Tag: Technical

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Products Sangoma Software

Zulu UC – The Ultimate Desktop and Softphone integration for your Business

Zulu UC Desktop and softphone integration unifies the most popular business communication tools & applications enhancing user productivity and mobility. Designed specifically for FreePBX and PBXact phone systems, Zulu enables features such as:

  • Zulu Softphone enabling users to make/receive phone calls from their desktop or mobile stations, including Chat for team collaboration.
  • Faxing directly from the Zulu widget & softphone.
  • Click-to-call to make calls directly from your web browser and /or email client.
  • Call Pop for CRM and help desk integration.

Get Your Free Zulu 2 User FreePBX License, FreePBX Zulu UC 2 User Package is Free of Charge. Each User Package comes with 2 users good for 12 months.

Features

Click to Call

With Click-to-Call integration, users can instantly call any phone number that is seen on their web browser or MS Outlook client which a click!. Simply click on the phone number and Zulu will initiate an outbound call via the softphone client or your desk phone, whichever is with you at the time. Great for mobile users who come and go from their workstation.

Click-to-Call also recognizes extensions and phone number prefixes, so you never have to worry about having to modify the phone number or extension you wish to click to dial.

Call Pop

Ideal for CRM and Help Desk Integration, Call Pop will automatically open your desktop web browser on an inbound call with all the information of the caller. This feature helps users expedite phone calls and provide the caller with the best customer service experience.

* For additional CRM integration check out our CRM Link Module

Presence

Improve communication between staff members by allowing them to see each other’s presence via the Zulu Softphone. This feature can save your employees time by reaching out to members who they know are available to take their request.

Users can set their presence using a variety of pre-set statuses or create their own. And because Presence is server side, a user’s presence will be updated across all communication endpoints automatically too.

Zulu Softphone

At the Center of the Zulu is the all new softphone enabling users to take their office with them and never miss a call.

  • Make and receive phone calls using Desktop
  • Send and receive FAX*
  • Control Presence status which will update your status on all your devices
  • Flexible Calling Options- generate a phone call from either the client on your desktop or your desk phone. Great for mobile users who come and go from their workstation.

Faxing requires the Fax Pro Module

Chat

The Zulu UC Softphone features integrated Chat functionality so that staff members can communicate with each other more effectively. Features like 1-to-1 messaging, group chat, file transfer and auto-archiving will improve employee collaboration and improve business results. Finally, one tool to do it all!

*Compatibility
Operating System: Zulu UC is compatible with Windows, Mac and Linux operating Systems. Browsers: Click-to-call and Screen POP work with Firefox and Chrome (Safari coming soon).

Categories
Blog Design FreePBX Knowledge Base

Voice recognition and Asterisk.

This is primarily about Googles new Cloud Speech API and Asterisk recordings.

Having worked on many Voice rec systems including Mitels attendant system, Oranges Wildfire virtual assistance and Lumenvox’s add on for Digium’s Asterisk system one thing none could do was transcribe speech such as voicemails and this is what people want. There was a startup in the UK called Spinvox  but as anyone knows this wasn’t all it seems and when I questioned them while working on a project they clammed up and withdrew our testing account and the rest is history as they say.

So now we are many years on and Google have their second API for this service. The first API was a little flaky to say the least and came up with some amusing translations. The cloud version is much better and does a good job with most voice and also can be localised.

So what have we done. Well we have mixed together some existing code we use and created a “mini voicemail” that records your message converts it to text saves it as a voicemail and emails the resultant Text and recording to you.  In the process we did find a few “gotchas” with the API for example a pause of more than a couple of seconds will result in the translation stopping there, also a big one is that the translation takes as long as the recording is, and the API has a 60 second limit. Both of these can be overcome by limiting the record time in Asterisk to 60 seconds and using sox to remove silence of more than a second.

exten => s,n,Record(catline/${UNIQUEID}.wav,3,60,kaq)

/usr/bin/sox /var/lib/asterisk/sounds/catline/${origdir}.wav ${PATH}${origmailbox}/INBOX/${FILENAME}.flac  lowpass -2 2500 silence -l 1 0.1 1% -1 0.8 1%

As you can see from these snippits of code above we have used variables where possible to that it can be incorporated easily with existing asterisk systems using GUIs such as Freepbx, We use the voicemail greetings that the user recorded and also use the email address thats linked with their mailbox for simplicity of management.

Now having Voicemails as text is nice but where it comes into its own is with structured mailboxes or simply put questionnaires where the caller is asked a number of predefined questions and these are recorded as one single voicemail. We already do this for some customers but they still have to have some one transcribe teh voicemail to text to input it. The quality of the Google translation means that soon they will be able to just copy the text over. Other applications are only limited by your imagination, Such as automated voice menus for Takeaways or Taxi firms.

To be Continued…HERE

Categories
Blog Knowledge Base

Do you hate having to use Module admin to update Freepbx

One of my pet hates is having to use module admin to update the Freepbx modules via the GUI. Its not a big deal but as we use SSH to connect to servers and then tunnels to connect to the GUI. Which is all fine unless you have multiple SSH sessions open and things get complicated..

So I have written a small “dirty” Bash script to prompt you through the fwconsole method of updating all or just one module of your choice.

