This is what is shown when the call connects with: sip show channel
The conference suite from another provider on internal IP is waiting for an ACK on port 5605, but * is sending it back to port 2289 Internal between Asterisk and another Conference suite: * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 14 Non-Codec Capability: 1 Their Codec Capability: 4 Joint Codec Capability: 4 Format ulaw Theoretical Address: 192.168.45.183:5605 Received Address: 192.168.45.183:2289 NAT Support: Always Audio IP: 192.168.45.196 (local) Our Tag: as31c610d6 Their Tag: t1122b SIP User agent: Username: slee Peername: slee Original uri: sip:[EMAIL PROTECTED]:5605 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5605 DTMF Mode: rfc2833 SIP Options: (none) Inbound from SIP Provider: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] <------ REMOVED Our Codec Capability: 14 Non-Codec Capability: 1 Their Codec Capability: 14 Joint Codec Capability: 14 Format gsm Theoretical Address: 193.111.201.32:5060 Received Address: 193.111.201.32:5060 NAT Support: Always Audio IP: xx.xx.xx.xx (local) <------ REMOVED Our Tag: as65c31c43 Their Tag: as26378dd7 SIP User agent: Asterisk PBX Original uri: sip:[EMAIL PROTECTED] <------ REMOVED Caller-ID: 01XXXXXXXXX <------ REMOVED Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:193.111.201.32;lr=on;ftag=as26378dd7 DTMF Mode: rfc2833 SIP Options: (none) _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
