01-02-2014 12:54 AM - edited 03-16-2019 09:04 PM
Hi all ! ,
Im kind off new to sip calling and cisco telephony , but here goes ,: i have a 2821 router with CME installed
IOS : C2800NM-IPVOICEK9-M
Sofware version : 15.1(4)M4 / CME 8.6
Attached to GE0/0 is a CISCO 3750 switch
GEO - consisfts of 3 VLANS , the native
172.22.1.X
172.22.100.X VOICE
172.22.101.X DATA
my tftpserver = 172.22.1.150
i need some help configuring a sip trunk , i have 10 testing phonenumbers from vodafone , but i do not know where to start to get this working
i have tried
but im getting stuck with what to fill in where .. is there anyone form NL whom has the same setup ? or similar ? or can give me some guidance on how to make the test calls
01-02-2014 02:22 AM
Hi Vincent.
With conf you uploaded, your are at good point.
Now you should receive authetication credentials from you TSP for both receive and place calls.
Once you have your account, you have to add it undre sip-ua configuration section.
Eg.
sip-ua
authentication username "user" password "password" realm tspdomain.com (valid to authenticate for outgoing calls)
credentials username (or number) "username" password "password realm tspdomain.com (valid to receive calls)
registrar 1 dns:sipregistrar.com:5060 (you can add up to 6 registrar in case of different services)
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
01-02-2014 02:32 AM
01-02-2014 02:41 AM
Hi Vincent.
Yes sure could be the telephone number and, in this case, to authenticate for incoming calls, you should add.
credentials number
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
01-02-2014 03:09 AM
Hi Carlo ,
Ok . i have added the info which vodafone provided me , is there any way to trace this ? eg
picking up the horn ?
i have tried debug callmanager events .. which shows me picking up the horn but im unable to verify the connection to sip?
01-02-2014 03:21 AM
Try using the show commands in the document below:
http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-vrfy_trouble.pdf
1. show sip service
2. show sip-ua register status
3. show sip-ua statistics
4. show sip-ua status
5. show sip-ua timers
01-02-2014 03:48 AM
aha !
thanks james for the command , however i think i have done something wrong :
Router#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
0262610290 -1 138 no
2626102901 20002 138 no
901 20001 138 no
Router#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is ipv4
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio video image
Network types supported: IN
Address types supported: IP4 IP6
Transport types supported: RTP/AVP udptl
01-02-2014 04:38 AM
Hi Vincent.
Please post relevant sip-ua configuration
thanks
Regards
Carlo
Sent from Cisco Technical Support iPhone App
01-02-2014 05:09 AM
Hi Carlo ,
i dont quite understand what u mean by relevant , as u know im from the netherlands , and the CME config i spitted in there is from a USA config posted on the cisco site , i duplicated that and now trying to get sip to work
i have 10 testing phone numbers , eg 0262610290 through 9
i have connected 1 ephone with the explained 901 extention as shown in the example
if i dial the number from my phone , the call is received on the modem , but does not get to the 2821
i can ping the vodafone RTR from my 2821 console , apperently im missing something
i can use the number as sip user , but im uncertain how to go from there?
01-02-2014 06:08 AM
after enabling the debug ccsip messages the following log appears
Router#debug ccsip messages
SIP Call messages tracing is enabled
Router#
Jan 2 14:28:00.875: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:62.140.159.225:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.51:5060;branch=z9hG4bK5B1A
From: <2626102901>;tag=64348-10982626102901>
To: <2626102901>2626102901>
Date: Thu, 02 Jan 2014 14:28:00 GMT
Call-ID: F7CC4D1-72F011E3-8003C222-F14BE05E
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1388672880
CSeq: 4 REGISTER
Contact: <2626102901>2626102901>
Expires: 3600
Supported: path
Content-Length: 0
Jan 2 14:28:00.883: //50/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable - registrar unavail or not enabled
Via: SIP/2.0/UDP 172.22.1.51:5060;branch=z9hG4bK5B1A
From: <2626102901>;tag=64348-10982626102901>
To: <2626102901>;tag=42F4ABC0-9E12626102901>
Date: Thu, 02 Jan 2014 13:51:55 GMT
Call-ID: F7CC4D1-72F011E3-8003C222-F14BE05E
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Timestamp: 1388672880
CSeq: 4 REGISTER
Content-Length: 0
Jan 2 14:28:04.871: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:62.140.159.225:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.1.51:5060;branch=z9hG4bK62467
From: <901>;tag=652E4-216E901>
To: <901>901>
Date: Thu, 02 Jan 2014 14:28:04 GMT
Call-ID: F307A93-72F011E3-8002C222-F14BE05E
