MiVoice Office 250 not dialling out via SIP Trunks -

[mattybrownuk]

We're in the UK, currently using ISDN30 to make and receive calls on our MiVoice Office 250.

We're attempting to switch a limited number of phones over to making their outgoing calls via 5 SIP Trunks so that we can trial making and receiving calls via SIP before increasing that to 20 SIP trunks, moving all outgoing calls to SIP Trunks and eventually porting over all of our incoming numbers and disconnecting the ISDN30.

We have:

• Subscribed to a BT WHC (Wholesale Hosted Communications) provider (5 SIP trunks + 10 temporary DDI's)
• Added 5 SIP Trunk licenses to our PBX
• Had the company that looks after our Mitel PBX configure it for us using the SIP Device Configuration Guide, May 2019

And now:

• We can make incoming calls to the first number in the block of 10 DDI's
• We cannot receive any incoming calls to the other 9 numbers in that block
• We cannot make any outgoing calls

When attempting to make an outgoing call, I can see from the packet capture on our firewall that we're getting a 404 User not Found response, but I don't know whether that means the number dialled should have started with the country code 44 instead of 0 (and how to fix that, if that's the problem) or whether this means there's a configuration problem at BT's end, or some other reason entirely why this isn't working.

I also have no clue as to why our system is only receiving calls for the first trunk number and not the other 9... they're not programmed anywhere in our PBX other than the Call Routing Table - I presumed that when our system registers for the first trunk number, that automatically routes all the numbers on our account to us... is that not the case?

Hope someone can make sense of this?  :

INVITE sip:07123456789@62.7.201.169:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.144.2:5060;branch=z9hG4bK2392623975-949
Route: <sip:62.7.201.169:5060;lr>
Max-Forwards: 70
Allow: NOTIFY,REGISTER,REFER,SUBSCRIBE,INFO,INVITE,ACK,OPTIONS,CANCEL,BYE
User-Agent: Mitel-5000-ICP-6.3.7.99
P-Asserted-Identity: "Our Company Name" <sip:441912345789@62.7.201.169>
From: "Our Company Name" <sip:441912345789@62.7.201.169:5060>;tag=Mitel-5000_2392624714-949
To: 07123456789 <sip:07123456789@62.7.201.169:5060>
Call-ID: 2392620560-949
CSeq: 1 INVITE
Contact: "Our Company Name" <sip:441912345789@192.168.144.2:5060>
Content-Type: application/sdp
Content-Length: 297

v=0
o=Mitel-5000-ICP 1091902712 1604593585 IN IP4 192.168.144.2
s=SIP Call
c=IN IP4 192.168.144.2
t=0 0
m=audio 6014 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:30
a=sqn:0
a=cdsc:1 image udptl t38
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.144.2:5060;branch=z9hG4bK2392623975-949
From: "Our Company Name" <sip:441912345789@62.7.201.169:5060>;tag=Mitel-5000_2392624714-949
To: 07123456789 <sip:07123456789@62.7.201.169:5060>
Call-ID: 2392620560-949
CSeq: 1 INVITE
Content-Length: 0

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.144.2:5060;branch=z9hG4bK2392623975-949
To: 07123456789 <sip:07123456789@62.7.201.169>;tag=3813582385-232866451
From: "Our Company Name" <sip:441912345789@62.7.201.169>;tag=Mitel-5000_2392624714-949
Call-ID: 2392620560-949
CSeq: 1 INVITE
Allow: PUBLISH,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:07123456789@62.7.201.169:5060>
Reason: Q.850;cause=1;text="User not Found"
Content-Length: 0

ACK sip:07123456789@62.7.201.169:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.144.2:5060;branch=z9hG4bK2392623975-949
Route: <sip:62.7.201.169:5060;lr>
Max-Forwards: 70
From: "Our Company Name" <sip:441912345789@62.7.201.169:5060>;tag=Mitel-5000_2392624714-949
To: 07123456789 <sip:07123456789@62.7.201.169:5060>;tag=3813582385-232866451
Call-ID: 2392620560-949
CSeq: 1 ACK
Contact: "Our Company Name" <sip:441912345789@192.168.144.2:5060>
Content-Length: 0

[Tech Electronics]

mattybrownuk,

It looks as though the user you are acting as is not a valid user.

