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SIP - ShoreTel CISCO SIP Trunk Configuration

Solution

Shoretel Side:

Create SIP ports on your Switch (assuming you already added a switch to director)

1. Log into ShoreTel Director
2. Navigate to Administration > Platform Hardware... > Voice Switches/Service Appliances > Primary
3. Select the switch you want configured and use the drop down selections to choose "5 SIP Trunks". Choose as many ports
as you want configured or are licensed for.
4. Save your work.

-------------------------------------

Create a SIP Profile:

1. Log into ShoreTel Director
2. Naigate to Administration > Trunks > SIP Profiles
3. Select New, then name your SIP profile
4. Type in .* in the User Agent field.
5. Priority should be se to 100
6. Check the "Enable" box.
7. Add the following custom Parameters:

acceptMWI=notify
Accept302=sip
HoldSupport=1
AddrSupport=diversion
EnableSymmetricDtmf=1
UseSipProxyOut=1
OAEMedialessPort=8600
AllowedCodecs=PCMU/8000
OptionsPing=0
EnableP-AssertedIdentity=1

8. Save your work.

----------------------------------

Create a SIP Trunk Group

1. Log into ShoreTel Director
2. Navigate to Administration > Trunks > Trunk Groups
3. Add new trunk group at your site, select SIP as type and hit Go.
4. Name your Trunk Group.
5. Select your Language.
6. Uncheck "Enable SIP info for G.711 DTMF Signaling" if checked.
7. Select the SIP profile created earlier.
8. Set the Digest Authentication to "None". Leave the Username/Password field blank.

Inbound Section:

9. Number of Digits from CO: should be the number of digits you are using on your ShoreTel extensions.
10.Uncheck "DNIS" and "DID" if selected.
11.Check the "Extension" box.
12.Translations table: If you are using the same amount of digits on Cisco extension as you are with ShoreTel extension then
you will NOT need a translations table. Otherwise, you will need to translate those digits accordingly.(let me know if you need
assistance with this and I will be happy to help)

Outbound Section:

13. Check the "Outbound" section.
14. Access code can be configured with "9", however you may not want to give access to the PSTN via the user group. You can also
use another number as an access code if you prefer to go that route. (let me know if you need
assistance with this and I will be happy to help)
15. Uncheck all the boxes under Trunk Services except the Caller ID not blocked by default box. That box should be
the only one checked.

Trunk Manipulation Section:

16. Uncheck all boxes.
17. Create off system extensions that will be used to reach the Cisco extensions.If you are going to use and access code,
then you may skip this step.(let me know if you need assistance with this and I will be happy to help)
18. Translations table: If you are using the same amount of digits on Cisco extensions as you are with ShoreTel extensions then
you will NOT need a translations table. Otherwise, you will need to translate those digits accordingly. (let me know if you need
assistance with this and I will be happy to help)
19. Save your work. (select cancel on the notification so that this trunk group doesnt get added to all user groups)


-------------------------

Create SIP trunks

1. Log into ShoreTel Director
2. Navigate to Administration > Trunks > Individual Trunks
3. Select your site, then the trunk group you just created. Hit Go.
4. Name the trunk.
5. Select the Switch you configured for SIP ports
6. Type in the IP address of the Cisco Publisher.
7. Add as many trunks as needed/configured/licensed
8. Save your work.

-------------------------

Once created make sure your extension has access to that trunk group via user groups for testing purposes. You can create a new
user group to include solely your sip trunks if preferred.


CISCO Side:

Create a SIP Security Profile
1. Navigate to System>Security>SIP Trunk Security Profile
2. Click on Add New
3. Name: Give it a name like ShoreTel SIP Security Profile
4. Description: Give it a description like SIP connection to ShoreTel
5. Device Security Mode: Non Secure
6. Incoming Transport Type: TCP+UDP
7. Outgoing Transport Type: UDP
8. Enable Digest Authentication: NO CHECK
9. Nonce Validity Timer (mins): Should be grayed out
10. X.509 Subject Name: Leave Blank
11. Incoming Port: 5060
12. Enable Application level authorization: No CHECK
13. Accept presence subscription: CHECK
14. Accept out-of-dialog refer: CHECK
15. Accept unsolicited notification: CHECK
16. Accept replaces header: CHECK
17. Transmit security status: NO CHECK
18. Allow charging header: NO CHECK
19. SIP V.150 Outbound SDP Offer Filtering: Use Default Filter
20. Save your work!
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Create a SIP Device Profile

SIP Profile Information
1. Navigate to Device>Device Settings>SIP Profile
2. Click Add New
3. Name: Give it a name like ShoreTel SIP Profile
4. Description: Give it a description like ShoreTel SIP Profile
5. Default MTP Telephony Event Payload Type: 101
6. Early Offer for G.Clear Calls: Disabled
7. User-Agent and Server header information: Send Unified CM Version Information as User-Agent Header
8. Version in User Agent and Server Header: Major and Minor
9. Dial String Interpretation: Phone number Consist of characters 0-9,*,#, and + (others treated as URI Addresses)
10. Confidential Access Level Headers: Disabled
11. Redirect by Application: NO CHECK
12. Disable Early Media on 180: CHECK
13. Outgoing T.38 INVITE include audio mline: NO CHECK
14. Use Fully Qualified Domain Name in SIP Requests: NO CHECK
15. Assured Services SIP conformance: No CHECK

SDP Information:
1. SDP Session-level Bandwidth Modifier for Early Offer and Re-invites: TIAS and AS
2. SDP Transparency Profile: Pass all unknown SDP attributes
3. Accept Audio Codec Preferences in Received Offer: Default
4. Require SDP Inactive Exchange for Mid-Call Media Change: NO CHECK

Parameters Used in Phone
1. Leave all Parameters at their Default (no changes needed in this section)

Normalization Script
1. Normalization Script: NONE
2. Enable Trace: NO CHECK
3. Leave Parameter fields blank

Incoming Requests FROM URI Settings
1. Caller ID DN: Leave Blank
2. Caller Name: Leave Blank

Trunk Specific Configuration
1. Reroute Incoming Request to new Trunk based on: NEVER
2. RSVP Over SIP: Local RSVP
3. Resource Priority Namespace List: NONE
4. Fall back to local RSVP: NO CHECK
5. SIP Rel1XX Options: Disabled
6. Video Call Traffic Class: Mixed
7. Calling Line Identification Presentation: Default
8. Session Refresh Method: Invite
9. Enable ANAT: NO CHECK
10. Deliver Conference Bridge Identifier: NO CHECK
11. Early Offer support for voice and video calls (insert MTP if needed): CHECK
12. Allow Passthrough of Configured Line Device Caller Information: NO CHECK
13. Reject Anonymous Incoming Calls: NO CHECK
14. Reject Anonymous Outgoing Calls: NO CHECK
15. Send ILS Learned Destination Route String: NO CHECK

SIP Options PING
1. Enable OPTIONS Ping to monitor destination status for Trunks with Service Type "None (Default)": NO CHECK
2. All other selections in this area should be grayed out.

SDP Information
1. Send send-receive SDP in mid-call INVITE: NO CHECK
2. Allow Presentation Sharing using BFCP: NO CHECK
3. Allow iX Application Media: NO CHECK
4. Allow multiple codecs in answer SDP: NO CHECK

 
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Article details
Article ID: 38
Category: SIP Stuff in General
Rating (Votes): Article rated 3.0/5.0 (16)

 
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