[Asterisk-Users] DTMF modes and external IVR systems over ISDN

[Paul Cheng]

Stefano,

I've come across this problem as well using SIP devices and asterisk. 
As far as I can tell, the IVR systems are deliberately not answering in 
order to not pay the telco for call charges. Ironically, they are 
sending audio before they answer the call. Depending on what device you 
are using, you may or may not receive the audio on your phone.

For example, using a Cisco ATA186 via SIP and Asterisk, I can dial 
pretty much any IVR system and even if they don't answer, I can hear 
the audio. However, passing DTMF is another issue. The only reliable 
connection type that I've found is using IAXTEL. IAXTEL successfully 
passes DTMF to IVR systems (using the ATA186 via SIP) before the call 
is answered. Using other VoIP systems (Cisco GW, local alternative 
telco calling card, etc.) I've not been able to pass DTMF before the 
call is answered.

There is a kind of chicken and egg problem here. The IVR system doesn't 
"answer" the call until after the user has selected the first level 
menu prompts. However, if it is only a DTMF system (no voice 
recognition) and the user can't pass DTMF, then the call will  never go 
answered and time out.

Using the same connection methods and different SIP devices, however, 
yields different results:

1) Grandstream Budgetone 102 - According to their tech support, early 
audio is not supported in the current firmware. The phone keeps ringing 
and ringing.
2) Voicefinder GW - Using my office *, early audio is successfully 
processed and with IAXTEL, DTMF can be passed. However, using the EXACT 
SAME configuration of Asterisk (same CVS, same drivers except i4L) at a 
machine hosted at a data center, no audio, just ringing and ringing. I 
suspect this is a NAT issue, but the Cisco ATA works in this exact 
configuration and NAT.
3) Same symptons for 2 other ATA devices. Early audio works in one 
case, not in the other.

Interoperability of equipment, codecs, etc. all add up to some things 
not working correctly in certain cases. I guess this is why VoIP hasn't 
become mainstream--yet...

On Wednesday, August 6, 2003, at 05:10  AM, James Sizemore wrote:

> Tones are to short.
>
> Stefano Finetti wrote:
>
>> Mark,
>>
>> I'm now able to send proper DTMF tones checking on the isdn driver 
>> and using
>> "rfc2833" as dtmf mode for sip.conf and phones.
>>
>> But there is a question that i think only you can check and answer:
>>
>> Why * often when calling via outside line some number that has IVR 
>> systems,
>> doesn't recognize the answer?
>>
>> It stays there, waiting, even if i'm sure the other side of the line 
>> has
>> answered the call (tried in the same time from * and using my mobile 
>> phone).
>>
>> I can't figure out what kind of problem can be, I encountered it in 
>> many *
>> installation...
>>
>> --
>> Stefano Finetti
>>
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>>
>
>
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