Invalid.gsm: "I'm sorry that's not a valid extension

I’m trying to break down what occurred when the call was placed:

[Apr  6 17:41:07] VERBOSE[12604] manager.c: [Apr  6 17:41:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:41:44] VERBOSE[2874] chan_sip.c: [Apr  6 17:41:44]     -- Registered SIP '200' at 192.168.0.24:5060
[Apr  6 17:41:44] VERBOSE[2874] chan_sip.c: [Apr  6 17:41:44]        > Saved useragent "YATE/5.4.2" for peer 200
[Apr  6 17:41:47] VERBOSE[2812] dnsmgr.c: [Apr  6 17:41:47]        > Refreshing DNS lookups.
[Apr  6 17:41:47] VERBOSE[2812] srv.c: [Apr  6 17:41:47]        > ast_get_srv: SRV lookup for '_sip._udp.nat5.babytel.ca' mapped to host nat5.babytel.ca, port 5065
[Apr  6 17:41:47] ERROR[2812] netsock2.c: getaddrinfo("nat5.babytel.net", "(null)", ...): Name or service not known
[Apr  6 17:41:47] WARNING[2812] acl.c: Unable to lookup 'nat5.babytel.net'
[Apr  6 17:42:03] VERBOSE[12704] manager.c: [Apr  6 17:42:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:42:03] VERBOSE[12706] manager.c: [Apr  6 17:42:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:42:03] VERBOSE[12706] manager.c: [Apr  6 17:42:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:42:03] VERBOSE[12704] manager.c: [Apr  6 17:42:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:42:04] VERBOSE[2874] netsock2.c: [Apr  6 17:42:04]   == Using SIP RTP CoS mark 5
[Apr  6 17:42:04] NOTICE[2874] chan_sip.c: Call from '200' (192.168.0.24:5060) to extension '202' rejected because extension not found in context 'local_200'.
[Apr  6 17:42:08] VERBOSE[12721] manager.c: [Apr  6 17:42:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:42:08] VERBOSE[12721] manager.c: [Apr  6 17:42:08]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:42:35] VERBOSE[2874] netsock2.c: [Apr  6 17:42:35]   == Using SIP RTP CoS mark 5
[Apr  6 17:42:35] VERBOSE[12757] pbx.c: [Apr  6 17:42:35]     -- Executing [200@default:1] Dial("SIP/202-00000007", "SIP/200,60,") in new stack
[Apr  6 17:42:35] VERBOSE[12757] netsock2.c: [Apr  6 17:42:35]   == Using SIP RTP CoS mark 5
[Apr  6 17:42:35] VERBOSE[12757] app_dial.c: [Apr  6 17:42:35]     -- Called SIP/200
[Apr  6 17:42:36] VERBOSE[12757] app_dial.c: [Apr  6 17:42:36]     -- SIP/200-00000008 is ringing
[Apr  6 17:42:41] VERBOSE[12757] app_dial.c: [Apr  6 17:42:41]     -- SIP/200-00000008 answered SIP/202-00000007
[Apr  6 17:42:41] DEBUG[12757] channel.c: setting peeraccount to 202 for SIP/200-00000008 from data on channel SIP/202-00000007
[Apr  6 17:42:41] DEBUG[12757] channel.c: setting peeraccount to 202 for SIP/202-00000007 from data on channel SIP/200-00000008
[Apr  6 17:42:41] VERBOSE[12757] rtp_engine.c: [Apr  6 17:42:41]     -- Locally bridging SIP/202-00000007 and SIP/200-00000008
[Apr  6 17:42:41] WARNING[12757] res_rtp_asterisk.c: RTP Read too short
[Apr  6 17:42:52] VERBOSE[12757] pbx.c: [Apr  6 17:42:52]     -- Executing [h@default:1] AGI("SIP/202-00000007", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----11") in new stack
[Apr  6 17:42:52] VERBOSE[12757] res_agi.c: [Apr  6 17:42:52]     -- <SIP/202-00000007>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----11 completed, returning 0
[Apr  6 17:42:52] VERBOSE[12757] pbx.c: [Apr  6 17:42:52]   == Spawn extension (default, 200, 1) exited non-zero on 'SIP/202-00000007'
