thinktel sip trunks on ucp and emg80-p2 · thinktel sip trunks on ucp and emg80-p2 – december...

ThinkTel SIP Trunks on

UCP & eMG80-P2

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[email protected] Updated June 2015

Table of Contents

I. SIP-Trunks - Summary .................................................................... 2

II. Software and Network Environment ............................................... 2

Network Configuration ............................................................................................................. 2

Software Versions ................................................................................................................... 2

SIP Accounts .......................................................................................................................... 2

III. Setup Information ........................................................................... 3

eMG80-P2 .............................................................................................................................. 3

Carrier Information - IMPORTANT ............................................................................................ 3

Troubleshooting Tools ............................................................................................................. 3

LAN/WAN and Port Forwarding ................................................................................................ 4

Bandwidth and Capacity .......................................................................................................... 4

Additional Information ............................................................................................................. 5

IV. SIP Trunks on the eMG80 / UCP ...................................................... 6

Pre-Requisites for configuring the eMG80 / UCP ........................................................................ 6

Configuring the CO/IP Lines - 1 ............................................................................................... 7

Configuring the CO/IP Lines – 2 ............................................................................................... 8

Non-DID Trunks Lines ............................................................................................................. 9

Station CLI ............................................................................................................................. 9

Configuring SIP Account Parameters ...................................................................................... 10

Configuring SIP Parameters - 1 .............................................................................................. 11

Configuring SIP Parameters – 2 ............................................................................................. 12



Configuring Stations to make SIP Calls ................................................................................... 15

V. Appendix 1 ..................................................................................... 16

SIP Trunk DID Programming ................................................................................................. 16

Virtual Trace Settings: ........................................................................................................... 18

THINKTEL SIP TRUNKS ON UCP AND EMG80-P2 – DECEMBER 2013 2

SIP-Trunks - Summary

SIP Trunks are becoming more widespread however getting the configuration correct can sometimes be a

bit trickier than setting up a standard T1-PRI. With this document I’m hoping that it will enable you to get

the SIP services up and running quickly and easily.

Software and Network Environment

This document is prepared on an eMG80-P2 system. The SIP call platform that I am testing on is

ThinkTel.

Network Configuration The system is on 10.1.42.0 subnet which is connected to the internet via gateway of 10.1.42.1. Our DNS

settings are using our in-house DNS server 10.1.42.22. The CALL SERVER resides at tor.trk.tprm.ca

Software Versions The eMG80 / UCP is running Phase 2 software version T1.2.5, I am using only digital terminals

SIP Accounts The CALL SERVER is using account 4389683123 which has the following numbers:

4389683123 / 4389683124 / 6044491813 / 6044491815


Setup Information

We know what to expect when making a call from a standard T1-PRI service, OUTGOING CLI and

INCOMING CLI but we can also get additional features such as NAME PRESENTATION and CALL

PROGRESS indication. In this configuration we will achieve individual station CLI as well as incoming and

outgoing name presentation.

eMG80-P2 The eMG80 will be only using the default on-board VoIP channels with no additional hardware or

licenses. The flexible numbering plan is default as are all other parameters.

Carrier Information - IMPORTANT When setting up SIP accounts you need to make sure that you receive the following MANDATORY

information before commencing the configuration:

1. SIP Authorization Name

2. SIP User ID

3. SIP Password

4. SIP Proxy

5. Domain Name (most UCP relay the carrier e.g. tor.trk.tprm.ca)

6. Carrier DNS Server (This is depending if your SIP Proxy is resolvable publicly as some are not)

7. Preferred CODEC (G.711u , G.711a , G.729a)

There are also other factors involved such as REGISTER or PROVISION. Some carriers do one or the

other. In this instance we are using REGISTER which keeps a heartbeat between the UCP and the SIP

carrier. PROVISION implies that the carrier will ONLY send the SIP call to your IP address, there is no

monitoring between the UCP and Carrier. My personal preference is using the REGISTER method and this

is what we will be using in this example.

