Categories
Blog Elastix Support

Elastix changes and what it means

This week, significant changes at Elastix were announced, including the involvement of 3CX and the removal of key Elastix versions for download. Since those announcements, many things have been written by many people, and this has left some folks wondering what happened. Sangoma would like to reinforce its commitment to open source, this open letter from Sangoma, will provide our own clarity about how these events affect or involve Sangoma. Sangoma are a professional, global, growing, profitable, engineering-focused, publicly traded company, and this is the only reliable source of information to understand how those recent events affect or involve Sangoma. Other commentary released by other third parties about Sangoma, is not to be relied upon.

Everyone comes to open source software for their own reasons: software developers to do what they love; some to earn a livelihood; manufacturers to augment the project and sell their wares; and most importantly community members to find flexible/cost effective/well-supported solutions to their ‘business problem’ (in our case, for UC/Telecom/PBX needs). In the end, the good projects build something bigger than themselves… a community, a solution, and an opportunity for end users to utilize the project to build their own businesses. Over the course of a project many people will enter and exit those communities as their needs change.

As the primary investor in and developers of FreePBX, Sangoma actively works with many different members of the Open Source Telephony (OST) community, including Asterisk Developers, other FreePBX-based distros (including Elastix!), and many third-party hardware/software developers and manufacturers. As just one example, we have a great relationship with Digium and talk with them on an almost weekly basis, even though many consider us competitors. This may seem surprising to some, as many folks would think we might be bitter enemies. In fact, the opposite is true…we encourage and help those products to compete in the marketplace on their own merits. And this is entirely consistent with the commitment Sangoma has demonstrated to open source for many, many years over the time when we worked hard to also make Asterisk better. When Sangoma took over stewardship of FreePBX, we reiterated this statement clearly and unequivocally.

So Sangoma continues to work very hard every day, and invests many millions of dollars each year, in order to build strong relationships and to benefit to the entire open source telephony community. There is a saying that ‘a rising tide lifts all boats.’ Thus, it is usually counter-productive for open source contributors to battle with each other. In other words, there is no reason for them to fight over the same slice of pie, when there is an entire cake that no one is touching.

Their approach was no different with Elastix. For over a decade, Sangoma has been a direct supporter of Elastix, in many, many different ways, visiting them in Ecuador many times. They supported the project financially, They attended/exhibited/supported/spoke at multiple ElastixWorld events over many years, They cooperated with their distribution partners who also supported Elastix, They invested in R&D to ensure their products (software and hardware) were compatible with Elastix, etc. The list goes on and on.They had (and hope, still have), excellent relationships between the companies, in all parts of the organizations right up to the CEO level of both companies.

With recent changes at Elastix, some people/blogs/websites have made comments which claim that the removal of Elastix downloads of version 4 or MT, was in some way caused by Sangoma/FreePBX, due to concerns about compliance with GPL conditions. That is not true and They wish to set the story straight.  Sangoma hold ourselves to high ethical standards, and as a publicly traded company as well, setting the record straight with facts and not rumours, is both important and required.

While it is indeed true that Sangoma pointed out to Elastix some time ago, that there was a copyright issue,They did so in a very friendly manner, with words carefully chosen to be respectful of the long term relationship between the companies, and critically, to ensure that this important relationship continued. It was a 2015 letter from CEO to CEO, and certainly did not suggest any legal action, since it was not that kind of letter at all…it was a positive, complementary letter seeking to deepen the relationship, not harm it. That letter was sent shortly after Sangoma acquired FreePBX, when they made it a priority to reach out to PaloSanto to reinforce that the Elastix Project was a valuable strategic partner to Sangoma. It was in no way threatening, did not ask for, was not intended to, and given it was 2015, did not cause any versions of Elastix to be withdrawn. Elastix decision this week to shutdown these versions is a business decision not a response to Sangoma. While it seems that these days, the number of open source projects that remain truly open source is definitely on the decline, Sangoma’s commitment to open source remains as true today, as always.

