Lack of information in 'core show channels verbose'

Asterisk Asterisk Support
1

We use the dialer software.
120 - dialer extension
630 - interviewer extension
600600600 - respondent number

Problem: no idea how to know what number the interviewer is speaking with, there is no correlation between 630 and 600600600

Question: should I modify dialplan somehow?

Below indicates the call in progress (dialer with SIP 120 called 600600600 number and transfer the call to SIP 630):
asterisk -rx ‘core show channels verbose’
[size=80][color=#0080FF]Channel[/color] [color=#0000BF]Context[/color] Extension Prio [color=#8b0f12]State[/color] Application [color=#804000]Data[/color] [color=#BFBF00]CallerID[/color] Duration Accountcode PeerAccount [color=#008040]BridgedTo[/color]
[color=#0080FF]SIP/120-0000fe1d[/color] [color=#0000BF]macro-dial-one[/color] s 42 [color=#8b0f12]Up[/color] Dial [color=#804000]SIP/630,TtrI[/color] [color=#BFBF00]120[/color] 00:51:18 [color=#008040]SIP/630-0000fe1e[/color]
[color=#0080FF]SIP/630-0000fe1e[/color] [color=#0000BF]from-internal[/color] 1 [color=#8b0f12]Up[/color] AppDial [color=#804000](Outgoing Line)[/color] [color=#BFBF00]630[/color] 00:51:18 [color=#008040]SIP/120-0000fe1d[/color]
[color=#0080FF]SIP/120-00010429[/color] [color=#0000BF]macro-dialout-trunk[/color] s 2 [color=#8b0f12]Up[/color] Dial [color=#804000]DAHDI/g0/600600600,300,T[/color] [color=#BFBF00]120[/color] 00:00:25 [color=#008040]DAHDI/i1/600600600-15cb2[/color]
[color=#0080FF]DAHDI/i1/600600600-15cb2[/color] [color=#0000BF]from-pstn[/color] 1 [color=#8b0f12]Up[/color] AppDial [color=#804000](Outgoing Line)[/color] [color=#BFBF00]48600600600[/color] 00:00:25 [color=#008040]SIP/120-00010429[/color][/size]

2

core show channels is an engineering diagnostic; it is not a tool for doing CTI. Also, the summary level doesn’t show the bridged channel, so you won’t be able to put the call together (core show channel xxxx might, but I haven’t checked.) If you want to do CTI use AMI.

Also note that the context names suggest you may be using FreePBX, and these forums are not suitable for questions about the intricacies of the FreePBX dialplan.