Cisco CME i dolazni pozivi

Unapred hvala momci

Evo dela konfiguracije koji se odnosi na dolazne i odlazne pozive
Odlazni pozivi funkcionisu ali dolazni ne

Ne reaguje mi na dolazni dial-peear 5000

i ako u njemu stoji
incoming called-number 381xxxxx0100


dodao sam i sadržaj toroute.tcl scrite i debug za dva dolazna poziva
ako može bez scriipte još bolje

verujem da mi nije dobar dial-peer


cenim svaku pomoć momci



!
voice translation-rule 1

rule 1 /100/ /381xxxxx0100/

rule 2 /101/ /381xxxxx0101/

rule 3 /102/ /381xxxxx0102/

rule 4 /103/ /381xxxxx0103/

rule 5 /104/ /381xxxxx0104/

rule 6 /105/ /381xxxxx0105/

rule 7 /106/ /381xxxxx0106/

rule 8 /108/ /381xxxxx0108/

rule 9 /109/ /381xxxxx0109/

rule 10 /110/ /381xxxxx0110/

rule 11 /111/ /381xxxxx0111/

rule 12 /112/ /381xxxxx0112/

rule 13 /113/ /381xxxxx0113/

rule 14 /114/ /381xxxxx0114/

rule 15 /115/ /381xxxxx0115/

rule 16 /116/ /381xxxxx0116/

rule 17 /117/ /381xxxxx0117/

rule 18 /118/ /381xxxxx0118/

rule 19 /119/ /381xxxxx0119/

rule 39 /199/ /381xxxxx0100/

rule 40 /^...$/ /381xxxxx0100/
!
voice translation-profile SIP-OUTGOING

translate calling 1

!
application
service
toroute flash:toroute.tcl

!
dial-peer voice 1004
voip
corlist outgoing POZIVmobilni

translation-profile outgoing SIP-OUTGOING

destination-pattern 06T

session protocol sipv2

session target dns:ims.telenor.rs

codec g711alaw
!
dial-peer voice 5000 voip

service toroute

session protocol sipv2

session target sip-server

incoming called-number 381xxxxx0100

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad


toroute.tcl
###################################################
proc setup { } {
# local settings
set areaCode 1
set leadNum 00
# end local settings
leg proceeding leg_incoming
set To [infotag get leg_proto_headers "To"]
set numero $To
puts "\n >>>>> TCL-SCRIPT Translate1: To = $To ; numero = $numero \n"
regexp {sip:[0-9]+([0-9]{2})@} $To w numero
set numero $areaCode$numero
if { $numero == 100 } { set numero 100 }
if { $numero == 101 } { set numero 101 }
if { $numero == 102 } { set numero 102 }
if { $numero == 103 } { set numero 103 }
if { $numero == 104 } { set numero 104 }
if { $numero == 105 } { set numero 105 }
if { $numero == 106 } { set numero 106 }
if { $numero == 107 } { set numero 107 }
if { $numero == 108 } { set numero 108 }
if { $numero == 109 } { set numero 109 }
if { $numero == 110 } { set numero 110 }
if { $numero == 111 } { set numero 111 }
if { $numero == 112 } { set numero 112 }
if { $numero == 113 } { set numero 113 }
if { $numero == 114 } { set numero 114 }
if { $numero == 115 } { set numero 115 }
if { $numero == 116 } { set numero 116 }
if { $numero == 117 } { set numero 117 }
if { $numero == 118 } { set numero 118 }
if { $numero == 119 } { set numero 199 }


puts "\n >>>>> TCL-SCRIPT Translate2: To = $To ; numero = $numero \n"
if { [regexp {tel:\+[0-9]+} $To telBroj] } { set numero $areaCode$leadNum }
leg setup $numero callInfo leg_incoming
puts "\n >>>>> TCL-SCRIPT Translate3: To = $To ; numero = $numero \n"
}

proc setup_done { } {
# Handle SETUP DONE.
}


proc cleanup { } {
call close
}


requiredversion 2.0


#----------------------------------
# State Machine
#----------------------------------

set fsm(any_state,ev_disconnected) "cleanup same_state"
set fsm(CALL_INIT,ev_setup_indication) "setup GETDEST"
set fsm(GETDEST,ev_setup_done) "setup_done CALLACTIVE"
set fsm(CALLACTIVE,ev_disconnected) "cleanup CALLDISCONNECT"
set fsm(CALLDISCONNECT,ev_disconnected) "cleanup same_state"
set fsm(CALLDISCONNECT,ev_disconnect_done) "cleanup same_state"

fsm define fsm CALL_INIT
######################################################



debug ccsip messages

*Dec 30 21:39:07.772: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:



*Dec 30 21:39:14.144: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Max-Forwards: 64
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKgbi7r8q75r6gf2mm7vq4pm1cd
To: <sip:[email protected];user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a0185709
Call-ID: [email protected]
CSeq: 14849 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:109.245.15.185;transport=udp;lr>
Min-Se: 900
P-Asserted-Identity: <sip:[email protected];user=phone>
Session-Expires: 3600
Supported: histinfo
Content-Type: application/sdp
Content-Length: 345
Allow: UPDATE, BYE, REFER, ACK, INVITE, REGISTER, PRACK, NOTIFY, OPTIONS, CANCEL
Accept: application/sdp

v=0
o=VpnImsGw 576742 576742 IN IP4 109.245.15.185
s=VpnImsGw_Session
c=IN IP4 109.245.15.201
t=0 0
a=sendrecv
m=audio 33312 RTP/AVP 8 0 18 96
c=IN IP4 109.245.15.201
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:40

