NEC 8500 - Cox Communications

www.CoxBusiness.com
SIP Trunking Configuration Guide
for
UNIVERGE SV8500
SV8500 S05 ISSUE 03.00
© 2012, Cox Communications, Inc. All rights reserved.
This documentation is the confidential and proprietary intellectual property of Cox Communications, Inc.
Any unauthorized use, reproduction, preparation of derivative works, performance, or display of this
document, or software represented by this document is strictly prohibited.

Table of Contents
1 Audience ................................................................................................................................................ 4
2 Introduction ............................................................................................................................................ 4
2.1 tekVizion Labs ................................................................................................................................ 5
3 SIP Trunking Network Components ...................................................................................................... 6
3.1 Hardware Components ................................................................................................................... 7
3.2 Software Requirements .................................................................................................................. 7
4 Features ................................................................................................................................................. 8
4.1 SIP Registration Method ................................................................................................................. 8
4.2 Features Supported ........................................................................................................................ 8
4.3 Features Not Supported ................................................................................................................. 8
5 Caveats and Limitations......................................................................................................................... 9
6 Configuration ........................................................................................................................................ 10
6.1 Configuration Checklist ................................................................................................................. 10
6.2 IP Address Worksheet .................................................................................................................. 11
6.3 UNIVERGE SV8500 Detailed Configuration Steps ...................................................................... 11
6.3.1 MAC Data .............................................................................................................................. 11
6.3.2 DRS Data .............................................................................................................................. 12
6.3.3 Common Config Data ............................................................................................................ 12
6.3.4 Port Config Data .................................................................................................................... 16
6.3.5 SIP Config Data ..................................................................................................................... 17
6.3.6 Routing Config Data .............................................................................................................. 27
7 Appendix: SIP Call Traces ..................................................................... Error! Bookmark not defined.
7.1 Terminating Call Trace ................................................................... Error! Bookmark not defined.
7.2 Originating Call Trace ..................................................................... Error! Bookmark not defined.
7.3 Call Forward Trace ......................................................................... Error! Bookmark not defined.
7.4 Call Transfer Trace ......................................................................... Error! Bookmark not defined.
7.5 Anonymous Caller ID Trace ........................................................... Error! Bookmark not defined.
Table of Figures
Figure 1 - Cox Fiber Network ........................................................................................................................ 4
Figure 2 - SIP Trunk Lab Reference Network ............................................................................................... 6
Figure 3 Terminating Call Trace ................................................................... Error! Bookmark not defined.
Figure 4 Terminating Call Trace cont. ........................................................... Error! Bookmark not defined.
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Figure 5 Originating Call Trace ..................................................................... Error! Bookmark not defined.
Figure 6 Originating Call Trace cont. ............................................................ Error! Bookmark not defined.
Figure 7 Call Forwarding Trace..................................................................... Error! Bookmark not defined.
Figure 8 Call Forwarding Trace cont. ............................................................ Error! Bookmark not defined.
Figure 9 Call Forwarding Trace cont. ............................................................ Error! Bookmark not defined.
Figure 10 Call Transfer Trace ....................................................................... Error! Bookmark not defined.
Figure 11 Call Transfer Trace cont. .............................................................. Error! Bookmark not defined.
Figure 12 Call Transfer Trace cont. .............................................................. Error! Bookmark not defined.
Figure 13 Anonymous Call Trace ................................................................. Error! Bookmark not defined.
Figure 14 Anonymous Call Trace cont. ......................................................... Error! Bookmark not defined.
Table of Tables
Table 1 – PBX Configuration Steps ............................................................................................................ 10
Table 2 – IP Addresses ............................................................................................................................... 11
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1 Audience
This document is intended for the SIP trunk customer’s technical staff and Value Added Retailer (VAR)
having installation and operational responsibilities.
2 Introduction
This Configuration Guide describes configuration steps for Cox SIP trunking to a UNIVERGE SV8500
PBX. Cox SIP trunking is a scalable and efficient IP trunking telecommunication solution for your
business that provides all the traditional services such as Direct Inward Dialing, Hunting, Calling Name,
Calling Number, Local/Long Distance and Business Continuity options, including:
Burstable Trunk Capacity – Dynamically increases call capacity during peak busy periods so your
customers never receive a busy signal.
