AASTRA 6730i - Key tones don't work

Hi guys,

New install of FreePBX bistro.

I bought one commercial module : the EndPoint Manager.

I set up my pfSense with option 66 to feed tftp:// (the IP of my FreePBX server).

I have AASTRA phones, different models, and all of them got provisioned automatically, without any kind of intervention… just resetting to factory default, they upgraded the firmware, and auto configured everything…

But i’m experimenting a very weird problem with the 6730i phones.

They key tones don’t seem to work??


  1. If a user dials *98 for VM, then they won’t be able to put their extension number and password… the keytones don’t work!
  2. If a user dials an external number and then has to dial an extension, the key tones don’t work either.

Anything typed after hitting “dial” button will not be sent.

How can i fix this?


Your DTMF protocol needs to match between your phone and your asterisk configuration. I recommend setting both to RFC2833

Where do I set the protocol for the phone, and for asterisk?

The only place I saw something like that is in the trunk configuration, and it is forced to RFC2833.

However, the internal call still don’t work (between a phone and FreePBX, for VM) so the trunk isn’t even involved here.

In your extension settings, look for DTMF signalling. You will see the drop down selection with RFC2833 in there. Additionally, depending on how you are using the endpoint manager, there should be a configuration file that is read in to your phones when they boot. There should be a way to define DTMF signalling in there.

The sip trunk is really only concerned about how your phone system interfaces with your provider.

DTMF signalling for all extensions was and still is set to RFC2833 in FreePBX.

As for the EndPoint Manager setup, there are only two things I can seem to configure:

  1. The template:

The only thing I see is something called “Tone Set” and the choices are by region of the world.
Right now it is in to US/Canada and that is where we are so it should be ok.

