od kilku dni usiłuję skonfigurować SIP Trunk z routera Cisco 891 z CME do operatora Voip: supervoip.pl. Jako jedyni, których znalazłem oferują darmowego SIP Trunka.
Mam jeden telefon Cisco 7960 i chciałem móc docelowo dzwonić z niego z 3 numerów zewnętrznych i na tyle samo nr. odbierać połączenia.
W supervoip.pl mam wykupione 3 nr:
04220XXX91
03234XXX13
04220XXX59
Oraz usługę SIP Trunk z parametrami:
Trunk: SIP
Numer będzie pobierany z nagłówka SIP: From
Mój config z routera:
Kod: Zaznacz cały
Router_CME#sh run
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router_CME
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
!
crypto pki token default removal timeout 0
!
!
ip source-route
!
!
ip dhcp excluded-address 10.1.50.1 10.1.50.10
ip dhcp excluded-address 10.1.40.1 10.1.40.10
!
ip dhcp pool Voice
!
ip dhcp pool DATA
network 10.1.50.0 255.255.255.0
default-router 10.1.50.1
option 150 ip 10.1.1.1
!
ip dhcp pool VOICE
network 10.1.40.0 255.255.255.0
default-router 10.1.40.1
option 150 ip 10.1.1.1
!
!
ip cef
ip domain name CrissNET
ip name-server 192.168.1.1
no ipv6 cef
!
!
multilink bundle-name authenticated
!
!
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 3600 min 3600
!
voice class codec 1
codec preference 1 g711ulaw
!
!
voice translation-rule 1
rule 1 /04220XXX91/ /101/
!
voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/
rule 15 /^...$/ /04220XXX91/
!
voice translation-rule 3
rule 1 /101/ /04220XXX91/
rule 15 /^...$/ /04220XXX91/
!
voice translation-rule 4
rule 1 /^9/ //
!
voice translation-rule 70
!
voice translation-rule 411
rule 1 /^9\(.*\)/ /ABCD9\1/
!
voice translation-rule 412
rule 1 /^ABCD\(.*\)/ /\1/
!
voice translation-rule 422
rule 15 /^ABCD\(.*\)/ /\1/
!
!
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 3
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 4
!
voice translation-profile PSTN_CallForwarding
translate redirect-target 2
translate redirect-called 2
!
voice translation-profile PSTN_Outgoing
translate calling 3
translate called 4
translate redirect-target 2
translate redirect-called 2
!
voice translation-profile SIP_Incoming
translate called 411
!
voice translation-profile SIP_Passthrough
translate called 412
!
voice translation-profile SIP_Passthrough_CallBlocking
translate called 422
!
voice translation-profile line202_Called_101
translate called 1
!
!
interface Loopback1
ip address 10.1.1.1 255.255.255.0
!
interface FastEthernet0
no ip address
!
interface FastEthernet1
no ip address
!
interface FastEthernet2
no ip address
!
interface FastEthernet3
no ip address
!
interface FastEthernet4
no ip address
!
interface FastEthernet5
no ip address
!
interface FastEthernet6
no ip address
!
interface FastEthernet7
no ip address
!
interface FastEthernet8
ip address dhcp // Adres po DHCP 192.168.1.85
duplex auto
speed auto
!
interface GigabitEthernet0
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0.40
encapsulation dot1Q 40
ip address 10.1.40.1 255.255.255.0
!
interface GigabitEthernet0.50
encapsulation dot1Q 50
ip address 10.1.50.1 255.255.255.0
!
interface Vlan1
no ip address
!
interface Async1
no ip address
encapsulation slip
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
control-plane
!
!
mgcp profile default
!
!
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming line202_Called_101
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9.........
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
!
sip-ua
credentials username 12XXX01 password 7 0XXXXXXXXXXXXXXXXX5 realm sip.supervoip.pl
keepalive target dns:sip.supervoip.pl
authentication username 12XXX01 password 7 0XXXXXXXXXXXXXXXXX5 realm sip.supervoip.pl
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar dns:sip.supervoip.pl expires 3600
sip-server dns:sip.supervoip.pl
host-registrar
!
!
telephony-service
no auto-reg-ephone
max-ephones 5
max-dn 50
ip source-address 10.1.1.1 port 2000
calling-number initiator
max-conferences 0 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Oct 08 2013 22:28:55
!
!
ephone-dn 1 dual-line
number 101 secondary 04220XXX91 no-reg primary
!
!
ephone 1
device-security-mode none
mac-address 000A.XXXX.XXXX
type 7960
button 1:1
!
!
!
line con 0
logging synchronous
line 1
modem InOut
stopbits 1
speed 115200
flowcontrol hardware
line aux 0
line vty 0 4
login
transport input all
!
ntp master
end
Kod: Zaznacz cały
Oct 8 23:39:02.375: //120/000000000000/SIP/Msg/ccsipDisplayMsg:
Sent:
OPTIONS sip:sip.supervoip.pl:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK92F3
From: <sip:192.168.1.85>;tag=A22504-D90
To: <sip:sip.supervoip.pl>
Date: Tue, 08 Oct 2013 23:39:02 GMT
Call-ID: A57E8218-2FA911E3-8095B401-2023014E@192.168.1.85
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: <sip:192.168.1.85:5060>
Content-Length: 0
Oct 8 23:39:02.399: //120/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
Router_CME#SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK92F3;rport=62320;received=195.88.29.101
From: <sip:192.168.1.85>;tag=A22504-D90
To: <sip:sip.supervoip.pl>;tag=c80d6311acdbb780cfb4ea1c37dbab4a.4259
Call-ID: A57E8218-2FA911E3-8095B401-2023014E@192.168.1.85
CSeq: 101 OPTIONS
Proxy-Authenticate: Digest realm="192.168.1.85", nonce="UlR88VJUe8XHKj1NPgKuZZTr/8pwZv0g"
Server: Supervoip
Content-Length: 0
Kod: Zaznacz cały
Line peer expires(sec) registered P-Associ-URI
1256801 -1 1435 yes
Na obecną chwilę chciałem skonfigurować jeden numer i sprawdzić czy zadziała. Niestety poległem.
Proszę o pomoc i wskazówki co i gdzie może być źle skonfigurowane lub czego brakuje. Mnie skończyły się pomysły. Druga sprawa, że robię to pierwszy raz w życiu, więc doświadczenie w tym zakresie raczej zerowe