Digital receptionist options not working..!

[keith.bucknall]

Hi again,

Just to advise i have configured a Digital receptionist menu system, i can hear this by dialing my Inbound SIP number. the menu displays.

I have 4 options routing to 4 internal ext numbers, when pressing 1,2,3 or 4 nothing happens. I can't even manually dial an internal ext number.

Can someone please help.

thanks

keith

 

[Anonymous]

Some initial questions:

Do you have incomming and outgoing SIP calls on the same port? You can only have one conversation on one port. So if you call in on your sip number and your config routes the number you divert to through the same SIP provider you might run into troubles.

PSTN ---> 3cx can do
PSTN ---> 3cx ---> VoIP (either internal or external via SIP) can do
SIP1 ---> 3cx ---> SIP1 can not do, due to only one line available for incomming and out going, which will be the same.
SIP1 ---> 3cx ---> SIP2 can do (if you have more than one sip lines).

Also your numbers you divert to need to be prefixed the way you set it up in your outbound rules. You can do this two ways (lets say if your outbound rule states to press 0 prior call)
1. 0+phonenumber
2. 0,+phonenumber (have not tried this, but in the old days a comma allowed for a bit of thinking time )

 

[keith.bucknall]

thanks for the reply, but I not that good with the SIP systems and can not really understand what you mean.

At present the setup i have 2 Trunks or SIP providers - Sipgate and VoIPUser. I have configured Outgoing for Sipgate dial 7 then the number required and for VoIPUser dial 8.

They both have incoming but I guess they are on the same port as i have not changed the default settings. The incoming VoIPUser number I only use as a backup.

Primary number we use all the time is the Sipgate number. How should it be configured as i only walked through then manual when setting these up.

 

[keith.bucknall]

itfarmer

I hope you managed to read my reply, can you please assist.

Keith

 

[Alexander]

Hi Keith !

It looks like DTMF recognition problem.
The most probable reasons are:
1. Your provider does not support DTMF.
2. You are using codec which does not support DTMF.
3. The DTMF settings of your phone ain't correct (you have to define either Inband DTMF or RFC2833).

There are also a lot of messages concerning DTMF recognition in this forum. Please have a look.

Thanks

 

[keith.bucknall]

Alexander

Many thanks for the reply, my SIP provider is Sipgate - do you know if they support this? Plus what codec should I use.

Sipgate worked fine with my old Trixbox / A@H system???

Keith

 

[Anonymous]

RFC2833 is what you should use as astrix is using that.

Astrix voices are recorded in GSM.

So to sum it up:
Use DTMF RFC2833
Use Codecs g729 or g729a

These settings you can setup on your phone (I use IP phones, not sure actually if you can do all of that on a USB phone)

Re your first post (sorry for the late reply but work needs to be done before play :lol: )

Your config should work, if you dial from a phone into your SIP so it looks like you done the config ok. Now I hope that the info from me and Alexander will help you sort this DTMF thing.

I posted a reply to your other new post aswell, pretty much allong the same lines.

Sorry again for the late reply.

 

[keith.bucknall]

I am happy changing the DTMF and codec's on my Cisco 7940's but i can not see why dialling into the digital receptionist pressing 1 does not work.

The original WAV was recorded for Asterisk so do i need to re-record this in the format specified for 3cx.

Otherwise I can not see why this does not work if it worked on asterisk with the same RFC.

The Cisco 7940's and soff phones work dialing out and internally the only issue I have is pressing 1 or 2 from the digital reciptionist.

Keith

 

[Anonymous]

Use Codecs g729 or g729a this is a licensed codecs.

You use Cisco, they are not always providing the licensed codecs. Astrixs uses GSM codes G711

CHeck if your Cisco has G729 Codecs support. For that reason I steered away from cisco (to clever for my liking), you need firmware for this firmware for that. On top of that they are not SIP phones out of the box you need Call Manager to support all functions.

If you want the functions and do not have call manager you have to play with the firmware to get some basics going.

I think we can narrow it down to Codecs.

 

[keith.bucknall]

i am not sure it is, but i have a grandstream 101 and will use that for testing. The X-lite softphones should also work then????

The cisco 7940 is already on the SIP firmware???

Like i said if the phones work internally and externally then why would this affect the digital receptionist - surely this is inbound and not outbound...

I thought if i dailed my Sipgate number it would route to sipgate then route down to the 3CX box, the voice menu appears I press 1, this gets transmitted back and at this point 3CX will then dial the internal extenstion number.

Please forgive if i am wrong but if i can dial from softphone to cisco phone, dial externally on both why the digital receptionist does not work

 

[Anonymous]

i am not sure it is, but i have a grandstream 101 and will use that for testing. The X-lite softphones should also work then????

That is a fair assumption, i really have to get my head arround these softphones one day.

The cisco 7940 is already on the SIP firmware???

Do not know, hope you could tell me anyway this is how you display it:
Using the phone top interface:
Press the settings key. The Settings menu is displayed.
Highlight Status.
Press the Select soft key. The Setting Status menu is displayed.
Highlight Firmware Versions.
Press the Select soft key. The Firmware Versions panel is displayed.

I thought if i dailed my Sipgate number it would route to sipgate then route down to the 3CX box, the voice menu appears I press 1, this gets transmitted back and at this point 3CX will then dial the internal extenstion number.

Hold that thought as I think that is correct. (you have to dial your sipgate number from a phone other than connect to your ATA and 3cx otherwise you might get a SIP port 5060 problem (two processes on one port))

Question:
Setup a line(extension) and have all the calls routed to the voice mail. When done that, call the extension. The question is can you hear the announcement "please leave your message..."?

You have to make sure that you use G729a codecs. Cisco uses MoH G711 and voice G729a, on the phone you can set these settings try to set them both to G729a and see what happens.

 

[keith.bucknall]

this works fine now in the new release and was related to sipgates DTMF - SIP_INFO setting