Summary:	ASTERISK-04007: DTMF features do not work for calling party
Reporter:	xrobau (xrobau)	Labels:
Date Opened:	2005-04-28 01:47:52	Date Closed:	2011-06-07 14:10:07
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Resources/res_features
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	With /etc/asterisk/features.conf having: atxfer => ** and two extensions, 301 and 302. If 301 calls 302, 302 can hit '**' and get the 'Transfer' prompt. 301 can't. If 302 then hangs up, and calls 301, 301 is then able to use **, and 302 isn't.
Comments:	By: petersv (petersv) 2005-04-28 02:25:50 This is probably the same bug as ASTERISK-3969. That bug note was closed with the usual "Contact Digium Technical Support". It is unfortunate since we are no longer able to discuss the problem here, nor have I received any information back from Digium. DTMF seems to be broken for outgoing call legs on zap. The dsp seems to be enabled, but non-functional. It may possibly affect other channels that depend on the dsp for digit decoding as well. By: xrobau (xrobau) 2005-04-28 02:28:58 I originally thought that, but this has nothing to do with Zap - this is SIP->SIP or IAX->IAX or any combination. In fact, the only thing I haven't tried is zap channels, because after reflashing phones to try to isolate it, and verifying that it's 'when you call out, DTMF doesn't work', I was pretty certain it was a core bug, rather than a module. By: Diego Ercolani (dercol) 2005-04-28 02:36:23 Have you used t, T options to dial? By: xrobau (xrobau) 2005-04-28 02:51:30 Well. Isn't *my* first bug report looking bloody stupid. Sigh. I humbly apologise for wasting your time. Is there a 'close bug because idiot user wasn't *really* watching the debug logs, but was, in fact, totally imagining putting options onto the dial string'? If not, I think there should be. You could use it for this one.