-Length: 445
Message Body
eXtensible Markup Language
<?xml
version=“1.0”
encoding=“UTF-8”
?>




open



<dm:person
id=“75e4ef07f933e640da”>
dm:note
On the phone
</dm:note>
rpid:activities
rpid:on-the-phone/
</rpid:activities>
</dm:person>

No. Time Source Destination Protocol Length Info
16 5.971489 192.168.239.133 192.168.239.1 SIP 438 Status: 489 Bad Event |

Frame 16: 438 bytes on wire (3504 bits), 438 bytes captured (3504 bits) on interface 0
Ethernet II, Src: Vmware_fd:27:46 (00:0c:29:fd:27:46), Dst: Vmware_c0:00:08 (00:50:56:c0:00:08)
Internet Protocol Version 4, Src: 192.168.239.133, Dst: 192.168.239.1
User Datagram Protocol, Src Port: 5060, Dst Port: 54369
Session Initiation Protocol (489)
Status-Line: SIP/2.0 489 Bad Event
Status-Code: 489
[Resent Packet: False]
[Request Frame: 15]
[Response Time (ms): 0]
Message Header
Via: SIP/2.0/UDP 192.168.239.1:54369;rport=54369;received=192.168.239.1;branch=z9hG4bK-524287-1—013cee25d8138077
Transport: UDP
Sent-by Address: 192.168.239.1
Sent-by port: 54369
RPort: 54369
Received: 192.168.239.1
Branch: z9hG4bK-524287-1—013cee25d8138077
Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI
[Generated Call-ID: 99262YWFhMTA4OTMwNTlmMjkzZWE4NzkyMTM5ZGI3YjM0ODI]
From: “test1” sip:500@192.168.239.133;tag=b9337e17
SIP Display info: “test1”
SIP from address: sip:500@192.168.239.133
SIP from tag: b9337e17
To: sip:500@192.168.239.133;tag=z9hG4bK-524287-1—013cee25d8138077
SIP to address: sip:500@192.168.239.133
SIP to tag: z9hG4bK-524287-1—013cee25d8138077
CSeq: 2 PUBLISH
Sequence Number: 2
Method: PUBLISH
Server: FPBX-15.0.16.72(16.11.1)
Content-Length: 0

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@Mesh wrote:

How can I check what audio codec was used during a session? Asterisk and a SIP phone have a choice of codecs to use. But is there a way to find which codec they actually used for a particular connection?
Is there a log file with this info? I ran Asterisk in verbose but still could’t see codecs in any of the FreePBX’s Asterisk Log Files .

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@dwsiemens wrote:

I think I ran over the edge, the system basically won’t respond, Lots of reloads Like 5 from apply configs where happening.

In the logs i have

[2020-09-16 09:19:57] WARNING[15078][C-00003593] taskprocessor.c: The ‘stasis/p:channel:all-0001a8fc’ task processor queue reached 500 scheduled tasks.

[2020-09-16 09:19:57] WARNING[16500][C-0000355a] taskprocessor.c: The ‘stasis/p:channel:all-0001a8d3’ task processor queue reached 500 scheduled tasks.

[2020-09-16 09:19:57] WARNING[25666] taskprocessor.c: The ‘stasis/p:channel:all-0001a8fe’ task processor queue reached 500 scheduled tasks.

[2020-09-16 09:19:57] WARNING[9091] taskprocessor.c: The ‘stasis/p:channel:all-0001a73a’ task processor queue reached 500 scheduled tasks.

[2020-09-16 09:19:57] WARNING[9091] taskprocessor.c: The ‘stasis/p:channel:all-0001a8e6’ task processor queue reached 500 scheduled tasks.

So what should I tune up is the next question. Load average was about of 13-14.
Sar info
08:30:01 AM all 8.92 0.01 5.38 5.79 0.00 79.90
08:40:01 AM all 9.99 0.01 5.32 6.01 0.00 78.68
08:50:01 AM all 10.01 0.01 5.26 6.46 0.00 78.25
09:00:01 AM all 10.80 0.01 5.63 5.92 0.00 77.64
09:10:01 AM all 12.41 0.01 6.33 5.75 0.00 75.50
09:20:01 AM all 14.13 0.01 6.64 7.70 0.00 71.52
09:30:01 AM all 5.60 0.00 1.57 0.01 0.00 92.81

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@smcdonald0370 wrote:

I have been handed the controls to a FreePBX system, as our previous tech suddenly left the company. My question is, what regular maintenance is recommended to keep the system running clean and error free. I have worked on telephone switches in the past (Nortel, Lucent, Motorola, etc…) but none of them were soft switches. If anyone can point me in the right direction, website help, documentation, I would greatly appreciate it. I honestly do not know much about this system and want to keep any catastrophic events to a minimum, if not completely prevent them. I appreciate any and all suggestions.

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@kwriley87 wrote:

I am running FreePBX 15.0.16.75 using the primary/warm spare setup as documented here: https://wiki.freepbx.org/pages/viewpage.action?pageId=185631299

My nightly backup/restore job on the primary server seems to be getting stuck for some reason. When I attempt to go re-run the job, I am getting the error:

Running with: /usr/sbin/fwconsole backup --backup=‘1f38c816-b792-492f-be5e-1429b472a1d9’ --warmspare --transaction=‘50e96e57-34be-4d91-8561-7ab8b323b4d6’ >> /var/log/asterisk/backup_50e96e57-34be-4d91-8561-7ab8b323b4d6_out.log 2> /var/log/asterisk/backup_50e96e57-34be-4d91-8561-7ab8b323b4d6_err.log & echo $!
The command is already running in another process.