#!/bin/bash
echo ssh freepbx update tool. 2016 cyber-cottage.eu
echo "Welcome"
echo "We will check for upgrades"

read -p "Do You want to check upgrade status of freepbx modules? (y/n) " RESP
if [ "$RESP" = "y" ]; then
  echo "Glad to hear it"
 fwconsole ma showupgrades
else
  exit
fi

echo "We will now apply all upgrades"

read -p "Do You want to upgrade all freepbx modules? (y/n) " RESP
if [ "$RESP" = "y" ]; then
  echo "Glad to hear it"
 fwconsole ma upgradeall
else
 echo "OK We will just upgrade the module you choose"
  read -p "Please enter the name of the module you want to upgrade " MODU
  echo "We Will Now Upgrade $MODU"
  fwconsole ma upgrade $MODU 
fi

read -p "Do You want to update permissions? (y/n) " RESP
if [ "$RESP" = "y" ]; then
 echo "Glad to hear it"
fwconsole chown
else
echo "Dont forget to apply changes on GUI then"
fi

read -p "Do You want to apply the changes? (y/n) " RESP
if [ "$RESP" = "y" ]; then
  echo "Glad to hear it"
 fwconsole reload
else
  echo "Dont forget to apply changes on GUI then"
  exit
fi

As I said it was quick and “dirty” but it does work and can save a bit of time.

Categories
Knowledge Base

Resetting the Polycom Soundpoints admin password to default

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:

  1. 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.
  2. Power down the phone.
  3. Power up the phone.
  4. 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.
  5. 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
  6. The Set will restart. You may need to restart again to get access to the menus with 456 password.

 

Categories
Knowledge Base Sangoma

Building FreePBX CallCenters

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

WEBINAR: Building Your Contact Center with FreePBX / PBXact UC from Sangoma on Vimeo.

Categories
Knowledge Base Technical

Changing the root or any other mysql password

MySQL stores username and passwords in the user table inside MySQL database. You can directly update or change the password using the following method:

Login to your server, type the following command at prompt:

$ mysql -u root -p

Use the mysql database;

mysql> use mysql;

Change password for user root, enter:

mysql> update user set password=PASSWORD("NEW-PASSWORD") where User='root';

Finally, you need to reload the privileges:

mysql> flush privileges;


mysql> quit
Categories
Asterisk Support Knowledge Base Security

Catching the IP of anonymous callers on Asterisk servers

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.
Categories
System Status

DNS issues affecting calls and routing

On 21-10-2016 there had been a widespread DDOS attack initially in the USA. This has affected service of some of our key voice and DNS service suppliers.

We monitor many sites and run monitoring ourselves and receive status updates from suppliers.

Below are some of the recent ones and some sites reporting the issue

http://www.diario4v.com/tendencias/2016/10/21/ataque-hacker-afecta-twitter-amazon-spotify-reddit-11816.html (you will need to translate)

http://money.cnn.com/2016/10/21/technology/ddos-attack-popular-sites/

https://www.dynstatus.com/incidents/nlr4yrr162t8

Update
Dyn Managed DNS advanced service monitoring is currently experiencing issues. Customers may notice incorrect probe alerts on their advanced DNS services. Our engineers continue to monitor and investigate the issue.

Customers with questions or concerns are encouraged to reach out to our Technical Support Team.
Posted 4 minutes ago. Oct 21, 2016 - 18:23 UTC
Update
Our engineers continue to investigate and mitigate several attacks aimed against the Dyn Managed DNS infrastructure.
Posted 34 minutes ago. Oct 21, 2016 - 17:53 UTC
Update
This DDoS attack may also be impacting Dyn Managed DNS advanced services with possible delays in monitoring. Our Engineers are continuing to work on mitigating this issue.
Posted about 2 hours ago. Oct 21, 2016 - 16:48 UTC
Investigating
As of 15:52 UTC, we have begun monitoring and mitigating a DDoS attack against our Dyn Managed DNS infrastructure. Our Engineers are continuing to work on mitigating this issue.
Posted about 2 hours ago. Oct 21, 2016 - 16:06 UTC

Gradwell:

Our upstream supplier is investigating a DNS issue, which is believed to be causing the problem.

Magrethea

We are now able to confirm that two nodes on our network where impacted by DNS issues between 17:13 and 17:45 today. As many of you will be aware there have been some major DOS attacks today which impacted a number of key sites at this time so we are attributing this issue to that attack.

We will continue to monitor and apologise for the inconvenience this outage has caused our customers.

As can be seen this is out of our control and is affecting many users worldwide.

Categories
Blog Knowledge Base

BT outage on 20th July 2016

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. Nagios monitor screen

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.

Categories
Knowledge Base Technical

Fortigate issues such as one way audio on Call Pickup With Hosted Asterisk and other problems.

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.

  1. Open the Fortigate CLI from the dashboard.
  2. Enter the following commands in FortiGate’s CLI:
    • config system settings
    • set sip-helper disable
    • set sip-nat-trace disable
    • reboot the device
  3. 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.
  4. Disable RTP processing as follows:
    • config voip profile
    • edit default
    • config sip
    • set rtp disable
  5. 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.

diagnose system session clear

Taken from

http://kb.fortinet.com/kb/documentLink.do?externalID=FD36405