*Dec 30 21:39:14.160: //14253/9AD048C3991D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKgbi7r8q75r6gf2mm7vq4pm1cd
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a0185709
To: <sip:[email protected];user=phone>
Date: Wed, 30 Dec 2015 21:39:14 GMT
Call-ID: [email protected]
CSeq: 14849 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0


*Dec 30 21:39:14.160: //14253/9AD048C3991D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKgbi7r8q75r6gf2mm7vq4pm1cd
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a0185709
To: <sip:[email protected];user=phone>;tag=176F8254-18D2
Date: Wed, 30 Dec 2015 21:39:14 GMT
Call-ID: [email protected]
CSeq: 14849 INVITE
Allow-Events: telephone-event
Warning: 399 10.135.70.209 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=1
Content-Length: 0


*Dec 30 21:39:14.168: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKgbi7r8q75r6gf2mm7vq4pm1cd
To: <sip:[email protected];user=phone>;tag=176F8254-18D2
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a0185709
Call-ID: [email protected]
CSeq: 14849 ACK
Content-Length: 0





ili



*Dec 30 21:41:27.348: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Max-Forwards: 64
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKlfn5qr4d6txn16olf13ey81xf
To: <sip:[email protected];user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a10ce039
Call-ID: [email protected]
CSeq: 15009 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:109.245.15.185;transport=udp;lr>
Min-Se: 900
P-Asserted-Identity: <sip:[email protected];user=phone>
Session-Expires: 3600
Supported: histinfo
Content-Type: application/sdp
Content-Length: 345
Allow: UPDATE, BYE, REFER, ACK, INVITE, REGISTER, PRACK, NOTIFY, OPTIONS, CANCEL
Accept: application/sdp

v=0
o=VpnImsGw 776279 776279 IN IP4 109.245.15.185
s=VpnImsGw_Session
c=IN IP4 109.245.15.201
t=0 0
a=sendrecv
m=audio 32616 RTP/AVP 8 0 18 96
c=IN IP4 109.245.15.201
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:40

*Dec 30 21:41:27.360: //14256/EA359965992C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKlfn5qr4d6txn16olf13ey81xf
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a10ce039
To: <sip:[email protected];user=phone>
Date: Wed, 30 Dec 2015 21:41:27 GMT
Call-ID: [email protected]
CSeq: 15009 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0


*Dec 30 21:41:27.360: //14256/EA359965992C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKlfn5qr4d6txn16olf13ey81xf
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a10ce039
To: <sip:[email protected];user=phone>;tag=17718AA8-1302
Date: Wed, 30 Dec 2015 21:41:27 GMT
Call-ID: [email protected]
CSeq: 15009 INVITE
Allow-Events: telephone-event
Warning: 399 10.135.70.209 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=1
Content-Length: 0


*Dec 30 21:41:27.368: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 109.245.15.185:5060;branch=z9hG4bKlfn5qr4d6txn16olf13ey81xf
To: <sip:[email protected];user=phone>;tag=17718AA8-1302
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_a10ce039
Call-ID: [email protected]
CSeq: 15009 ACK
Content-Length: 0


*Dec 30 21:41:37.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

Bez skripte - potrebno je da napravis translaciju za dolazne pozive slicno kao sto si pravio za odlazne.
Znaci tipa:

voice translation-rule 10
rule 1 /\+381XXXXX0\(101\)/ /101/

i tako za sve unutrasnje lokale.

voice translation-profile TELENOR-IN
translate called 10

I povezes u dial-peer:

dial-peer voice 200 voip
description === OD TELENORA ===
translation-profile incoming TELENOR-IN
session protocol sipv2
incoming called-number +381xxxxx01XX - mislim da ce ti telenor slati poziv po lokalu!!!
dtmf-relay rtp-nte sip-notify
codec g711alaw
fax rate 14400
no vad

Normalno setuj da ti SIP poruke budu ka Telenoru sa source interfejsom kojim si povezan na njihov SIP...

I to bi trebalo da radi sa mozda jos malo nekog sitnog setovanja koje ces videti na osnovu novog debug-a koji dobijes...

Pozdrav,
Dragan

Potpis

Dragan

Probam odmah,


Samo jedna mala sitnice

kad si napisao
Normalno setuj da ti SIP poruke budu ka Telenoru sa source interfejsom kojim si povezan na njihov SIP...

nisam siguran da ovo znam može mala pomoć


hvala unapred

Ono pod "voice service voip" pa "sip" pa "bind...." - tu stavi intf koji imas ka Telenoru za SIP.

Mislim da ce da prihvate pozive samo sa te adrese ali posto ti odlazni pozivi rade mislim da ti ovo cak i ne treba...opet videcemo.
Ajde pusti ovo u test pa da vidimo sta se desava...

Pozdrav,
Dragan

Potpis

Dragan

Da malo ozivim temu sa aktuelnim desavanjima...

Cisco CME, 12.4T... dolazni pozivi ne funkcionisu... pretpostavka je da je problem u znaku ";" u To polju...

To: <sip:[email protected];user=phone>


Kada poziv stize bez toga, prolazi...

To: <sip:[email protected]>

Potpis

YU2MP

Takodje, ni odlazni pozivi nisu bez problema...

... poziv prodje, dobijam ring i cim se javim sa druge strane, veza se prekida:

"CSeq: 102 BYE
Reason: Q.850;cause=96"

Osnovna podesavanja dial-peer-a, jedna translacija (interni lokali na javni broj).


Da napomenem da je isti hardver i isti IOS pre 2 godine sa Telenorom radio bez greske, bez ikakvih intervencija.

Potpis

YU2MP