CiscoForums.it

Ciao a tutti
Sto configurazione il voip un una rete cond sedi collegate tra di lo tramite vpn, il voip internamente funziona, ed entrano anche le telfonate...
l'unico problema e che non escono , nel senso che mi da sempr eil segnale di occupao...

per tagliare la testa la toro ho fatto le provde direttamente collegato ad una porta fxs evitando cosi problemi di centralino o altro....

Facendo il debug del ccsip

mi viene restituito questo

//-1/000000000000/SIP/State/sipSPIChangeState: 0x84778C88 : State change from (STATE_NONE, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
1d23h: //-1/000000000000/SIP/State/sipSPIChangeState: 0x84778C88 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
1d23h: //-1/000000000000/SIP/State/sipSPIChangeState: 0x84778C88 : State change from (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE) to (SIP_STATE_OUTGOING_REGISTER, SUBSTATE_NONE)
1d23h: //-1/000000000000/SIP/Error/act_register_new_message: Message Code Class 4xx Method Code 100 not implemented for REGISTER


questo messaggio esce dopo che ho inserito nel sip-ua autenticazioe con uno dei numeri di telefono.




version 12.4
no service pad
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname +++++++++++
!
boot-start-marker
boot-end-marker
!
logging buffered 32768 notifications
enable password 7 ++++++++++++
!
no aaa new-model
memory-size iomem 15
clock timezone CEST 1
clock summer-time CEDT recurring last Sun Mar 2:00 last Sun Oct 3:00
clock save interval 8
voice-card 1
!
voice-card 2
!
ip cef
!
!
ip telnet source-interface Loopback0
!
!
ip domain lookup source-interface Loopback0
ip name-server 151.99.125.2
!
!
password encryption aes
!
voice call send-alert
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
modem passthrough nse codec g711alaw
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
!
!
voice class codec 1
codec preference 1 g711alaw
!
!
!
!
!
!
!
!
!
!
!
username +++++++ privilege 15 password 7 +++++++
!
!
class-map match-any VoIP
match ip precedence 5
class-map match-any Multimediale
match ip precedence 3
class-map match-any Professional
match ip precedence 0 1 2 4
match ip precedence 7
class-map match-any Management
match ip precedence 6
class-map match-any LAN
match access-group 110
!
!
policy-map CLIENTE-OUT
class VoIP
priority percent 80
class Management
bandwidth percent 5
class Multimediale
bandwidth percent 5
policy-map ETH-IN
class class-default
!
!
translation-rule 2
Rule 0 ^9* 0
!
!
!
!
!
!
interface Loopback0
description Interfaccia nat della LAN
ip address 62.++++++++ 255.255.255.248
!
interface Tunnel1
ip address 172.16.1.2 255.255.255.252
tunnel source 62.++++++++
tunnel destination 62.+++++++
tunnel mode ipip
!
interface ATM0/0
no ip address
no atm auto-configuration
no atm ilmi-keepalive
no atm address-registration
no atm ilmi-enable
dsl operating-mode auto
hold-queue 224 in
!
interface ATM0/0.835 point-to-point
ip address 193.+++++++++ 255.255.255.252
no ip unreachables
ip nat outside
ip virtual-reassembly
no ip mroute-cache
no snmp trap link-status
pvc 8/35
oam-pvc manage
encapsulation aal5snap
!
!
interface FastEthernet0/0
description Interfaccia LAN Cliente
ip address 192.168.1.254 255.255.255.0
ip nat inside
ip virtual-reassembly
speed auto
full-duplex
hold-queue 100 out
!
ip route 0.0.0.0 0.0.0.0 ATM0/0.835
ip route 192.168.3.0 255.255.255.0 172.16.1.1
!
!
no ip http server
no ip http secure-server
ip nat service sip udp port 8000
ip nat service sip udp port 8001
