[jstraten]

I am using Trixbox on VMWare and everything works great except that I can't access voice mail (or any other functionality requiring DTMF) from external calls coming in through the SPA-3102. Calls coming in through SIP/IAX2 and internal phones connected to the SPA-3102 (tested by dialing 7777) work fine.

I already played around with the DTMF Gain, Length and Transfer Method to no avail (eventually 7777 on internal phones stopped working, but I got it back to work). It is almost as if the SPA-3102 never really forwards any DTMF to Trixbox...

Do I miss any setting? Any localization settings? I am in the western region of the US.

I guess from my point of view the most puzzling thing is that it all works fine for internal phones and I basically use the same phone for my tests to call my regular PSTN phone number through VOIP. In addition I also tried multiple cell phones to no avail...

Maybe somebody using the same configuration could help me?

Thanks,
Jens

[hwittenb]

Originally Posted by jstraten

Do I miss any setting? Any localization settings? I am in the western region of the US.

You didn't mention it and perhaps you have it set but you should have the DTMF Tx Method set to AVT on the Sipura adapter and on the asterisk config you should have dtmfmode=rfc2833.

[jstraten]

hwittenb,

Thanks for your reply.

My setting was actually set to "Auto". I changed it to "AVT", but that didn't solve the problem. Basically the SPA-3102 is still not forwarding any DTMF to Asterisk for incoming calls from the PSTN line...

Any other suggestions?

Thanks,
Jens

[hwittenb]

To confirm things, you could use the Sipura debug trace facility. If the debug level is set to 3, I believe the debug trace will show the out of band AVT (rfc2833) dtmf activity being sent or received.

[jstraten]

hwittenb,

I know where to change the debug level, but I don't know where it shows me the results. Do I need to install a syslog server to get this to work?

I am also wondering if it could be related to my dial plan on the PSTN line. It currently says: <S0:600>

600 is a valid extension in my Trixbox. Do I need to specify anything else?

Thanks,
Jens

[hwittenb]

Originally Posted by jstraten

I know where to change the debug level, but I don't know where it shows me the results. Do I need to install a syslog server to get this to work?

You download a program that runs on your pc from the sipura web site and put the address of your pc in the box on the system tab. On the line tab set debug to full.
Sipura Frequently Asked Questions

I am also wondering if it could be related to my dial plan on the PSTN line. It currently says: <S0:600>
600 is a valid extension in my Trixbox. Do I need to specify anything else?

I don't believe there is anything else you could specify in the dial plan. As mentioned before, in your asterisk system config for the extension I would have dtmfmode=rfc2833.

[jstraten]

hwittenb,

I ran a syslog session and here is what I got:

...
FXO:Start CNDD
FXO:CNDD name=, number=
FXO:Stop CNDD
FXO:CNDD Name= Phone=
Calling:500@127.0.0.1:5060
[1:0]AUD ALLOC CALL (port=20641)
[1:0]RTP Rx Up
Calling:600@192.168.1.34:0
[0]FM Alert Stop RxTx (c=0023f2a0;a=0)
[1:0]AUD Rel Call
SIPLI:REDIR 600@29295c:0
[1:0]AUD ALLOC CALL (port=20641)
[1:0]RTP Rx Up
[1:0]ENC INIT 0
[1:0]RTP Tx Up (pt=0->c0a80122:19092)
[1:0]RTCP Tx Up
CC:Remote Resume
AUD:Stop PSTN Tone
CC:Connected
AUD:Stop PSTN Tone
FXO:Off Hook
FXO:Stop CNDD
Sess Terminated
[0:0]ENC INIT 18
[0:0]RTP Tx Up (pt=18->c0a80122:16360)
[0:0]RTCP Tx Up
CC:Remote Resume
CC:Connected
[0:0]RTP Rx Up
[0:0]RTP Rx 1st PKT @20639(2)
[0:0]DEC INIT 18
[1:0]RTP Rx 1st PKT @20641(2)
[1:0]DEC INIT 0
[0:1]AUD ALLOC CALL (port=20643)
[0:1]RTP Rx Up
[1]RegOK. NextReg in 58 (1)
AUD:Stop PSTN Tone
AVT Tx 3 start
2. Report digit 3 (6)(90 ms)
AVT Tx 3 end
FXOigit=3
AVT Tx 3 start
AUD:Stop PSTN Tone
AVT Tx 3 end
AUD:Stop PSTN Tone
AVT Tx 3 start
2. Report digit 3 (6)(90 ms)
AVT Tx 3 end
FXOigit=3
AVT Tx 3 start
AUD:Stop PSTN Tone
AVT Tx 3 end
[0:0]LAT-- 6(2)
[0]RegOK. NextReg in 58 (1)
AVT Tx 3 start
AVT Tx 3 end
FXOigit=3
AVT Tx 3 start
AUD:Stop PSTN Tone
AVT Tx 3 end
[1:0]LAT-- 6(2)
AVT Tx 3 start
2. Report digit 3 (6)(90 ms)
AVT Tx 3 end
FXOigit=3
AVT Tx 3 start
AUD:Stop PSTN Tone
AVT Tx 3 end
[0:0]LAT-- 5(2)
[0]FM Alert Stop RxTx (c=0023c998;a=0)
[0:1]AUD Rel Call
CC:Ended
[1:0]LAT++ 7(2)
[1:0]LAT++ 8(2)
[0]On Hook
[0]FM Alert Stop RxTx (c=0023a2d0;a=0)
[0:0]AUD Rel Call
DLG Terminated 291080
FXO:CPC
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
[0]FM Alert Stop RxTx (c=0023f2a0;a=0)
[1:0]AUD Rel Call
DLG Terminated 291114
Sess Terminated
Sess Terminated
DLG Terminated 2911a8
Sess Terminated
CC:Clean Up
...