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1388672884
CSeq: 4 REGISTER
Contact: <901>901>
Expires: 3600
Supported: path
Content-Length: 0
Jan 2 14:28:04.879: //51/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service Unavailable - registrar unavail or not enabled
Via: SIP/2.0/UDP 172.22.1.51:5060;branch=z9hG4bK62467
From: <901>;tag=652E4-216E901>
To: <901>;tag=42F4BB5C-D6C901>
Date: Thu, 02 Jan 2014 13:51:59 GMT
Call-ID: F307A93-72F011E3-8002C222-F14BE05E
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Timestamp: 1388672884
CSeq: 4 REGISTER
Content-Length: 0
Jan 2 14:28:08.667: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.22.1.51:5060 SIP/2.0
Via: SIP/2.0/UDP 62.140.159.226:5060;branch=z9hG4bK3F1B21AD1
From: <62.140.159.226>;tag=42F4CA2C-1A1462.140.159.226>
To: <172.22.1.51>172.22.1.51>
Date: Thu, 02 Jan 2014 13:52:02 GMT
Call-ID: E4A0F55B-72EB11E3-B23DD474-C3882D2E@62.140.159.226
User-Agent: Vodafone-NL-SIP-Gateway-V1.0
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: <62.140.159.226:5060>62.140.159.226:5060>
Content-Length: 0
Jan 2 14:28:08.671: //52/EF79772E803E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.140.159.226:5060;branch=z9hG4bK3F1B21AD1
From: <62.140.159.226>;tag=42F4CA2C-1A1462.140.159.226>
To: <172.22.1.51>;tag=661B8-F5C172.22.1.51>
Date: Thu, 02 Jan 2014 14:28:08 GMT
Call-ID: E4A0F55B-72EB11E3-B23DD474-C3882D2E@62.140.159.226
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 166
v=0
o=CiscoSystemsSIP-GW-UserAgent 3073 3407 IN IP4 172.22.1.51
s=SIP Call
c=IN IP4 172.22.1.51
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 172.22.1.51
Jan 2 14:28:09.655: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.22.1.51:5060 SIP/2.0
Via: SIP/2.0/UDP 62.140.159.226:5060;branch=z9hG4bK3F1B38C9
From: <62.140.159.226>;tag=42F4CE08-F6762.140.159.226>
To: <172.22.1.51>172.22.1.51>
Date: Thu, 02 Jan 2014 13:52:03 GMT
Call-ID: E537B92C-72EB11E3-B23ED474-C3882D2E@62.140.159.226
User-Agent: Vodafone-NL-SIP-Gateway-V1.0
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: <62.140.159.226:5060>62.140.159.226:5060>
Content-Length: 0
Jan 2 14:28:09.659: //53/F010389D803F/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.140.159.226:5060;branch=z9hG4bK3F1B38C9
From: <62.140.159.226>;tag=42F4CE08-F6762.140.159.226>
To: <172.22.1.51>;tag=66594-16BD172.22.1.51>
Date: Thu, 02 Jan 2014 14:28:09 GMT
Call-ID: E537B92C-72EB11E3-B23ED474-C3882D2E@62.140.159.226
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 166
v=0
o=CiscoSystemsSIP-GW-UserAgent 2096 2958 IN IP4 172.22.1.51
s=SIP Call
c=IN IP4 172.22.1.51
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 172.22.1.51
Jan 2 14:28:09.667: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.22.1.51:5060 SIP/2.0
Via: SIP/2.0/UDP 62.140.159.225:5060;branch=z9hG4bK3F1B4CEB
From: <62.140.159.225>;tag=42F4CE14-51162.140.159.225>
To: <172.22.1.51>172.22.1.51>
Date: Thu, 02 Jan 2014 13:52:03 GMT
Call-ID: E5398E3D-72EB11E3-B23FD474-C3882D2E@62.140.159.225
User-Agent: Vodafone-NL-SIP-Gateway-V1.0
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: <62.140.159.225:5060>62.140.159.225:5060>
Content-Length: 0
Jan 2 14:28:09.671: //54/F0120DAE8040/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.140.159.225:5060;branch=z9hG4bK3F1B4CEB
From: <62.140.159.225>;tag=42F4CE14-51162.140.159.225>
To: <172.22.1.51>;tag=665A0-11172.22.1.51>
Date: Thu, 02 Jan 2014 14:28:09 GMT
Call-ID: E5398E3D-72EB11E3-B23FD474-C3882D2E@62.140.159.225
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 165
v=0
o=CiscoSystemsSIP-GW-UserAgent 7454 235 IN IP4 172.22.1.51
s=SIP Call
c=IN IP4 172.22.1.51
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 172.22.1.51
Jan 2 14:28:10.755: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.22.1.51:5060 SIP/2.0
Via: SIP/2.0/UDP 62.140.159.225:5060;branch=z9hG4bK3F1B5214B
From: <62.140.159.225>;tag=42F4D254-D2562.140.159.225>
To: <172.22.1.51>172.22.1.51>
Date: Thu, 02 Jan 2014 13:52:05 GMT
Call-ID: E5DF95C5-72EB11E3-B240D474-C3882D2E@62.140.159.225
User-Agent: Vodafone-NL-SIP-Gateway-V1.0
Max-Forwards: 70
CSeq: 101 OPTIONS
Contact: <62.140.159.225:5060>62.140.159.225:5060>
Content-Length: 0
Jan 2 14:28:10.759: //55/F0B811668041/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.140.159.225:5060;branch=z9hG4bK3F1B5214B
From: <62.140.159.225>;tag=42F4D254-D2562.140.159.225>
To: <172.22.1.51>;tag=669E4-32E172.22.1.51>
Date: Thu, 02 Jan 2014 14:28:10 GMT
Call-ID: E5DF95C5-72EB11E3-B240D474-C3882D2E@62.140.159.225
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 166
v=0
o=CiscoSystemsSIP-GW-UserAgent 7723 2893 IN IP4 172.22.1.51
s=SIP Call
c=IN IP4 172.22.1.51
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 172.22.1.51
can anyone shed some light on this ?
