Walter Klomp
2005-Mar-21 02:12 UTC
[Asterisk-Users] DTMF doesn't seem to get through incoming ZAP channels
Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting voicemail also doesn't work through ZAP (the selection of menu-options that is...) So now I definately know that DTMF through ZAP doesn't work (anymore, it used to in the past). Is there any way I can troubleshoot this ? I have already set the relaxdtmf=yes option in /etc/asterisk/zapata.conf, which looks like this: [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=2.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 context=default signalling=pri_net; this is connected to voice switch channel => 1-15 channel => 17-31 Any suggestions and assistance would be very welcome. Thanks in advance Walter Klomp Singapore.> >
Peter Svensson
2005-Mar-21 03:54 UTC
[Asterisk-Users] DTMF doesn't seem to get through incoming ZAP channels
On Mon, 21 Mar 2005, Walter Klomp wrote:> I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium > TE410P card. > > Calling into meeting rooms that have been configured with the p option > works fine. > > From ZAP extensions the # key does not work to exit, however from SIP > extensions the # key works fine. This makes me believe that somehow the > DTMF doesn't get through the ZAP interface. After furter experimenting > voicemail also doesn't work through ZAP (the selection of menu-options > that is...)I tried CVS on 2005-03-18 and we found a similar problem with Dial with the transfer options enabled. The calling phone could transfer but the called phone could not. Identical phones etc, and the results were the same when the two endpoints were interchanged. We placed debug logging code at various places, including all the way down in zt_read in chan_zap. It seems that the dsp code got called, but did not detect digits on the outbound leg. Perhaps some state in the dsp code is not initialized properly for DTMF detection? If the called phone was a sip phone set to rfc2833 then transfers work. I ran out of time to test further and reverted to an older cvs release. (Much older). Peter