Hi, we've got a CIX200 with an IP card for SIP extensions. We also have a mixture of digital and analogue extensions. The CO lines are analogue. I recently got a Linksys SPA2102 to register on the CIX200, one can make internal calls, receive calls etc. That works fine. However there is a problem with DTMF on the CO lines. If one dials "0", to get an outside line, then wait for external dial tone and then one tries to dial a number you hear the DTMF tones, but dial tone continues. Like the CIX is not passing the DTMF tones onto the line. If one punches the number into the phone and then presses "dial" it works fine. I think the digits are probably passed as part of the SIP call setup in that case.

I've tried "in band", "rfc2833", "SIP INFO", "AVR" etc, with no success.

I believe the problem here would be the same as if you tried to interact with some IVR from the SIP extension.

I'm using eManager to configure the CIX.

Has anybody got a SPA2102 or similar to work with a CIX200?

Thanks in advance!

I've never used the SPA2102, but am curious how you're using it.

Are you taking the existing analog ports on the system and sending them elsewhere, or are you using SIP service to generate dial-tone from the SPA2102?

Hi Tony, the setup is that the SPA2102 is in another building connected to the building that the CIX200 is in via fibre. It is IP from the SPA2102 to the CIX200. The SPA2102 generates dial tone. However if you dial a "0" then the CIX200 connects to an outside line and the dial tone is then coming from the phone company. It is here that the DTMF digits dialed from the phone connected to the SPA2102 don't seem to be sent out onto the outside line.

Dialing other extensions from the SPA2102 works fine. Also if one presses all the digits first (in this case a cordless phone) and then press the "dial button" on the cordless so that it sends the "0" and the number all in quick succession then that works fine too. It is only when you dial a "0" wait for external dial tone and then continue dialing that it fails.

I ahevn't used an FXS device on the CIX, but I have setup cordless SIP phones. Per the manual, rfc2833 is the only supported method.

Just out of curiousity, I would call another land line (I wouldn't use a cell phone) and after the call is established, press some digits on the cordless phone to see how the digits sound, if there is any sound.

I don't know if this will tell you a lot. Normally the caller on the PBX side is muted until a valid number is dialed, but it may tell you what the Toshiba hears as far as DTMF after the SIP call is established.

The CIX is lacking the ability to show the SIP messages. I don't know if the SPA2102 will show you the SIP emssages in a log or not.

Wireshark the xIPU card on the CIX and you will be able to see the communication between the SIP handset and the xIPU card.

Hi David and Carl.

Thanks for your suggestions; sorry for the delayed response I've come down with the flu. I should be back on site next week and will try setting up a call:
* by dialing the complete external number with the leading zero
* in contrast to dialing "0" and then waiting for external dial tone and then dialing the number.
* Once I have a connection will press buttons and see if anything is heard.

I did do a similar test where I dialed another extension from the SIP ATA and then pressed numbers and they came out the other end loud and clear; but that case could be handled differently by the CIX200.

Regarding the packet capture; yes, I was planning on that; just got to find a switch with port mirroring.

I was also wondering if the dial plan on the ATA could be doing something? I have a |0S0| in there so that 0 is dialed immediately, but does that prevent further DTMF signals being sent? Imagine interacting with a booking system etc?

Craig

Normally with SIP handsets you dial the complete number with the line access code then press the call/dial/green button. Once the call has been established you should be able to use DTMF without any problems.
Set the DTMF to RFC2833 as well.