E2t orRtc

[blackberry1980]

I have a question on sip trunking using an lx controller.
When opening ports on my firewall for nat, Do i need to open ports 5060 and 50000-50255 for my e2t address or rtc address or both ?

 

[Lundah]

Depends on the network layout and call type. It would be far easier just to open the ports to both addresses than to map out all the scenarios to make the decision.

 

[blackberry1980]

The reason why i ask this is that the 3300 engineering guidelines quote :

Note: The ports on the LX and MXe are associated with the E2T (voice gateway) IP address
rather than the RTC IP Address. Other platforms use the common RTC/E2T IP address.

Do I even need to associate the sip trunk providers media and signalling gateways with my RTC address or just E2T ?
Its kinda sounds like I dont need to associate the RTC but i am uncertain!

 

[IrwinMFletcher]

Only the sip port needs to be opened up for the RTC address and the voice ports are only requried for the E2T address. The exceptions to this rule are MXe-LX versions that use T38 (as the T38 streams are supplied by the RTC controller card) and the MXe server which uses both the RTC and E2T for voice calls.

 

[blackberry1980]

Does this mean that i need to open port 5060 egress/ingress to my rtc address and ports 50000 to 50255 egress/ingress to my e2t address ?

 

[Mitelmatt]

Yes i believe that is correct. As long as you have a controller that has a seperate e2t card. (some controllers for the lower end have a joint RTC and E2T)

 

[IrwinMFletcher]

Correct, as Matt indicated, as long as you have an E2T card, if not then open the 50000 range to the RTC address.

Use the Maint/Diag->Hardware Module->Compute cards to see how many cards you have if you're not sure.

 

[blackberry1980]

Strangely enough i have done exactly that but everytime i try to do an inbound call i get a software error message :

Session 0x8f4d000 NULL peer data pointer

The call doesnt even connect as i get an error as if im dialling a wrong number. If i try to dial out through the sip trunks, i get trunks busy.

any ideas ?

 

[Mitelmatt]

Inbound yeah? Check in the Trunk service assignment for that sip trunk and make sure your have a "0" in Absorb Digits field

Just out of interest, what software release are you on?

Do you know how to use wireshark? could take a trace and see what's going on?

 

[blackberry1980]

The inbound and outbound dont work at all. I have tried absorbing 0 and uri translation. Its on release 9. This is surely a firewall problem. Any pointers for firewalls ? I think null peer data pointer means that media gateway and signalling gateway are 'crossed' with the rtc and e2t ?
Will try getting a trace as well.

 

[Mitelmatt]

Try 50000 to 50555 on UDP ports

it changed in 8 ur3 slightly

 

[IrwinMFletcher]

NULL peer means that the 3300 doesn't know who it's talking to. It has nothing to do with the E2T, if it's a NAT problem with the E2T then the call would 'go through', you'd just hear no or one-way audio.

It's also most likely nothing to do with the NAT at all, but a programming error. It sounds like that when you go to make a call, say outgoing, the 3300 doesn't know who to send the offer to. What's in your Peer Profile Assignment form for the outgoing trunk?

 

[blackberry1980]

Policies
Trunk Service: 3
Interconnect Restriction: 1
Maximum Simultaneous Calls: 1
Session Timer: 90
Zone: 2
SMDR Tag: 0
NAT Keepalive: False
Enable Mitel Proprietary SDP: Yes
Use P-Asserted Identity Header: No
Use Restricted Character Set For Authentication: No
Disable Reliable Provisional Responses: Yes
Use Alternate Destination Domain: No
FQDN or IP Address:
Ignore Incoming Loose Routing Indication: No
Suppress Use of SDP Inactive Media Streams: No
Enable Special Re-invite Collision Handling: No
Enable sending '+' for E.164 numbers: No
Force sending SDP in initial Invite message: Yes
Use To Address in From Header on Outgoing Calls: No
Force Answer - send SDP in initial Invite: No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages: No
Use P-Preferred Identity Header: No
Route Call Using To Header: No
Private SIP Trunk: No
Public Calling Party Number Passthrough: No
Use Diverting Party Number as Calling Party Number: No
Build Contact Using Request URI Address: No
Renegotiate SDP To Enforce Symmetric Codec: No
Repeat SDP Answer If Duplicate Offer Is Received: No
Allow Peer To Use Multiple Active M-Lines: No


Authentication
User Name:
Password: *******
Confirm Password: *******

Authentication Option for Incoming Calls: No Authentication


The trunk providers is ip direct connect. The link is established but as i mentioned outgoing calls give you trunks busy, Incoming calls give number unobtainable. I have tried many changes in the peer profile. And everytime i put a call into the number, it generates the software log null peer/null data pointer. The trunk service is absorbing 0 and sip ddi is in uri translation to my extension.

This is why i think its a nat issue.

 

[IrwinMFletcher]

I'm not familiar with all of the settings, but if you think it's a NAT problem, shouldn't you have NAT Keepalive enabled?

Unfortunately, I don't have enough experience with this to help much further, but I still think it's an internal problem. My logic is that I don't think the 3300 would produce a log like that if it actually sent an Offer out, but didn't get a reply. Calls fail quietly in the system all the time.

One sure way to tell is to do a Wireshark capture at the output of the 3300. Make an outgoing call and make sure that you see the SIP Offer going out (and an Answer coming back). If you do, then keep following the trace through the various pieces of your network. If you don't see the Offer, then it's still some programming in the 3300 where the call control doesn't know where to send the request.

 

[TheMitelGuy]

Have you been able to make any successful calls at all? Even one?

What happens when you open the SIP Trunking Profile for that provider and SAVE it AGAIN (without any changes)... any differences?

 

[blackberry1980]

Well, we were able to make a call into the system with audio bothways working fine but, whenever sip calls werent being processed, the 3300 would have a critical sip alarm, but if a call is made, the alarm goes away until the end of the call, but since playing with some firewall settings, now there is no audio and no call that reaches the 3300. The firewall being used is a checkpoint firewall and im pretty sure its a NAT and rule problem here. Anybody have any knowledge of a checkpoint NGX R65 and what NAT and rule settings should be for SIP trunks ?

Hehe......this might be asking for too much!