ip nat service sip udp port 8002
ip nat service sip udp port 8003
ip nat service sip udp port 3478
ip nat inside source list 101 interface Loopback0 overload
ip nat inside source static tcp 192.168.1.230 21 62.++++++++ 21 extendable
ip nat inside source static tcp 192.168.1.230 25 62.++++++++ 25 extendable
ip nat inside source static tcp 192.168.1.230 80 62.+++++++++ 80 extendable
ip nat inside source static tcp 192.168.1.230 110 62+++++++++ 110 extendable
ip nat inside source static tcp 192.168.1.230 3389 62.++++++++ 3389 extendable
ip nat inside source static tcp 192.168.1.230 32000 62.++++++++ 32000 extendable
!
access-list 10 permit 62.123.248.130
access-list 97 permit 213.234.128.169
access-list 97 permit 213.234.128.165
access-list 97 permit 213.234.128.134
access-list 101 permit ip 192.168.1.0 0.0.0.255 any
access-list 150 permit ip 213.234.152.0 0.0.1.255 host 62.123.248.130
access-list 150 permit ip 10.16.251.0 0.0.0.255 host 62.123.248.130
access-list 150 permit ip 10.16.252.0 0.0.0.255 host 62.123.248.130
access-list 150 deny tcp 62.123.0.0 0.0.0.255 host 62.123.248.130 eq telnet
access-list 150 deny udp 62.123.0.0 0.0.0.255 host 62.++++++ eq snmp
access-list 150 permit ip 62.123.0.0 0.0.0.255 host 62.+++++++
access-list 150 deny ip any host +++++++++.130
access-list 150 permit ip any any
!
route-map SET-PREC permit 10
match ip address 100
set ip precedence internet
!
route-map SET-PREC permit 20
match ip address 10
set ip precedence critical
!
route-map SET-PREC permit 30
set ip precedence routine
!
!
!
control-plane
!
!
!
voice-port 1/0
disc_pi_off
timeouts call-disconnect 0
timeouts wait-release 1
timing guard-out 300
timing sup-disconnect 50
!
voice-port 1/1
disc_pi_off
timeouts call-disconnect 0
timeouts wait-release 1
timing guard-out 300
timing sup-disconnect 50
!
voice-port 2/0
echo-cancel coverage 32
playout-delay nominal 100
playout-delay mode fixed
cptone IT
timeouts initial 120
timeouts interdigit 3
timeouts call-disconnect 0
!
voice-port 2/1
echo-cancel coverage 32
playout-delay nominal 100
playout-delay mode fixed
cptone IT
timeouts initial 120
timeouts interdigit 3
timeouts call-disconnect 0
caller-id enable
!
!
!
!
!
!
dial-peer voice 1 pots
destination-pattern 51
port 1/0
!
dial-peer voice 2 pots
destination-pattern 52
port 1/1
!
dial-peer voice 3 pots
destination-pattern 53
port 2/0
!
dial-peer voice 4 pots
destination-pattern 54
port 2/1
!
dial-peer voice 5 voip
destination-pattern 55
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 6 voip
destination-pattern 56
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 7 voip
destination-pattern 57
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 8 voip
destination-pattern 58
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 1000 voip
destination-pattern .T
translate-outgoing called 2
voice-class codec 1
session protocol sipv2
session target ras
session transport udp
incoming called-number .T
dtmf-relay h245-alphanumeric
!
dial-peer voice 12 pots
destination-pattern num tel
port 2/0
authentication username ++++++++ password ++++++++
!
dial-peer voice 13 pots
destination-pattern num tel
port 1/1
authentication username +++++++ password ++++++++
!
dial-peer voice 11 pots
destination-pattern num tel
port 2/0
authentication username <num tel> password +++++++++++++
!
sip-ua
authentication username <numtel> password ++++++++++++++
retry invite 4
retry response 3
retry bye 2
retry cancel 2
registrar dns:voip.eutelia.it:5060 expires 3600
sip-server dns:voip.eutelia.it:5060
!