Call Forward Always – On the trunk group pilot number for all calls in case of an outage (i.e.,
flood, fire, loss of power, etc.).
Call Forward Not Reachable – On the trunk group pilot number that operates on a per-call
contingency basis to forward the call to any PSTN number (i.e., call center or alternate office
location) during temporary call completion impairments.
Route Exhaustion – Automatic reroute of trunk group calls to any PSTN phone number (i.e., a call
center) if calls can’t be completed to the PBX.
Support for geo-redundant PBX deployments and automatic reroute of SIP trunks to the backup
customer data center.
All calls are routed over Cox’s national fiber network with guaranteed Quality of Service (QoS); calls never
traverse the internet.
Figure 1 - Cox Fiber Network
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2.1 tekVizion Labs
tekVizion Labs TM is an independent testing and Verification facility offered by tekVizion PVS, Inc.
(“tekVizion”). tekVizion Labs offers several types of testing services including:
Remote Testing – provides secure, remote access to certain products in tekVizion Labs for pre-
Verification and ad hoc testing
Verification Testing – Verification of interoperability performed on-site at tekVizion Labs between
two products or in a multi-vendor configuration (“solution Verification”)
Product Assessment – independent assessment and verification of product functionality, interface
usability, assessment of differentiating features as well as suggestions for added functionality,
stress and performance testing, etc.
tekVizion is a systems integrator specifically dedicated to the telecommunications industry. Our core
services include consulting/solution design, interoperability/Verification testing, integration, custom
software development and solution support services. Our services helps service providers achieve a
smooth transition to packet-voice networks, speeding delivery of integrated services. While we have
expertise covering a wide range of technologies, we have extensive experience surrounding our
FastForward>> practice areas which include: SIP Trunking, Packet Voice, Service Delivery, and
Integrated Services.
The tekVizion team brings together experience from the leading service providers and vendors in
telecom. Our unique expertise includes legacy switching services and platforms, and unparalleled product
knowledge, interoperability and integration experience on a vast array of VoIP and other next-generation
products. We rely on this combined experience to do what we do best: help our clients advance the
rollout of services that excite customers and result in new revenues for the bottom line. tekVizion
leverages this real-world, multi-vendor integration and test experience and proven processes to offer
services to vendors, network operators, enhanced service providers, large enterprises and other
professional services firms. tekVizion’s headquarters, along with a state-of-the-art test lab and Executive
Briefing Center, is located in the Telecom Corridor® in Richardson, Texas.
(For more information on tekVizion and its practice areas, please visit tekVizion Labs’s web site at
www.tekVizionlabs.com.)
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3 SIP Trunking Network Components
The network for the SIP trunk reference configuration is illustrated below and is representative of a
UNIVERGE SV8500 configuration.
Figure 2 - SIP Trunk Lab Reference Network
Note: The UNIVERGE SV8500 does not offer DHCP server for dynamic IP address assignment for the
SIP phones; however, the Cox Enterprise Session Border Controller (E-SBC) requires a static LAN IP
address that must be manually assigned by the LAN network administrator. The DHCP server is
provisioned on the Ethernet switch. The DHCP’s IP address pool is constrained so that the E-SBC can
be assigned an IP address outside of the pool.
The lab network consists of the following components:
ShoreTel ShoreWare PBX for voice features, SIP proxy and SIP trunk termination.
Various SIP phones on the local LAN.
The Cox E-SBC is the Edgewater Networks (www.edgewaternetworks.com) EdgeMarc appliance.
The EdgeMarc is the service demarcation point between customer’s LAN network and Cox’s
WAN network and provides firewall/NAT traversal, B2BUA and SIP Application-level gateway.
The EdgeMarc has diverse routes to a primary and secondary Acme SBC.
Acme Packet Net-Net 9200 Session Border Controllers (SBC).
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3.1 Hardware Components
UNIVERGE SV8500
Analog fax machine
EdgeMarc 4550 E-SBC
MG-SIP
3.2 Software Requirements
SV8500 S05 Issue 03.00
EdgeMarc 4550 11.6.5 Release
MG-SIP SP.No 3988 Issue 02.02.02.07
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4 Features
4.1 SIP Registration Method
Cox Network requires SIP REGISTER support to allow the IP-PBX to originate calls from the IP-PBX and
to send calls to the PBX from the PSTN. ShoreTel ShoreWare supports SIP Register with authentication
via MG-SIP. Cox implementation team provides the Pilot number and the authentication key, which
should be provisioned in the UNIVERGE SV8500. How to configure these in the UNIVERGE SV8500 are
shown in Section 6.3.1.
