Quoting Myself :( > 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.
Further to my earlier post... I've replicated this problem using a passive Fritz PCI card, so my bargain B1 card appears not to be the problem. I can only replicate the lock ups when bridging a call directly from chan_capi to chan_sip (or vica versa) and I use in-band DTMF detection in the SIP channel. I have tried various other combinations, including SIP>SIP, CAPI>CAPI, CAPI>MEETME>CAPI and CAPI>MEETME>SIP, and they work without a problem. The SIP client I am using is SJPhone (which does only in-band DTMF), and if I switch off the in-band detection in sip.conf it works fine, but of course I can no longer signal to * from the SIP phone. I can't test whether out of band signalling is ok as SJPhone doesn't support this. I find it hard to believe that this is a capi problem - I would have expected see some other capi related problems aside from this single (easily repeatable) error. Jamie Neil Versado I.T. Services Ltd. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
