initial commitHEAD master

author: arf20 <aruizfernandez05@gmail.com> 2025-07-11 20:36:01 +0200
committer: arf20 <aruizfernandez05@gmail.com> 2025-07-11 20:36:01 +0200
commit: e661974fbe9ecf8e7eed0e52a41855fd1eaa684d (patch)
tree: 2c0085f58a2feb14566df084ded539bf61aa22db
download: arfnet2-asterisk-e661974fbe9ecf8e7eed0e52a41855fd1eaa684d.tar.gz
arfnet2-asterisk-e661974fbe9ecf8e7eed0e52a41855fd1eaa684d.zip
10 files changed, 1456 insertions, 0 deletions
diff --git a/README.md b/README.md
new file mode 100644
index 0000000..c21ea3b
--- /dev/null
+++ b/README.md
@@ -0,0 +1,4 @@
+# arfnet2-asterisk
+
+Asterisk configuration for ARFNET
+
diff --git a/asterisk-conf/acl.conf b/asterisk-conf/acl.conf
new file mode 100644
index 0000000..7b7976c
--- /dev/null
+++ b/asterisk-conf/acl.conf
@@ -0,0 +1,96 @@
+;
+; Named Access Control Lists (ACLs)
+;
+; A convenient way to share acl definitions
+;
+; This configuration file is read on startup
+;
+; CLI Commands
+; -----------------------------------------------------------
+;   acl show                         Show all named ACLs configured
+;   acl show <name>                  Show contents of a particular named ACL
+;   reload acl                       Reload configuration file
+;
+; Any configuration that uses ACLs which has been made to be able to use named
+; ACLs will specify a named ACL with the 'acl' option in its configuration in
+; a similar fashion to the usual 'permit' and 'deny' options. Example:
+; acl=my_named_acl
+;
+; Multiple named ACLs can be applied by either comma separating the arguments or
+; just by adding additional ACL lines. Example:
+; acl=my_named_acl
+; acl=my_named_acl2
+;
+; or
+;
+; acl=my_named_acl,my_named_acl2
+;
+; ACLs specified by name are evaluated independently from the ACL specified via
+; permit/deny. In order for an address to pass a given ACL, it must pass both
+; the ACL specified by permit/deny for a given item as well as any named ACLs
+; that were specified.
+;
+;[example_named_acl1]
+;deny=0.0.0.0/0.0.0.0
+;permit=209.16.236.0
+;permit=209.16.236.1
+;
+;[example_named_acl2]
+;permit=0.0.0.0/0.0.0.0
+;deny=10.24.20.171
+;deny=10.24.20.103
+;deny=209.16.236.1
+;
+; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
+; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
+; The following example explains how combining the ACLs works:
+; <in another configuration>
+; [example_item_with_acl]
+; acl=example_named_acl1
+; acl=example_named_acl2
+;
+; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
+; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
+; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
+; either, so it passes.
+;
+; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
+; First, example_named_acl1 is evaluated and the address is allowed.
+; However, it is blocked by example_named_acl2, so the address is blocked from the combined
+; ACL.
+;
+; Similarly, the permits/denies in specific configurations that make up an ACL definition
+; are also treated as a separate ACL for evaluation. So if we change the example above to:
+; <in another configuration>
+; [example_item_with_acl]
+; acl=example_named_acl1
+; acl=example_named_acl2
+; deny=209.16.236.0
+;
+; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
+; though it passes the two named components.
+;
+;
+; Named ACLs can use ipv6 addresses just like normal ACLs.
+;[ipv6_example_1]
+;deny = ::
+;permit = ::1/128
+;
+;[ipv6_example_2]
+;permit = fe80::21d:bad:fad:2323
+
+[internal]
+deny=0.0.0.0/0.0.0.0
+permit=192.168.0.0/255.255.0.0
+
+[external]
+permit=0.0.0.0/0.0.0.0
+
+[sdf]
+deny=0.0.0.0
+permit=205.166.94.7/255.255.255.255
+
+[pstn]
+deny=0.0.0.0/0
+permit=176.56.119.0/255.255.255.0
+
diff --git a/asterisk-conf/extensions.conf b/asterisk-conf/extensions.conf
new file mode 100644
index 0000000..47dfc20
--- /dev/null
+++ b/asterisk-conf/extensions.conf
@@ -0,0 +1,233 @@
+[globals]
+tandmxkey=l2408fy4dlia3SqONVFSLD0LVkNmuerd209181Rl8228829DYIB
+tandmxhostid=110
+
+arf20=PJSIP/1051
+all=PJSIP/1001&PJSIP/1002&PJSIP/1051&SCCP/1101
+
+
+[from-internal]
+
+exten => _1XXX,1,Goto(internal-users,${EXTEN},1)       ; 1xxx users
+exten => _2XXX,1,Goto(internal-services,${EXTEN},1)    ; 2xxx services
+exten => _8XXXXXXX,1,Goto(tandmx,${EXTEN:1},1)         ; tandmx prefixed by 8
+exten => _733XXXX,1,Set(number=${EXTEN:3})             ; SDF numbers prefixed by 733
+ same => n,Goto(sdf,${EXTEN:3},1)
+
+; PSTN
+exten => _0ZX,1,Goto(pstn,${EXTEN},1)                  ; emergency numbers (3 numbers starting with 0 followed by non-zero)
+exten => _112,1,Goto(pstn,${EXTEN},1)                  ; emergency service (fixed)
+exten => _XXXXXX,1,Goto(pstn,968${EXTEN},1)            ; regional numbers (6 digits)
+exten => _[6789]XXXXXXXX,1,Goto(pstn,${EXTEN},1)       ; national numbers (9 digits)
+exten => _0ZXXX,1,Goto(pstn,${EXTEN:1},1)              ; short numbers prefixed by 0
+exten => _00ZX!,1,Goto(pstn,${EXTEN},1)                ; international numbers prefixed by 00
+
+; catch invalid extension
+exten => _X!,1,Goto(bogons,s,1)
+
+; catch timeouts
+exten => t,1,Playback(press-button-again)
+ same => n,Hangup()
+
+
+[internal-users]
+
+; arf20 alias
+exten => 1000,1,Progress()
+ same => n,Dial(${arf20}, 20)
+ same => n,Goto(fail-to-internal,${EXTEN},1)
+
+; internal calls between extensions (10xx SIP)
+exten => _10XX,1,Progress()
+ same => n,Dial(PJSIP/${EXTEN}, 20)
+ same => n,Goto(fail-to-internal,${EXTEN},1)
+
+; internal calls between extensions (11xx SCCP)
+exten => _11XX,1,Progress()
+ same => n,Dial(SCCP/${EXTEN}, 20)
+ same => n,Goto(fail-to-internal,${EXTEN},1})
+
+
+[internal-services]
+
+; test
+exten = 2222,1,Answer()
+ same = n,Wait(1)
+ same = n,Playback(hello-world)
+ same = n,Hangup()
+; test2
+exten = 2223,1,Answer()
+ same = n,Wait(1)
+ same = n,Playback(digits/10)
+ same = n,Hangup()
+; Echo test
+exten => 2101,1,NoOp(Echo test)
+same  => n,Playback(demo-echotest)
+same  => n,Echo()
+same  => n,Playback(demo-echodone)
