I dont think thta its serious. so will continue on
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Setup pjsip with tls on asterisk 16, freepbx 14
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Unable to receive calls
Logs, Logs always post logs!
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The queue still ring and identified the agent is Not in use. But this agent is offline(unkown/UNREACHABLE)
this is my option:
with Agent Restrictions, I tried all 3 but same.
Watchers 4?
line 309 is not register(i turn off my phone)
the queue still identified agent 309 Not in use like agent online, when 309 is offline.
Any help for situation. I think the point of problem is the “Watchers” not = 0 when the line 309 not register! I need some help!
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Setup pjsip with tls on asterisk 16, freepbx 14
Those can be ignored.
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FreePBX design discussion
Code contributions are always welcome.
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App_voicemail maxsilence should be less than minsecs or you may get empty messages
This is a issue with some sort of default setting,
if you reload app_voicemail you will get a message in console of
voice*CLI> module reload app_voicemail.so
Module ‘app_voicemail.so’ reloaded successfully.
[2020-01-17 02:35:44] WARNING[39573]: app_voicemail.c:14229 actual_load_config: maxsilence should be less than minsecs or you may get empty messages
If I go to config edit, the file is not writeable. any thing that I should do to fix the warning?
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Changing vm-login file from "Comedian Mail, Mailbox" to "Mailbox"
Test it, dial *98 in the PBX.
Get a copy of the vm-login.sln16 file from the directory (/var/lib/asterisk/sounds/en/vm-login.sln16).
Then using WinSCP (https://winscp.net/eng/download.php) We were using AWS so I logged a certificate in Pageant which is in the PuTTY suite (https://www.putty.org/). I was able to log in.
After getting a copy of that file I had to modify it.
I used Audacity (https://www.audacityteam.org/download/) to modify the existing file. (/var/lib/asterisk/sounds/en/vm-login.sln16).
Import it using RAW data, you have to change the Sample rate to 16000. I cut off the comedian portion and saved it in a .wav file format.
Change directory permissions and ownership if needed (use the id command in linux to find your group (mine was ec2-user).
These were the commands i used for permissions and ownership of the directory:
sudo chgrp ec2-user /var/lib/asterisk/sounds/en/
sudo chmod 775 /var/lib/asterisk/sounds/en
sudo chown ec2-user /var/lib/asterisk/sounds/en
Using WinSCP backup all of the vm-login.* files to your machine just in case!
I found out we currently only use the vm-login.ulaw file.
Execute this command: sox vm-login.wav --rate 8000 --channels 1 --type ul vm-login_new.ulaw lowpass 3400 highpass 300
Backup the current file: cp vm-login.ulaw vm-login_old.ulaw
Delete it: rm vm-login.ulaw
Copy the new one: cp vm-login_new.ulaw vm-login.ulaw
Done! Dial *98 to test it.
Then cleanup your temp file: rm vm-login_new.ulaw
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Changing vm-login file from "Comedian Mail, Mailbox" to "Mailbox"
these steps will work! People never make it clear!
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Changing vm-login file from "Comedian Mail, Mailbox" to "Mailbox"
(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)
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App_voicemail maxsilence should be less than minsecs or you may get empty messages
That has been there forever. I’ve never bothered to check if there is an open Asterisk bug on this or not.
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Function PJSIP_HEADER not registered
Hi all,
Wasn’t there a module update for SIP a few days ago ? I have noticed that systems that only use ChanSIP have suddenly started showing the following error in the Full log from about 13 Jan20:
ERROR[22850][C-00000000] pbx_functions.c: Function PJSIP_HEADER not registered
They come in blocks of 12 and could be inbound calls. I have been through all the settings and nothing has changed for a couple of years.
Given that PJSIP is not available on these boxes I am a little curious on how to rollback to an older module to check.
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Pi 4 Distro with a working Endpoint Manager?
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Users and Extensions
Thanks for the suggestions. We are using chan_pjsip.
Wouldn’t it be best to have 1 extension number and share it among all their devices and soft phones?
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Users and Extensions
That is what option 3, above means.
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Users and Extensions
@sorvani thanks. One part was not clear to me:
How do I increase the “extension limit”? So if a user has 4 extensions currently I would increase to 6? Why is this?