I might not be an expert, but it seems to me that the SPA-3102 clearly recognizes me pushing the "3" button. However it doesn't seem to pass it on to Trixbox.

Any ideas?

Thanks,
Jens

[hwittenb]

Originally Posted by jstraten

I might not be an expert, but it seems to me that the SPA-3102 clearly recognizes me pushing the "3" button. However it doesn't seem to pass it on to Trixbox.

I believe, as you say, it recognizes you pushing the "3" button multiple times however I believe it is sending it. I would look on the receiving asterisk side for solution to your problem. Maybe you can run some traces there.

I setup a test on my SPA3000 calling my pstn number with another voip phone and with the SPA3000 set to automatically dial another pstn number. The answering pstn number required dtmf input which could validate the passthru. When the SPA3000 automatically answered the incoming pstn call it automatically dialed a pstn number using the voip provider setup on the pstn tab. My voip provider on the SPA3000 was Voxee. I believe this tested the passthru of dtmf from the calling phone over the pstn thru the SPA-3000.

I had the trace up and running so that I could see the syslog make a record when I push a key on the calling telephone.

The test results showed that with the pstn line dtmf transmit setting of AVT, AVT+INFO, INBAND or just INFO, they all worked, however with INBAND the codec needed to be set to G711u. INBAND would not work with the codec set to G729 which is what I would expect. I also tried it with the process AVT and process INFO set to NO. Setting these variables to NO did not seem to have any effect. I could not find an explanation in the manual as to what they do.

The thing I found that did make a big difference was the setting for dtmf transmittal used by the phone I was calling from. I was calling using voip to a pstn number. If I set dtmf transmit to INBAND on the calling phone it worked in all cases. If I set it to AVT, it was spotty at best missing digits unless the key was held a long time. It would not work with AVT using VBuzzer at all, spotty using GizmoProject.

[jstraten]

hwittenb,

Encouraged by your posting and test results I did some more testing myself.

My testing shows that DTMF also works if an incoming call goes directly into Asterisk (e.g. SIP/IAX2 call). This actually works no matter what codec I select (729a, GSM, etc.).

It seems that my problem is really limited to calls coming into Asterisk through the PSTN line of the SPA-3102.

Since it works for all other connections I am almost certain that the SPA-3102 never passes on any DTMF from the PSTN line, but I will try to confirm this by doing some tracing (not sure if that functionality is available) in Trixbox/Asterisk.

Just to clarify: You can use DTMF in your system for calls coming into Trixbox/Asterisk through the SPA-3102?

Thanks,
Jens

[hwittenb]

Originally Posted by jstraten

Just to clarify: You can use DTMF in your system for calls coming into Trixbox/Asterisk through the SPA-3102?

I'm not running asterisk/trixbox. I tried to setup a test to validate the passthru of dtmf thru the SPA. I think my test showed me that if I saw the report on the trace, it was passed thru, if I didn't it wasn't. I don't have a 3102, just a 3000, but I think they are similiar. The SPA passes the call thru whatever setup you have on the pstn tab. I had a voip provider proxy setup, you just have an ip address but it is still sip protocol and rtp packets that are passed over an ethernet circuit.

If you have a cheap hub that you can run the ethernet thru so that your pc sees all the packets you can setup ethereal on your pc and look at the packets. If you are sending dtmf inband you won't see anything but if you are sending the dtmf out of band (AVT (rfc2833)) you will see the packets.