01-02-2014 06:45 AM
ok .. i figured out the register ,
so i have configured 5 phones , which are reachable from the outside ,
trying to dial myself back is not working .. i think it has something to do with the dial rules
01-02-2014 06:11 AM
Hi Vincent.
My question was if you can upload here the router config after changes that I suggested.
Thanks
Carlo
Sent from Cisco Technical Support iPhone App
01-02-2014 06:51 AM
ok.. here goes
Building configuration...
Current configuration : 9721 bytes
!
! Last configuration change at 15:26:14 CET Thu Jan 2 2014
! NVRAM config last updated at 15:26:14 CET Thu Jan 2 2014
! NVRAM config last updated at 15:26:14 CET Thu Jan 2 2014
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
clock timezone CET 1 0
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-select 1 E1 0/0/0
network-clock-select 2 E1 0/0/1
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
ip dhcp pool VOICE
network 172.22.100.0 255.255.255.0
option 150 ip 172.22.1.150
default-router 172.22.100.1
!
ip dhcp pool DATA
network 172.22.101.0 255.255.255.0
default-router 172.22.101.1
!
!
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-net5
!
!
!
voice service voip
ip address trusted list
ipv4 172.22.1.50
ipv4 172.22.1.51
ipv4 172.22.100.1
ipv4 172.22.101.1
ipv4 62.140.159.225
callmonitor
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
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
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
!
voice register global
!
!
!
voice translation-rule 1
rule 1 /5123781291/ /601/
rule 2 /5123781290/ /600/
!
voice translation-rule 2
rule 1 /^112$/ /112/
!
voice translation-rule 3
rule 1 /^.*/ /0262610290/
!
voice translation-rule 4
rule 2 /600/ /5123788000/
rule 3 /601/ /5123788001/
rule 4 /^2(..)$/ /51237812\1/
!
!
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
!
voice translation-profile PSTN_CallForwarding
translate redirect-target 4
translate redirect-called 4
!
voice translation-profile PSTN_Outgoing
translate calling 3
translate called 2
translate redirect-target 4
translate redirect-called 4
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
!
!
!
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-31
!
controller E1 0/0/1
framing NO-CRC4
pri-group timeslots 1-31
!
ip ftp username cisco
ip ftp password cisco123
ip tftp source-interface GigabitEthernet0/0.1
!
!
!
!
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
no keepalive
!
interface GigabitEthernet0/0.1
encapsulation dot1Q 1 native
ip address 172.22.1.51 255.255.255.0
!
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 172.22.101.1 255.255.255.0
!
interface GigabitEthernet0/0.100
encapsulation dot1Q 100
ip address 172.22.100.1 255.255.255.0
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex full
speed 100
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
!
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
!
ip forward-protocol nd
!
ip http server
ip http authentication local
no ip http secure-server
ip http max-connections 16
ip http path flash:gui
!
ip route 0.0.0.0 0.0.0.0 172.22.1.50
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
tftp-server flash:7941/apps41.8-4-1-23.sbn alias apps41.8-4-1-23.sbn
tftp-server flash:7941/cnu41.8-4-1-23.sbn alias cnu41.8-4-1-23.sbn
tftp-server flash:7941/dsp41.8-4-1-23.sbn alias dsp41.8-4-1-23.sbn
tftp-server flash:7941/jar41sccp.8-4-1-23.sbn alias jar41sccp.8-4-1-23.sbn
tftp-server flash:7941/cvm41sccp.8-4-1-23.sbn alias cvm41sccp.8-4-1-23.sbn
tftp-server flash:7941/SCCP41.8-4-2S.loads alias SCCP41.8-4-2S.loads
tftp-server flash:7941/term41.default.loads alias term41.default.loads
tftp-server debug
!
control-plane
!