+same  => n,Hangup()
+
+
+; IVR
+exten => 2000,1,Goto(ivr,s,1)
+; conference
+exten => 2001,1,Answer()
+ same => n,ConfBridge(1)
+; asterisk time
+exten => 2002,1,Answer()
+ same => n,DateTime()
+ same => n,Hangup()
+; voicemail
+exten => 2003,1,VoiceMailMain()
+
+
+[ivr]
+; prompt
+exten => s,1,Answer(500)
+ same => n(loop),Background(arfnet/ivr_hello)
+ same => n,WaitExten()
+; on 1
+exten => 1,1,Read(entered_ext,arfnet/please-enter-ext,4)
+ same => n,Goto(from-internal,${entered_ext},1)
+; on 2
+exten => 2,1,Goto(internal-users,1000,1)
+; on 3
+exten => 3,1,Goto(internal-services,2001,1)
+; on 4
+exten => 4,1,VoiceMail(1000@default)
+
+exten = i,1,Playback(option-is-invalid)
+ same = n,Goto(ivr,s,1)
+
+exten = t,1,Playback(are-you-still-there)
+ same = n,Goto(ivr,s,1) 
+
+
+[from-external]
+
+exten => _1XXX,1,Goto(internal-users,${EXTEN},1)
+exten => _2XXX,1,Goto(internal-services,${EXTEN},1)
+
+exten => _X!,1,Goto(bogons,s,1)
+
+
+[from-anonymous]
+
+exten => _X!,1,Verbose(*** Anonymous direct IP inbound call from: ${CALLERID(all)})
+ same => n,Goto(from-external,${EXTEN},1)
+
+
+[sdf]
+
+; to SDF
+exten => _XXXX,1,Verbose(*** Dialing outbound via SDF ${EXTEN})
+ same => n,Set(CALLERID(num)=2063)
+ same => n,Dial(PJSIP/${EXTEN}@sdf)
+ same => n,Goto(fail-to-external,${EXTEN},1)
+
+
+; external trunk calls from sdf
+[from-sdf]
+
+exten => 2063,1,Answer()
+ same => n,Verbose(*** Inbound call via SDF from: ${CALLERID(all)})
+ same => n,Dial(${arf20})
+ same => n,Hangup()
+
+
+[tandmx]
+
+; to Tandmx
+exten => _XXXXXXX,1,Verbose(*** Dialing outbound via Tandmx to: ${EXTEN})
+ same => n,Set(number=${EXTEN})
+ same => n,Set(CALLERID(num)=2736${CALLERID(num):1})
+ same => n,Set(lookup=${SHELL(curl "http://api.tandmx.net/api/v1/?auth=${tandmxkey}&number=${number}&ani=${CALLERID(num)}&host=${tandmxhostid}")})
+ same => n,Dial(${lookup}) 
+ same => n,Goto(fail-to-external,${EXTEN},1)
+
+
+[from-tandmx]
+
+exten => _2736XXX,1,Answer()
+ same => n,Verbose(*** Inbound call via Tandmx from: ${CALLERID(all)})
+ same => n,Goto(internal-users,1${EXTEN:4},1)
+
+exten => _2737XXX,1,Answer()
+ same => n,Verbose(*** Inbound call via Tandmx from: ${CALLERID(all)})
+ same => n,Goto(internal-services,2${EXTEN:4},1)
+
+
+[pstn]
+
+; to PSTN
+exten => _X!,1,Verbose(*** Dialing outbound via PSTN to: ${EXTEN})
+ same => n,Set(CALLERID(all)=868925554)
+ same => n,Dial(PJSIP/${EXTEN}@pstn)
+ same => n,Goto(fail-to-external,${EXTEN},1)
+
+[from-pstn]
+
+exten => 868925554,1,Answer()
+ same => n,Verbose(*** Inbound call via PSTN from: ${CALLERID(all)})
+ same => n,Progress()
+ same => n,Wait(1)
+ same => n,Ringing()
+ same => n,Dial(${all},45,d)
+ same => n,VoiceMail(1000@default,u)
+ same => n,Hangup()
+
+exten => 1,1,Goto(ivr,s,1)
+
+
+[bogons]
+
+exten => s,1,Answer()
+ same => n,Playback(pbx-invalid)
+ same => n,Hangup()
+
+
+[fail-to-external]
+
+exten => _X!,1,Goto(s-${DIALSTATUS},1)	; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => s-NOANSWER,1,Playback(vm-nobodyavail)
+ same => n,Hangup()
+
+exten => s-BUSY,1,Playback(all-circuits-busy-now)
+ same => 1,Busy(5)
+ same => n,Hangup()
+
+exten => s-CHANUNAVAIL,1,Playback(ss-noservice)
+ same => n,Hangup()
+
+exten => s-CONGESTION,1,Congestion(5)
+ same => n,Hangup()
+
+exten => _s-.,1,Goto(s-NOANSWER,1)
+
+
+[fail-to-internal]
+
+exten => _X!,1,Set(MAILBOX=${EXTEN})
+ same => n,Goto(s-${DIALSTATUS},1)	   ; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => s-NOANSWER,1,VoiceMail(1000@default,u) ; If unavailable, send to voicemail w/ unavail announce
+ same => n,Hangup()                        ; If they press #, return to start
+
+exten => s-BUSY,1,VoiceMail(1000@default,b)     ; If busy, send to voicemail w/ busy announce
+ same => n,Hangup()                        ; If they press #, return to start
+
+exten => _s-.,1,Goto(s-NOANSWER,1)         ; Treat anything else as no answer
+
+exten => a,1,VoicemailMain(${MAILBOX})        ; If they press *, send the user into VoicemailMain
+
+
+[send-text]
+exten => s,1,Verbose(*** Sending text to: ${ARG1})
+ same => n,Set(PEER=${CUT(CUT(CUT(MESSAGE(from),@,1),<,2),:,2)})
+ same => n,Set(FROM=${SHELL(asterisk -rx ‘pjsip show endpoint ${PEER}’ | grep ‘callerid ‘ | cut -d’:’ -f2- | sed ‘s/^ *//’ | tr -d ‘‘)})
+ same => n,Set(CALLERID_NUM=${CUT(CUT(FROM,>,1),<,2)})
+ same => n,Set(FROM_SIP=${STRREPLACE(MESSAGE(from),
+ same => n,MessageSend(pjsip:${ARG1},${FROM_SIP})
+ same => n,Hangup()
+
diff --git a/asterisk-conf/iax.conf b/asterisk-conf/iax.conf
new file mode 100644
index 0000000..56d73df
--- /dev/null
+++ b/asterisk-conf/iax.conf
@@ -0,0 +1,33 @@
+;==========================================================================================================;
+;     These values define how IAX connections are made. Only adjust them if you know what you're doing     ;
+;==========================================================================================================;
+
+[general]
+bindaddr=0.0.0.0
+bindport=4569
+qualify=1000
+relaxdtmf=yes
+bandwidth=high
+;disallow=all
+;allow=g722
+;allow=ulaw
+;allow=alaw
+allow=all
+jitterbuffer=no
+tos=0x12
+calltokenoptional=0.0.0.0/0.0.0.0
+requirecalltoken=no
+delayreject=yes
+autokill=yes
+encryption=yes
+
+;======================================================;
+;     Adjust this section as part of TandmX setup!     ;
+;======================================================;
+
+[from-tandmx]
+type=user
+context=from-tandmx
+auth=md5
+secret=REDACTED
+
diff --git a/asterisk-conf/pjsip.conf b/asterisk-conf/pjsip.conf
new file mode 100644
index 0000000..faf8575
--- /dev/null
+++ b/asterisk-conf/pjsip.conf
@@ -0,0 +1,278 @@
+;===============TRANSPORT
+
+[simpletrans]
+type=transport
+protocol=udp
+bind=0.0.0.0
+local_net=192.168.4.0/24
+local_net=192.168.5.0/24
+local_net=192.168.6.0/24
+local_net=192.168.9.0/24
+external_media_address=2.59.235.35
+external_signaling_address=2.59.235.35
+
+
+;===============ENDPOINT TEMPLATES
+
+[endpoint-basic](!)