Ciao, qualche domanda:
- nel dial peer voip 1000 chi è ras? se vuoi fare uscire le tue chiamate dal voip devi mettergli ad esempio: session target dns:voip.eutelia.it oppure visto che lo hai specificato session target sip-server
- i numeri di telefono che autentichi nei dial peer sono tutti uguali oppure diversi?
- tu hai authenticato i tuoi numeri tramite i dial-peer dove hai specificato lo username e la password; perchè ripeti l'autenticazione anche nel sip-ua?
Se poi sbaglio specifica meglio la tua situazione.

allora per quanto riguarda il session target del dial peer voip 1000 è un mio errore , ho postato una parte di configurazione. è impostato ora sip-server

i numeri di telefocno che autentico sono tutti diversi


e ho messo l'autenticazione al sip-ua
perchè senza facendo il debug ccsip error , mi usciva fuori
" Error getting credentials"

Togli l'authentication dal sip-ua perchè è primaria rispetto a quelle dei dial-peer e quindi te le annulla.
Puoi postare il debug dei ccsip messages di quando fai una chiamata in uscita?

ecco quà

se vuoi ci possiamo sentire su messenger


SIP Call messages tracing is enabled
Router#
00:28:39: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:voip.eutelia.it:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK2D162A
From: "13" <sip:[email protected]>;tag=1A3E40-18CC
To: "13" <sip:[email protected]>
Date: Fri, 24 Feb 2006 15:15:23 GMT
Call-ID: 93A4438D-A47B11DA-800CDF87-DA7BBE40
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1140794123
CSeq: 14 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Content-Length: 0



00:28:39: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:voip.eutelia.it:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK2E31D
From: "11" <sip:[email protected]>;tag=1A3E50-D1C
To: "11" <sip:[email protected]>
Date: Fri, 24 Feb 2006 15:15:23 GMT
Call-ID: 93932CCC-A47B11DA-800ADF87-DA7BBE40
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1140794123
CSeq: 14 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Content-Length: 0



00:28:39: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:voip.eutelia.it:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK2F373
From: "12" <sip:[email protected]>;tag=1A3E60-1024
To: "12" <sip:[email protected]>
Date: Fri, 24 Feb 2006 15:15:23 GMT
Call-ID: 93A1362D-A47B11DA-800BDF87-DA7BBE40
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1140794123
CSeq: 14 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Content-Length: 0



00:28:39: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.31:5060;rport=51875;received=79.13.130.41;branch=z9hG4bK2D162A
From: "13" <sip:[email protected]>;tag=1A3E40-18CC
To: "13" <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Call-ID: 93A4438D-A47B11DA-800CDF87-DA7BBE40
CSeq: 14 REGISTER
WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4678f08a81be9f5bd2f9a2a2e30c63c6f092aea8", qop="auth"
Server: SPS EUT GW 01 (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 83.211.227.14:5060 "Noisy feedback tells: pid=24106 req_src_ip=83.211.227.21 req_src_port=5060 in_uri=sip:voip.eutelia.it:5060 out_uri=sip:voip.eutelia.it:5060 via_cnt==2"


00:28:39: //-1/00000000-0000-0000-0000-000000000000/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
00:28:40: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.31:5060;rport=51875;received=79.13.130.41;branch=z9hG4bK2F373
From: "12" <sip:[email protected]>;tag=1A3E60-1024
To: "12" <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Call-ID: 93A1362D-A47B11DA-800BDF87-DA7BBE40
CSeq: 14 REGISTER
WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4678f08a81be9f5bd2f9a2a2e30c63c6f092aea8", qop="auth"
Server: SPS EUT GW 01 (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 83.211.227.14:5060 "Noisy feedback tells: pid=24109 req_src_ip=83.211.227.21 req_src_port=5060 in_uri=sip:voip.eutelia.it:5060 out_uri=sip:voip.eutelia.it:5060 via_cnt==2"


00:28:40: //-1/00000000-0000-0000-0000-000000000000/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
00:28:40: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:voip.eutelia.it:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK2E31D
From: "11" <sip:[email protected]>;tag=1A3E50-D1C
To: "11" <sip:[email protected]>
Date: Fri, 24 Feb 2006 15:15:24 GMT
Call-ID: 93932CCC-A47B11DA-800ADF87-DA7BBE40
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1140794124
CSeq: 14 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 3600
Content-Length: 0



