DTMF detection issues when Mobile->SIP

[Anish Mangal]

Hi,

I am running asterisk 11.22.0 (thanks to Terry) on an AR150, and I seem to be having a strange DTMF tone detection issue.

I built a simple IVR menu where Background (and/or WaitExten) (I tried both) are used to redirect the user calling on the mobile network to a user on the SIP/Softphone network.

My dialplan looks something like this:
exten => s,1,Answer(500)
exten => s,n,Playback(please-dial-the-extension)
exten => s,n,WaitExten(10)
exten => s,n,Hangup()

I also created a test extension with exactly similar user-flow
exten => _6666,1,Answer(500)                                                   
exten => _6666,n,Playback(please-dial-the-extension)                           
exten => _6666,n,WaitExten(10)                                                 
exten => _6666,n,Hangup() 

Now when I test internally on the SIP network by dialing 6666, I am prompted to press an extension. I can enter an extension, such as 3-0-0 and then I get connected to that user.

However in the real world case (incoming call from cellphone), the moment I press 3 (out of 3-0-0) asterisk tries to dial the extension 3, which obviously doesnt work.

Log: [Feb 19 05:47:04] WARNING[14105][C-0000000d]: pbx.c:6846 __ast_pbx_run: Invalid extension '3', but no rule 'i' or 'e' in context 'incoming-dongle'

I am guessing this is some issue with the DTMF tone detection when running on the PSTN network. I am testing this with a nokia candybar phone.

I am wondering if anyone has come across something similar before, and what are the steps that were taken to fix this.

Best,

Anish

[Anish Mangal]

For what it's worth, I am also using chan_dongle to connect the 3G dongle (Huawei K3765) to the router. dongle.conf relevant settings are below.

language=en                     ; set channel default language                                                                              
smsaspdu=yes                    ; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'      
mindtmfgap=45                   ; minimal interval from end of previews DTMF from begining of next in ms                                    
mindtmfduration=80              ; minimal DTMF tone duration in ms                                                                          
mindtmfinterval=200             ; minimal interval between ends of DTMF of same digits in ms  

These exact same settings work well in another asterisk+freepbx setup on an x86_64 machine, so I dont know if this is where things are going wrong, or the issue is elsewhere.

Any pointers in helping to isolate would help :)

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Anish

[Anish Mangal]

Well, a bit more looking, and I have found a workaround ... atleast.

On SIP dtmf detection is rfc2833, while on dongle.conf, it is either relax/inband/off.

Now since DTMF detection seems to be working on SIP channels (and I dont know how to configure dtmf detection in the 'dialplan' extension - if that can be done at all), for now I routed the inbound GSM call first on the SIP extension (6666), where dtmf seems to work

So the flow now is

dongle -> SIP(6666) (IVR) -> (DTMF detection)

as opposed to

dongle -> (DTMF detection)

I guess this might eat up a few more CPU cycles but seems to work for now. :)

--

Anish

[Sammy Dominguez]

Hi Anish try testing with read() func. After Wait() func. And let me know.

exten=>6123,1,Read(Digits,enter-ext-of-person)
exten=>6123,n,Playback(you-entered)
exten=>6123,n,SayNumber(${Digits

Note: sometimes is just that you need to give more time at wait func because the opened channel takes longer on giving you dtmf.

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[Anish Mangal]

Hi,

Thanks for the reply. I didnt have the Read app in my build but now I have one. I will let you know how it goes.

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Anish