+type=endpoint
+context=from-internal
+message_context=send-text
+disallow=all
+;allow=g722
+allow=ulaw
+;allow=alaw
+;allow=opus
+;allow=all
+
+[auth-userpass](!)
+type=auth
+auth_type=userpass
+
+[aor-single-reg](!)
+type=aor
+max_contacts=1
+
+;===============ATA site A port 1
+
+[1001](endpoint-basic)
+auth=auth1001
+aors=1001
+acl=internal
+
+[auth1001](auth-userpass)
+password=REDACTED
+username=1001
+
+[1001](aor-single-reg)
+
+;===============ATA site A port 2
+
+[1002](endpoint-basic)
+auth=auth1002
+aors=1002
+acl=internal
+
+[auth1002](auth-userpass)
+password=REDACTED
+username=1002
+
+[1002](aor-single-reg)
+
+;===============ATA site B port 1
+
+[1011](endpoint-basic)
+auth=auth1011
+aors=1011
+acl=internal
+
+[auth1011](auth-userpass)
+password=REDACTED
+username=1011
+
+[1011](aor-single-reg)
+
+;===============ATA site B port 2
+
+[1012](endpoint-basic)
+auth=auth1012
+aors=1012
+acl=internal
+
+[auth1012](auth-userpass)
+password=REDACTED
+username=1012
+
+[1012](aor-single-reg)
+
+
+;===============soft phone 1
+
+[1021](endpoint-basic)
+auth=auth1021
+aors=1021
+acl=internal
+
+[auth1021](auth-userpass)
+password=REDACTED
+username=1021
+
+[1021](aor-single-reg)
+
+;===============soft phone 2
+
+[1022](endpoint-basic)
+auth=auth1022
+aors=1022
+acl=internal
+
+[auth1022](auth-userpass)
+password=REDACTED
+username=1022
+
+[1022](aor-single-reg)
+
+
+;===============remote phone 1
+
+[1031](endpoint-basic)
+auth=auth1031
+aors=1031
+rtp_symmetric=yes
+force_rport=yes
+direct_media=no
+rewrite_contact=yes
+acl=internal ; these are currently disabled for external registration so they arent really remote
+
+[auth1031](auth-userpass)
+password=REDACTED
+username=1031
+
+[1031](aor-single-reg)
+
+;===============remote phone 2
+
+[1032](endpoint-basic)
+auth=auth1032
+aors=1032
+rtp_symmetric=yes
+force_rport=yes
+direct_media=no
+rewrite_contact=yes
+acl=internal
+
+[auth1032](auth-userpass)
+password=REDACTED
+username=1032
+
+[1032](aor-single-reg)
+
+;===============Cisco 3911 SIP phone 1
+
+[1051](endpoint-basic)
+auth=auth1051
+aors=1051
+rtp_symmetric=yes
+force_rport=yes
+direct_media=no
+rewrite_contact=yes
+;acl=internal
+
+[auth1051](auth-userpass)
+password=REDACTED
+username=1051
+
+[1051](aor-single-reg)
+
+
+
+
+; ============= anonymous inbound calls
+[anonymous]
+type=endpoint
+context=from-anonymous
+;disallow=all
+;allow=speex,g726,g722,ilbc,gsm,alaw,ulaw
+allow=all
+acl=external
+
+
+
+;=============== SDF TRUNK
+
+[sdf]
+type=registration
+outbound_auth=sdf
+server_uri=sip:sip.sdf.org
+client_uri=sip:2063@sip.sdf.org
+retry_interval=120
+forbidden_retry_interval=180
+auth_rejection_permanent=no
+max_retries=3
+contact_user=2063
+expiration=1200
+
+[sdf]
+type=auth
+auth_type=userpass
+username=2063
+password=REDACTED
+
+[sdf]
+type=aor
+contact=sip:sip.sdf.org:5060
+qualify_frequency=58
+
+[sdf]
+type=endpoint
+context=from-sdf
+disallow=all
+allow=ulaw
+allow=g729
+allow=alaw
+allow=opus
+;allow=all
+outbound_auth=sdf
+aors=sdf
+;rtp_symmetric=yes
+;force_rport=yes
+direct_media=no
+;rewrite_contact=yes
+acl=sdf
+
+[sdf]
+type=identify
+endpoint=sdf
+match=sip.sdf.org
+
+;=============== AVANZA PSTN TRUNK
+
+[pstn]
+type=registration
+outbound_auth=pstn
+server_uri=sip:sip.avanzafibra.net
+client_uri=sip:mta69987-1@sip.avanzafibra.net
+retry_interval=120
+forbidden_retry_interval=180
+auth_rejection_permanent=no
+max_retries=3
+contact_user=868925554
+expiration=1200
+
+[pstn]
+type=auth
+auth_type=userpass
+username=mta69987-1
+password=REDACTED
+
+[pstn]
+type=aor
+contact=sip:sip.avanzafibra.net:5060
+qualify_frequency=58
+
+[pstn]
+type=endpoint
+transport=simpletrans
+context=from-pstn
+disallow=all
+allow=ulaw
+allow=g729
+allow=alaw
+allow=opus
+;allow=all
+outbound_auth=pstn
+aors=pstn
+;rtp_symmetric=yes
+;force_rport=yes
+direct_media=no
+;rewrite_contact=yes
+acl=pstn
+
+[pstn]
+type=identify
+endpoint=pstn
+match=sip.avanzafibra.net
+
+
+
+
diff --git a/asterisk-conf/rtp.conf b/asterisk-conf/rtp.conf
new file mode 100644
index 0000000..19f3486
--- /dev/null
+++ b/asterisk-conf/rtp.conf
@@ -0,0 +1,162 @@
+;
+; RTP Configuration
+;
+[general]
+;
+; RTP start and RTP end configure start and end addresses
+;
+; Defaults are rtpstart=5000 and rtpend=31000
+;
+rtpstart=10000
+rtpend=10100
+;
+; Whether to enable or disable UDP checksums on RTP traffic
+;
+;rtpchecksums=no
+;
+; The amount of time a DTMF digit with no 'end' marker should be
+; allowed to continue (in 'samples', 1/8000 of a second)
+;
+;dtmftimeout=3000
+; rtcpinterval = 5000 	; Milliseconds between rtcp reports
+			;(min 500, max 60000, default 5000)
+;
+; Enable strict RTP protection.  This will drop RTP packets that do not come
+; from the recognized source of the RTP stream.  Strict RTP qualifies RTP
+; packet stream sources before accepting them upon initial connection and
+; when the connection is renegotiated (e.g., transfers and direct media).