00:28:40: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.31:5060;rport=51875;received=79.13.130.41;branch=z9hG4bK2E31D
From: "11" <sip:[email protected]>;tag=1A3E50-D1C
To: "11" <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Call-ID: 93932CCC-A47B11DA-800ADF87-DA7BBE40
CSeq: 14 REGISTER
WWW-Authenticate: Digest realm="voip.eutelia.it", nonce="4678f08bd8461ef53e9da5948016d391daad6bb2", qop="auth"
Server: SPS EUT GW 01 (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 83.211.227.14:5060 "Noisy feedback tells: pid=24106 req_src_ip=83.211.227.21 req_src_port=5060 in_uri=sip:voip.eutelia.it:5060 out_uri=sip:voip.eutelia.it:5060 via_cnt==2"


00:28:40: //-1/00000000-0000-0000-0000-000000000000/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
00:28:52: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK30E55
From: <sip:voip.eutelia.it>;tag=1A6F48-18B3
To: <sip:[email protected]>
Date: Fri, 24 Feb 2006 15:15:36 GMT
Call-ID: [email protected]
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 907785618-2759791066-2149965703-3665542720
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1140794136
Contact: <sip:192.168.1.31:5060>
Call-Info: <sip:192.168.1.31:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 227

v=0
o=CiscoSystemsSIP-GW-UserAgent 4508 5683 IN IP4 192.168.1.31
s=SIP Call
c=IN IP4 192.168.1.31
t=0 0
m=audio 18926 RTP/AVP 0 8 19
c=IN IP4 192.168.1.31
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:19 CN/8000

00:28:52: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.31:5060;received=79.13.130.41;branch=z9hG4bK30E55
From: <sip:voip.eutelia.it>;tag=1A6F48-18B3
To: <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="4678f097cc3a05ded99e695c79012ae43dc95b61", qop="auth"
Server: SPS EUT GW 01 (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 83.211.227.14:5060 "Noisy feedback tells: pid=24109 req_src_ip=83.211.227.21 req_src_port=5060 in_uri=sip:[email protected]:5060 out_uri=sip:[email protected]:5060 via_cnt==2"


00:28:52: //25/361BB592-A47F-11DA-8025-DF87DA7BBE40/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
00:28:52: //25/361BB592-A47F-11DA-8025-DF87DA7BBE40/SIP/Error/act_sentinvite_new_message: Error handling AuthenticationChallenge
00:28:52: //25/361BB592-A47F-11DA-8025-DF87DA7BBE40/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x826645BC
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number :
Called Number : 3395684808
Source IP Address (Sig ): 192.168.1.31
Destn SIP Req Addr:Port : 83.211.227.21:5060
Destn SIP Resp Addr:Port : 83.211.227.21:5060
Destination Name : voip.eutelia.it

00:28:52: //25/361BB592-A47F-11DA-8025-DF87DA7BBE40/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 192.168.1.31
Source IP Port (Media): 18926
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0

00:28:52: //25/361BB592-A47F-11DA-8025-DF87DA7BBE40/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 407

00:28:52: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK30E55
From: <sip:voip.eutelia.it>;tag=1A6F48-18B3
To: <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Date: Fri, 24 Feb 2006 15:15:36 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

Ciao, intanto ti consiglio di mettere nei ephone-dn "num interno" il no-reg dopo il number cosi non lo registri sul sip server.
Altra cosa aggiungi nel dail peer 1000 voip il clid network-number + un numero di telefono di quelli che hai registrato.
Il problema sta nel fatto che quando fai una chiamata in uscita usi il numero dell'interno che ovviamente non è registrato sul server eutelia.