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Users and Extensions
How does voicemail work in this situation? It seems to me that voicemail is extension specific and not user specific, or am I mistaken?
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Yealink T4XG phones will not autoprovision over HTTPS with FreePBX 14
this also affects FreePBX 15.
- I just performed a new install of FreepBX 15
- obtained a LE cert within certman 15.0.18
- set cert as active in apache
- updated by DHCP to point to the new PBX for boot files
- defaulted a Yealink T42G
This is all the shows up in the PBX log. Same as in FreePBX 14. HTTP error 408.
[jbusch@freepbx tftpboot]$ sudo tail /var/log/httpd/access_log
64.53.199.99 - - [17/Jan/2020:04:15:41 +0000] "GET /admin/ajax.php?module=search&command=global HTTP/1.1" 200 9427 "https://pbx15.bundystl.com/admin/config.php?display=sipsettings" "Mozilla/5.0 (X11; Fedora; Linux x86_64; rv:72.0) Gecko/20100101 Firefox/72.0"
64.53.199.99 - - [17/Jan/2020:04:15:56 +0000] "POST /admin/config.php?display=sipsettings HTTP/1.1" 200 135281 "https://pbx15.bundystl.com/admin/config.php?display=sipsettings" "Mozilla/5.0 (X11; Fedora; Linux x86_64; rv:72.0) Gecko/20100101 Firefox/72.0"
64.53.199.99 - - [17/Jan/2020:04:15:56 +0000] "GET /admin/assets/less/cache/lessphp_9e018ea8f09ca04d937648f01e33eb196b09ae1d.css HTTP/1.1" 200 87893 "https://pbx15.bundystl.com/admin/config.php?display=sipsettings" "Mozilla/5.0 (X11; Fedora; Linux x86_64; rv:72.0) Gecko/20100101 Firefox/72.0"
64.53.199.99 - - [17/Jan/2020:04:15:57 +0000] "GET /admin/ajax.php?module=search&command=global HTTP/1.1" 200 9427 "https://pbx15.bundystl.com/admin/config.php?display=sipsettings" "Mozilla/5.0 (X11; Fedora; Linux x86_64; rv:72.0) Gecko/20100101 Firefox/72.0"
64.53.199.99 - - [17/Jan/2020:04:15:59 +0000] "POST /admin/ajax.php?command=reload HTTP/1.1" 200 93 "https://pbx15.bundystl.com/admin/config.php?display=sipsettings" "Mozilla/5.0 (X11; Fedora; Linux x86_64; rv:72.0) Gecko/20100101 Firefox/72.0"
64.53.199.99 - - [17/Jan/2020:04:25:35 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:26:23 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:26:55 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:27:23 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:32:06 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:32:35 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:33:04 +0000] "-" 408 - "-" "-"
64.53.199.99 - - [17/Jan/2020:04:33:34 +0000] "-" 408 - "-" "-"
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Using analogue line for Trunk
I would like to integrate my current ADSL phone line and connect it so that it calls my VoIP phones using inbound and outbound trunks, just like a SIP Trunk from a VoIP provider. After doing some research I found out that I need a special VoIP ATA. I know that this may not be advised by anyone however it would be great if anyone could help me if anyone knows how to do this.
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System Recordings not working
This is the output when I ran ls -i /var/lib/asterisk/sounds | grep en
drwxrwxr-x 3 asterisk asterisk 4096 Jan 13 04:04 en_US
drwxrwxr-x 3 asterisk asterisk 266240 Jan 15 03:08 en_GB
drwxrwxr-x 3 asterisk asterisk 4096 Jan 13 03:59 en
Running the commands below do not output anything as there is nothing in the directory to view its permissions.
li -i /var/lib/asterisk/sounds/custom | grep en
li -i /var/lib/asterisk/sounds/en/custom | grep en
li -i /var/lib/asterisk/sounds/en_GB/custom | grep en
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Unable to receive calls
You are describing an issue which seems to be an issue with the register string setting in the incoming section of the trunk. What is the format with the register string under the incoming tab? It should be something that is supported by your SIP Provider. Normally this is Username:Password@SIPProviderHost/Username_Or_DID. The last part of the string (Username_Or_DID) may not be needed, however with the SIP Trunks that I sell, I require clients to enter this for multiple numbers being registered on one PBX.
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