Inbound DTMF disabled?

[gil0109]

This is wierd but I wanted to see if some can confirm my suspicion.

Setup: Asterisk 1.4
Three Voip Accounts. One account with freephoneline.ca and two with another provider.

When calling my phone number and no one picks up, my Asterisk box is used for voicemail.
I added an extra line in my extensions.conf to allow you to go to voicemail if you click "*" just before leaving a message.

If I dial my two other provider numbers, then the call hits my asterisk box and I can get to the voicemailMain application to check my messages by clicking "*"
BUT if I dial the freephoneline.ca number, I get to my asterisk box but I am unable to get to the voicemailmain application by clicking "*"

I think that when freephoneline.ca connects to my asterisk box it is using info as dtmf which doesn't allow for tones with asterisk...

Here is some of my debug logs:

Working when dialing to other provider:

[2010-09-02 06:54:02] DTMF[2832]: channel.c:2226 __ast_read: DTMF begin '*' received on SIP/6049091446-005befd0
[2010-09-02 06:54:02] DTMF[2832]: channel.c:2230 __ast_read: DTMF begin ignored '*' on SIP/6049091446-005befd0

[2010-09-02 06:54:03] DTMF[2832]: channel.c:2148 __ast_read: DTMF end '*' received on SIP/6049091446-005befd0, duration 280 ms
[2010-09-02 06:54:03] DTMF[2832]: channel.c:2211 __ast_read: DTMF end passthrough '*' on SIP/6049091446-005befd0
-- Executing [a@sip:1] VoiceMailMain("SIP/6049091446-005befd0", "100") in new stack


Not Working when dialing to freephoneline.ca (does not detect the DTMF)


-- Executing [s@sip:2] VoiceMail("SIP/16042834274-005c68d0", "100") in new stack

-- <SIP/16042834274-005c68d0> Playing 'vm-intro' (language 'en')

[gil0109]

Ignore the above...

I added dtmfmode=auto to my sip.conf and all is well...very strange...