4.2 Features Supported
Basic calls using G.711ulaw
Anonymous call
International Call
Call Transfer
Call Forwarding
Call Hold and Resume
Call Pickup
Call Waiting
DND
Call Park
Do Not Disturb
Hunt groups (Simultaneous and Sequential Ring)
Three-Way Calling
PBX Account Codes
PBX Authorization Codes
Casual dialing
Fax Send
Fax Receive
Dial-Up Modem
900 Call blocking
E911 Call
PBX-Defined Caller ID (spoofing)**See Caveats and Limitations
4.3 Features Not Supported
Broadsoft Account Codes
Broadsoft Authorization Codes
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5 Caveats and Limitations
For all call forward test cases, the forwarded call sends Anonymous as the name in “From”
header. This causes Broadworks to send Anonymous or Private call out to the second leg of the
call.
Caller ID Spoofing can be done for the whole route. If the caller Id spoofing is activated for the
router, all the extensions in the whole route will have the same Caller ID. So the extensions for
which the caller IDs to be spoofed should be in a separate Route.
Broadsoft account codes and Broadsoft authorization codes have one way audio. Edgewater’s
current code cannot support the route in the re-invite that is generated from the network.
When the PBX sends the 200 OK with SDP to the public WAN IP address, the SBC is not
modifying the SDP which causes one way audio.
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6 Configuration
6.1 Configuration Checklist
In this section we present an overview of the steps that are required to configure UNIVERGE SV8500 for
SIP Trucking as well as all features that were tested.
Table 1 – PBX Configuration Steps
Step Description Reference
Step 1 MAC Data Section 6.3.1
Step 2 DRS Data Section 6.3.2
Step 3 Common Config Data Section 6.3.3
Step 4 Port Config Data Section 6.3.4
Step 5 Sip Config Data Section 6.3.5
Step 6 Routing Config Data Section 6.3.6
Step 7 International Call Section 6.3.7
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6.2 IP Address Worksheet
The specific values listed in the table below and in subsequent sections are used in the lab configuration
described in this document, and are for illustrative purposes only. The customer must obtain and use
the values for your deployment.
Table 2 – IP Addresses
Component Cox Lab Value Customer Value
EdgeMarc E-SBC
LAN IP Address 10.64.2.249
LAN Subnet Mask 255.255.255.0
UNIVERGE SV8500 IP PBX
System IP Address
The Internet Connection will typically be on the same
subnet as the LAN IP Address of the E-SBC. If this is
not the case, then Layer 3 routing must be in place.
Default Gateway
The Default Gateway must be the LAN Network
default Gateway. This will allow the administrator to
log in via his\her workstation if the workstation is on
a different network
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10.70.88.2
10.70.88.1
WAN MG-SIP 10.64.2.172
LAN MG-SIP 10.70.88.5
6.3 MG-SIP Detailed Configuration Steps
The SV8500 is an enterprise PBX. The SV8500 will be provisioned by a certified SV8500 engineer. The
information below represents the configuration information for the MG-SIP card as tested.
Equipment used for configuration setup:
MG-SIP.
6.3.1 MAC Data
1. The MAC address for the Ethernet 1 and Ethernet 2 are assigned by default.

Ether[1]
MAC Address : 00-60-b9-4e-db-92
Ether[2]
MAC Address : 00-60-b9-4e-db-52
6.3.2 DRS Data
1. Set DRS Data:
2. The command used to set the DRS Data is “set drsaddress”.
3. Set DRS IP Address for unit(0):10.70.88.2 is used for this example. This IP Address is the PBX
IP address. Please use the actual IP address of PBX for your network. The PBX address may/will
be different from this example.