+; Initial connection and renegotiation starts a learning mode to qualify
+; stream source addresses.  Once Asterisk has recognized a stream it will
+; allow other streams to qualify and replace the current stream for 5
+; seconds after starting learning mode.  Once learning mode completes the
+; current stream is locked in and cannot change until the next
+; renegotiation.
+; Valid options are "no" to disable strictrtp, "yes" to enable strictrtp,
+; and "seqno", which does the same thing as strictrtp=yes, but only checks
+; to make sure the sequence number is correct rather than checking the time
+; interval as well.
+; This option is enabled by default.
+; strictrtp=yes
+;
+; Number of packets containing consecutive sequence values needed
+; to change the RTP source socket address. This option only comes
+; into play while using strictrtp=yes. Consider changing this value
+; if rtp packets are dropped from one or both ends after a call is
+; connected. This option is set to 4 by default.
+; probation=8
+;
+; Enable sRTP replay protection. Buggy SIP user agents (UAs) reset the
+; sequence number (RTP-SEQ) on a re-INVITE, for example, with Session Timers
+; or on Call Hold/Resume, but keep the synchronization source (RTP-SSRC). If
+; the new RTP-SEQ is higher than the previous one, the call continues if the
+; roll-over counter (sRTP-ROC) is zero (the call lasted less than 22 minutes).
+; In all other cases, the call faces one-way audio or even no audio at all.
+; "replay check failed (index too old)" gets printed continuously. This is a
+; software bug. You have to report this to the creator of that UA. Until it is
+; fixed, you could disable sRTP replay protection (see RFC 3711 section 3.3.2).
+; This option is enabled by default.
+; srtpreplayprotection=yes
+;
+; Whether to enable or disable ICE support. This option is enabled by default.
+; icesupport=false
+;
+; Hostname or address for the STUN server used when determining the external
+; IP address and port an RTP session can be reached at. The port number is
+; optional. If omitted the default value of 3478 will be used. This option is
+; disabled by default. Name resolution will occur at load time, and if DNS is
+; used, name resolution will occur repeatedly after the TTL expires.
+;
+; e.g. stundaddr=mystun.server.com:3478
+;
+; stunaddr=
+;
+; Some multihomed servers have IP interfaces that cannot reach the STUN
+; server specified by stunaddr.  Blacklist those interface subnets from
+; trying to send a STUN packet to find the external IP address.
+; Attempting to send the STUN packet needlessly delays processing incoming
+; and outgoing SIP INVITEs because we will wait for a response that can
+; never come until we give up on the response.
+; * Multiple subnets may be listed.
+; * Blacklisting applies to IPv4 only.  STUN isn't needed for IPv6.
+; * Blacklisting applies when binding RTP to specific IP addresses and not
+; the wildcard 0.0.0.0 address.  e.g., A PJSIP endpoint binding RTP to a
+; specific address using the bind_rtp_to_media_address and media_address
+; options.  Or the PJSIP endpoint specifies an explicit transport that binds
+; to a specific IP address.  Blacklisting is done via ACL infrastructure
+; so it's possible to whitelist as well.
+;
+; stun_acl = named_acl
+; stun_deny = 0.0.0.0/0
+; stun_permit = 1.2.3.4/32
+;
+; For historic reasons stun_blacklist is an alias for stun_deny.
+;
+; Whether to report the PJSIP version in a SOFTWARE attribute for all
+; outgoing STUN packets. This option is enabled by default.
+;
+; stun_software_attribute=yes
+;
+; Hostname or address for the TURN server to be used as a relay. The port
+; number is optional. If omitted the default value of 3478 will be used.
+; This option is disabled by default.
+;
+; e.g. turnaddr=myturn.server.com:34780
+;
+; turnaddr=
+;
+; Username used to authenticate with TURN relay server.
+; turnusername=
+;
+; Password used to authenticate with TURN relay server.
+; turnpassword=
+;
+; An ACL can be used to determine which discovered addresses to include for
+; ICE, srflx and relay discovery.  This is useful to optimize the ICE process
+; where a system has multiple host address ranges and/or physical interfaces
+; and certain of them are not expected to be used for RTP. For example, VPNs
+; and local interconnections may not be suitable or necessary for ICE. Multiple
+; subnets may be listed. If left unconfigured, all discovered host addresses
+; are used.
+;
+; ice_acl = named_acl
+; ice_deny = 0.0.0.0/0
+; ice_permit = 1.2.3.4/32
+;
+; For historic reasons ice_blacklist is an alias for ice_deny.
+;
+; The MTU to use for DTLS packet fragmentation. This option is set to 1200
+; by default. The minimum MTU is 256.
+; dtls_mtu = 1200
+;
+[ice_host_candidates]
+;
+; When Asterisk is behind a static one-to-one NAT and ICE is in use, ICE will
+; expose the server's internal IP address as one of the host candidates.
+; Although using STUN (see the 'stunaddr' configuration option) will provide a
+; publicly accessible IP, the internal IP will still be sent to the remote
+; peer. To help hide the topology of your internal network, you can override
+; the host candidates that Asterisk will send to the remote peer.
+;
+; IMPORTANT: Only use this functionality when your Asterisk server is behind a
+; one-to-one NAT and you know what you're doing. If you do define anything
+; here, you almost certainly will NOT want to specify 'stunaddr' or 'turnaddr'
+; above.
+;
+; The format for these overrides is:
+;
+;    <local address> => <advertised address>,[include_local_address]
+;
+; The following will replace 192.168.1.10 with 1.2.3.4 during ICE
+; negotiation:
+;
+;192.168.1.10 => 1.2.3.4
+;
+; The following will include BOTH 192.168.1.10 and 1.2.3.4 during ICE
+; negotiation instead of replacing 192.168.1.10.  This can make it easier
+; to serve both local and remote clients.