There is one other thing to point out when using the REGISTER method, some platforms vary in how the

RESISTER information needs to be sent. Some only require the REGISTRATION USER ID / AUTH

USERNAME / PASSWORD while some require REGISTRATION USER ID@domain / AUTH

USERNAME / PASSWORD. You sometimes need to experiment if you get a registration error.

Troubleshooting Tools Make sure that you have a laptop/pc with a copy of Wireshark running. You will need to speak to the

customers IT department in arranging to have PORT MIRRORING enabled on the ETHERNET switch or

if that is not available you can enable the SIP virtual trace on the CALL SERVER. This is not a detailed as

a packet capture but it will certainly assist you in trying to find the cause of your SIP problems. Another

useful tool in troubleshooting is a software program called X-Lite from Counterpath. It is a cross platform

SIP softphone client and aside from being free, it’s a very solid and reliable test tool to make sure that

you have indeed got the correct settings and that your SIP account is actually working.


LAN/WAN and Port Forwarding Depending on the customers network there may be varying levels of configuration required in order to

make SIP calls work successfully. In MOST scenarios you will simply have to put the EXTERNAL IP

address of the customers WAN into the CALL SERVER FIREWALL parameter and the NAT router

should take care of the rest. Most modern routers and firewalls have an ALG (Application Layer

Gateway) and are SIP aware which put simply, they recognize that it is a SIP VoIP call and allow it to

pass uninterrupted. In other situations it may be required to forward the specific ports for both the SIP

information (Port 5060) and the RTP traffic (Ports 10000 – 20000). Consult with the customers IT

department as they will be able to provide you with more detailed information.

Bandwidth and Capacity How many calls is the customer planning to put down their WAN link, make sure that they have both the

CO/IP capacity on the UCP side but also the WAN bandwidth? Each SIP CO call using G.711 will consume

90kbps BOTH WAYS so please ensure that they have the UPLOAD speed as well as the DOWNLOAD

capacity. Also please keep in mind that there is NO QOS implemented on the public internet so regardless

of them having a huge 100mbps WAN link if they are going via the public internet, there is ZERO

guarantee that it’s going to be ISDN quality.

[Intentionally Blank]

http://en.wikipedia.org/wiki/Application-level_gateway

http://en.wikipedia.org/wiki/Application-level_gateway


Additional Information



SIP Trunks on the eMG80 / UCP

Pre-Requisites for configuring the eMG80 / UCP Some assumptions are made in this guide, please make sure that you have all the correct relevant

documentation with you regarding your own SIP carrier.

i. The system has been properly initialized and all modules/terminals and gateways have been

upgraded to the latest software

ii. The CALL SERVER is connected to the LAN and has the correct WAN interface address

iii. The SIP account has been tested and confirmed active by the carrier

iv. The person configuring this system is qualified to work on the eMG80/UCP system



Configuring the CO/IP Lines - 1 PGM 140: Configure CO/IP lines 1-3 to be DID, ENBLOCK ON and set the VoIP Mode to SIP or SIP/RTP

Relay (SIP only will be fine for single site). It is also worth while adding a CO Name so that you can

remember which trunk is which. The arrows are pointing to parameters that you need to change.

PGM 142: Set the VoIP Trunk type to SIP or SIP/RTP-Packet Relay


Configuring the CO/IP Lines – 2 PGM 151: Configure the lines to use the correct COLP/CLIP tables, you will need to add the

corresponding COLP data into the system via PGM 201. This will ensure the correct CLI is sent out to

the receiving party. It is important that you ensure that the correct STATION CLI parameter is set, if

you are using a mixture of CO lines then this field is important as your call will possibly fail or send the

incorrect outgoing CLI to the trunk.

If you are not using a DID range then I recommend that you set the COLP/CLIP tables to Station CLI:


Non-DID Trunks Lines PGM 202 – If the service is NOT a DID range (as per the example I am using) then it may be easier to

simply use the MSN tables and route the calls that way. The MSN table will simply route the called

number to a reference point in the Flexible DID table (PGM 231). Make sure that you enter the full

range of VoIP trunks as well as the full number received by the eMG80.

