If the default Polycom password of 456 does not work, or if someone has changed the admin password on the phone, please do the following:
Find and write down the MAC address (serial number) of the phone you want to reset. It should be twelve characters, and look something like ‘0004F2ABCDEF’. If you can’t read the back label, you can find the MAC address by pressing Menu, Status, Network, Ethernet.
Power down the phone.
Power up the phone.
While powering up the phone (you have about 6-8 seconds to complete this step):
For SoundPoint IP 320, 321, 330. 331, 335, 430, and 450 press and hold the 1, 3, 5, and 7 on the dial pad at the same time.
For SoundPoint IP 301, 501, 550, 600, 601, and 650 press and hold the 4, 6, 8, * on the dial pad at the same time.
After holding down the numbers for few second, you will be prompted to enter the admin password. Enter the MAC address of the phone. No colons and the alpha characters must be entered as lowercase letters
The Set will restart. You may need to restart again to get access to the menus with 456 password.
Leo D’Alessandro, Product Marketing Manager at Sangoma, and Frederic Dickey, VP of Product Management at Sangoma, will in this webinar explain how to build an efficient contact center cost-effectively with Sangoma’s FreePBX / PBXact UC.
In this webinar, you’ll learn how the many ways FreePBX / PBXact UC can solve your contact center requirements:
• How calls are best routed using call queues
• Maximizing Agent Productivity and Customer Satisfaction with automated Queue Callbacks
• Integration with desktop and CRM
• Monitoring live call metrics
• Reporting tools to analyze overall performance
Hi just sharing a simple bit of dialplan to catch anon callers ip addresses when using freepbx and Anonymous callers is set to yes, which is needed for some suppliers.
Normally I would say lock your firewall to only known IPs, but in some cases this isn’t possible
Im sure if you have a Asterisk server with a public IP you will have seen calls on the console screen where the call is to a destination but the callers are exten@yourserver . Well this little bit of dialplan at the end of you default sip context should catch them and log them with the ip of the originating server
In extensions_custom.conf add the dialplan below
[catchall]exten => s,1,Noop(Dead calls rising)exten => s,n,Set(uri=${SIPCHANINFO(uri)})exten => s,n,Verbose(3,Unknown call from ${uri} to ${EXTEN})exten => s,n,System(echo "[${STRFTIME(${EPOCH},,%b %d %H:%M:%S)}] SECURITY[] Unknown Call from ${CALLERIDNUM} to ${FROM_DID} IPdetails ${uri}" >> /var/log/asterisk/sipsec.log)exten => s,n,Hangup()
Then in Custom Destinations add a destination as catchall,s,1
so you now get in your logs
[May 1 00:11:06] SECURITY[] Unknown Call from to 900441516014742 IPdetails sip:101@37.75.209.113:21896
I hope this is some help to you, It allows other scripts to pick up this address and add it to your firewall.
BT have confirmed that their recent outage has been ‘resolved and services restored’.
We can also confirm this as we have slowly seen all customer alarms clearing. As many customers are aware that we operate a 24×7 monitoring platform so saw this issue start and checked that there was nothing we could do in most cases but also contacted key customers to warn them that they might be issues.
Therefore, any issues that Customers have experienced this morning when connecting to services using BT connectivity (including quality issues) should now be resolved. In the event that issues are still occurring, please reboot equipment on the BT line such as Firewalls or Routers and retest.
If you have any questions whatsoever please do not hesitate to contact us, Also if you are a
Asterisk / Freepbx reseller or user and would like affordable monitoring please get in touch as we provide Asterisk Monitoring from £25 per year.
Introducing the FreePBX appliance! The FreePBX appliance is a purpose-built, high-performance PBX solution. Designed and rigorously tested for optimal performance, this is the only officially supported hardware solution for FreePBX. The appliance comes preloaded with the FreePBX Distro and includes a one-year warranty!