4. All the other options are set by default.
MG(SIP) > set drsaddress
DRS[0] IP address : 0.0.0.0 (default=0.0.0.0) : PortNo[3456] (default=3456)
DRS[1] IP address : 0.0.0.0 (default=0.0.0.0) : PortNo[3456] (default=3456)
DRS[2] IP address : 0.0.0.0 (default=0.0.0.0) : PortNo[3456] (default=3456)
DRS[3] IP address : 0.0.0.0 (default=0.0.0.0) : PortNo[3456] (default=3456)
DRS unit(0)
DRS IP address : 10.70.88.2
DRS port number (1024-65535) :
DRS unit(1)
DRS IP address :
DRS port number (1024-65535) :
DRS unit(2)
DRS IP address :
DRS port number (1024-65535) :
DRS unit(3)
DRS IP address :
DRS port number (1024-65535) :
DRS[0] IP address : 10.70.88.2 : PortNo[3456]
DRS[1] IP address : 0.0.0.0 : PortNo[3456]
DRS[2] IP address : 0.0.0.0 : PortNo[3456]
DRS[3] IP address : 0.0.0.0 : PortNo[3456]
6.3.3 Common Config Data
1. All the port numbers are assigned by default.
set signaling_port_no :
Signal port number : 61012
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--------------------------------------------
set registration_port_no :
Registration port number : 3456
--------------------------------------------
set pre_negotiation_port_no:
Voice path port number : 61014
--------------------------------------------
set ipx_rtp_port_no :
IPX side RTP port number : 51000
--------------------------------------------
set sip_rtp_port_no :
SIP side RTP port number : 50000
--------------------------------------------
set h245_base_port_no :
H245 base port number : 40000
--------------------------------------------
2. The DRS QOS parameters are set by default.
MG(SIP) > set drs_qos:
DRS QOS : 0xa0
PRECEDENCE : 5
DELAY : 0
THROUGHPUT : 0
RELIABILITY : 0
COST : 0
3. The RTP QOS parameters are set by default.
MG(SIP) > set rtp_qos :
RTP QOS : 0xa0
PRECEDENCE : 5
DELAY : 0
THROUGHPUT : 0
RELIABILITY : 0
COST : 0
4. The SIP QOS parameters are set by default.
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SIP QOS : 0xa0
PRECEDENCE : 5
DELAY : 0
THROUGHPUT : 0
RELIABILITY : 0
COST : 0
5. The Domain Name parameters are set by default.
MG(SIP) > set domain
Domain name : 0
6. The DNS IP Address parameters are set by default.
MG(SIP) > set dnsaddress
DNS IP address : 0.0.0.0
7. The H/C Alarm type parameters are set by default.
set hc_alarm :
H/C Alarm type : IPX and SIP Side enable
8. The H/C Timer value parameters are set by default.
set hc_timer :
H/C Timer value : 240
9. The Music Type parameters are set by default.
set musictype :
Music Type : Type1
10. Set Call Hold function:
11. The command used to change the Call Hold function is “set call_hold”.
12. Set Call hold function: 1(enable) is given for this example.
13. All the other options are set to default values.
MG(SIP) > set call_hold
>
Call Hold function : disable
Select the setting of Call Hold function.
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0 = disable (default)
1 = enable
Input : 1
Input RTP detect timer value.(sec)
(3 - 32(default=3)) : 3
Input SIP Error Response code to reject new session.
(400 - 606(default=488)) : 488
>
Call Hold function : enable
RTP detect timer value : 3sec
Response code to reject : 488
14. Set Name Display:
15. The command used to set name display is “set name_display”.
16. Set select the setting of Name Display: 1(enable) is set for this example.
MG(SIP) > set name_display
>
Name Display : enable
Select the setting of Name Display.
0 = disable (default)
1 = enable
Input : 1
>
Name Display : enable
17. The DRS IP Version parameters are set by default.
set drs_ip_version :
DRS IP version : 0
18. Set Multi-session:
19. The command used to change the Multi-session setting is “set multisession”.
20. Set Multi-session: 1(enable) is set for this example.
MG(SIP) > set multisession
The multi-session setting is disable.
Is the multi-session setting enable? (0:disable/1:enable(default=0)) : 1
The multi-session setting is enable.
set multisession :
Multi-session setting : enable.
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6.3.4 Port Config Data
1. Set Port IP Address:
2. The command used to set port IP address is “set ipaddress”
3. Set Use one-port only? Y/N = N.
4. Set Ether[1] IP Address: 10.70.88.5 is set for this example. This IP Address is the MG-SIP LAN
IP address. Please use the actual IP address of MG-SIP for your network. The MG-SIP LAN
address may/will be different from this example.