+;
+;192.168.1.10 => 1.2.3.4,include_local_address
+;
+; You can define an override for more than 1 interface if you have a multihomed
+; server. Any local interface that is not matched will be passed through
+; unaltered. Both IPv4 and IPv6 addresses are supported.
diff --git a/asterisk-conf/sccp.conf b/asterisk-conf/sccp.conf
new file mode 100644
index 0000000..4c94ef0
--- /dev/null
+++ b/asterisk-conf/sccp.conf
@@ -0,0 +1,182 @@
+;=========================================================================================
+;
+; general definitions
+;
+;=========================================================================================
+[general]
+debug = core, config, action, socket, device, line, channel
+servername = Asterisk
+keepalive = 60
+context = default
+dateformat = D.M.Y
+bindaddr = 0.0.0.0
+port = 2000
+disallow=all
+allow=alaw
+allow=ulaw
+allow=g729
+firstdigittimeout = 16
+digittimeout = 8
+autoanswer_ring_time = 1
+autoanswer_tone = Zip
+remotehangup_tone = Zip
+transfer=on
+transfer_tone = 0
+transfer_on_hangup = off
+dnd_tone = 0x0
+callwaiting_tone = Call Waiting Tone
+musicclass=default
+language=en
+deny=0.0.0.0/0.0.0.0
+permit=internal						; 'internal' is automatically converted to these private cidr address:
+							; 127.0.0.0/255.0.0.0, 10.0.0.0/255.0.0.0,	
+							; 172.0.0.0/255.224.0.0, 192.168.0.0/255.255.0.0
+;permit=192.168.1.0/255.255.255.0
+localnet = internal					; (MULTI-ENTRY) All RFC 1918 addresses are local networks, example '192.168.1.0/255.255.255.0'
+;externip = 77.44.22.33					; External IP Address of the firewall, required in case the PBX is running on a seperate host behind it. IP Address that we're going to notify in RTP media stream as the pbx source address.
+dndFeature = on
+sccp_tos = 0x68
+sccp_cos = 4
+audio_tos = 0xB8
+audio_cos = 6
+video_tos = 0x88
+video_cos = 5
+echocancel = on
+silencesuppression = off
+private = on
+callanswerorder=oldestfirst
+pickup_modeanswer = on
+hotline_enabled=yes					;can devices without configuration register
+hotline_context=default					; context for hotline
+hotline_extension=111					; extension will be dialed on offHook
+
+;=========================================================================================
+;
+; actual definitions
+;
+;=========================================================================================
+[SEP001BD4C70670]
+description = Cisco estudio
+addon = 7941
+devicetype = 7941
+park = off
+button = speeddial, IVR, 2000, 1101@hints  	; Add SpeedDial to Helpdesk
+button = line, 1101, default				; Assign Line 1101 to Device and use this as default line
+button = empty						; Assign an Empty Button
+button = line, 1102					; Assign Line 1102 to Device
+button = speeddial, ATA A p1, 1001, 1001@hints	; Add SpeedDial to Phone Number Two Line 1 (button labels allow special characters like 'Ã©')
+cfwdall = off
+type = device
+keepalive = 60
+;tzoffset = +2
+transfer = on
+park = on
+cfwdall = off
+cfwdbusy = off
+cfwdnoanswer = off
+deny=0.0.0.0/0.0.0.0
+permit=192.168.0.0/255.255.0.0
+dndFeature = on
+dnd = off
+directrtp=off
+earlyrtp = progress
+private = on
+mwilamp = on
+mwioncall = off
+setvar=testvar=value
+cfwdall = on
+
+[1101]
+id = 1101
+type = line
+;pin = 1234
+label = Cisco estudio
+description = 1101 estudio
+mailbox = 1101
+cid_name = 1101
+cid_num = 1101
+accountcode=1101
+;callgroup=1,3-4
+;pickupgroup=1,3-5
+directed_pickup = on
+directed_pickup_context = ""
+pickup_modeanswer = on
+;amaflags = 
+context = from-internal
+incominglimit = 2
+transfer = on
+vmnum = 2003
+meetme = on
+meetmeopts = qxd
+meetmenum = 700
+trnsfvm = 1000
+secondary_dialtone_digits = 9
+secondary_dialtone_tone = Outside Dial Tone
+musicclass=default
+language=en
+echocancel = on
+silencesuppression = off
+setvar=testvar2=my value
+dnd = reject
+parkinglot = myparkspace
+
+[1102]
+id = 1102
+type = line
+;pin = 1102
+label = Cisco estudio 2
+description = 1102 estudio
+mailbox = 1102
+cid_name = 1102
+cid_num = 1102
+accountcode=1102
+;callgroup=1,4-9
+;pickupgroup=1,3-9
+directed_pickup = on
+directed_pickup_context = "another"
+pickup_modeanswer = on
+echocancel = off
+context = from-internal
+incominglimit = 2
+transfer = on
+vmnum = 600
+trnsfvm = 1000
+secondary_dialtone_digits = 9
+secondary_dialtone_tone = Outside Dial Tone
+musicclass=default
+language=en
+echocancel = on
+silencesuppression = off
+silencesuppression = on
+dnd = silent
+
+
+
+
+
+;-----------------------------------------------------------------------------------
+;create a user defined softkeyset
+; - if not softkey is not specified in device the chan-sccp-b default mapping will be used
+; - multiple softkeysets are possible
+;
+;valid softkeys:
+;redial, newcall, cfwdall, cfwdbusy, cfwdnoanswer, pickup, gpickup, conflist, dnd, hold, endcall, park, select
+;idivert, resume, newcall, transfer, dirtrfr, answer, transvm, private, meetme, barge, cbarge, conf, back, join, empty, dial, vidmode
+
+; using this softkey set on the 69xx phones solves transfer issues using the
+; hard key on the phone.
