DTMF problem with SkypeIn

[saul jogui]

Hi everyone.
I'm having problems to detect DTMF input from an user that is connected to my asterisk server from a SkypeIn line. When my asterisk answers a call from the SkypeIn and then asks the user to input the extension (I'm using WaitExten() cmd), my asterisk server does not recognize anything the user may input and sends him to the "t" extension (timeout state and no input detected at all).
Can anyone help me with this?

I'm using asterisk 1.6.0.13

Here is a copy of my config files. (I haven't done any other major modification in any config file other than the 2 below)

**** chan_skype.conf ****

[skypein.user]
context=incoming_skype
secret=XXXX
disallow=all
allow=ulaw
direction=incoming

**** extensions.conf ****

[incoming_skype]

exten => skypein.user,1,Answer()
exten => skypein.user,n,NoOp(Call from ${CALLERID(all)})
exten => skypein.user,n,Set(TIMEOUT(digit)=5)
exten => skypein.user,n,WaitExten()

exten => 10,1,BackGround(beep)
exten => 10,n,NoOp(Answered exten 10)

**** The output from the asterisk panel ****

-- Executing [skypein.user@incoming_skype:1] Answer("Skype/skypein.user-b006b488", "") in new stack
-- Executing [skypein.user@incoming_skype:2] NoOp("Skype/skypein.user-b006b488", "CallerID -> "" <>") in new stack
-- Executing [skypein.user@incoming_skype:3] Set("Skype/skypein.user-b006b488", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing [skypein.user@incoming_skype:4] WaitExten("Skype/skypein.user-b006b488", "") in new stack
-- Timeout on Skype/skypein.user-b006b488, going to 't'
-- Executing [t@incoming_skype:1] NoOp("Skype/skypein.user-b006b488", "Timeout - t") in new stack
-- Executing [t@incoming_skype:2] Hangup("Skype/skypein.user-b006b488", "") in new stack
== Spawn extension (incoming_skype, t, 2) exited non-zero on 'Skype/skypein.user-b006b488'
[Aug 24 17:07:19] NOTICE[24128]: core.cpp:2103 sfa_call_hangup: ending call

[saul jogui]

I also tested with 1.6.1.5 and 1.4.26.1 and nothing worked as well.
This mean I've tested with 1.6.0, 1.6.1 and 1.4 (all supported versions).

After that I thought it could be the license or even the coded I was using.
Since I was using the beta license, I was using "ulaw" or "alaw" codecs.
So I bought a license from digium, which comes with a g729 license and tried them. Well, I'm now using the official/final license and the g729 codec but still, no DTMF.
I've also noticed that the SkypeIn is detecting the Caller ID but it is not passing it to the asterisk ${CALLERID} variable and until now, I don't see how I can get it in my dialplan.
I wonder if the very same issue is causing both DTMF and CallerID problems with SkypeIn....

Anyone?

[teidel]

I called digium and from what I hear its a known issue due to some codec problem between skype and the sfa. They are working on it (I hope)

[saul jogui]

Hey teidel, thanks a lot!

I'll keep my eyes on that and post any news I have here.

[brianberlin]

I'm having the same issue, has there been any update on this?

[saul jogui]

Hi brianberlin,

sorry but no sign of any update available to fix this.
I'm still waiting for it too.

[twilson]

This is actually a problem on Skype's backend. They are ignoring out-of-band dtmf from their PSTN providers, and therefore not passing it on to us. Some providers also send the dtmf inband, but because this is compressed to g.729, it cannot be reliably decoded.

Skype is aware of the problem and are working to fix it, but it isn't a simple fix. They estimate that it could take as long as 4 weeks to fix. I would expect things to just start working once they fix it, I don't *think* that we will need to make a new release for it. Of course, we'll try to let everyone know when they tell us that it is fixed.

[saul jogui]

Hey twilson,

thanks for the update.
Well, I've been waiting for this fix since I got my hands on the beta license, so 4 more weeks I can handle.

