Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: BobbyMitel on October 18, 2018, 07:20:45 PM
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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! ;D ;D
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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
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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
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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
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Thanks!!