!
voice-port 0/0/0:15
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/0/1:15
!
voice-port 2/0/0
!
voice-port 2/0/1
!
voice-port 2/0/2
!
voice-port 2/0/3
!
voice-port 2/0/4
!
voice-port 2/0/5
!
voice-port 2/0/6
!
voice-port 2/0/7
!
voice-port 2/0/8
!
voice-port 2/0/9
!
voice-port 2/0/10
!
voice-port 2/0/11
!
voice-port 2/0/12
!
voice-port 2/0/13
!
voice-port 2/0/14
!
voice-port 2/0/15
!
voice-port 2/0/16
!
voice-port 2/0/17
!
voice-port 2/0/18
!
voice-port 2/0/19
!
voice-port 2/0/20
!
voice-port 2/0/21
!
voice-port 2/0/22
!
voice-port 2/0/23
!
!
!
mgcp profile default
!
!
dial-peer voice 1 voip
description **Incomming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
session protocol sipv2
session target ipv4:172.22.1.50
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 ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description **Outgoing Call to SIP Trunk **
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[2-9]..[2-9]......
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 4 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[0-1][2-9]..[2-9]......
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 5 voip
description **911 Outgoing Call to SIP trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 911
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 6 voip
description **Emergency Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9911
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 7 voip
description **911/411 Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9[2-9]11
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 8 voip
description **International Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 9011T
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 9 voip
description **Star Code to SIP Trunk**
destination-pattern *..
session protocol sipv2
session target ipv4:172.22.1.50
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10 voip
description **CUE Voicemail**
translation-profile outgoing PSTN_CallForwarding
destination-pattern 600
b2bua
session protocol sipv2
session target ipv4:172.22.1.155
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 11 voip
description **CUE Auto Attendant**
translation-profile outgoing PSTN_CallForwarding
destination-pattern 601
b2bua
session protocol sipv2
session target ipv4:172.22.1.155
dtmf-relay sip-notify
codec g711ulaw
no vad
!
!
sip-ua
authentication username 0262610290 password 7 15020A1F173D24362C realm 62.140.1
59.225
authentication username 0262610290 password 7 021605481811003348
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:62.140.159.225 expires 3600
sip-server ipv4:62.140.159.224
host-registrar
!
!
telephony-service
max-ephones 58
max-dn 192
ip source-address 172.22.100.1 port 2000
calling-number initiator
system message testing
cnf-file location TFTP tftp://172.22.1.150/
load 7960-7940 P00307020200.loads
load 7941 SCCP41.8-4-2S.loads
load 7941GE SCCP41.8-4-2S
time-format 24
dialplan-pattern 1 26261029.. extension-length 3 extension-pattern 9..
voicemail 600
max-conferences 12 gain -6
call-forward pattern 9.T
moh music-on-hold.au
web admin system name admin password password
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
directory entry 1 101 name 101
create cnf-files version-stamp 7960 Jan 02 2014 08:40:49
!
!
ephone-dn 1
number 290 secondary 0262610290
name Phone 1
hold-alert 30 originator
!
!
ephone-dn 2
number 291 secondary 0262610291
name phone 2
hold-alert 30 originator
!
!
ephone-dn 3
number 292 secondary 0262610292
name Phone 3
hold-alert 30 originator
!
!
ephone-dn 4
number 293 secondary 0262610293
name Phone 4
hold-alert 30 originator
!
!
ephone-dn 5
number 294 secondary 0262610294
label Phone 5
hold-alert 30 originator
!
!
ephone 1
mac-address 0019.E88F.3BDD
button 1:1
!
!
!
ephone 2
mac-address 001E.4A92.0A27
type 7961
button 1:2
!
!
!
ephone 3
mac-address 0012.43F5.03AF
button 1:3
!
!
!
ephone 4
mac-address 000F.F7AC.502A
button 1:4
!
!
!
ephone 5
mac-address 0019.E851.090A
button 1:5
!
!
!
!
line con 0
line aux 0
line vty 0 4
login
transport input all
!
scheduler allocate 20000 1000
ntp master
end
01-02-2014 08:53 AM
Hi Vincent
As I suggested, to receive calls, you should add this configuration line under sip-ua section.
sip-ua
credentials number 0262610290 username 0262610290 password 7 021605481811003348 realm 62.140.159.225
Than post again a show sip-ua register status
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
01-02-2014 12:13 PM
i am able to receive calls , i am unable to call something
i have added additional ephones to my setup
0262610291
0262610292
0262610293
and they all can recieve calls , but i cannot place calls to an outside line ..
i will add the setup cfg u proposed
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