sip trunking configuration guide for univerge...

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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.

Property of Cox Communications, Inc.Version 0.1

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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 LabsTM

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).

http://www.edgewaternetworks.com/


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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.

10.70.88.2

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

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.


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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 unit(0)

DRS IP address : 10.70.88.2

DRS port number (1024-65535) :

DRS unit(1)

DRS IP address :


DRS unit(2)

DRS IP address :


DRS unit(3)

DRS IP address :


DRS[0] IP address : 10.70.88.2 : 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

<< Current Setting >>

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

<< Result >>

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.


1 = enable

Input : 1

<< Result >>

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



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 ?


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.


1 = enable

Input : 1

<< Result >>

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.


1 = enable

Input : 1

Select the setting of G711 Fax renegotiation.


1 = enable

Input : 1

<< Result >>

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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[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.


sip trunking configuration guide for univerge...

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