5. Set Ether[1] Subnet: 255.255.255.0.
6. Set Ether[2] IP Address: 10.64.2.172. This IP Address is the MG-SIP WAN IP Address. Please
use the actual IP address of MG-SIP for your network. The MG-SIP WAN address may/will be
different from this example.
7. Set Ether[2] Subnet: 255.255.0.0.
MG(SIP) > set ipaddress
Use all port
Ether [1]
IP address: 0.0.0.0 (default=0.0.0.0)
Subnet : 0.0.0.0 (default=0.0.0.0)
Ether [2]
IP address: 0.0.0.0 (default=0.0.0.0)
Subnet : 0.0.0.0 (default=0.0.0.0)
Use one-port only? Y/N = N
Ether [1]
IP Address: 10.70.88.5
Subnet : 255.255.255.0
Ether[2]
IP Address: 10.64.2.172
Subnet : 255.255.0.0
Use all port
Ether[1]
IP address : 10.70.88.5 (default=0.0.0.0)
Subnet : 255.255.255.0 (default=0.0.0.0)
Ether[2]
IP address : 10.64.2.172 (default=0.0.0.0)
Subnet : 255.255.0.0 (default=0.0.0.0)
Please choose a reference number: 3
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6.3.5 SIP Config Data
1. Set SIP Server:
2. The command used to set SIP Server is “set sip_server”.
3. Set SIP Server IP Address: 10.64.2.249 is set for this example. This IP Address is the E-SBC
LAN IP Address. Please use the actual IP address of E-SBC LAN for your network. The E-SBC
IP address may/will be different from this example.
4. Set SIP server port number: 5060 is set by default.
5. Set Strict or Loose(0/1): 0 is set for this example.
6. Set RFC3581(rport) function is disable or enable (0/1): 0 is set for this example.
7. Set Domain Name(MAX128): 0 is set for this example.
MG(SIP) > set sip_server
SIP server type : IP Address (default=IP Address)
SIP server IP address : 0.0.0.0(default=0.0.0.0)
SIP server port number : 5060 (default=5060)
SIP server Strict or Loose : 0 (default=0)
rport(RFC3581) function : disable (defaut=disable)
Domain Name : 0 (default=0)
SIP server type (0=IPAddress/1=FQDN): 0
SIP server IP address : 10.64.2.249
SIP server port number (1024-65535) : 5060
Strict or Loose (0/1) : 0
RFC3581(rport) function is disable or enable (0/1) : 0
Domain Name (MAX128) : 0
SIP server type : IP Address
SIP server IP address : 10.64.2.249
SIP server port number : 5060
SIP server Strict or Loose : 0
rport(RFC3581) function : disable
Domain Name : 0
8. Set SIP register:
9. The command used to set SIP register is “set sip_register”.
10. Set SIP register type(0: IP address/1: FQDN): IP Address is given for this example.
11. Set SIP register IP Address: 10.64.2.249 is set for this example. This IP Address is the E-SBC
LAN IP Address. Please use the actual IP address of E-SBC for your network. The E-SBC IP
address may/will be different from this example.
12. All other parameters are set to default values.
MG(SIP) > set sip_register
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SIP register type : IP Address (default=IP Address)
SIP register IP address : 10.64.2.249 (default=0.0.0.0)
SIP register port number : 5060 (default=5060)
SIP register expires time : 60 (default=3600)
SIP register type (0:IP address/1:FQDN) : 0
SIP register IP address : 10.64.2.249
SIP register port number (1024-65535) : 5060
SIP register expires time(1-65535) : 60
SIP register type : IP Address
SIP register IP address : 10.64.2.249
SIP register port number : 5060
SIP register expires time : 60
13. Set Key Number:
14. The command used to set SIP register is “set keynumber”.
15. Set Key number (MAX32): 6782383600 is given for this example.
MG(SIP) > set keynumber
Key number : 0 (default=0)
Key number (MAX32) : 6782383600
Key number : 6782383600
16. Set Authorization:
17. The command used to set authorization is “set auth”.
18. Set user ID (MAX32): 6782383600.
19. Set Password (MAX32): 6782383600.
MG(SIP) > set auth
User ID : 0 (default=0)
Password : 0 (default=0)
User ID (MAX32) : 6782383600
Password (MAX32) : 6782383600
User ID : 6782383600
Password : 6782383600
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20. The Privacy Pattern parameters are set by default.
set privacy_pattern :
Privacy pattern : DisplayName
21. Set Session Timer:
22. The command used to set Session Timer is “set session_timer”.
23. Set session-expires(0-86400(default=180sec)): 1800 is set for this example.
24. All the other options are default settings.
MG(SIP) > set session_timer
Session-timer is enable.
