Hello all, I've got an Asterisk system I'm working on here, and we often dial remote IVR systems, where our end must enter an extension to get to a remote user. We're using Polycom hardphones here, speaking SIP, and Asterisk sends these out over a PRI line with Zaptel hardware.
I've used rtp debug on the phone, and I've got output, but I can't tell if it's correct or not -- I was dialing extension 221, but the remote system lost one or more of the digits. I'd appreciate another few pairs of eyes checking out the rtp debug... [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '2' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 60 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end accepted with begin '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end emulation of '2' queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin '2' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF begin ignored '2' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 60 ms [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end '2' has duration 60 but want minimum 80, emulating on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[27394] channel.c: DTMF end emulation of '2' queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' received on SIP/199-b31ddc00, duration 222 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '2' put into dtmf queue on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin emulation of '2' with duration 100 queued on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin '1' received on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF begin passthrough '1' on SIP/199-b31ddc00 [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' received on SIP/199-b31ddc00, duration 80 ms [Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' put into dtmf queue on SIP/199-b31ddc00 Thanks! Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users