+[69xxsoftkeys]
+type=softkeyset
+onhook=redial,newcall,cfwdall,dnd,pickup,gpickup,private
+offhook=redial,endcall,cfwdall,dnd,pickup,gpickup,private
+offhookfeat=redial,endcall
+digitsfoll=back,endcall,dial
+ringout=empty,endcall,transfer
+connected=hold,endcall,transfer,idivert,conf,conflist,park,monitor,vidmode
+conntrans=empty,endcall,transfer,monitor
+connconf=hold,endcall,conf,conflist,join,vidmode,monitor
+onhold=resume,newcall,endcall,idivert
+ringin=answer,idivert
+onhint=barge,newcall,redial,pickup,gpickup
+onstealable=redial,newcall,cfwdall,pickup,gpickup,dnd,intrcpt
+holdconf=resume,newcall,endcall,join
diff --git a/asterisk-conf/voicemail.conf b/asterisk-conf/voicemail.conf
new file mode 100644
index 0000000..77b2202
--- /dev/null
+++ b/asterisk-conf/voicemail.conf
@@ -0,0 +1,52 @@
+[general]
+format=wav49|gsm|wav
+serveremail=asterisk@arf20.com
+attach=yes
+;maxmsg=100
+;maxsecs=180
+;minsecs=3
+;maxgreet=60
+skipms=3000
+maxsilence=10
+silencethreshold=128
+maxlogins=3
+;moveheard=yes
+;forward_urgent_auto=no
+;userscontext=default
+aliasescontext=aliases
+charset=UTF-8
+pbxskip=no
+fromstring=ARFNET PBX
+usedirectory=yes
+;odbcstorage=asterisk
+;odbctable=voicemessages
+emaildateformat=%A, %d %B %Y at %H:%M:%S
+pagerdateformat=%A, %B %d, %Y at %r
+;mailcmd=/usr/sbin/sendmail -t
+sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
+
+[zonemessages]
+eastern=America/New_York|'vm-received' Q 'digits/at' IMp
+central=America/Chicago|'vm-received' Q 'digits/at' IMp
+central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
+military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
+european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
+
+[default]
+1000 => 1000,arf20,arf20@arf20.com
+
+[aliases]
+1000 => 1000@default
+1001 => 1000@default
+1002 => 1000@default
+1011 => 1000@default
+1012 => 1000@default
+1021 => 1000@default
+1022 => 1000@default
+1031 => 1000@default
+1032 => 1000@default
+1051 => 1000@default
+1052 => 1000@default
+1101 => 1000@default
+
+
diff --git a/tftp-sepfiles/SEP001AA11B4EC1.cnf.xml b/tftp-sepfiles/SEP001AA11B4EC1.cnf.xml
new file mode 100644
index 0000000..e7dd55b
--- /dev/null
+++ b/tftp-sepfiles/SEP001AA11B4EC1.cnf.xml
@@ -0,0 +1,311 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<device  xsi:type="axl:XIPPhone" ctiid="8" uuid="{18ed15c3-5782-5fd1-0ab8-6a1ddf47179e}">
+<fullConfig>true</fullConfig>
+<portalDefaultServer>192.168.9.2</portalDefaultServer>
+<deviceProtocol>SIP</deviceProtocol>
+<sshUserId></sshUserId>
+<sshPassword></sshPassword>
+<ipAddressMode>0</ipAddressMode>
+<allowAutoConfig>true</allowAutoConfig>
+<dadEnable>true</dadEnable>
+<redirectEnable>false</redirectEnable>
+<echoMultiEnable>false</echoMultiEnable>
+<ipPreferenceModeControl>0</ipPreferenceModeControl>
+<ipMediaAddressFamilyPreference>0</ipMediaAddressFamilyPreference>
+<tzdata>
+<tzolsonversion>2020a-2025</tzolsonversion>
+<tzupdater>tzdatacsv.csv</tzupdater>
+</tzdata>
+<mlppDomainId>000000</mlppDomainId>
+<mlppIndicationStatus>Off</mlppIndicationStatus>
+<preemption>Disabled</preemption>
+<executiveOverridePreemptable>false</executiveOverridePreemptable>
+<devicePool  uuid="{1b1b9eb6-7803-11d3-bdf0-00108302ead1}">
+<revertPriority>0</revertPriority>
+<name>SITEA</name>
+<dateTimeSetting  uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}">
+<name>CMLocal</name>
+<dateTemplate>D/M/Y</dateTemplate>
+<timeZone></timeZone>
+<olsonTimeZone>Europe/Madrid</olsonTimeZone>
+<ntps>
+<ntp>
+<name>192.168.9.1</name>
+<nameV6></nameV6>
+<ntpMode>Unicast</ntpMode>
+</ntp>
+</ntps>
+</dateTimeSetting>
+<callManagerGroup>
+<name>Default</name>
+<tftpDefault>true</tftpDefault>
+<members>
+<member  priority="0">
+<callManager>
+<name>192.168.9.2</name>
+<description>cucm</description>
+<ports>
+<ethernetPhonePort>2000</ethernetPhonePort>
+<sipPort>5060</sipPort>
+<securedSipPort>5061</securedSipPort>
+<sipOAuthPort>5090</sipOAuthPort>
+<sipMRAOAuthPort>5091</sipMRAOAuthPort>
+<mgcpPorts>
+<listen>2427</listen>
+<keepAlive>2428</keepAlive>
+</mgcpPorts>
+</ports>
+<processNodeName>192.168.9.2</processNodeName>
+</callManager>
+</member>
+</members>
+</callManagerGroup>
+<srstInfo  uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}">
+<name>Disable</name>
+<srstOption>Disable</srstOption>
+<userModifiable>false</userModifiable>
+<ipAddr1></ipAddr1>
+<port1>2000</port1>
+<ipAddr2></ipAddr2>
+<port2>2000</port2>
+<ipAddr3></ipAddr3>
+<port3>2000</port3>
+<sipIpAddr1></sipIpAddr1>
+<sipPort1>5060</sipPort1>
+<sipIpV6Addr1></sipIpV6Addr1>
+<sipIpAddr2></sipIpAddr2>
+<sipPort2>5060</sipPort2>
+<sipIpAddr3></sipIpAddr3>
+<sipPort3>5060</sipPort3>
+<isSecure>false</isSecure>
+</srstInfo>
+<connectionMonitorDuration>120</connectionMonitorDuration>
+</devicePool>
+<sipProfile>
+<sipProxies>
+<backupProxy>USECALLMANAGER</backupProxy>
+<backupProxyPort>5060</backupProxyPort>
+<emergencyProxy>USECALLMANAGER</emergencyProxy>
+<emergencyProxyPort>5060</emergencyProxyPort>
+<outboundProxy>USECALLMANAGER</outboundProxy>
+<outboundProxyPort>5060</outboundProxyPort>
+<registerWithProxy>true</registerWithProxy>
+</sipProxies>
+<sipCallFeatures>
+<cnfJoinEnabled>true</cnfJoinEnabled>
+<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
+<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
+<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
+<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
+<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
+<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
+<rfc2543Hold>false</rfc2543Hold>
+<callHoldRingback>2</callHoldRingback>
+<URIDialingDisplayPreference>1</URIDialingDisplayPreference>
+<localCfwdEnable>true</localCfwdEnable>
+<semiAttendedTransfer>true</semiAttendedTransfer>
+<anonymousCallBlock>2</anonymousCallBlock>
+<callerIdBlocking>2</callerIdBlocking>
+<dndControl>0</dndControl>
