Configuring SIP on Avaya IP Office 500V2 9.0

Hi,
We are looking ahead to migrate existing phone system with Avaya IP Office 500V2 version 9.0. Currently we are using Cisco UCM, Phone vendor is Fusion Connect, Internet service provider is CEN (Connecticut Education Network).
We know how to setup PRI with Avaya IP Office but not sure about SIP. Could you please let us know what information we will need to collect from service provider to configure SIP, also apart from below things what need to buy from Avaya for SIP.
- Endpoint license
- Voicemail pro license
- IP Office control unit
- SD card
- Rack mounting kit
- Essential edition license
Thank you,
Prashant Tayade
ACS.
This topic first appeared in the Spiceworks Community

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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

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An exception of type Microsoft.SharePoint.Upgrade.SPUpgradeException was thrown. Additional exception information: Failed to call GetTypes on assembly Microsoft.Office.InfoPath.Server, Version=14.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c.
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specified. Could not load file or assembly 'Microsoft.Office.InfoPath, Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c' or one of its dependencies. The system cannot find the file specified. Could not load file or assembly 'Microsoft.Office.InfoPath,
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or one of its dependencies. The system cannot find the file specified. Could not load file or assembly 'Microsoft.Office.InfoPath, Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c' or one of its dependencies. The system cannot find the file
specified. Could not load file or assembly 'Microsoft.Office.InfoPath, Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c' or one of its dependencies. The system cannot find the file specified. Could not load file or assembly 'Microsoft.Office.InfoPath,
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specified. Could not load file or assembly 'Microsoft.Office.InfoPath, Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c' or one of its dependencies. The system cannot find the file specified. Could not load file or assembly 'Microsoft.Office.InfoPath,
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Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c' or one of its dependencies. The system cannot find the file specified. Could not load file or assembly 'Microsoft.Office.InfoPath, Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c'
or one of its dependencies. The system cannot find the file specified. Could not load file or assembly 'Microsoft.Office.InfoPath, Version=15.0.0.0, Culture=neutral, PublicKeyToken=71e9bce111e9429c' or one of its dependencies. The system cannot find the file
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About configuring sales organization and sales office

how can i create sales organization and sales office in SAP ??

Hi
where to create new sales orgs and sales office is well explained by the other friend
After creating the new sales org there are a lot of assignments to be done in img-enterprise structure assignment-sales and distribution
1.   assign sales org to company code
2.   Assign sales org to distribution channel
3.   Assign sales org to division
4.   Assign sales org distribution channel to plant
5.   Set up sales area
6.   Assign sales office to sales area
Like that a lot of enterprise structure assignments needs to be completed
Plus
T code VOR1 define common distribution channels
T code VOR2 define common divisions
T code OVAO combine sales organizations
T code OVAM combine distribution channel
T code OVAN combine divisions
T code OVAZ Assign sales order types to sales areas
(Right now i dont have SAP access so that i am not able to tell you every thing
But these are must assignments which needs to be done if you have to use your new sales org
All the best
Regards
Raja

Configuring Level3 SIP trunk with Lync 2013

Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
Level 3 provided us with one signaling IP and two RTP IPs.
I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
Questions:
1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
Thanks, and let me know if I should provide additional info.

Hi,
On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Configuring SIP on Avaya IP Office 500V2 9.0

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