fatto mi viene restituito questo


01:53:20: //38/0227FB62-A48B-11DA-8036-DF87DA7BBE40/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
01:53:20: //38/0227FB62-A48B-11DA-8036-DF87DA7BBE40/SIP/Error/act_sentinvite_new_message: Error handling AuthenticationChallenge
01:53:20: //38/0227FB62-A48B-11DA-8036-DF87DA7BBE40/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x82679F34
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number :
Called Number : 339568+++
Source IP Address (Sig ): 192.168.1.31
Destn SIP Req Addr:Port : 83.211.227.21:5060
Destn SIP Resp Addr:Port : 83.211.227.21:5060
Destination Name : voip.eutelia.it

01:53:20: //38/0227FB62-A48B-11DA-8036-DF87DA7BBE40/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 192.168.1.31
Source IP Port (Media): 17856
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0

01:53:20: //38/0227FB62-A48B-11DA-8036-DF87DA7BBE40/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 407

riposta tutta la conf aggiornata.

sto facendo ora le provce su un cisco 827 4v

sto collegato dietro alla rete nattata con port forwordig dal router collegato al cisco 5060 e il range di porte udp

se faccio show sip-ua register summary

il numero voip risulta registrato, se provo a fare una chiamata in entrara funziona.

o meglio arriva la chiamata sip con il debug, ma visto hce l'interfaccia cisco è su un ip privato, nel URL invite mi da errore. ma normale hce faccia cosi.




version 12.3
no service pad
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$PlGM$DqYlSoiNQSf2Wyf.DSQD/0
!
no aaa new-model
ip subnet-zero
ip dhcp excluded-address 10.10.10.1
ip dhcp excluded-address 192.168.21.2
!
!
ip name-server 213.234.128.211
ip name-server 208.67.220.220
!
voice call send-alert
voice call convert-discpi-to-prog
voice call debug full-guid
voice rtp send-recv
!
voice service voip
sip
bind control source-interface Ethernet0
bind media source-interface Ethernet0
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
!
!
!
!
!
!
!
!
!
username Router password 7 105C060C111200
!
!
no crypto isakmp ccm
!
!
!
!
interface Ethernet0
ip address 192.168.1.31 255.255.255.0
ip nat inside
ip virtual-reassembly
no ip mroute-cache
hold-queue 100 out
!
interface ATM0
no ip address
shutdown
atm vc-per-vp 64
no atm ilmi-keepalive
dsl operating-mode auto
!
interface ATM0.1 point-to-point
ip address 213.199.6.28 255.255.255.0
ip nat outside
ip virtual-reassembly
pvc 8/35
encapsulation aal5snap
!
!
interface Dialer0
no ip address
!
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.1.1
ip http server
no ip http secure-server
!
ip nat inside source list 102 interface ATM0.1 overload
!
access-list 102 permit ip 192.168.1.0 0.0.0.255 any
access-list 102 permit udp any any eq 3478
access-list 102 permit udp any any eq 5060
access-list 102 permit udp 83.211.227.0 0.0.0.255 any
access-list 102 permit udp 83.211.2.0 0.0.0.255 any
!
!
control-plane
!
!
voice-port 1
no echo-cancel enable
cptone IT
caller-id enable
!
voice-port 2
cptone IT
caller-id enable
!
voice-port 3
cptone IT
!
voice-port 4
cptone IT
!
dial-peer voice 1 pots
destination-pattern 11
port 1
no register e164
!
dial-peer voice 2 pots
destination-pattern 12
port 2
no register e164
!
dial-peer voice 3 pots
destination-pattern 13
port 1
no register e164
!
dial-peer voice 21 pots
destination-pattern 0+++++++
port 1
authentication username 0+++++++ password ++++++++++++
!
dial-peer voice 1000 voip
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay sip-notify
clid network-number 0+++++++
!
dial-peer voice 4 pots
no register e164
!
gateway
timer receive-rtp 1200
!
sip-ua
retry invite 3
retry response 3
retry bye 2
retry cancel 2
retry register 3
timers expires 60000
timers notify 100
timers register 150
registrar dns:voip.eutelia.it:5060 expires 3600
sip-server dns:voip.eutelia.it:5060
!
!
line con 0
exec-timeout 120 0
stopbits 1
line vty 0 4
exec-timeout 120 0
login local
length 0
!
scheduler max-task-time 5000
end