Session-expires : 180
Refresher : UAC
Invite Refresher : disable
Forced : disable
Is a session-timer enable ? (0:enable/1:disable(default=0)) : 0
Session-expires(0-86400(default=180sec)) : 1800
Refresher(0:UAC/1:UAS(default=0)) : 0
Add "refresher=uac" in INVITE ? (0:disable/1:enable(default=0)) : 0
forced mode ? (0:disable/1:enable(default=0)) : 0
Session-timer is enable.
Session-expires : 1800
Refresher : UAC
Invite Refresher : disable
Forced : disable
25. The Prack parameters are set by default.
100rel : enable.
Function(stop cyclical-send of 18x) :disable.
26. The CDN Pattern parameters are set by default.
Called-number pattern : To:userinfo
27. The SIP port number parameter is set by default.
set sip_port_no :
SIP port number : 5060
28. The 184 Privacy function parameter is set by default.
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set 184ToPrivacy :
184->Privacy function : disable
29. Set SIP server slip response code:
30. The command used to set SIP Server response code is “set slipresp”
31. Set Server slip response code(default=488): 503 is given for this example.
MG(SIP) > set slipresp
SIP Server slip response code(default=488) : 488
SIP Server slip response code ? (400-699) : 503
SIP Server slip response code : 503
32. Set Self SIP response domain mode:
33. The command used to set Self SIP response domain mode is “set self_sip_domain”
34. Set self SIP domain mode is disable?(0:disable/1:enable(default=0)): 1 is given for this
example.
8. Set self SIP domain string:10.64.2.172 is set for this example. This IP Address is the MG-SIP
WAN IP Address. Please use the actual IP address of MG-SIP for your network. The MG-SIP
WAN address may/will be different from this example.
MG(SIP) > set self_sip_domain
Self SIP domain mode is disable.
Self SIP domain string : 10.64.2.172
Self SIP domain mode is disable ? (0:disable/1:enable(default=0)) : 1
Please input a self SIP domain string. (MAX128) : 10.64.2.172
Self SIP domain mode is enable.
Self SIP domain string : 10.64.2.172
35. Set DTMF mode:
36. The command used to set DTMF mode is “set dtmf_mode”.
37. Set DTMF Mode (0=Negotiation, 1=Out-Band, 2=In Band) (default=0): 1 is set for this
example.
38. All the other settings are set by default.
MG(SIP) > set dtmf_mode
DTMF Mode : Negotiation
DTMF Duration : 120ms
DTMF Pause : 100ms
Payload type : 101
DTMF Mode (0=Negotiation,1=Out-Band,2=In-Band) (default=0) : 0
DTMF Duration (50-240(default=120)) : 120
DTMF Pause (30-240(default=100)) : 100
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Paylaod type(RFC2833) (96-127(default=101)) : 101
DTMF Mode : Out-Band(RFC2833)
DTMF Duration : 120ms
DTMF Pause : 100ms
Payload type : 101
39. The RBT add parameter is set by default.
set 183rbt
RBT add (183w/oSDP) : enable
40. Set RBT Pathon:
41. The command used to set RBT Pathon is “set rtp_pathon”.
42. Set That-on 183 after 180? (0:disable/1:enable(default=0)): 1 is set for this example.
MG(SIP) > set rtp_pathon
Path-on 183 after 180 : disable
Path-on 183 after 180 ? (0:disable/1:enable(default=0)) : 1
Path-on 183 after 180 : enable
43. The another key-number parameters are set by default.
set another_keynumber
Another keynumber 1 : 0
Another keynumber 2 : 0
Another keynumber 3 : 0
44. Thermulti registration mode parameters are set by default.
set multi_regist
Multi-registration mode : disable
45. Set SIP Telephone services:
46. The command used to set multi registration mode is “set sip_tel_service”.
47. Set SIP telephone service function: enable is selected for this example.
48. Set Replaces function: enable is selected for this example.
49. Set hold[a=inactive] function: enable is selected for this example.
50. Set hold[put off hold] function: disable is selected for this example.
51. Set REFER/NOTIFY function: disable is selected for this example.
52. Set 3PCC function: disable is selected for this example.
53. Set 3PCC function(codec): AUTO is selected for this example.
MG(SIP) > set sip_tel_service
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SIP telephony service function: disable
Replaces function : enable
hold[a=inactive] function : enable
hold[put off hold] function : disable
REFER/NOTIFY function : disable
3PCC function : disable
3PCC function (codec) : AUTO
Is SIP telephony service function disable ?