[Apr  6 17:43:03] VERBOSE[12814] manager.c: [Apr  6 17:43:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:43:03] VERBOSE[12815] manager.c: [Apr  6 17:43:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:43:03] VERBOSE[12815] manager.c: [Apr  6 17:43:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:43:03] VERBOSE[12814] manager.c: [Apr  6 17:43:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:43:08] VERBOSE[12830] manager.c: [Apr  6 17:43:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:43:08] VERBOSE[12830] manager.c: [Apr  6 17:43:08]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:44:00] VERBOSE[12903] manager.c: [Apr  6 17:44:00]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:44:00] VERBOSE[12903] netsock2.c: [Apr  6 17:44:00]   == Using SIP RTP CoS mark 5
[Apr  6 17:44:02] VERBOSE[12928] manager.c: [Apr  6 17:44:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:44:02] VERBOSE[12929] manager.c: [Apr  6 17:44:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:44:02] VERBOSE[12929] manager.c: [Apr  6 17:44:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:44:03] VERBOSE[12928] manager.c: [Apr  6 17:44:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:44:04] VERBOSE[12903] pbx.c: [Apr  6 17:44:04]        > Channel SIP/200-00000009 was answered.
[Apr  6 17:44:04] VERBOSE[12937] pbx.c: [Apr  6 17:44:04]     -- Executing [8600051@default:1] MeetMe("SIP/200-00000009", "8600051,F") in new stack
[Apr  6 17:44:04] VERBOSE[12937] config.c: [Apr  6 17:44:04]   == Parsing '/etc/asterisk/meetme.conf': [Apr  6 17:44:04] VERBOSE[12937] config.c: [Apr  6 17:44:04]   == Found
[Apr  6 17:44:04] VERBOSE[12937] config.c: [Apr  6 17:44:04]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr  6 17:44:04] VERBOSE[12937] config.c: [Apr  6 17:44:04]   == Found
[Apr  6 17:44:04] VERBOSE[12937] app_meetme.c: [Apr  6 17:44:04]     -- Created MeetMe conference 1023 for conference '8600051'
[Apr  6 17:44:04] VERBOSE[12937] file.c: [Apr  6 17:44:04]     -- <SIP/200-00000009> Playing 'conf-onlyperson.gsm' (language 'en')
[Apr  6 17:44:04] WARNING[12937] res_rtp_asterisk.c: RTP Read too short
[Apr  6 17:44:05] VERBOSE[12903] manager.c: [Apr  6 17:44:05]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:44:07] VERBOSE[12943] manager.c: [Apr  6 17:44:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:44:07] VERBOSE[12943] manager.c: [Apr  6 17:44:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:44:09] VERBOSE[12949] manager.c: [Apr  6 17:44:09]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:44:09] VERBOSE[12950] pbx.c: [Apr  6 17:44:09]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000003;2", "8600051,F") in new stack
[Apr  6 17:44:09] VERBOSE[12949] pbx.c: [Apr  6 17:44:09]        > Channel Local/8600051@default-00000003;1 was answered.
[Apr  6 17:44:09] VERBOSE[12951] pbx.c: [Apr  6 17:44:09]   == Starting Local/8600051@default-00000003;1 at default,919876543210,1 failed so falling back to exten 's'
[Apr  6 17:44:09] VERBOSE[12951] pbx.c: [Apr  6 17:44:09]   == Starting Local/8600051@default-00000003;1 at default,s,1 still failed so falling back to context 'default'
[Apr  6 17:44:09] VERBOSE[12951] pbx.c: [Apr  6 17:44:09]     -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-00000003;1
[Apr  6 17:44:09] VERBOSE[12951] pbx.c: [Apr  6 17:44:09]     -- Executing [i@default:1] Playback("Local/8600051@default-00000003;1", "invalid") in new stack