Domanda, la tua route statica 0.0.0.0 va al 192.168.1.1 mentre la ethernet del router è 192.168.1.31 ed ha un ip nat inside ed un overload sull'atm0.1; da ciò deduco che la tua connessione ad internet è gestita da altro router quindi puoi togliere l'ip nat inside ed anche l'overload sull'atm0.1. Guarda cosa risponde il debug ora

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK299166A
From: <sip:voip.eutelia.it>;tag=18ADF98-32D
To: <sip:[email protected]>
Date: Fri, 24 Feb 2006 21:58:02 GMT
Call-ID: [email protected]
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 1843880644-2763461082-2151800711-3665542720
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1140818282
Contact: <sip:+++++++++:5060>
Call-Info: <sip:+++++++++:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 227

v=0
o=CiscoSystemsSIP-GW-UserAgent 8172 7885 IN IP4 192.168.1.31
s=SIP Call
c=IN IP4 192.168.1.31
t=0 0
m=audio 18516 RTP/AVP 0 8 19
c=IN IP4 192.168.1.31
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:19 CN/8000

07:11:18: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.31:5060;received=79.13.130.41;branch=z9hG4bK299166A
From: <sip:voip.eutelia.it>;tag=18ADF98-32D
To: <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="46794ee9bb3fd9eccac78f450ae4629ccb57275f", qop="auth"
Server: SPS EUT GW 01 (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 83.211.227.14:5060 "Noisy feedback tells: pid=24108 req_src_ip=83.211.227.21 req_src_port=5060 in_uri=sip:[email protected]:5060 out_uri=sip:[email protected]:5060 via_cnt==2"


07:11:18: //43/6DE762C4-A4B7-11DA-8041-DF87DA7BBE40/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
07:11:18: //43/6DE762C4-A4B7-11DA-8041-DF87DA7BBE40/SIP/Error/act_sentinvite_new_message: Error handling AuthenticationChallenge
07:11:18: //43/6DE762C4-A4B7-11DA-8041-DF87DA7BBE40/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x8267E44C
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number :
Called Number : ++++++++++
Source IP Address (Sig ): 192.168.1.31
Destn SIP Req Addr:Port : 83.211.227.21:5060
Destn SIP Resp Addr:Port : 83.211.227.21:5060
Destination Name : voip.eutelia.it

07:11:18: //43/6DE762C4-A4B7-11DA-8041-DF87DA7BBE40/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 192.168.1.31
Source IP Port (Media): 18516
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0

07:11:18: //43/6DE762C4-A4B7-11DA-8041-DF87DA7BBE40/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 407

07:11:18: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.31:5060;branch=z9hG4bK299166A
From: <sip:voip.eutelia.it>;tag=18ADF98-32D
To: <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Date: Fri, 24 Feb 2006 21:58:02 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

Hai qualche problema di nat da qualche parte probabilmente sul 192.168.1.1 perchè non è normale, daltronde come hai notato anche tu che il server di eutelia veda il tuo ip privato.
Che cosa ti da la connessione internet, un atlro cisco? Se si posta la conf.
Ciao

Ciao ndiego
ho risolto il problema
elimando tutti i dial-peer
e inserendo per prima quello voice .T

e poi tuttiquelli interni
la differenza che nella configurazione vengono scritte in ordine di inserimento

ed ora le telefonate escono ed entrano tranquillamente
l'unico problema rimasto
si sente l'audio in entrata e non in uscita...