+<remoteCcEnable>true</remoteCcEnable>
+<retainForwardInformation>false</retainForwardInformation>
+</sipCallFeatures>
+<sipStack>
+<sipInviteRetx>6</sipInviteRetx>
+<sipRetx>10</sipRetx>
+<timerInviteExpires>180</timerInviteExpires>
+<timerRegisterExpires>3600</timerRegisterExpires>
+<timerRegisterDelta>5</timerRegisterDelta>
+<timerKeepAliveExpires>120</timerKeepAliveExpires>
+<timerSubscribeExpires>120</timerSubscribeExpires>
+<timerSubscribeDelta>5</timerSubscribeDelta>
+<timerT1>500</timerT1>
+<timerT2>4000</timerT2>
+<maxRedirects>70</maxRedirects>
+<remotePartyID>true</remotePartyID>
+<userInfo>None</userInfo>
+</sipStack>
+<autoAnswerTimer>1</autoAnswerTimer>
+<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
+<autoAnswerOverride>true</autoAnswerOverride>
+<transferOnhookEnabled>false</transferOnhookEnabled>
+<enableVad>false</enableVad>
+<preferredCodec>none</preferredCodec>
+<dtmfAvtPayload>101</dtmfAvtPayload>
+<dtmfDbLevel>3</dtmfDbLevel>
+<dtmfOutofBand>avt</dtmfOutofBand>
+<kpml>3</kpml>
+<phoneLabel></phoneLabel>
+<stutterMsgWaiting>2</stutterMsgWaiting>
+<callStats>true</callStats>
+<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
+<T302Timer>3000</T302Timer>
+<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
+<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
+<poundEndOfDial>false</poundEndOfDial>
+<startMediaPort>16384</startMediaPort>
+<stopMediaPort>32766</stopMediaPort>
+<startVideoPort>0</startVideoPort>
+<stopVideoPort>0</stopVideoPort>
+<organizationTopLevelDomain></organizationTopLevelDomain>
+<sipLines>
+<line  button="1" lineIndex="1">
+<featureID>9</featureID>
+<featureLabel></featureLabel>
+<proxy>USECALLMANAGER</proxy>
+<port>5060</port>
+<name>1051</name>
+<displayName>cisco 3911 estudio</displayName>
+<autoAnswer>
+<autoAnswerEnabled>2</autoAnswerEnabled>
+</autoAnswer>
+<recordingOption>disable</recordingOption>
+<callWaiting>3</callWaiting>
+<authName>1051</authName>
+<authPassword>REDACTED</authPassword>
+<sharedLine>false</sharedLine>
+<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
+<messageWaitingAMWI>0</messageWaitingAMWI>
+<messagesNumber></messagesNumber>
+<ringSettingIdle>4</ringSettingIdle>
+<ringSettingActive>5</ringSettingActive>
+<contact>43c89e6e-f728-02fe-80a0-525af6457cb4</contact>
+<forwardCallInfoDisplay>
+<callerName>false</callerName>
+<callerNumber>false</callerNumber>
+<redirectedNumber>false</redirectedNumber>
+<dialedNumber>false</dialedNumber>
+</forwardCallInfoDisplay>
+<maxNumCalls>2</maxNumCalls>
+<busyTrigger>1</busyTrigger>
+</line>
+</sipLines>
+<externalNumberMask></externalNumberMask>
+<voipControlPort>5060</voipControlPort>
+<dscpForAudio  tcl="conversational.audio.aq:admitted">184</dscpForAudio>
+<dscpVideo  tcl="conversational.video.avconf.aq:admitted">136</dscpVideo>
+<dscpAudioForVideo  tcl="conversational.audio.avconf.aq:admitted">136</dscpAudioForVideo>
+<dscpForTelepresence  tcl="conversational.video.immersive.aq:admitted">128</dscpForTelepresence>
+<dscpAudioForTelepresence  tcl="conversational.audio.immersive.aq:admitted">128</dscpAudioForTelepresence>
+<dscpForPriorityAudio>180</dscpForPriorityAudio>
+<dscpForImmediateAudio>176</dscpForImmediateAudio>
+<dscpForFlashAudio>164</dscpForFlashAudio>
+<dscpForFlashOverrideAudio>168</dscpForFlashOverrideAudio>
+<dscpForExecutiveOverrideAudio>168</dscpForExecutiveOverrideAudio>
+<dscpForPriorityVideo>156</dscpForPriorityVideo>
+<dscpForImmediateVideo>148</dscpForImmediateVideo>
+<dscpForFlashVideo>140</dscpForFlashVideo>
+<dscpForFlashOverrideVideo>132</dscpForFlashOverrideVideo>
+<dscpForExecutiveOverrideVideo>132</dscpForExecutiveOverrideVideo>
+<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
+<dialTemplate></dialTemplate>
+<softKeyFile>SKb0ec918f-b9ee-994b-57ae-345883c1fde8.xml</softKeyFile>
+<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
+<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
+</sipProfile>
+<commonProfile>
+<phonePassword></phonePassword>
+<backgroundImageAccess>true</backgroundImageAccess>
+<callLogBlfEnabled>3</callLogBlfEnabled>
+</commonProfile>
+<loadInformation>SIP3951.8-1-4a</loadInformation>
+<vendorConfig>
+<garp>0</garp><webAccess>0</webAccess><telnetAccess>1</telnetAccess></vendorConfig>
+<versionStamp>1735953486-cebe5d8a-030a-4ad4-84db-5af644f25211</versionStamp>
+<userLocale>
+<name>English_United_States</name>
+<uid>1</uid>
+<langCode>en_US</langCode>
+<version>12.0.0.0(1)</version>
+<winCharSet>iso-8859-1</winCharSet>
+</userLocale>
+<networkLocale>United_States</networkLocale>
+<networkLocaleInfo>
+<name>United_States</name>
+<uid>64</uid>
+<version>12.0.0.0(1)</version>
+</networkLocaleInfo>
+<sipOAuthMode>0</sipOAuthMode>
+<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
+<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
+<dscpForCm2Dvce>96</dscpForCm2Dvce>
+<transportLayerProtocol>2</transportLayerProtocol>
+<dialToneSetting>1</dialToneSetting>
+<dndCallAlert>5</dndCallAlert>
+<phonePersonalization>0</phonePersonalization>
+<rollover>0</rollover>
+<singleButtonBarge>0</singleButtonBarge>
+<joinAcrossLines>0</joinAcrossLines>
+<autoCallPickupEnable>false</autoCallPickupEnable>
+<blfAudibleAlertSettingOfIdleStation>0</blfAudibleAlertSettingOfIdleStation>
+<blfAudibleAlertSettingOfBusyStation>0</blfAudibleAlertSettingOfBusyStation>
+<capfAuthMode>0</capfAuthMode>
+<capfList>
+<capf>
+<phonePort>3804</phonePort>
+<processNodeName>192.168.9.2</processNodeName>
+</capf>
+</capfList>
+<certHash></certHash>
+<encrConfig>false</encrConfig>
+<advertiseG722Codec>1</advertiseG722Codec>
+<mobility>
+<handoffdn></handoffdn>
+<dtmfdn></dtmfdn>
+<ivrdn></ivrdn>
+<dtmfHoldCode>*81</dtmfHoldCode>
+<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
+<dtmfResumeCode>*83</dtmfResumeCode>
+<dtmfTxfCode>*84</dtmfTxfCode>
+<dtmfCnfCode>*85</dtmfCnfCode>
+</mobility>