[Apr  6 17:44:09] VERBOSE[12951] file.c: [Apr  6 17:44:09]     -- <Local/8600051@default-00000003;1> Playing 'invalid.gsm' (language 'en')
[Apr  6 17:44:10] VERBOSE[12949] manager.c: [Apr  6 17:44:10]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:44:13] VERBOSE[12951] pbx.c: [Apr  6 17:44:13]     -- Executing [i@default:2] Hangup("Local/8600051@default-00000003;1", "") in new stack
[Apr  6 17:44:13] VERBOSE[12951] pbx.c: [Apr  6 17:44:13]   == Spawn extension (default, i, 2) exited non-zero on 'Local/8600051@default-00000003;1'
[Apr  6 17:44:13] VERBOSE[12951] pbx.c: [Apr  6 17:44:13]     -- Executing [h@default:1] AGI("Local/8600051@default-00000003;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr  6 17:44:13] VERBOSE[12951] res_agi.c: [Apr  6 17:44:13]     -- <Local/8600051@default-00000003;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr  6 17:44:13] VERBOSE[12950] pbx.c: [Apr  6 17:44:13]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000003;2'
[Apr  6 17:44:13] VERBOSE[12950] pbx.c: [Apr  6 17:44:13]     -- Executing [h@default:1] AGI("Local/8600051@default-00000003;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr  6 17:44:13] VERBOSE[12950] res_agi.c: [Apr  6 17:44:13]     -- <Local/8600051@default-00000003;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr  6 17:45:02] VERBOSE[13063] manager.c: [Apr  6 17:45:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:45:02] VERBOSE[13064] manager.c: [Apr  6 17:45:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:45:02] VERBOSE[13064] manager.c: [Apr  6 17:45:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:45:03] VERBOSE[13063] manager.c: [Apr  6 17:45:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:45:07] VERBOSE[13077] manager.c: [Apr  6 17:45:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:45:07] VERBOSE[13077] manager.c: [Apr  6 17:45:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:46:02] VERBOSE[13170] manager.c: [Apr  6 17:46:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:46:02] VERBOSE[13171] manager.c: [Apr  6 17:46:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:46:02] VERBOSE[13171] manager.c: [Apr  6 17:46:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:46:03] VERBOSE[13170] manager.c: [Apr  6 17:46:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:46:07] VERBOSE[13185] manager.c: [Apr  6 17:46:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:46:07] VERBOSE[13185] manager.c: [Apr  6 17:46:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr  6 17:46:07] VERBOSE[2874] chan_sip.c: [Apr  6 17:46:07]     -- Registered SIP '200' at 192.168.0.99:5060
[Apr  6 17:46:07] VERBOSE[2874] chan_sip.c: [Apr  6 17:46:07]        > Saved useragent "YATE/5.0.0" for peer 200
[Apr  6 17:46:47] VERBOSE[2812] dnsmgr.c: [Apr  6 17:46:47]        > Refreshing DNS lookups.
[Apr  6 17:46:47] VERBOSE[2812] srv.c: [Apr  6 17:46:47]        > ast_get_srv: SRV lookup for '_sip._udp.nat5.babytel.ca' mapped to host nat5.babytel.ca, port 5065
[Apr  6 17:46:47] ERROR[2812] netsock2.c: getaddrinfo("nat5.babytel.net", "(null)", ...): Name or service not known
[Apr  6 17:46:47] WARNING[2812] acl.c: Unable to lookup 'nat5.babytel.net'
[Apr  6 17:47:02] VERBOSE[13281] manager.c: [Apr  6 17:47:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr  6 17:47:02] VERBOSE[13282] manager.c: [Apr  6 17:47:02]   == Manager 'sendcron' logged on from 127.0.0.1

