Hi, I'm having a problem with DTMF tones from my SIP client apparently crashing the chan_capi driver. However I'm not sure whether this is a bug or misconfiguration on my part: if I set "softdtmf=1" in /etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support DTMF detection?
The set up I have is using latest CVS (3 days old) running RH8 on a 933MHz P3. SIP client is SJPhone (G711) on Windows connected to * via a 100Mbit switched LAN. PSTN connection is using latest chan_capi on an AVM B1 PCI v4 card (a steal at �1.20 on ebay :)). I can call the * box using the SIP client and interact with the voicemail app with no problems using in-band DTMF. I can also call in from the PSTN through the capi interface and interact with the IVR menu with no problems. Finally I can bridge the CAPI and SIP channels and hear DTMF digits entered on the PSTN phone with no problems (they are also detected and displayed on the console). However when the CAPI and SIP channels are bridged, entering more than a couple of DTMF digits into the _SIP_ client appears to crash the channel: neither party gets disconnected, but there is no longer any audio in either direction and new calls (inbound or outbound) trying to use the CAPI channel fail. Once locked if I enter "capi info" in the * console it return nothing and trying to autocomplete capi commands e.g. "capi [TAB]" just locks the console up. Entering capiinfo and lsmod at the command prompt suggests the driver is ok. The only way of getting it working again is to restart *. When I switch to softdtmf, everything seems to work fine, but I noticed that even though DTMF signalling works fine on the IVR menu, once the call is bridged DTMF digits entered on the PSTN phone are not displayed on the console like before. Jamie Neil Versado I.T. Services Ltd. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
