SIP Trunk Channels 1

[EddieV3]

Hi All.

I have a major issue, and I'm not getting much assistance from Mitel, who have told me they are still investigating....

I have an MXe III with 300 SIP Trunk Licenses. MXe is running MCD 5.0 SP1 PR1. The proxy to the peer is via an MBG 7.1. I have 300 SIP Trunk Licenses available here too. MBG is in server gateway mode, connected on the WAN side via the provider's router.

The problem I have is that I can only make or receive 2 calls. The next call in or out receives busy tone. The SIP provider is adamant that they are currently providing 100 channels, so I've naturally set the maximum simultaneous calls on the SIP peer to 100. Below I've added a copy of my peer profile details. Any help would be so greatly appreciated.

SIP Peer Profile Label Sip
Network Element Sip

Local Account Information
Registration User Name xxxxxxxxx
Address Type IP Address: 192.168.100.10


Administration Options
Interconnect Restriction 1
Maximum Simultaneous Calls 100
Outbound Proxy Server MBG Main
SMDR Tag 0
Trunk Service 3
Zone 1


Authentication Options
User Name xxxxxxxxx
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 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

Allow Peer To Use Multiple Active M-Lines No
Allow Using UPDATE For Early Media Renegotiation No
Avoid Signaling Hold to the Peer Yes
Enable Mitel Proprietary SDP Yes
Force sending SDP in initial Invite message Yes
Force sending SDP in initial Invite - Early Answer No
Limit to one Offer/Answer per INVITE No
NAT Keepalive No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages No
Renegotiate SDP To Enforce Symmetric Codec No
Repeat SDP Answer If Duplicate Offer Is Received No
RTP Packetization Rate Override No
RTP Packetization Rate 20ms
Special handling of Offers in 2XX responses (INVITE) No
Suppress Use of SDP Inactive Media Streams No

Trunk Group Label SIP
Allow Display Update No
Build Contact Using Request URI Address No
De-register Using Contact Address not * No
Disable Reliable Provisional Responses Yes
Disable Use of User-Agent and Server Headers No
E.164: Enable sending '+' No
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
Ignore Incoming Loose Routing Indication No
Only use SDP to decide 180 or 183 No
Require Reliable Provisional Responses on Outgoing Calls No
Use Privacy: none No
Use P-Asserted Identity Header Yes
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

Keep-Alive (OPTIONS) Period 120
Registration Period 120
Registration Period Refresh (%) 50
Registration Maximum Timeout 90
Session Timer 90
Subscription Period 3600
Subscription Period Minimum 300
Subscription Period Refresh (%) 80
Invite Ringing Response Timer 0

Allow Inc Subscriptions for Local Digit Monitoring No
Allow Out Subscriptions for Remote Digit Monitoring No
Force Out Subscriptions for Remote Digit Monitoring No
Request Outbound Proxy to Handle Out Subscriptions No
KPML Transport default
KPML Port 0

 

[dryaquaman]

You may have to turn on SIP TCPDUMP, create the failure and then read the dump in Wireshark.
You can do this on both the 3300 and the MBG.

I suspect you'll see a "busy" coming from the carrier.

Dry Aquaman

http://www.university-music-on-hold.com

 

[LoopyLou]

Just a stupid question but in the 3300 licensing are the SIP trunk license applied to the 3300. Have seen where the licenses are shown but not applied to a particular system.

I'd tell you a UDP joke but I'm afraid you won't get it. TCP jokes are the best because you always get them.

 

[EddieV3]

Hi chaps.

Thanks for responding.

The TCPDUMP was very useful, proved it to the provider. "Oh yes, we can see that now, we'll activate all 100 channels asap." How do these people sleep at night.

And the license application question. Very valid, I did check my config, just in case. All licenses were applied.

Thanks

 

[LoopyLou]

its the sleep of the innocent or in their case the stupid

I'd tell you a UDP joke but I'm afraid you won't get it. TCP jokes are the best because you always get them.