Mitel and Skype For Business

[BobbyMitel]

All - I am posting on behalf of a coworker.  He is setting up Skype for Business with Mitel and having issues.  He brought Microsoft into it to work with the PBX vendor.  MS is saying that some ports are being blocked by the PBX.  When he sent the request to the vendor, their answer is that the PBX doesn't drop/block any ports.  Below is the exchange.... Can anyone clear this up for him?

Microsoft's email that was fwded to the vendor along with my coworkers email:

.....just know if you had a range of let’s say 5070 – 5090, then you are telling your telecom person that you can send TCP/SIP and Media through that particular range; but on this particular trunk you are going to send SIP and Media through a particular port that you specified on the “Listening port for IP/PSTN gateway” section of the trunk configuration.
 

The Mediation server can handle multiple connections over the SIP trunk that are created to the IP/PSTN gateway.  In this example, we are saying that the PBX is listening on port TCP/5068 and the Mediation server is listening on port TCP/5068.  Besides these ports the Mediation server uses ports 60,000 – 65,536 (UDP) for the audio traffic as well. Make sure to open firewall ports 60,000 – 65,536 between the Mediation server and the PBX.

Coworker:   I’ve been working with Microsoft support.  When we did a pcap from the mediation server we got the following results.  We need to have ports 50000 to 65535 tcp and udp opened on the pbx to allow media traffic.

PBX Response:

No ports are blocked on the PBX. We are not using the mediation server,.

Any help is appreciated!

[johnp]

I set this up for a customer and had no issues from what I remember. This was a sip trunk between the 2, no mediation (whatever that is) server was involved. I know there may be some sip uri  translation form settings that may be needed, but not for what I did

[VinceWhirlwind]

I setup a SIP trunk between a Mitel controller and a Skype server.
The only thing that had me stumped for a bit was on the Trunk Attributes form under Dial In Trunks Incoming Digit Modification - Absorb I forgot to put in a 0. Once I got that right, it worked fine.
 
If you can do a packet capture of the unsuccessful call, then in Wireshark, click on Telephony, Voip calls, highlight the call (could be on multiple lines, highlight them all) and then click on the "Flow Sequence" button.
 
This should help you figure out which bit is missing.
 
Just in case it helps, here is my Trunk Attributes:
Trunk Service Number   9
Release Link Trunk   No
Call Recognition Service   Off
Direct Inward Dialing Service   Off
Class of Service   55
Class of Restriction   5
Baud Rate   9600
Intercept Number   1
Non-dial In Trunks Answer Point - Day   
Non-dial In Trunks Answer Point - Night 1   
Non-dial In Trunks Answer Point - Night 2   
Dial In Trunks Incoming Digit Modification - Absorb   0
Dial In Trunks Incoming Digit Modification - Insert   
Dial In Trunks Answer Point   
Dial In Trunks Insert Forwarding Information   No

[VinceWhirlwind]

Just looking through my SIP Peer Profile.
Registration User Name   <Controller_Name>
Address Type   IP Address: <Controller_IP_Address>
Interconnect Restriction   1
Maximum Simultaneous Calls   50
Minimum Reserved Call Licenses   50
Outbound Proxy Server   
SMDR Tag   0
Trunk Service   9
Zone   19
User Name   
Password   
Confirm Password   
Authentication Option for Incoming Calls   No Authentication
Subscription User Name   
Subscription Password   
Subscription Confirm Password   
Alternate Destination Domain Enabled   No
Alternate Destination Domain FQDN or IP Address   
Enable Special Re-invite Collision Handling   No
Only Allow Outgoing Calls   No
Private SIP Trunk   No
Reject Incoming Anonymous Calls   No
Route Call Using P-Called-Party-ID (if present)   Yes
Route Call Using To Header   No
Default CPN   
Default CPN Name   
CPN Restriction   No
Public Calling Party Number Passthrough   No
Strip PNI   No
Use Diverting Party Number as Calling Party Number   No
Use Original Calling Party Number If Available   No
Allow Peer To Use Multiple Active M-Lines   Yes
Allow Using UPDATE For Early Media Renegotiation   Yes
Avoid Signaling Hold to the Peer   No
AVP Only Peer   Yes
Enable Mitel Proprietary SDP   No
Force sending SDP in initial Invite message   Yes
Force sending SDP in initial Invite - Early Answer   No
Ignore SDP Answers in Provisional Responses   No
Limit to one Offer/Answer per INVITE   Yes
NAT Keepalive   Yes
Prevent the Use of IP Address 0.0.0.0 in SDP Messages   Yes
Renegotiate SDP To Enforce Symmetric Codec   Yes
Repeat SDP Answer If Duplicate Offer Is Received   No
Restrict Audio Codec   G.711u/G.711a
RTP Packetization Rate Override   Yes
RTP Packetization Rate   60ms
Special handling of Offers in 2XX responses (INVITE)   No
Suppress Use of SDP Inactive Media Streams   No
Trunk Group Label   
Allow Display Update   Yes
Build Contact Using Request URI Address   No
De-register Using Contact Address not *   No
Disable Reliable Provisional Responses   No
Disable Use of User-Agent and Server Headers   No
Domain for Trunk Context   
E.164: Enable sending '+'   Yes
E.164: Add '+' if digit length > N digits   0
E.164: Do not add '+' to Emergency Called Party   No
E.164: Do not add '+' to Called Party   No
Force Max-Forward: 70 on Outgoing Calls   No
If TLS use 'sips:' Scheme   No
Ignore Incoming Loose Routing Indication   No
Include Diversion Header for EHDU   No
Multilingual Name Display   No
Only use SDP to decide 180 or 183   No
Prefer From Header for Caller ID   No
Require Reliable Provisional Responses on Outgoing Calls   No
Signal Privacy (if enabled) on Emergency Calls   No
Suppress Redirection Headers   All
Use Fixed Retry Time for 491   No
Use Privacy: none   No
Use P-Asserted Identity Header   Yes
Use P-Asserted Identity for Billing   No
Use P-Call-Leg-ID Header   No
Use P-Early-Media Header   No
Use P-Preferred Identity Header   No
Use Restricted Character Set For Authentication   No
Use To Address in From Header on Outgoing Calls   No
Use user=phone   No
Use user=phone for Diversion Header   No
Keep-Alive (OPTIONS) Period   120
Registration Period   3600
Registration Period Refresh (%)   50
Registration Maximum Timeout   90
Session Timer   90
Session Timer: Local as Refresher   No
Subscription Period   3600
Subscription Period Minimum   300
Subscription Period Refresh (%)   80
Invite Ringing Response Timer   0
Allow Inc Subscriptions for Local Digit Monitoring   Yes
Allow Out Subscriptions for Remote Digit Monitoring   Yes
Force Out Subscriptions for Remote Digit Monitoring   No
Request Outbound Proxy to Handle Out Subscriptions   No
KPML Transport   UDP
KPML Port   5060

[BobbyMitel]

Thanks!!