First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension:
Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works great. I dial the extension on the 7941, Asterisk parses DTMF just fine. In the 2nd scenario, DTMF digits do not interrupt the "Please enter conf #" announcement, and when I do finish entering the same conf # used for scenario #1, it announces "Invalid Conf #". With channel debugging enabled on Asterisk, I see that I often get duplicate DTMF entries. So where I might have dialed 1234#, Asterisk sees 112344# or similar, under scenario 2. All dial-peers involved on the 2821's are configured for dtmf-relay rtp-nte, and debugging shows this to be the negotiated dtmf method. Interestingly, there is a 3rd scenario which adds to the confusion: Call placed from Boston to European Asterisk Meetme *DID*: (note the trip out to the PSTN is simply a hairpin in this case) Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-PRI-> PSTN <-PRI-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) This is using Asterisk 1.6.0.5 I've tried altering relaxdtmf in sip.conf to no avail. dtmfmode is set to auto. Any suggestions? Regards, --phil _______________________________________________ -- 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