+<TLSResumptionTimer>3600</TLSResumptionTimer>
+<userId  serviceProfileFile="SPDefault.cnf.xml">siteaphone1</userId>
+<ownerId  serviceProfileFile="SPDefault.cnf.xml">siteaphone1</ownerId>
+<phoneServices  useHTTPS="true">
+<provisioning>0</provisioning>
+<phoneService  type="1" category="0">
+<name>Missed Calls</name>
+<url>Application:Cisco/MissedCalls</url>
+<vendor></vendor>
+<version></version>
+</phoneService>
+<phoneService  type="2" category="0">
+<name>Voicemail</name>
+<url>Application:Cisco/Voicemail</url>
+<vendor></vendor>
+<version></version>
+</phoneService>
+<phoneService  type="1" category="0">
+<name>Received Calls</name>
+<url>Application:Cisco/ReceivedCalls</url>
+<vendor></vendor>
+<version></version>
+</phoneService>
+<phoneService  type="1" category="0">
+<name>Placed Calls</name>
+<url>Application:Cisco/PlacedCalls</url>
+<vendor></vendor>
+<version></version>
+</phoneService>
+<phoneService  type="1" category="0">
+<name>Personal Directory</name>
+<url>Application:Cisco/PersonalDirectory</url>
+<vendor></vendor>
+<version></version>
+</phoneService>
+<phoneService  type="1" category="0">
+<name>Corporate Directory</name>
+<url>http://192.168.9.2/cisco/phonebook.xml</url>
+<vendor></vendor>
+<version></version>
+</phoneService>
+</phoneServices>
+<ssospServers>
+<ssospServer  name="192.168.9.2">
+<url>https://192.168.9.2:8443/ssosp</url>
+</ssospServer>
+</ssospServers>
+<mraTrustHash></mraTrustHash>
+<allowMraMode>false</allowMraMode>
+</device>
diff --git a/tftp-sepfiles/SEP001BD4C70670.cnf.xml b/tftp-sepfiles/SEP001BD4C70670.cnf.xml
new file mode 100644
index 0000000..0999d89
--- /dev/null
+++ b/tftp-sepfiles/SEP001BD4C70670.cnf.xml
@@ -0,0 +1,105 @@
+<device>
+
+  <sshUserId>cisco</sshUserId>
+  <sshPassword>cisco</sshPassword>
+
+  <devicePool>
+                <dateTimeSetting>
+                        <name>Spain</name>
+                        <dateTemplate>D/M/Y</dateTemplate>
+                        <timeZone>W. Europe Standard/Daylight Time</timeZone>
+                        <ntps>
+                                <ntp>
+                                        <name>192.168.9.1</name>
+                                        <ntpMode>Unicast</ntpMode>
+                                </ntp>
+                        </ntps>
+                </dateTimeSetting>
+                <callManagerGroup>
+                        <members>
+                                <member priority="0">
+                                  <callManager>
+                                        <ports>
+                                          <ethernetPhonePort>2000</ethernetPhonePort>
+                                        </ports>
+                                        <processNodeName>192.168.9.2</processNodeName>
+                                  </callManager>
+                                </member>
+                        </members>
+                </callManagerGroup>
+
+   <!--  <srstInfo>
+      <name>Enable</name>
+      <srstOption>Enable</srstOption>
+      <userModifiable>true</userModifiable>
+      <ipAddr1>x.x.x.x</ipAddr1>
+      <port1>2000</port1>
+      <ipAddr2/>
+      <port2>2000</port2>
+      <ipAddr3/>
+      <port3>2000</port3>
+    </srstInfo> -->
+  </devicePool>
+
+  <vendorConfig>
+    <disableSpeaker>false</disableSpeaker>
+    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
+    <forwardingDelay>1</forwardingDelay>
+    <pcPort>0</pcPort>
+    <settingsAccess>1</settingsAccess>
+    <garp>0</garp>
+    <voiceVlanAccess>0</voiceVlanAccess>
+    <videoCapability>0</videoCapability>
+    <autoSelectLineEnable>0</autoSelectLineEnable>
+        <autoCallSelect>0</autoCallSelect>
+    <webAccess>0</webAccess>
+    <daysDisplayNotActive>1,7</daysDisplayNotActive>
+    <displayOnTime>08:30</displayOnTime>
+    <displayOnDuration>11:30</displayOnDuration>
+    <displayIdleTimeout>01:00</displayIdleTimeout>
+        <minimumRingVolume>0</minimumRingVolume>
+
+  </vendorConfig>
+
+  <userLocale>
+                <name>English</name>
+                <uid>1</uid>
+                <langCode>en</langCode>
+                <version>10.5.3.0</version>
+                <winCharSet>iso-8859-1</winCharSet>
+        </userLocale>
+        <networkLocale>Spain</networkLocale>
+        <networkLocaleInfo>
+                <name>Spain</name>
+                <uid>64</uid>
+                <version>10.5.3.0</version>
+        </networkLocaleInfo>
+
+
+
+        <loadInformation>SCCP45.9-4-2SR3-1S</loadInformation>
+        <idleTimeout>3600</idleTimeout>
+        <authenticationURL></authenticationURL>
+        <directoryURL>http://192.168.9.2/cisco/directory.xml</directoryURL>
+        <idleURL></idleURL>
+
+        <informationURL></informationURL>
+        <messagesURL></messagesURL>
+        <proxyServerURL></proxyServerURL>
+        <servicesURL>http://192.168.9.2/cisco/directory.xml</servicesURL>
+        <phonePersonalization>1</phonePersonalization>
+        <deviceSecurityMode>1</deviceSecurityMode>
+        <dscpForSCCPPhoneConfig>104</dscpForSCCPPhoneConfig>
+        <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
+        <dscpForCm2Dvce>184</dscpForCm2Dvce>
+        <transportLayerProtocol>4</transportLayerProtocol>
+        <capfAuthMode>0</capfAuthMode>
+        <capfList>
+                <capf>
+                  <phonePort>3804</phonePort>
+                </capf>
+        </capfList>
+        <certHash/>
+        <encrConfig>false</encrConfig>
+</device>
+
author	arf20 <aruizfernandez05@gmail.com>	2025-07-11 20:36:01 +0200
committer	arf20 <aruizfernandez05@gmail.com>	2025-07-11 20:36:01 +0200
commit	e661974fbe9ecf8e7eed0e52a41855fd1eaa684d (patch)
tree	2c0085f58a2feb14566df084ded539bf61aa22db
download	arfnet2-asterisk-e661974fbe9ecf8e7eed0e52a41855fd1eaa684d.tar.gz arfnet2-asterisk-e661974fbe9ecf8e7eed0e52a41855fd1eaa684d.zip