  1. The basefile:

Basefile Settings for Aastra 6730i Using Template aastra

admin password: "__adminPass__"
user password: "__userPass__"
paging group listening: "__multicastAddress__"
sip dial plan: "__dialpattern__"
auto resync mode: "3"
auto resync time: "05:00"
sip line1 auth name: "__line1Ext__"
sip line1 password: "__line1Pass__"
sip line1 user name: "__line1Ext__"
sip line1 display name: "__line1Ext__"
sip line1 screen name: "__line1Name__"
sip line1 proxy ip: "__line1Dest__"
sip line1 proxy port: "__line1sipPort__"
sip line1 registrar ip: "__line1Dest__"
sip line1 registrar port: "__line1ServerPort__"
sip line1 vmail: "__voicemail__"
sip line1 mode: "0"
sip line2 auth name: "__line2Ext__"
sip line2 password: "__line2Pass__"
sip line2 user name: "__line2Ext__"
sip line2 display name: "__line2Ext__"
sip line2 screen name: "__line2Name__"
sip line2 proxy ip: "__line2Dest__"
sip line2 proxy port: "__line2sipPort__"
sip line2 registrar ip: "__line2Dest__"
sip line2 registrar port: "__line2ServerPort__"
sip line2 vmail: "__voicemail__"
sip line2 mode: "0"
sip line3 auth name: "__line3Ext__"
sip line3 password: "__line3Pass__"
sip line3 user name: "__line3Ext__"
sip line3 display name: "__line3Ext__"
sip line3 screen name: "__line3Name__"
sip line3 proxy ip: "__line3Dest__"
sip line3 proxy port: "__line3sipPort__"
sip line3 registrar ip: "__line3Dest__"
sip line3 registrar port: "__line3ServerPort__"
sip line3 vmail: "__voicemail__"
sip line3 mode: "0"
sip line4 auth name: "__line4Ext__"
sip line4 password: "__line4Pass__"
sip line4 user name: "__line4Ext__"
sip line4 display name: "__line4Ext__"
sip line4 screen name: "__line4Name__"
sip line4 proxy ip: "__line4Dest__"
sip line4 proxy port: "__line4sipPort__"
sip line4 registrar ip: "__line4Dest__"
sip line4 registrar port: "__line4ServerPort__"
sip line4 vmail: "__voicemail__"
sip line4 mode: "0"
switch focus to ringing line: "__focus__"
sip allow auto answer: "__answer__"
sip intercom mute mic: "__mute__"
sip intercom allow barge in: "__barge__"
directed call pickup: "1"
directed call pickup prefix: "**"
auto offhook: "__offhook__"
stutter disabled: "__stutter__"
sip digit timeout: "__timeout__"
tone set: "__toneset__"
download protocol: "__protocol__"
ftp server: "__ftpserver__"
tftp server: "__ftpserver__"
ftp path: ""
ftp username: "__ftpuser__"
ftp password: "__ftppass__"
http server: "__ftpserver__"
http path: "/"
http port: "__provisionport__"
time server disabled: "0"
time server1: "__timeserver1__"
time server2: "__timeserver2__"
time server3: "__timeserver3__"
time zone code: "__timezone__"
time zone name: "__timezonename__"
time format: "0"
date format: "0"
call forward disabled: "1"
sip gruu: "0"
sip instance id: "0"
sip accept out of order requests: "1"
sip blf subscription period: "600"
sip contact matching: "2"
dynamic sip: "1"
firmware server: "__firmware__"
sip xml notify event: "1"
action uri registered: "http://__line1Dest__:__restapps__/sync.php?user=__line1Ext__"
services script: "http://__line1Dest__:__restapps__/applications.php/restapps/main?user=__line1Ext__"
xml application URI: "http://__line1Dest__:__restapps__/applications.php/restapps/main?user=__line1Ext__"
xml application title: "Applications"
sip customized codec: "payload=9;ptime=20;silsupp=on,payload=0;ptime=20;silsupp=on,payload=8;ptime=20;silsupp=on"
dhsg: "__missed__"
audio mode: "__volume__"
background image: "__background__"
sip xml notify event: "1"
DST_ENABLE: "__daylight__"
sip line5 auth name: "__line5Ext__"
sip line5 password: "__line5Pass__"
sip line5 user name: "__line5Ext__"
sip line5 display name: "__line5Ext__"
sip line5 screen name: "__line5Name__"
sip line5 proxy ip: "__line5Dest__"
sip line5 proxy port: "__line5sipPort__"
sip line5 registrar ip: "__line5Dest__"
sip line5 registrar port: "__line5ServerPort__"
sip line5 vmail: "__voicemail__"
sip line5 mode: "0"
sip line6 auth name: "__line6Ext__"
sip line6 password: "__line6Pass__"
sip line6 user name: "__line6Ext__"
sip line6 display name: "__line6Ext__"
sip line6 screen name: "__line6Name__"
sip line6 proxy ip: "__line6Dest__"
sip line6 proxy port: "__line6sipPort__"
sip line6 registrar ip: "__line6Dest__"
sip line6 registrar port: "__line6ServerPort__"
sip line6 vmail: "__voicemail__"
sip line6 mode: "0"
sip line7 auth name: "__line7Ext__"
sip line7 password: "__line7Pass__"
sip line7 user name: "__line7Ext__"
sip line7 display name: "__line7Ext__"
sip line7 screen name: "__line7Name__"
sip line7 proxy ip: "__line7Dest__"
sip line7 proxy port: "__line7sipPort__"
sip line7 registrar ip: "__line7Dest__"
sip line7 registrar port: "__line7ServerPort__"
sip line7 vmail: "__voicemail__"
sip line7 mode: "0"
sip line8 auth name: "__line8Ext__"
sip line8 password: "__line8Pass__"
sip line8 user name: "__line8Ext__"
sip line8 display name: "__line8Ext__"
sip line8 screen name: "__line8Name__"
sip line8 proxy ip: "__line8Dest__"
sip line8 proxy port: "__line8sipPort__"
sip line8 registrar ip: "__line8Dest__"
sip line8 registrar port: "__line8ServerPort__"
sip line8 vmail: "__voicemail__"
sip line8 mode: "0"
sip line9 auth name: "__line9Ext__"
sip line9 password: "__line9Pass__"
sip line9 user name: "__line9Ext__"
sip line9 display name: "__line9Ext__"
sip line9 screen name: "__line9Name__"
sip line9 proxy ip: "__line9Dest__"
sip line9 proxy port: "__line9sipPort__"
sip line9 registrar ip: "__line9Dest__"
sip line9 registrar port: "__line9ServerPort__"
sip line9 vmail: "__voicemail__"
sip line9 mode: "0"
sip line10 auth name: "__line10Ext__"
sip line10 password: "__line10Pass__"
sip line10 user name: "__line10Ext__"
sip line10 display name: "__line10Ext__"
sip line10 screen name: "__line10Name__"
sip line10 proxy ip: "__line10Dest__"
sip line10 proxy port: "__line10sipPort__"
sip line10 registrar ip: "__line10Dest__"
sip line10 registrar port: "__line10ServerPort__"
sip line10 vmail: "__voicemail__"
sip line10 mode: "0"
sip line11 auth name: "__line11Ext__"
sip line11 password: "__line11Pass__"
sip line11 user name: "__line11Ext__"
sip line11 display name: "__line11Ext__"
sip line11 screen name: "__line11Name__"
sip line11 proxy ip: "__line11Dest__"
sip line11 proxy port: "__line11sipPort__"
sip line11 registrar ip: "__line11Dest__"
sip line11 registrar port: "__line11ServerPort__"
sip line11 vmail: "__voicemail__"
sip line11 mode: "0"
sip line12 auth name: "__line12Ext__"
sip line12 password: "__line12Pass__"
sip line12 user name: "__line12Ext__"
sip line12 display name: "__line12Ext__"
sip line12 screen name: "__line12Name__"
sip line12 proxy ip: "__line12Dest__"
sip line12 proxy port: "__line12sipPort__"
sip line12 registrar ip: "__line12Dest__"
sip line12 registrar port: "__line12ServerPort__"
sip line12 vmail: "__voicemail__"
sip line12 mode: "0"
language 1: ""
language 2: ""
language 3: ""
language 4: ""
language: ""
input language: ""
web language: ""
directory script: "__directory__"
sip line1 registration period: "60"
sip line2 registration period: "60"
sip line3 registration period: "60"
sip line4 registration period: "60"
sip line5 registration period: "60"
sip line6 registration period: "60"
sip line7 registration period: "60"
sip line8 registration period: "60"
sip line9 registration period: "60"
sip line10 registration period: "60"
sip line11 registration period: "60"
sip line12 registration period: "60"

