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Looking for any advice here, a bit wet behind the ears on this one...

Looking at a large site with about 4500 Shoretel users spread out across several separate systems, and a couple of dozen legacy PBX sites that have a variety of systems.

Plan is to slowly implement Cisco at the legacy sites and have them be able to 5 digit dial the main Shoretel site.

My first thought is SIP trunks between sites (MPLS is in place already).

Any other (easier) ways of doing this?

Where could I gen some info on the Shoretel side of this setup?

The entire site will eventually be Cisco, but the the complete takeover of the Shoretel could take 5-7 years.

D

Doghart, This is a tough one. I went through this a few months back. There are 2 ways to do this. My VAR tried to get me to pay for a trunk on both ends as this was the "easier" way to do it and recommended not doing the sip route as it was "hard". I was able to use SIP between our shoretel and cisco locations using SIP over MPLS with no issues. I currently do not have access to those systems or my documentation but, will late next week. PM me and I can send you some more info and some screen shots on how the SIP trunks were set up on each side. You will need SIP trunking licenses on the Shoretel side. Think about how many simultaneous calls you think they will connect. This will be the amount of licenses you will need. I would consider padding it with a few extra to be safe.

Thanks,

-Mike

PM sent, email me when you have the time.

Thanks!!!

D

Depending on the volume of calls, and the location and availability of equipment, QSIG PRI's may be an easier way to go.

Justin

OK, I found the following information, ad so far it seems to be working when dialing from the Cisco to the Shoretel.

But, when I dial from the Shoretel to the Cisco, the call will ring, dislpay user name/ext info, but drop when answered.

Logs show that the Shoretel is sending a "BYE" when the call is answered... I am on 12.3 currently, will be moving to 14.2 once (if ever) it's stable.

:::::::::::::::::::::::::::::::::::::::::::::::::::

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 set 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.
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.
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.
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.
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!
-------------------------------------------------------------------------------------------------------------------
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

Save Your WORK!!!


----------------------------------------------------------------------------------------------------------------------
Create a Sip Trunk
1. Navigate to Device>Trunk
3. Click on Add New
2. Select SIP Trunk for Trunk Type and SIP for Device Protocol (should auto-select). Trunk Service type can be
left as None (Default)

Device Information
1. Device Name: Give it a name like ShoreTel_SIP
2. Description: Give it a description like SIP_Trunk
3. Device Pool can be left to Default
4. Common Device Configuration: None
5. Call Classification: OnNet
6. Media Resource Group: None
7. Location: None
8. AAR Group: None
9. Tunneled Protocol: None
10. QSIG Variant and ASN.1 ROSE OID Encoding should be grayed out. If not, select No Changes.
11. Packet Capture Mode: None
12. Packet Capture Duration: 0
13. Media Termination Point Required: CHECKED
14. Retry Video Call as Audio: No CHECK
15. Path Replacement Support: NO CHECK
16. Transmit UTF-8 for Calling Party Name: NO CHECK
17. Transmit UTF-8 Names in QSIG APDU: NO CHECK
18. Unattended Port: NO CHECK
19. SRTP Allowed: NO CHECK
20. Route Class Signaling Enabled: Default (off)
21. Use Trusted Relay Point: Default (off)
22. PSTN Access: No CHECK
23. Run On All Active Unified CM Nodes: NO CHECK

Intercompany Media Engine (IME)
1. E.164 Transformation Profile < None >

MLPP and Confidential Access Level Information
1. MLPP Domain < None >
2. Confidential Access Mode < None >
3. Confidential Access Level < None >

Call Routing Information
1. Remote-Party-Id: No CHECK
2. Asserted-Identity: NO CHECK
3. Asserted-Type Default
4. SIP Privacy Default

Inbound Calls
1. Significant Digits: All
2. Connected Line ID Presentation: Default
3. Connected Name Presentation: Default
4. Calling Search Space: Select your desired CSS
5. AAR Calling Search Space: NONE
6. Prefix DN: NONE
7. Redirecting Diversion Header Delivery - Inbound: NO CHECK
8. Incoming Calling Party Settings and Icoming Called Party Settings: Leave as is (Default)
9. Connected Party Transformation CSS: NONE
10. Use Device Pool Connected Party Transformation CSS: CHECK

Outbound Calls
1. Called Party Transformation CSS: NONE
2. Use Device Pool Called Party Transformation CSS: CHECK
3. Calling Party Transformation CSS: NONE
4. Use Device Pool Calling Party Transformation CSS: CHECK
5. Calling Party Selection: Originator
6. Calling Line ID Presentation: Default
7. Calling and Connected Party Info Format: Deliver DN only in connected party
8. Redirecting Diversion Header Delivery - Outbound: No CHECK
9. Redirecting Party Transformation CSS: None
10. Use Device Pool Redirecting Party Transformation CSS: CHECK

Caller Information
1. Caller ID DN: Leave Blank
2. Caller Name: Leave Blank
3. Maintain Original Caller ID DN and Caller Name in IDentity Headers: NO CHECK

SIP Information
1. Destination Address in an SRV: NO CHECK
2. 1* Destination Address: Your ShoreTel Switch IP Address, Destination Address IPv6: Leave Blank, Destination Port: 5060
3. MTP Preferred Originating Codec: 711ulaw
4. BLF Presence Group: Standard Presence Group
5. SIP Security Profile: CHoose the ShoreTel SIP security profile you created
6. Rerouting CSS: NONE
7. Out-of-Dialog Refer CSS: NONE
8. SUBSCRIBE CSS: None
9. SIP Profile: Choose the ShoreTel SIP Profile you created
10. DTMF Signaling Method: No Preference
11. Normalization Script: NONE
12. Enable Trace: NO CHECK, Leave Parameters blank
13. Recording Information: None

Geolocation Configuration
1. Geolocation: None
2. Geolocation Filter: NOne
3. Send Geolocation Information: NO Check

I probably can't offer too much help, but I would like to mention that Shoretel made a lot of improvements to SIP trunks starting with version 13.1.

Ver 14.2 had some pretty rough builds with some major issues for a couple of customers. It has been out for a while, so most of the major problems should be fixed by now.

I know that an upgrade of that size is a pretty big task, and I can't promise that the upgrade will fix your problem.

I have all of our stand-alone sites (about 30) upgraded from 12.2 to 14.2 build 19.44.7900. That's what I plan on rolling out here at the main site, which is at 12.3 currently.

Coordinating the upgrade for the main site however is going to take some major PM skills... haha. I have about 4500 users across the entire state and would want a person available at each site for testing when we push the upgrade out.
SG appliances have been known to get stuck on the reboot and need a power cycle at times.

Still hashing this one out, but I think we're close.

Will update once we reach a solution, thanks!


EDIT: for FYI we have licensing for 30 SIP trunk connections in the ShoreTel for the setup between the ShoreTel & Cisco UCM.

D

UPDATE:
Upgraded the Shoretel from 12.3 to 14.2 and SIP connection between the the Shoretel & Cisco is now working flawlessly using the settings listed above.

D

Thanks for sharing. I had a feeling that 14.2 would work better then 12.X with SIP trunks.

I just tied our Shoretel to Toshiba in a lab using SIP trunks and got it working pretty easily. A few years ago I tried to do the same thing and failed. Either I did not know as much back then about SIP trunks or the Shoretel and Toshiba software improved with SIP trunking. I think it may be a combination of both.