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Topic # 139454 9-Feb-2014 21:53
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I have a small business with an analog Panasonic Kx-TES 824 pbx. We have installed 2 ATAs (Linksys sp2102), so can have 4 concurrent calls. While dialing out and receiving calls is fine, for incoming calls, when the dialer dials a tone to select a department from the auto attendant, the Panasonic fails to pick up the dial tone. This is frustrating as it then makes all the extension phones ring when it has timed out due to it thinking the caller has failed to dial a tone.  When I call my company and simulated the dial tones myself, they sound scratchy and not as smooth as the good old copper lines. Are there settings I need 2 talk to adjust?  Please advise.

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  Reply # 983024 9-Feb-2014 22:09
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What codec are you using? Are you on ADSL, VDSL or fibre?

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  Reply # 983084 9-Feb-2014 23:38
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Hmm thats a bit strange. Normally it is the other way around.

Maybe just check that your ATA is using RFC2833 for DTMF and also be sure you aren't on 2talk's old platform - just make sure it is registering to 'sip.2talk.co.nz'. The old platform had issues with dtmf often.

The only other thing I can think of is the Panasonic isn't listening for dtmf. But I assume this was working on the standard POTS lines so that should already be set up.
I will have a look through my panasonic book for other dtmf related settings tomorrow and post up if I find anything helpful.

My only other recommendation is get a new PABX and terminate the SIP trunks natively ;-)

 
 
 
 


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  Reply # 983157 10-Feb-2014 08:51
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Did you auto provision the 2102's or just configure them yourself? If you did it yourself there are around 30+ settings in the SPA that need to even get close to making it work perfectly.




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  Reply # 983174 10-Feb-2014 09:38
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sbiddle: Did you auto provision the 2102's or just configure them yourself? If you did it yourself there are around 30+ settings in the SPA that need to even get close to making it work perfectly.





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Definitely take advantage of of 2Talk's provisioning tool for your SPA devices. It will get things how they should be for a pleasant VoIP experience.  

Had a quick skim through the programming book, didn't find too much....

Have a look at these programs though....

103 - Voicemail port DTMF
210 - DTMF time - Outgoing
401 - Trunk dial mode


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