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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