(0:disable/1:enable(default=0)) : 1
Is Replaces function enable ? (0:enable/1:disable(default=0)) : 0
Is hold[a=inactive] function enable? (0:enable/1:disable(default= 0)) : 0
Is hold[put off hold] function enable ?
(0:disable/1:enable(default=0)) : 1
Is the REFER/NOTIFY function enable ?
(0:enable/1:disable(default=1)) : 0
Is the 3PCC function enable ?
(0:enable/1:disable(default=1)) : 0
Select 3PCC function(codec).
(0:PCMU/1:PCMA/2:AUTO(default=2)) : 2
SIP telephony service function : enable
Replaces function : enable
hold[a=inactive] function : enable
hold[put off hold] function : enable
REFER/NOTIFY function : enable
3PCC function : enable
3PCC function(codec) : AUTO
54. The Register interval parameter is set by default.
set reg_interval
REGISTER Interval timer : 5min
55. The Update parameter is set by default.
set update
UPDATE Method support function : disable
56. The Authorization Header parameter is set by default.
set auth_header
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Authorization header cache function: enable.
57. Set Cause table:
58. The command used to set Cause table is “set cause_table”.
59. Set cause_table: 3(Route Advance) is selected for this example.
MG(SIP) > set cause_table
Cause_table is Route Advance
Select cause_table(SIP_Error_response to Error_cause)
0 = normal (default)
1 = alternate routing
2 = customized pattern1
3 = Route Advance
60. The Response table parameters are set by default.
MG(SIP) > set response_table
Response_table is normal.
Select Response_table(Error_cause to SIP_Error_response)
(0:normal/1:revised_table(default=0)) :
61. Set CPN Pattern:
62. The command used to set CPN Pattern is “set cpn_pattern”.
63. Set CPN Pattern: 1(Userinfo) is selected for this example.
MG(SIP) > set cpn_pattern
Priority setting of calling party number : Displayname
What calling party number is given to priority?
(0:Displayname/1:Userinfo/2:Anonymous/3:P-Asserted-ID(default=0)) : 1
Priority setting of calling party number : Userinfo
64. The CC Convert parameters are set by default.
set cc_convert
The country-code convert function : disable
65. The Check number parameter is set by default.
set check_number
Number check function : disable
66. The Error response code when out of range parameter is set by default.
set out_of_area_code
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Error response code when the out of area : 408(default)
67. Set SDP style:
68. The command used to set SDP style is “set sdp_style”.
69. Set SDP style setting: 1(RFC4566) is selected for this example.
70. Set G729 annexb: 1(enable) is selected for this example.
MG(SIP) > set sdp_style
SDP style setting : normal
G729 annexb=no : disable
Please select SDP style setting(0:normal/1:RFC4566(default=0)) : 1
Please select G729 annexb=no setting(0:disable/1:enable(default=0)) : 1
SDP style setting : RFC4566
G729 annexb=no : enable
71. Set Caller-ID relay mode:
72. The command used to set Caller-ID relay mode is “set call_id_relay”.
73. Set Caller-ID relay mode: 1(enable) is selected for this example.
MG(SIP) > set call_id_relay
>
Call-ID relay mode : disable
Select function of Call-ID relay mode.
0 = disable (default)
1 = enable
Input : 1
>
Call-ID relay mode : enable
74. The SIP Access Filter parameter is set by default.
set sip_accfilter
SIP Access Filter : disable
75. The SIP Transport Protocol parameter is set by default.
set sip_transport
SIP transport protocol : UDP
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76. The DNS Option parameter is set by default.
set dns_option
DNS type : A Record resolve(single IP Address)
77. Set Codec change:
78. The command used to set Codec change is “set codec_change”.
79. Set Codec Change function: 1(enable) is given for this example.
80. Set G711 Fax renegotiation: 1(enable) is given for this example.
MG(SIP) > set codec_change
>
Codec Change function : enable
G711 Fax renegotiation : enable
Select the setting of Codec Change function.