with this sip.conf (relevant portion):

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
register => 12345678901@sip.babytel.ca:jfkldjf543543jdslfds:12345678901@nat5.babytel.ca:5065/12345678901

; VICIDIAL Carrier: BABYTEL - babytel
; Babytel
[babytel_in]
type=peer
qualify=no
host=nat5.babytel.ca
port=5065
context=inbound-calls

[babytel_out]
type=peer
username=12345678901
host=nat5.babytel.net
outboundproxy=nat5.babytel.ca:5065
secret=jfkldjf543543jdslfds
canreinvite=no
insecure=invite



[200]
username=200
secret=password
accountcode=200
callerid="" <200>
mailbox=200
context=local_200
type=friend
host=dynamic

[201]
username=201
secret=password
accountcode=201
callerid="" <201>
mailbox=201
context=default
type=friend
host=dynamic

[202]
username=202
secret=password
accountcode=202
callerid="" <202>
mailbox=202
context=default
type=friend
host=dynamic

[gs102]
username=gs102
secret=password
accountcode=gs102
callerid="Test Admin Phone" <>
mailbox=102
context=default
type=friend
host=dynamic


; END OF FILE    Last Forced System Reload: 2015-04-03 17:14:22

and the extension as:

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
TRUNKloop = IAX2/ASTloop:password@127.0.0.1:40569
TRUNKblind = IAX2/ASTblind:password@127.0.0.1:41569
TRUNKplay = IAX2/ASTplay:password@127.0.0.1:42569
TESTSIPTRUNK = SIP/babytel_out



; agent phones restricted to only internal extensions
[default---agent]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----default---agent-------------------------NO)
exten => s,n,Set(INVCOUNT=0) 
exten => s,n,Background(sip-silence)
exten => s,n,WaitExten(20)


; hangup
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => i,1,Goto(s,4)
exten => i,n,Hangup()
; hangup
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; custom dialplan entries
include => vicidial-auto-internal
include => vicidial-auto-phones




; logging of all outbound calls from agent phones
[defaultlog]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----defaultlog-------------------------NO)
exten => s,n,Set(INVCOUNT=0) 
exten => s,n,Background(sip-silence)
exten => s,n,WaitExten(20)


; hangup
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => i,1,Goto(s,4)
exten => i,n,Hangup()
; hangup
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; custom dialplan entries
exten => _X.,1,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => _X.,n,Goto(default,${EXTEN},1)




[vicidial-auto-external]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Local Server: 192.168.0.19
exten => _192*168*000*019*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*000*019*.,2,Hangup()
exten => _**192*168*000*019*.,1,Goto(default,${EXTEN:18},1)
exten => _**192*168*000*019*.,2,Hangup()

; Agent session audio playback meetme entry
exten => _473782178600XXX,1,Meetme(${EXTEN:8},q)
exten => _473782178600XXX,n,Hangup()
; Agent session audio playback loop
exten => _473782168600XXX,1,Dial(${TRUNKplay}/47378217${EXTEN:8},5,To)
exten => _473782168600XXX,n,Hangup()
; Agent session audio playback extension
exten => 473782158521111,1,Answer
exten => 473782158521111,n,ControlPlayback(${CALLERID(name)},99999,0,1,2,3,4)
exten => 473782158521111,n,Hangup()
; SendDTMF to playback channel to control it
exten => _473782148521111.,1,Answer
exten => _473782148521111.,n,SendDTMF(${CALLERID(num)},250,250,IAX2/ASTplay-${EXTEN:15})
exten => _473782148521111.,n,Hangup()
; Silent wait channel for DTMFsend
exten => 473782138521111,1,Answer
exten => 473782138521111,n,Wait(5)
exten => 473782138521111,n,Hangup()
; VICIDIAL Carrier: BABYTEL - babytel
; Babytel
[general]

[inbound-calls]
 exten => 12345678901,1,Dial(SIP/200)

[local_200]
exten => _9x.,1,Set(CALLERID(all)="Ali Baba" <1234567890>)
exten => _9x.,1,Dial(SIP/${EXTEN:1}@babytel_out)
exten => 201,1,Dial(SIP/201)

[local_201]
exten => 200,1,Dial(SIP/200)

[vicidial-auto-internal]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Voicemail Extensions:
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,n,Voicemail(${EXTEN:14},u)
exten => _85026666666666.,n,Hangup()
exten => _85026666666667.,1,Wait(1)
exten => _85026666666667.,n,Voicemail(${EXTEN:14},su)
exten => _85026666666667.,n,Hangup()
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
exten => 8500,3,Hangup()
exten => 8501,1,VoicemailMain(s${CALLERID(num)})
exten => 8501,2,Hangup()