Featuring the FreePBX Distro, this appliance is an ideal fit for businesses looking to get more from a PBX. With millions of deployments throughout the world, FreePBX is relied upon daily by everyone from enterprises to startups. Leveraging the power of FreePBX has enabled businesses to grow while keeping communication expenses minimal. The FreePBX Distro has made deploying, configuring and using a PBX system easier than ever! With an easy-to-use GUI (Graphical User Interface), getting started is a breeze!
We have noted that with some Fortigate routers and firewalls come with SIP helpers enabled by default.
The customer may initially not think that there is any issue and inbound and outbound calls work as expected, But we had noted on one customer site that when they did a call pickup on another phone that was ringing in the office they would not be able to hear the caller. The caller could hear them and if they put the call on and off hold they could speak normally.
On further investigation with wireshark we noted that the RTP port changed when the pickup took place. We tested this on other sites not using the Fortigate hardware and did not have this issue.
Below are listed the commands to clear the SIP helper settings from the Fortigate hardware.
Open the Fortigate CLI from the dashboard.
Enter the following commands in FortiGate’s CLI:
config system settings
set sip-helper disable
set sip-nat-trace disable
reboot the device
Reopen CLI and enter the following commands – do not enter the text after //:
config system session-helper
show //locate the SIP entry, usually 12, but can vary.
delete 12 //or the number that you identified from the previous command.
Disable RTP processing as follows:
config voip profile
edit default
config sip
set rtp disable
And finally:
config system settings
set default-voip-alg-mode kernel-helper based
End
on a fortigate 200D the following is the method to use
Step 1) Removing the session helper.
Run the following commands:
config system session-helper show
Amongst the displayed settings will be one similar to the following example:
edit 13 set name sip set protocol 17 set port 5060
In this example the next commands would be:
delete 13 end Step 2) Change the default –voip –alg-mode.
Run the following commands:
config system settings set default-voip-alg-mode kernel-helper based end Step 3) Either clear sessions or reboot to make sure changes take effect
a) To clear sessions
The command to clear sessions applies to ALL sessions unless a filter is applied, and therefore will interrupt traffic.
With over 1 MILLION production systems worldwide and 20,000 new systems installed monthly, the FreePBX community continues to out-perform the industry’s commercial efforts. The FreePBX EcoSystem has developed over the past decade to be the most widely deployed open source PBX platform in use across the world. The openness of the project allows users, resellers, enthusiasts and Partners to utilize the FreePBX EcoSystem to build robust communications solutions that are powerful but at the same time easy to implement and support. Sangoma is proud to be the sponsor of FreePBX project. If you are new to FreePBX you can get started quickly by downloading and installing the FreePBX Distro. The FreePBX Distro is an all in one platform that installs everything you need to build a phone system. Once You have a basic PBX in place you can add commercial modules to add advanced features to an already feature rich base install of FreePBX.
As an open source GPL, web-based PBX solution, FreePBX is easy to customize and adapt to your changing needs. FreePBX can run in the cloud or on-site, and is currently being used to manage communications of all sizes and types of environments from small one person SOHO (Small Home, Small Office) businesses, to multi-location corporations and call centers. The FreePBX ecosystem provides you with the freedom and flexibility to custom design business communications around your needs.
FreePBX Commercial Modules are add-ons that enhance the already feature rich base install of FreePBX! These modules are not Open Source GPL and are only designed to work with CentOS or RHEL systems. The FreePBX Distro is already preconfigured to work with these modules. For custom installations please see: Install Commercial Modules on CentOS and RHEL based systems
The FreePBX appliance is a purpose-built, high-performance PBX solution. Designed and rigorously tested for optimal performance, this is the only officially supported hardware solution for FreePBX. The appliance comes preloaded with the FreePBX Distro and includes a one-year warranty!
Featuring the FreePBX Distro, this appliance is an ideal fit for businesses looking to get more from a PBX. With millions of deployments throughout the world, FreePBX is relied upon daily by everyone from enterprises to startups. Leveraging the power of FreePBX has enabled businesses to grow while keeping communication expenses minimal. The FreePBX Distro has made deploying, configuring and using a PBX system easier than ever! With an easy-to-use GUI (Graphical User Interface), getting started is a breeze!