And while it is admittedly a little unusual for companies to do so, in this case, for full transparency to the open source communities that they respect so very much (and to dispel any untrue rumours or claims), the entire letter is available. They share it for those who need confirmation of the above statements, and to reassure the Elastix community that Sangoma continues to be committed to you as well as to the entire Latin America region (and would be honored to have you consider joining the family)

This page is a shorted and edited version of Sangoma’s announcement at https://www.freepbx.org/what-happened-to-elastix/  follow the link for the full version.

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
FreePBX Software

FreeHMS , A Web based call management package for small Hotels and Serviced offices

FreeHMS is a web based call management package for small Hotels, Guest Houses and managed offices. It is designed to work with FreePBX and Asterisk.

It allows owners to bill guests or users for the phone usage allowing guests to make calls, setup wakeup calls and access voicemail. Rooms are initially blocked from calling other rooms but can call Admin extensions with out being checked in. When a room is checked in they can make trunk calls and set up wakeup calls. When checking out any wakeup calls are removed and the voicemail is defaulted and all Voicemail messages are deleted.

Setting up the system is simple for Installer with minimal changes to the dialplan which are included in the custom configuration file. The system can be set to any language as all text is from a single configuration file which also includes currency and tax rate for billing. Users are created in FreePBX user admin so are easily changed and added.

Call rates are set using the rates page only available to the Admin users

Administration is simple and password controlled using the ‘User Managment’ module of Freepbx so changes to rates and rooms can only be carried out by the admin users, Reception users can log guests in and out, Create Bills and mark rooms for cleaning as well as set or cancel wakeup calls, The Housekeeping login only allows setting of rooms clean or not*. If a room is not marked as clean then that room cannot be checked in.

The software is fully web based and can be used on PC, Mac, Tablet or even smartphone.

iphonehms

The software is OSS and can be customised to suit most customers.

Features Include:

Checkin /Checkout

Billing : Rates are set by the admin user only, Bill can printed with relevant sales tax added.

Cleaning : Rooms are marked unclean on checkout and can only be checked in when marked as clean by reception or the cleaning staff. A cleaning list can be printed off for stall without a tablet or smartphone.

The software is here to download  and as its released as OSS you can modify and extend it as you wish

For more details and pricing for complete system please email info@cyber-cottage.co.uk or call us.

If you just want the software its free to download,  Limited email support will be available, All we ask is if you add a feature or make a change let us have it so everyone can benefit from it.

Finally if there is a feature you want let us know how we can work with you to make it come about.

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

Categories
Asterisk Support Elastix Support Knowledge Base Support Technical

One way audio with Yealink T23 and Gamma Sip trunks on Freepbx

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.

Categories
Elastix Support Knowledge Base Support

Converting recordings to MP3 in FreePBX and updating mysql CDR records

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

/usr/local/sbin/postrecord.sh ^{CDR(linkedid)} to “

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.

Hope this is of help to you and your users

Categories
Asterisk Support Elastix Support Knowledge Base Support

Multiple Dynamic features with Asterisk Applicationmaps

Dynamic features are very useful for allowing users access to custom features during calls. These can be loaded individually via the dialplan, but in freepbx based solutions this will mean a bit of hacking of the dialplan using overides and making sure all still works afterwards, or as a global varible.

The easiest way is to load them as a global as is done with apprecord, But if you want to add lots of features then you will have to use a Application Map group. This is done by editing the features_applicationmap_custom.conf  file so it looks like below for example, at the top are your application maps then your group

testfeature => #9,callee,Playback,tt-monkeys 
calleehangup => #8,callee,Hangup()
callerhangup => #7,caller,Hangup()
[mymapgroup]
testfeature => #9
calleehangup => #8
callerhangup => #7
apprecord => *1

DO NOT FORGET to add the apprecord to your group.

You then need to edit the globals_custom.conf file and add a line like below

DYNAMIC_FEATURES => mymapgroup

Then reload asterisk and issue the command “features show”

Dynamic Feature           Default Current
---------------           ------- -------
callerhangup              no def  #7     
calleehangup              no def  #8     
testfeature               no def  #9     
apprecord                 no def  *1     
Feature Groups:
---------------
===> Group: mymapgroup
===> --> apprecord (*1,caller,Macro,one-touch-record)
===> --> callerhangup (#7)
===> --> calleehangup (#8)

and to check that they are loaded as a global variable do “dialplan show globals” and near or at the top you will see:-

 DYNAMIC_FEATURES=mymapgroup

And thats all there is to it.