I’ve tried doing an fwconsole restart to no avail. The only way that I’ve been able to re-run the job manually is to reboot the server.

Can anyone point me in the right direction in determining why the backup job is stuck in a running process?

Thank you.

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@myr1de wrote:

Went to go do a module update and I am getting a connection error to both mirror2.freepbx.org and mirror1.freepbx.org.

It appears these are down at the moment. AS well as LE is timing out with the following ‘There was an error updating the certificate: Connection timed out after 10001 milliseconds’ even though ALL ports both in and out are opened up to the FreePBX server.

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@Rom1906 wrote:

Hello everyone, I want to be able to make a call from my website through my FreePBX. I can’t find an API that allows you to create a call by sending the number to call and the destination extension.
Can you help me ?
Thanks

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@myr1de wrote:

Went to go do a module update and I am getting a connection error to both mirror2.freepbx.org and mirror1.freepbx.org.

It appears these are down at the moment. AS well as LE is timing out with the following ‘There was an error updating the certificate: Connection timed out after 10001 milliseconds’ even though ALL ports both in and out are opened up to the FreePBX server.

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@BlueOcean3 wrote:

We got our voicemail setup for inbound calls. Our receptionist sounds like she’s inside of a box, the unedited sound file sounds great.

How do I get the sound quality in my voicemail to sound decent if the FreePBX wav requirements shrink the file down to a low quality?

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@Edge2020 wrote:

Hello Folks,

i need some advice or someone who is putting me in the right direction:
Setup a Freepbx, everything is working, incoming and outgoin calls are working well. Now some people want to redirect incoming calls to their mobile device, so i told them to redirect the call via a setting on their Yealink T54W.
Unfortunalety the call does not get routed to my provider (i’ve asekd them if they send the originate CallerID and they said yes) and freepbx only shows me a
“TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks”) in new stack"
I don’t know what i did wrong. I also changed my outgoing Trunk and the dial pattern to only “.” so it won’t make any Problems at this Point.
Here is the Log from a Call:
https://pastebin.freepbx.org/view/19f9e63e

I hope someone can give me a hint.

Thanks in advance!

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@psdk wrote:

Hi to all,

We are using FPBX in our call center. We have big instances on version 13 with Asterisk 11 without any issues.
Recently we upgraded one of them to version 15 with Asterisk 16.

We installed from ISO and updated all the things.
But after 1 hour working in the production, FPBX UI doesn’t open and everything has lag and also some users receives SIP 488 message when they call out.

FastAgi also is enable.
Server resources is big enough for system capacity.
Average concurrent call is 60.

Do you have any idea what is going on? what would be the problem?

We checked the process and see lots of ttendedtransfer-rec-restart.php.
We comment this line int the function php file and it helped a little.

Please advise.

BR

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@michaelcrapse wrote:

Our SIP trunk provider is charging us for every false outbound attempt by our FreePBX system.
Currently we have several ring groups set up with 10-20 extensions each. Some of them cell phone numbers. When there is an extension that is no longer registered, our system is dialing the trunk with a Prefix of LC+EXT@example.trunk.c
We get 100-200 errors a day when we get a lot of calls, and it’s starting to hit calls per second limits.
The installation is fairly basic, with nothing advanced everything seems to be set up correctly, but i cannot fathom why local unavailable extensions would be called using an outside trunk line.

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@loveman wrote:

Hello everyone,
I am setup IVR and recording voice so When call from outside by gsm the line open but without any Announcement and dial pad not work like without activities, I need to solve it how can that!!
I can call from outside by gsm to inside “local” my server and ring to extension. But I need to enable IVR!
I enable all extension with SIP protocol.
Created Announcement, Created IVR and added the Announcement and added IVR Entries, Created Inbound Route with DID number, and added Set destination to “IVR”.
My gsm devices “Goip 1 sim” And config. below:

And other picture can see it by reply

Thank you

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@johnjces wrote:

In checking my system for general info today, I did an

lspci -knn | egrep -i ‘Eth|net|wire’ -A 3

and I knew my NIC was a realtek 8111/8168/8411. It is using the r8169 driver and works well. However, I do believe that I can use the more appropriate r8168 driver.

Would updating this driver be beneficial? And if it can be updated is it a KMOD driver? and could it be installed with:

yum install kmod-r816*

Would I have to change repositories?

Thanks!

John

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@amardeeprana wrote:

Hi
Asterisk Service Taking High CPU USAGE 100% ( 4 Cores ).

FreePBX 14.0.13.34

Please suggest.

Thanks
Amardeep

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@dwsiemens wrote:

Hi It appears I’m hitting some issues on Task processors,

eg
stasis/m:cache_pattern:1/channel:all-00000678 16272515 0 588 450 500

stasis/m:channel:all-00000679 13036468 0 646 450 500

stasis/m:devicestate:all-00000003 963655 0 734 450 500

should I increase the statis side some more?

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@starzen wrote:

We have been using FreePBX for years and besides phones also have 2 small programs that use the api to monitor queues/calls and one to route calls.

not sure exactly what happened as i wasnt involved but now we are running into an issue with the web interface and api becoming unresponsive after a few hours. Also you cannot change queues from the phones any longer either

calls are still processed normally without any issue

any ideas what i can look for that could cause this?

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@Wallygator67 wrote:

Importing extensions works fine exept that my devinfo_dial field is set to defaul instead of what I need.
ex
in my csv file for extension 103, devinfo_dial = sip/V1086B00G0L0@200.0.1.103 but after importing, dial is set to sip/103…any way to keep my weird dial number?

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