G'day mate, I've got 15 7960/7940 in my office with firmware 7.4 and have no problems.
>I can make calls from the 7960. When I get a call placed to the 7960 the call >is setup but there is no audio in either direction. Is call from 7960 to 7960? > I have tried firmware versions 6 & 7 on the Cisco phones, same result. Means - something wrong is with your config... [9001] >type=friend ; either "friend" (peer+user), "peer" >context=extensions >secret=9001 >fromuser=Cisco ; overrides the callerid, e.g. required by FWD >callerid=9001 >host=dynamic ; we have a static but private IP address >nat=never ; there is not NAT between phone and >dtmfmode=rfc2833 ^^^^^^^^^^^^^^^ see below, in phone section. > canreinvite=no ; allow RTP voice traffic to bypass Asterisk Hmm... something tells me that RTP stream goes to asterisk, instead of phones' ip addresses. Could you check, what parameter do you have in global and second 7960's section? > progressinband=yes Why do you need this? >disallow=all >allow=ulaw Seconf 7960 has the same config in SIP.CONF? >dhcp_server : 192.168.10.254 >my_ip_addr : 192.168.10.17 >subnet_mask : 255.255.255.0 >defaultgw : 192.168.10.254 >tftp_addr : 192.168.11.2 Does phone receive sipdefault.cnf and SIPxxxx.cnf file from TFTP? >dtmf_outofband : avt >dtmf_avt_payload : 101 >dtmf_db_level : 3 >dtmf_inband : 1 You've set in Asterk's config DTMF as out_of_band, while in phone's config you'set as in_band. Corret it first. >proxy1_address : "192.168.10.106" do sip debug ip ip_address_of_7960 ad take a look. Good luck! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