Categories
Calls and Lines Connectivity

SIP2SIM Mobile extensions

The SIP2SIM service is a very simple concept which provides you with control of your mobile communications. It is ideal for an office of any size and even for more technical home users.

The service consists of a SIM card, which you put in a mobile phone and it makes that phone appear as if it is a SIP extension (e.g. SIP phone) on a phone system of your choice.

meerkat

Not a SIP application

It is important to realise that we are not talking about a SIP application on a smart phone which then uses mobile data or WiFi. With this service your mobile phone is working as a mobile phone on the GSM mobile network making and receiving proper mobile voice calls. The SIP part is what we do in the back end to pass the calls to and from your SIP server.

An extension on your office phone system

The basic service allows you to specify, on our control pages, the server name, login and password for a SIP server. This could be your office phone system whether an asterisk box, or a FireBrick or whatever. As long as it handles normal UDP SIP with G.711 a-law audio then we will register as a phone and allow calls both ways.

This means your phone can simply be an office extension, like any other.

  • Call office extensions using short extension numbers from your mobile phone.
  • Office policies on callable numbers, such as premium rate, enforced like any other extension.
  • Office voicemail system working, just like any other extension.
  • Office call logging, just like any other extension.
  • Office call recording, just like any other extension.
  • Use in hunt groups, just like any other extension.
  • Even use features like call steal to transfer calls if you want, or in-band DTMF to control call transfer and related features.

Manage costs

The costs are very simple for using the SIM in the UK on O2. Higher costs apply for roaming SIMs, even roaming to other UK networks.

  • £5+VAT to purchase the SIM card.
  • £2+VAT pcm once activated (activates on first use).
  • 2p+VAT per minute for calls either way.
  • 2p+VAT per text either way.
  • 2p+VAT per MB for data usage either way.
  • No minimum term – just monthly rolling contract.

Calls are charged per second, and data per kilobyte. Charges apply to the SIP2SIM service separately from any charges you may have for text or voice calls using your VoIP/SMS provider.

In some cases a SIM will be set up with a free trial which allows some usage without assigning to an account or setting up an account. For these trial SIMs, once assigned to an account an activation fee of 1p+VAT is charged and usage charges commence as normal.

OFCOM call charges

The SIP2SIM service is not a service that allows calls to telephone numbers in the national dialling plan. It allows calls to be passed to a VoIP/SIP gateway of your choice. Any ability to make calls to normal telephone numbers is provided by that VoIP provider (which may be our VoIP service). As such, special rules on costs of 01, 02, 03 numbers, rules on 0800 numbers being free, and rules on charges for other special and premium rate numbers do not apply to the SIP2SIM part of the service. The cost or the SIP2SIM service applies regardless of the number you dial.

Telephone numbers

Just like a SIP handset, the service does not come with any sort of telephone number.

You can, if you wish, have the phone register on a VoIP provider’s service. This would mean you get calls to a number operated by the VoIP provider, and can make calls from that number, just like any other SIP phone. If registered with your own telephone system, it would have internal extension numbering, and even direct dial in numbering as you have chosen to configure on your phone system.

There is no question of porting numbers to or from the service, it has no numbers.

We can, of course, provide telephone numbers as part of our VoIP service if you wish, and we can even pre-configure these to connect with your SIM as part of the order process.

International roaming

The SIMs are available with world wide roaming. Costs are higher when roaming, obviously. The following are charges for use within EU. See full roaming price list for more details.

  • £5+VAT to purchase the SIM card.
  • £2+VAT pcm once activated (activates on first use).
  • 10p+VAT per minute for calls either way.
  • 5p+VAT per text either way.
  • 10p+VAT per MB for data usage either way.
  • No minimum term – just monthly rolling contract.

In the UK and rest of EU, calls are charged per second, and data per kilobyte. Charges apply to the SIP2SIM service separately from any charges you may have for text or voice calls using your VoIP/SMS provider.

