Trouble with call monitoring

[ajax1515]

So I've followed some of the suggestions on a few of the other posts regarding this topic but to no avail. I can't seem to get the Monitor function to work.

Here's what I'm doing:
I've got a SIP phone 201 and it has Admin Monitor set to 1
I go into Real Time Reporting and activate Monitor
I click Listen next to an agent's name...

Here is the asterisk CLI output I get:

Executing [192*168*001*021*08600060@default:1] Dial("Local/192*168*001*021*08600060@default-0000003d;2", "IAX2/cs-dial6:*******@192.168.1.21:port/08600060,55,oT") in new stack
-- Called IAX2/cs-dial6:********@192.168.1.21:port/08600060
-- Call accepted by 192.168.1.21 (format ulaw)
-- Format for call is ulaw
-- IAX2/cs-dial1-7978 answered Local/192*168*001*021*08600060@default-0000003d;2
> Channel Local/192*168*001*021*08600060@default-0000003d;1 was answered.
-- Executing [201@default:1] Dial("Local/192*168*001*021*08600060@default-0000003d;1", "SIP/201|60|") in new stack
[Jan 5 14:40:04] WARNING[10642]: pbx.c:1475 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/201|60|)) <----- not sure what this is about might be the problem right here.
== Using SIP RTP CoS mark 5
[Jan 5 14:40:04] ERROR[10642]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("201|60|", "(null)", ...): Name or service not known
[Jan 5 14:40:04] WARNING[10642]: chan_sip.c:5711 create_addr: No such host: 201|60|
[Jan 5 14:40:04] WARNING[10642]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) <--- Obviously this and the preceding lines are where the issue is.
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [201@default:2] Goto("Local/192*168*001*021*08600060@default-0000003d;1", "default,85026666666666201,1") in new stack
-- Goto (default,85026666666666201,1)
-- Executing [85026666666666201@default:1] Wait("Local/192*168*001*021*08600060@default-0000003d;1", "1") in new stack
-- Executing [h@default:1] AGI("Local/192*168*001*021*08600060@default-0000003d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0") in new stack
-- <Local/192*168*001*021*08600060@default-0000003d;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---0-----0 completed, returning 0
== Spawn extension (default, 192*168*001*021*08600060, 1) exited non-zero on 'Local/192*168*001*021*08600060@default-0000003d;2'
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [85026666666666201@default:2] VoiceMail("IAX2/cs-dial1-7978", "201,u") in new stack
-- <IAX2/cs-dial1-7978> Playing 'vm-theperson.gsm' (language 'en')
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- <IAX2/cs-dial1-7978> Playing 'digits/2.gsm' (language 'en')
-- <IAX2/cs-dial1-7978> Playing 'digits/0.gsm' (language 'en')
-- <IAX2/cs-dial1-7978> Playing 'digits/1.gsm' (language 'en')
-- <IAX2/cs-dial1-7978> Playing 'vm-isunavail.gsm' (language 'en')
-- <IAX2/cs-dial1-7978> Playing 'vm-intro.gsm' (language 'en')
-- <IAX2/cs-dial1-7978> Playing 'beep.gsm' (language 'en')


It's worth noting I have six dial servers and the admin phone (201) is on dial server 6 while the agent's I'm trying to listen to are on dial server 1, although the IAX2 calls make that pretty clear...

Thanks in advance for any help here.

[ajax1515]

Ok, so I tried putting the admin phone on the same server as the agent im trying to listen to and it seemed to work. I'd still like to be able to monitor across servers, so I'm hoping there is a way to solve that issue.

[mflorell]

Do you have "Generate Cross-Server Phone Extensions" enabled in system settings?

[ajax1515]

No I did not... Was not aware of that setting, thank you as always.