Station CLI This is a section of the STATION CLI field that relates to your outgoing CLI. This parameter is found in

PGM113. Note that we are only using the last 4 digits of the CO trunk, the rest of the CLI is populated

by the COLP/CLIP table.

If you are NOT matching a DID range then you should enter the specific CLI for each station. If this is

not set then each phone will only send the PRIME CLI.


Configuring SIP Account Parameters PGM126: The first thing we have to do is configure our SIP account information into the UCP, take the

information provided by the SIP carrier and populate the fields accordingly UCPe in the example below.

As you can see in the example below we are using REGISTER and the REGISTRATION USER ID is in

the format of USERID@Domain.

Keep in mind the SIP User ID index number (this is SIP UID 1) in the UCP as well as the Authorized

Rep ID Table Index, these are important later on in the programming.

The SIP Trunks follow the same Flex DID Table as normal ISDN/CO trunks and they program exactly the

same. See Appendix 1

Non-DID Version, using the MSN Table first.


Configuring SIP Parameters - 1 PGM133: This is where you setup ThinkTel’s proxy settings. You will most likely be given a DNS name

e.g. tor.trk.tprm.ca however as stated earlier we will be using a direct IP address. The DOMAIN will also

be given to you in a DNS format but you can if required use the IP address. The two really important

parts to be aware of are:

A. Registration UID Range – This is the parameter we configured earlier with the

user/authentication details.

B. DTMF Type – Most carriers will work with RCF 2833 however this parameter may need to be

adjusted to suit.


Configuring SIP Parameters – 2 PGM133: Most of these parameters can and should be left at default and should only be adjusted one at

a time as to not cause confusion when making changes. The most important parameters to be adjusted

are the options in the ID Individuality table, these options dictate what account is used and what CLI is

sent to the carrier. If these settings are wrong there is a very high likelihood of the call failing or sending

the incorrect CLI information. It is very important that you have programmed the STATION CLI and

CLIP/COLP tables.



Configuring SIP Parameters – 3 PGM113: The rest of these parameters are to do with codec negotiation which is part of the SDP used in

negotiating with the carrier what codecs you can and cannot use. The important parts to note here are.

I. The SIP UID Fixed Table

II. SIP User ID Selection

III. Codec Priority

The SIP User ID relates back to PGM126 that we programmed earlier, this relates to the ID

Individuality information that we looked at in the previous page. You should not need to adjust any

other parameters.


Configuring SIP Parameters – 4 PGM113: Once the other options are done there is one last part to configure which is the trunk group

the SIP trunks are going to use. There is also an option to be aware of which is Ignore INBAND DTMF

which should you have problems with IVR’s or CCR then set that to YES and the system will only listen to

RFC2833 or SIP-INFO dial information.

When you are ready to go simply press the SAVE button making sure that you have chosen the SELECT

ALL button at the top of the page otherwise you will have to re-enter the data. When you are ready,

press the REGISTER button and then have a look at the SIP Trunk information STATUS page.


Configuring Stations to make SIP Calls PGM114: Once you have your trunks registered now all that is required is for the IP terminals to be able

to use the SIP account and send the correct CLI out to the trunk. In this example below we are

configuring station 5601 so that when it makes calls out via the SIP trunk it sends 438-968-3124 or 302

which also matches it’s DID number.

Non-DID Version:

PGM111~113: It is also important that each station can use the correct SIP account. This relates to the

SIP User ID table covered in PGM133 earlier. This parameter makes sure that the IP terminal

authenticates to the SIP proxy using the correct account.


Appendix 1

SIP Trunk DID Programming PGM233: As mentioned earlier, SIP trunks follow the same Flex DID Table programming as normal ISDN

lines as well as digit translation tables, here are screenshots taken from our example system.

Flex-DID Table:

Digit Translation Table:


CLIP/COLP Tables:


Virtual Trace Settings: This is set from the SYSTEM MANAGEMENT -> TRACE and may assist you in troubleshooting your SIP

trunk configuration. To access the trace information you need to enable trace information via TCP/IP in

PGM175.

Virtual Trace Settings:


Serial Port Selections:

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