UK roaming for the best coverage of the UK

The SIMs can even roam in the UK. This means the SIM can change its identity automatically – being an O2 SIM when O2 is available (at the lowest costs), but switching to a foreign identity (Dutch Vodafone) to roam on to other UK networks. Whist costs are then higher this means you stay in touch even when there is no O2 signal.

When the SIM is using the O2 profile and on O2 in the UK, the UK prices apply. When using the EU profile on any UK network (including O2) or elsewhere in the EU, the EU roaming prices (as above) apply. If on EU profile on O2, it will normally switch back to O2 profile automatically within a few minutes. Currently the SIM will not roam to Three in the UK.

Text

Texts are operated separately. Our text interface using HTTP can send texts to the phone, and texts from the phone can be posted to an HTTP gateway of your choice. If you have an A&A VoIP telephone number then texts can be associated with that number directly (not all of our numbers ranges can handle inbound texts).

Mobile Data

Data currently allows simple NAT, unfiltered, Internet access. We hope to offer data via A&A in the future.

Third Party SIP services

The service involves entering SIP registration details in to our control pages. Where these are the details of your own SIP server such as an office phone system, you can make the decision as to whether or not you trust us with those SIP details in order to provide the SIP2SIM service. We will, of course, use all reasonable skill and care to ensure the details remain confidential and are not disclosed.

We have, however, designed the service so that it can work with a wide variety of third party SIP services, not just in the UK but in various countries. There are a lot of unusual systems out there and we continue to work to ensure that such services operate with SIP2SIM. However, using third party SIP details may well be in breach of your terms with the third party SIP provider as it means giving us your SIP details. It is up to you to check the terms and we would not suggest anyone breaks a contract they have. You may find that the provider is happy to trust us, especially if they do any other work with us, so it is worth asking. We are also happy to discuss contracts with other providers for the SIP2SIM service and we may be able to provide them with branded SIMs to sell to their customers.

Unfortunately we cannot guarantee that the service will always work with all other providers, and whilst we aim to resolve any technical issues (with reference to the standards), if a provider simply will not deal with our service and blocks us there is little we can do.

The Sip2Sim service is provided by Andrews and Arnold Ltd  and can be purchased direct from them or via ourselves where we will assist with setting up the service for you.

Categories
Elastix Support Knowledge Base Technical

Setting the server domain in elastix correct for scripted email

We run many scripts on customer servers to email cdrs, backups etc, one problem with some mail servers is the mail gets rejected as it comes from root@elastixserver.yourdomain.com by default to fix this is simple and only takes a few lines.

Postfix MTA offers smtp_generic_maps parameter. You can specify lookup tables that replace local mail addresses by valid Internet addresses when mail leaves the machine via SMTP.

Open your main.cf file

# vi /etc/postfix/main.cf

Append following parameter

smtp_generic_maps = hash:/etc/postfix/generic

Save and close the file. Open /etc/postfix/generic file:

# vi /etc/postfix/generic

Make sure root@elastixserver.yourdomain.com change to elastixserver@yourdomain.com add :

root@elastixserver.yourdomain.com  elastixserver@yourdomain.com

Save and close the file. Create or update generic postfix table:

# postmap /etc/postfix/generic

Restart postfix:

# /etc/init.d/postfix restart

When mail is sent to a remote host via SMTP this replaces root@elastixserver.yourdomain.com by elastixserver@yourdomain.com mail address. You can use this trick to replace address with your ISP address if you are connected via local SMTP.

To set up gmail for delivery look at this

Categories
Handsets Products

A510H End of Life Notice

A510H entry level cordless handset will be going end of life from approximately the end of February 2015. There will be no further production of the A510H after this.

Announcing the New Gigaset A540H as the A510H replacementA540H

We are pleased to announce the imminent arrival of the Gigaset A540H as the new entry level handset and A510H replacement. This new handset has all the same features as the A510H and comes with a new sleek moulding that makes it look more attractive and nicer to handle.

We are expecting stock to arrive around early February and pricing will be very similar to the A510H. We will keep you informed of the delivery date as soon as it is confirmed.