0 = disable (default)
1 = enable
Input : 1
Select the setting of G711 Fax renegotiation.
0 = disable (default)
1 = enable
Input : 1
>
Codec Change function : enable
G711 Fax renegotiation : enable
81. The Response code when unsupported codec is set by default
set no_media_code
Response code(unsupported codec) : 488
82. The Paid transmission function parameter is set by default.
set ppi_to_pai
PAID transmission function : disable
83. The SIP server 2 parameters are set by default.
set sip_server2
SIP server 2 type : IP Address
SIP server 2 IP address : 0.0.0.0
SIP server 2 portno : 5060
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SIP server 2 Strict or Loose : 0
Domain Name : 0
84. The SIP register 2 parameters are set by default.
set sip_register2
SIP register 2 type : IP Address
SIP register 2 IP address : 0.0.0.0
SIP register 2 port number : 5060
SIP register 2 expires time : 3600
85. The Connection parameter is set by default.
set connection
Connection type : 0
86. The Wait timer parameter is set by default.
set 1xx_wait_timer
1xx wait timer(sec) : 6sec
87. The DECIDED codec in SDP for re-INVITE function parameter is set by default.
set sdp_decided_codec
Set only Decided codec in SDP : disable
88. The Info message parameter is set by default.
set cftinfo
CFT Info : disable
89. Set Header configuration:
90. The command used to set Header Configuration is “set header_config”.
91. Set User of contact header: 1(Keynumber).
92. Set Delete 100rel from Supported-header after connected:0(disable).
MG(SIP) > set header_config
[Contact header]
User of contact header : Not Configration
[Supported header]
Not delete 100rel from Supported-header after connected.
Select user of contact header
(0:Not Configration/1:Keynumber/2:Random(default=0)) : 1
Delete 100rel from Supported-header after connected.
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(0:disable/1:enable(default=0)) : 0
[Contact header]
User of contact header : Keynumber
[Supported header]
Not delete 100rel from Supported-header after connected.
93. The Session redirect parameter is set by default.
set session_redirect
The session redirect function : disable
94. The SIP Server no answer timeout timer parameter is set by default.
set noanswer_timer
SIP Server no answer timer(second) : 86400s(1day)
95. The Redial suppression parameter is set by default.
set redial_suppression
Redial suppression : disable
96. The No escape character function parameter is set by default.
set no_escape_uri
No escape character function : disable
97. The PCM codec select pattern parameter is set by default.
set pcm_codec
PCM codec select pattern : SDP
6.3.6 Routing Config Data
1. Set SIP Route:
2. The command used to set SIP Route is “set sip_route”.
3. Set Route type: 2(Static).
4. Set Select command: 1(Set).
5. Set Gateway type:2(Default Gateway)
6. Set Gateway IP address: 10.64.2.249 is set for this example. This IP Address is the E-SBC LAN
IP Address. Please use the actual IP address of E-SBC LAN for your network. The E-SBC IP
address may/will be different from this example.
MG(SIP) > set sip_route
Route type : Static
Gateway Address | Destination network/Prefix
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0. 0. 0. 0 | 0. 0. 0. 0 / 0
Route type (0:Exit/1:None/2:Static) : 2
Select command (0:Exit/1:Set/2:Delete) : 1
Select gateway type (0:Exit/1:Gateway/2:Default gateway) : 2
Gateway IP address (0.0.0.0) : 10.64.2.249
Route type : Static
Gateway address | Destination network/Prefix
10. 64. 2.249 | 0. 0. 0. 0 / 0
Are you sure ? Y/N = Y
6.3.7 International Call
The configuration steps for the international call by setting header configuration.
1. The command used to set header configuration is “set header_config”.
2. Set Select user of contact header: 1: Keynumber
3. Set Delete 100rel from Supported-header after connected: 0:disable.
MG(SIP) > set header_config
Select user of contact header
(0:Not Configration/1:Keynumber/2:Random(default=0)) : 1
Delete 100rel from Supported-header after connected.
(0:disable/1:enable(default=0)) : 0
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