Let's just hope that the folks at skype publish the fix as soon as possible.

[saul jogui]

Hello everyone.

Just an update.
I've updated the asterisk to 1.6.1.9 and updated the skypeforasterisk to this new release 1.0.6
The ${CALLERID} variable has been fixed.
But the DTMF is still not being recognized.

[jln17]

to this new release 1.0.6
The ${CALLERID} variable has been fixed.
But the DTMF is still not being recognized.

Connected to Asterisk 1.6.2.0-rc4 currently running on ns (pid = 19286)
ns*CLI> skype show version
Skype For Asterisk Components:
Channel Driver: 1.6.1_1.0.6
Library: 1.6.1_1.0.6

I can confirm. DTMF is still not working!

Bye,
Jean-Louis

[mag_aleksey]

I'm having the same issue with DTMF in SkypeIn calls from New Zealand and USA. DTMF is not working yet.

Code: Select all

asterisk*CLI> skype show version
Skype For Asterisk Components:
        Channel Driver: 1.6.0_1.0.6
        Library: 1.6.0_1.0.6

Any updates?

[fotofx]

I contacted Skype support and received the following email.

Hello ,

I am Mihkel from Skype support. As promised during our Chat Support
session 281149, I am contcating you concerning the problem you are
experiencing with Asterisk and DTMF tones.

I have further looked into this and we are working on this problem. It
will be fixed in the beginning of January 2010.

We apologize for the inconvenience.

Best regards,

Mihkel P.

Skype Customer Support

Visit http://www.skype.com for latest news, updates and tips.

[saul jogui]

Hi everyone.

The new SFA 1.0.9.2 has been released today (06 Jan 2010).
I've tested it with Asterisk 1.6.1.12 AND 1.6.2.0 but still no signs of progress on this matter.
I got the same information from the skype support so I'm checking constantly for new releases.
These past few days, Digium has published several new SFA releases so I'm hopping that, soon, they release one version that would finally solve this problem.
If I have any news, I'll post them here.

[fotofx]

Has anyone heard any update on this problem?

Thanks

[twilson]

Still waiting on a fix from Skype. They have a fix they are testing, but as it affects all of their gateways it takes a while to get through testing and deployment--so there isn't an ETA yet. But they are working on it. We talk to them about it every week. Sorry for the delay.

[jvictor]

Has anyone heard if there has been any progress on this issue?

[twilson]

They found a bug with their fix, so they've fixed that and it is back in testing. Shouldn't be too much longer, but still no ETA.

[mag_aleksey]

Hi to everybody.

I'm glad to inform you that new version of SFA (26-Mar-2010) is available to download. And DTMF works! Thanks.

Code: Select all

asterisk*CLI> skype show version
Skype For Asterisk Components:
        Channel Driver: 1.6.0_1.0.11
        Library: 1.6.0_1.0.11

[twilson]

Well, sort of. Skype is still pushing the update out to all of their servers. They won't be finished until Monday sometime. So it will work on *some* calls. 1.0.11 is required because their fix required us to rebuild as well. So, using 1.0.11 after Monday should result in consistently working DTMF on SkypeIn calls.

[saul jogui]

After reading twilson and mag_alekseys messages, I tried the new 1.6.2.0_1.0.11 and it worked like a charm, at least calling from my office. I'll try calling from other landline phones and mobiles from different carriers to see what happens and I'll post the result later.
And I would like to thank twilson for all his assistance on this topic.

[twilson]

twilson would like to thank everyone for their patience. It took a very long time to get this fix out to everyone. Skype has to be very careful with the code that gets put on their gateways, so there was tons of testing and lots of delays. I'm glad things apear to be working.

[mag_aleksey]

Hello, dear colleagues.

I'm faced again with the same DTMF problem for once with online JP number. I've tested with SFA 1.6.0_1.0.11 (26-Mar-2010) and 1.6.0_1.1.0.1 (08-Jun-2010) - unsuccessful. Engineers from Skype said the DTMF only support inband DTMF in Japan.

The support case I've just opened.