; Prompt Extensions:
exten => 8167,1,Answer
exten => 8167,2,AGI(agi-record_prompts.agi,wav-----720000)
exten => 8167,3,Hangup()
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----720000)
exten => 8168,3,Hangup()

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
;    Recording is limited to 1 hour, to make longer, just change the server
;    setting ViciDial Recording Limit
;     this is the WAV verison, default
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERID(name)})
exten => 8309,3,Wait(3600)
exten => 8309,4,Hangup()
;     this is the GSM verison
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERID(name)})
exten => 8310,3,Wait(3600)
exten => 8310,4,Hangup()

;     agent alert extension
exten => 83047777777777,1,Answer
exten => 83047777777777,2,Playback(${CALLERID(name)})
exten => 83047777777777,3,Hangup()
; This is a loopback dial-around to allow for immediate answer of outbound calls
exten => _8305888888888888.,1,Answer
exten => _8305888888888888.,n,Wait(${EXTEN:16:1})
exten => _8305888888888888.,n,Dial(${TRUNKloop}/${EXTEN:17},,To)
exten => _8305888888888888.,n,Hangup()
; No-call silence extension
exten => _8305888888888888X999,1,Answer
exten => _8305888888888888X999,n,Wait(3600)
exten => _8305888888888888X999,n,Hangup()

[vicidial-auto-phones]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Phones direct dial extensions:
exten => 200,1,Dial(SIP/200,60,)
exten => 200,2,Goto(default,85026666666666200,1)
exten => 200,3,Hangup()
exten => 201,1,Dial(SIP/201,60,)
exten => 201,2,Goto(default,85026666666666201,1)
exten => 201,3,Hangup()
exten => 202,1,Dial(SIP/202,60,)
exten => 202,2,Goto(default,85026666666666202,1)
exten => 202,3,Hangup()
exten => 102,1,Dial(SIP/gs102,60,)
exten => 102,2,Goto(default,85026666666666102,1)
exten => 102,3,Hangup()

[vicidial-auto]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

include => vicidial-auto-internal
include => vicidial-auto-phones
include => vicidial-auto-external


; END OF FILE    Last Forced System Reload: 2015-04-03 17:14:22

I think it’s related to prefixing a “9” when dialing out?

Admittedly, the SIP provider might not be being reached:

vici*CLI> 
vici*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport ACL Port     Status     
200/200                   192.168.0.24                             D   N             5060     OK (20 ms) 
201/201                   (Unspecified)                            D   N             0        UNKNOWN    
202/202                   192.168.0.15                             D   N             39012    OK (216 ms) 
babytel_in                198.38.7.34                                  N             5065     Unmonitored 
babytel_out/12345678901   (Unspecified)                                N             0        UNKNOWN    
gs102/gs102               (Unspecified)                            D   N             0        UNKNOWN    
6 sip peers [Monitored: 2 online, 3 offline Unmonitored: 1 online, 0 offline]
vici*CLI> 

You never will get the answer to your question if you dont ask them on the right forum. This problem is caused to a misconfiguration on your vicidial setting.

These are some of the requirements when you post a question on the vicidial forum

[quote]
When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager’s manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is “from scratch” you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is “Hosted” list the site name of the host.

If this is a “Cloud” or “Virtual” server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600 [/quote]

My advise is read the Paid version of the manager’s manual and move your questions to the vicidial forum vicidial.org/VICIDIALforum/

Fair enough. Posted to:

vicidial.org/VICIDIALforum/v … =4&t=34333

I was more asking about how the extensions work in the context of Asterisk, but ok.

In the world of telecommunications, the word extension usually refers to a numeric identifier that, when dialed, will ring a phone (or system resource such as voicemail or a queue). In Asterisk, an extension is far more powerful, as it defines the unique series of steps (each step containing an application) through which Asterisk will take that call.

asteriskdocs.org/en/3rd_Edit … ECT-1.html