Sangoma IP Phones Designed Exclusively for FreePBX are Designed to work with FreePBX, Sangoma IP phones are so smart you can quickly and easily use them right out of the box. Each phone in the series features industry standard Power over Ethernet, so no power cable or outlets required. They have full duplex speakerphones, dual Ethernet Ports, multi-way conference calling, high definition voice quality, and they’re Virtual Private Network (VPN) capable.
Full Integration with FreePBX, FreePBX phone apps are available right on the phone, straight out of the box with no requirement for additional licenses. Users can control complicated features directly from their phones. There’s no need to remember feature codes. User applications include: Call Parking, Follow Me, Do Not Disturb, Conference Rooms, Call Forwarding, Time Conditions, Presence, Queues, Transfer to Voice Mail, Visual Voice Mail, and Log in/out.
Why is Sangoma Zero Touch Better? VoIP telephones can be complex to install, and manually configuring many different parameters and hundreds of extensions can take hours. When you buy and install your Sangoma IP phones, the redirection server automatically points the phone to the Sangoma FreePBX for configuration. Other vendors have redirection servers, but they have to be programmed with the details of the IP PBX. Only Sangoma can provide Zero Touch provisioning with FreePBX.
EndPoint Manager Included When using a Sangoma phone, EndPoint Manager software inside FreePBX is automatically enabled. This lets your users control global settings, program their phone keys, map extensions, upload images, download new firmware, and much more.
In freepbx there is a feature that is quite well hidden but actually does a very useful job.
In The “Advanced Settings” page if you enable both “Post Call Recording Script” As the name suggests this is a script that run after a recorded call has ended. We created a script called postrecord.sh and in the text field on the menu we have put as below. This emails both inbound and outbound calls.
For calls to and from an extension we can pull the email address from the voicemail.conf and send the email to that address.
Its also set to delete the wav file away after a defined number of days.
The Script below will first convert the recording then email it to you or your customer.
A couple of prerequisites are required, these are sox and lame. sox is probably already installed, lame maybe not.
Installing Lame is simple for centos as below.
wget http://sourceforge.net/projects/lame/files/lame/3.99/lame-3.99.5.tar.gz tar -zxvf lame-3.99.5.tar.gz
cd lame-3.99.5
./configure
make
make install
The script is fairly simple as below. the main variables are passed to it but we build the directory structure on the fly and file extension is fixed as wav. you can set the file_age variable to delete the wav file messages over that many days old.
Be careful if cutting and pasting this scripty as wordpress may have wrapped some lines
#!/bin/bash
#This script emails the recorded call right after the call is hung up. Below arethe variables passed through asterisk
# $1 - Time String
# $2 - Source
# $3 - File
# $4 - unique id
# $5 - Destination
# $dt - Date and Time
/bin/nice /bin/sleep 3
dy=$(date '+%Y')
dm=$(date '+%m')
dd=$(date '+%d')
file_age=35
dtpath=/var/spool/asterisk/monitor/$dy/$dm/$dd/
/bin/nice /usr/local/bin/lame -b 16 -m m -q 9-resample $dtpath$3.wav$dtpath$3.mp3
/bin/nice /bin/chownasterisk:asterisk $dtpath$3.mp3
dt=$(date '+%m/%d/%Y %r');
id=$(mysql -uUser -pPassword -s -N -e "SELECT descr from asterisk.queues_config where extension = $5");
email=recordings@yourdomain.com
file=$dtpath$3
if [ "$id" = "" ]; then
direction=callers
id=$(mysql -uUser -pPassword -s -N -e "SELECT name from asterisk.users where extension = $2");
IN=$(/bin/grep "$2 =>" /etc/asterisk/voicemail.conf)
echo $IN
set -- "$IN"
IFS=","; declare -a Array=($*)
email=${Array[2]}
else
direction=customers fi
echo -e "You have a new call recording to listen to\n\n
The call date and time was: $dat \n\n
The call was from: $2 \n\n The call was to: $5 \n\n
The $direction name was: $id \n\n
And the unique call id was: $4 \n\n
Please see the attached file \n\n" | mail -a $file.mp3 -s "New Recording at $dt" $email
/bin/nice /usr/bin/find /var/spool/asterisk/monitor/-type f -mtime +"$file_age" |grep wav | \
while read I; do
/bin/rm"$I"
done
We recently had a very puzzling issue with a customer who we supplied some T23 Yealink handsets. When making outgoing calls over Gamma sip trunks on their Elastix server we were getting one way audio, This was not an issue with their existing Snom handsets or a problem for internal or incoming calls over the same trunks. It also wasn’t an issue when using iax2 trunks.
