[asterisk-users] SIP Dial Command to a non-Asterisk url

[Gavin Henry]

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:                0a5d773327e8d067370fa4da03097e58 at 192.168.45.196
 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:jNetX at 192.168.45.183:5605
 Need Destroy:           0
 Last Message:           Tx: ACK
 Promiscuous Redir:      No
 Route:                  sip:jNetX at 192.168.45.183:5605
 DTMF Mode:              rfc2833
 SIP Options:            (none)

Inbound from SIP Provider:

 * SIP Call
 Direction:              Incoming
 Call-ID:                7f73070762b13c4f2445c16f32b80c4b at xx.xx.xx.xx
 <------   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:01XXXXXXXXX at xx.xx.xx.xx     <------   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)