Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test setup (to avoid NAT complications, etc...)
Here's the local static one, which works fine: sip.conf: ---------- [general] context=default allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ; [ccme-inbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme allow=all insecure=port,invite canreinvite=no ; [ccme-outbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme trustrpid=yes sendrpid=yes allow=all canreinvite=no dtmfmode=rfc2833 And, in CME: ----------------- dial-peer voice 200 voip session protocol sipv2 incoming called-number 2155551212 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 101 voip description softphones 4-N destination-pattern 4[0-9] monitor probe icmp-ping session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua no remote-party-id registrar dns:sylvester.home.misty.com expires 3600 secondary sip-server dns:sylvester.home.misty.com I think if I want to use host=dynamic in sip.conf on asterisk, I need to do something like this in CME: ----------------------------------- dial-peer voice 101 voip destination-pattern [1-2][0-9] session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua authentication username foobar password 7 060F06233B583F4B00 realm NOTSURE registrar dns:sylvester.home.misty.com expires 3600 sip-server dns:sylvester.home.misty.com And, maybe for sip.conf, something like this: ----------------------------------------------- [foobar] type=friend context=from-ccme host=dynamic secret=notthis username=foobar dtmfmode=rfc2833 But, I'm really not getting far with this. There are tons of examples online of asterisk configurations to initiate connections to static hosts such as SIP providers, and CCME examples using static hosts, but I can't find anything like what I'm doing, even though it seems to me like a common kind of thing to set up. Any help would be greatly appreciated. Mark -- Mark G. Thomas (m...@misty.com) http://mail-cleaner.com/ _______________________________________________ -- 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