It seems that there is some interoperability issue when using sip trunks and these handsets. and seems to be a little known issue as only affects a few operators.
It seems to addressed in 44.80.0.20 version software that isn’t on the Yealink UK site yet but is available here and should be loaded on all T23 handsets as they are being delivered as 44.80.0.5 firmware at the moment.
In FreePBX users can listen to wav file recordings via the “Call Recordings” tab, This uses a field in the mysql cdr table to say where that recording is and what its called, They are now stored in year/month/day directory structure under /var/spool/asterisk/monitor so if the end user wants the recordings in mp3 format as many do its not just a case of converting them its also a case of updating the database.
Luckily this is fairly straight forward, its just a case of doing a quick query and then converting the file and the updating the database. First you have to install lame, This can be done simply with yum then write a script.
In FreePBX advanced settings, you need to enable “Display” and “Override” readonly settings and then add
The script I use is simple with a bit of basic logging.
#!/bin/bash
. postrecconfig.sh
date >> /var/log/asterisk/mp3.log
pcmwav=$(mysql -u$user -p$secret -s -N -D asteriskcdrdb<<<"select recordingfile from cdr where linkedid LIKE '$1' AND disposition = 'ANSWERED' ORDER by calldate DESC LIMIT 1");
mp3="$(echo $pcmwav | sed s/".wav"/".mp3"/)"
nice lame -b 16 -m m -q 9-resample "$path$pcmwav" "$path$mp3" >> /var/log/asterisk/mp3.log
touch -r "$path$pcmwav" "$path$mp3" >> /var/log/asterisk/mp3.log
mysql -u$user -p$secret -s -N -D asteriskcdrdb<<<"UPDATE cdr SET recordingfile='$mp3' WHERE recordingfile = '$pcmwav'" >> /var/log/asterisk/mp3.log
echo $pcmwav >> /var/log/asterisk/mp3.log
echo "--------||-------" >> /var/log/asterisk/mp3.log
date >> /var/log/asterisk/mp3.log
echo "Done" >> /var/log/asterisk/mp3.log
echo "--------||-------" >> /var/log/asterisk/mp3.log
exit 1
The postrecconfig.sh file looks like
user=freepbxuser
secret=secret
receptemail=info@youremailaddress.com
file_age=35
dy=$(date '+%Y')
dm=$(date '+%m')
dd=$(date '+%d')
path=/var/spool/asterisk/monitor/$dy/$dm/$dd/
As can be seen it steps through entry by entry converting and updating the DB, This example is cron'd to run hourly but does not delete the original wav file, this would be done in a separate script run weekly to remove old files. The reason to keep them is so that a backup of the original is held for a period in case of errors.
This website uses cookies to improve your experience. We'll assume you're ok with this, but you can opt-out if you wish.AcceptRejectRead More
Privacy & Cookies Policy
Privacy Overview
This website uses cookies to improve your experience while you navigate through the website. Out of these, the cookies that are categorized as necessary are stored on your browser as they are essential for the working of basic functionalities of the website. We also use third-party cookies that help us analyze and understand how you use this website. These cookies will be stored in your browser only with your consent. You also have the option to opt-out of these cookies. But opting out of some of these cookies may affect your browsing experience.
Necessary cookies are absolutely essential for the website to function properly. This category only includes cookies that ensures basic functionalities and security features of the website. These cookies do not store any personal information.
Any cookies that may not be particularly necessary for the website to function and is used specifically to collect user personal data via analytics, ads, other embedded contents are termed as non-necessary cookies. It is mandatory to procure user consent prior to running these cookies on your website.