Ciao, bene piccolo passo avanti. Comunque senza rifare i dial peer potei giocare anche con u discorso di preferenze.
Per l'altro problema io ne ho avuto uno simile con un altro provider ed in quel caso il problema era il loro, vedevo io il loro ip privato del server sip.
Verifica la conf del router 192.168.1.1 e semmai riposta i sip messages.

ip nat inside source list 101 interface Loopback0 overload
access-list 101 permit 192.168.1.0 0.0.0.0
access-list 101 permit udp any an

!
route-map SET-PREC permit 10
match ip address 100
set ip precedence internet
!
route-map SET-PREC permit 20
match ip address 10
set ip precedence critical
!
route-map SET-PREC permit 30
set ip precedence routine
!
!
!
control-plane
!
!
!
voice-port 1/0
disc_pi_off
timeouts call-disconnect 0
timeouts wait-release 1
timing guard-out 300
timing sup-disconnect 50
!
voice-port 1/1
disc_pi_off
timeouts call-disconnect 0
timeouts wait-release 1
timing guard-out 300
timing sup-disconnect 50
!
voice-port 2/0
echo-cancel coverage 32
playout-delay nominal 100
playout-delay mode fixed
cptone IT
timeouts initial 120
timeouts interdigit 3
timeouts call-disconnect 0
!
voice-port 2/1
echo-cancel coverage 32
playout-delay nominal 100
playout-delay mode fixed
cptone IT
timeouts initial 120
timeouts interdigit 3
timeouts call-disconnect 0
caller-id enable
!
!
!
!
!
!
dial-peer voice 1 pots
destination-pattern 51
port 1/0
no register e164
!
dial-peer voice 2 pots
destination-pattern 52
port 1/1
no register e164
!
dial-peer voice 3 pots
destination-pattern 53
port 2/0
no register e164
!
dial-peer voice 4 pots
destination-pattern 54
port 2/1
no register e164
!
dial-peer voice 5 voip
destination-pattern 55
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 6 voip
destination-pattern 56
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 7 voip
destination-pattern 57
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 8 voip
destination-pattern 58
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 12 pots
destination-pattern ++++++++
port 2/0
authentication username +++++++ password ++++++++++
!
dial-peer voice 20 voip
destination-pattern +++++++++
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 21 voip
destination-pattern ++++++++++
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 22 voip
destination-pattern ++++++++++
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 23 voip
destination-pattern ++++++++++
session target ipv4:172.16.1.1
codec g711alaw
!
dial-peer voice 11 pots
destination-pattern +++++++++
port 2/1
authentication username ++++++++ password +++++++++
!
dial-peer voice 10 voip
preference 1
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay sip-notify
clid network-number +++++++++



DEBUG CCSIP MESSEGES



2d01h: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP ++++++++++:5060;branch=z9hG4bK1866246D
From: <sip:[email protected]>;tag=AB1A708-26B8
To: <sip:+++++++++[email protected]>
Date: Fri, 22 Jun 2007 12:36:33 GMT
Call-ID: [email protected]
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 114364722-536089052-2156305128-2767801356
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:+++++++++++@++++++++++>;party=calling;screen=yes;privacy=off
Timestamp: 1182515793
Contact: <sip:+++++++++@++++++++++:5060>
Call-Info: <sip:++++++++++++:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 5723 5178 IN IP4 62.123.198.65
s=SIP Call
c=IN IP4 62.123.198.65
t=0 0
m=audio 16610 RTP/AVP 8 100 19
c=IN IP4 62.123.198.65
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:19 CN/8000
a=ptime:20

2d01h: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP +++++++++++++:5060;branch=z9hG4bK1866246D
From: <sip:[email protected]>;tag=AB1A708-26B8
To: <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="467bc37d7aa7a6cca24567ff8cf1dead9b4fabd2", qop="auth"
Server: SPS EUT GW 01 (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 83.211.227.14:5060 "Noisy feedback tells: pid=24107 req_src_ip=83.211.227.21 req_src_port=5060 in_uri=sip:[email protected]:5060 out_uri=sip:[email protected]:5060 via_cnt==2"


2d01h: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:++++++[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP +++++++++++:5060;branch=z9hG4bK1866246D
From: <sip:[email protected]>;tag=AB1A708-26B8
To: <sip:[email protected]>;tag=d5ce1f561fa195512de35a1851da2a84.37d1
Date: Fri, 22 Jun 2007 12:36:33 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0