Tones don't register after call has been dialed

[Ryandc]

Here's my setup:
Windows 2003 Server 32-bit
Grandstream 4108 FXO connected to several PSTN lines
Linksys SPA922s registered and active

I can dial other extensions within the network fine, and I can dial external calls just fine as well. The only thing I seem unable to do is use other numbers' automated menu systems (You know, "Press 1 for English, Press 2 for Espanol, Press 0 for the Operator"). I hit the buttons, but the system on the other end doesn't seem to register them.

Any ideas?

 

[Ralph]

Hi

Good afternoon,

So what you are saying is that when you call an external number that answers with its own call attendant you can't use the number pad (press 1, etc.) to make a selection?

This sounds like it may be a symptom of one way voice communication.
You can hear the called party but because they can't hear you the signal from the keypad isn't reaching their system.

When you call an external telephone number (not an extension) using your VoIP line do you get sound in both directions?

 

[Ryandc]

Yep, voice communication is A-OK internally and externally (beautiful thing about running PSTN lines-- you don't have to worry about poking holes in your firewall). I can send and receive voice transmissions on the phone. Just the keypad on the phone doesn't work.
Now, I haven't properly configured dialing rules yet. As it stands, I have to press the number (ie, 9-1-801-555-1234) with the lineout prefix; when the number is stored, I then press the dial soft button on the phone. The phone doesn't dial out automatically. Could this have anything to do with it?

 

[5qg4]

Hi Ryandc,

I can dial other extensions within the network fine, and I can dial external calls just fine as well. The only thing I seem unable to do is use other numbers' automated menu systems (You know, "Press 1 for English, Press 2 for Espanol, Press 0 for the Operator"). I hit the buttons, but the system on the other end doesn't seem to register them.

It seems the type of DTMF TX issue. Please open your Linksys SPA922s admin page, at "Admin Logon"= "EXT 1"= "DTMF Tx Method:". If exiting type is "Auto", then please try to change to "InBound+Info" to see it work or not.

 

[Ryandc]

That fixed it. Thanks for the help.

 

[5qg4]

Hi Ryandc,

Good to hear that your problem had been fixed.

Regards,
5QG4

 

[AlecM]

Hi 5qg4,

I am experiencing the same problem as RyanC, but I have different handsets.

Equipment:
GrandStream GXW4104
Grandstream BT200 handsets
3CX 3.1.2434 running on Win XP Pro SP2

Your fix for RyanC was specific to the LinkSys handset, but if I am correct in interpreting this to the equivalent on the BT200's, this would be the "Send DTMF" options. I have these set to:
In-Audio = ON
via RTP (RFC2833) = ON

But in the 3CX log I see the following type of entry:

10:28:30.112 MediaServerReporting:TMFhandler [MS211000] Call(36) Ext.102: DTMF (RTP) from 10.0.0.160:5004 arrived. in-band DTMF tone detection is turned off.
10:19:36.952 MediaServerReporting::RTPSender [MS104000] Call(33) Ext.120: (ACTIVE) RTP stream to 10.0.0.163:5004 restored.
10:18:48.012 MediaServerReporting::RTPSender [MS004000] Call(33) Ext.120: (ACTIVE): Can't send RTP stream to 0.0.0.0:5004 destination unreachable

At first I thought it might have been the gateway's DTMF digit transmit volume setting, but varying this made no difference to this problem.

Any ideas what else I might need to try?
Thanks.

Alec

 

[5qg4]

Have you check your GXW-4104 setting? If using default setting

Please try to goto Channels->Channel Specific Setting->DTMF Methods
Set string as ch1-4:3;
The default is ch1-4:1; (If in-audio only. Most of PSTN not support)

Let me know it works or not.

 

[AlecM]

Hi Ricky,

Thanks for your reply. Sorry, I should have mentioned that I already have that setting for DTMF Method on the 4104, but still get a problem.
My setting: DTMF Methods(1-7): ch1-4:3;
= in-audio and RFC2833.

It seems that the issue is a little intermittent. I'm testing the settings against my mobile's voicemail menu (Vodafone UK), which is a convenient mechanism, as it requests the full mobile number to be entered as the login process.

When doing this, I usually get a "Sorry that number has not been recognised" response, which is their method of politely responding that the tones didn't get recognised as a valid number (and yes, I have checked I've dialled my number correctly).

I wonder if it is speed-sensitive? I did get one response of "do not use more than 19 digits", which was puzzling! But on the next attempt, again the same number was not recognised.

Also, on the FXO Lines configuration page of the 4104, there are settings for DTMF Digit Volume (dB), DTMF Digit Length and DTMF Dial Pause, which I have at the defaults (for firmware version 1.0.1.2).

These would seem to be potential targets for tweaking, which I'm happy to try, but it would be helpful and a lot quicker if someone could suggest known working values (for in the UK).

Thanks

Alec

 

[LLangston]

Alec,
Set the DTMF in the gateway to ch 1-4:2 and in the phone check the RTP box. I had the same problem with the Grandstream products and when I set the phones and the gateway to use RTP(RFC2833) then the digital menus of called numbers worked.

 

[AlecM]

Hi Laura,

Unfortunately, I already have those settings on the handsets and the gateway, but the issue is still manifest.

But you give me an idea about possibly trying toggling the "Bind to Media Server" (PBX handles the audio) option for extensions.

Thanks for the input!

Alec

 

[LLangston]

Sorry Alec,
I thought you had already toggled that for the extensions. Yes, the Bind to Media server should be toggled for the extensions.