Which don’t seem to contain anything concerning DTMF or such.

The only thing that might be relevant is the “Tone Set” and it is the same.

I’m really starting to go nuts over this :’(

Are your “codecs allowed” genuinely acceptable ? (i.e licensed, particularly g729)

You mean in Asterisk SIP configuration?

I didn’t touch that since the install.

Right now those that are active are the following:


That would depend on how the call came in , g726 can be problematic, disable it.

Yeah but it’s not about the call coming in… it’s internal… like once you dial *98, you can’t type anything on the keypad, it will not react !

I disabled it anyway.

But in the trunk config I disallow all, and then enable only ulaw.

Also, i just noticed this in the phone config.


I’m not sure what it should be set to :S

Sorry, it’s all yours on your “phone config”

RTP on 3000 will unlikely coincide or be accepted by your /etc/asterisk/rtp.conf settings. By default it will accept rtp connections on 10000 to 20000

Ok so I researched around and it was suggested to uncheck the Out-of-band thing, then set mode to SIP info.

I did that and it WORKS !!!

But now next question: how can I set that in the EndPoint Manager?? I can’t find anywhere.

Endpoint manager is still broken in FreePBX 12 13, be conservative.

I don’t mean the OSS EndPoint Manager, I mean the commercial module.

Is that what you are referring to?

Yes the commercial one is a problem yet if you use the alpha/Beta version.


It’s the single, only module I bought for this install because some phone models were not supporter by the OSS EndPoint Manager.

I didn’t know it was broken…

So I made the configuration on each individual phone by connecting to the Web GUI by IP… what a waste of time :’(

But now it’s all good!!

Thanks for the pointers guys.

Using non production grade software is always going to be a problem, no?

I don’t mean to be arguing I just want to find out: how could have I known it was not production grade?

I downloaded the latest STABLE version of FreePBX, then I bought the EndPoint Manager commercial module!

You would have to take that to the “commercial” forum or perhaps the “Distro” one you posted without a category.

Endpoint Manager is most assuredly not broken in FreePBX 12. Version 12 is the current stable version and it’s recommended that all users be running 12 for production systems. Could it be that @dicko is referring to FreePBX 13?

In any case @Nucleus, what you are trying to achieve can be done with EPM in 12 without touching the phone GUI by editing the Aastra basefile to override the default setting for Out of Band DTMF therein.


That was indeed a typo .