If I had to guess the phone number being used is not one provided by the SIP Carrier, but one provided by your ISDN30 Carrier; 441912345789.

I am not familiar with how the Numbering Plan works in the UK, but is this a valid phone number; 07123456789? The other issue you may is if they require E.164 as you are not sending calls out using that.

Thanks,

TE

[mattybrownuk]

Hi TE, thanks for getting back to me.

mattybrownuk,

It looks as though the user you are acting as is not a valid user.

Could be - I actually don't have any "employees" created in the Hosted Voice portal.  Since these are trunks for use by any extensions that attempt to make a phone call, they're not specific to any one employee, so I don't even understand why I have that option in their portal.

If I had to guess the phone number being used is not one provided by the SIP Carrier, but one provided by your ISDN30 Carrier; 441912345789.

The real phone number in this capture is one from the SIP carrier, not the ISDN30 - they gave us 10 temporary DDI numbers to test with.  I've swapped out the number in this packet capture for a fake number formatted like a real one, so there's no identifiable info.

I am not familiar with how the Numbering Plan works in the UK, but is this a valid phone number; 07123456789? The other issue you may is if they require E.164 as you are not sending calls out using that.

Yes, numbers with the format 07xxxxxxxxx are valid UK mobile numbers.  I'm wondering whether the SIP provider is expecting the leading 0 to be 44 (UK country code), like 447xxxxxxxxx.  I've no idea where to change that to try it.
The configuration guide says "Use ITU-T E.164 Phone Number: If set to Yes, the MiVo250 handles ITU-T E.164 formatted phone numbers as part of the incoming SIP INVITE messages from the SIP peer. Set this to NO for WHC.", so no, they don't require E.164.

[Tech Electronics]

mattybrownuk,

Alright, since the number you are presenting in the FROM portion of your SIP Message is one of the test DDIs then that shouldn't be an issue.

Are you sure that they are allowing outbound calls to Out of Network phone numbers? Meaning do they allow calls going to another Carrier? Try calling a number that is In-Network and see if you have the same issue.

Also, you may also want to make sure that the Firewall/Router is not performing some sort of Transformation on your SIP messages.

The other thing I would try, like you stated, is making sure the 44 is in front of the dialed numbered. I noticed that you didn't put the 0 when you showed it with the 44 in front of it so my guess is that a leading 0 is only allowed when the country code isn't on there?

If you want to test that out then just setup a new Facility Group that removes the 07 and puts in the 447 into the dialed digits. Then use that Facility Group in a new Route Group that looks for your test number to be dialed out. For testing purposes you can set this to Route 1 since it should only work for a specific number being dialed. Don't leave this new Route Group at 1 though when you are done testing as that is the route that is used for Emergency calls. If you know which Route Group it would normally take then put it right before that one instead of Route 1.

Thanks,

TE

[mattybrownuk]

Hi TE,

Thanks for your help.  Much appreciated.

So today, the supplier has "made some changes on the SIP Trunk on the platform" and now all the incoming DDI numbers are working!!

Still can't make outgoing calls, but we're one step closer.

Are you sure that they are allowing outbound calls to Out of Network phone numbers? Meaning do they allow calls going to another Carrier? Try calling a number that is In-Network and see if you have the same issue.

Outbound calls certainly should be allowed... I can only assume that they've configured the service correctly their end!  Now I know all ten DDI's are working, I tried calling outbound to one of the other DDI numbers, but that fails the came as calling my mobile number.

Also, you may also want to make sure that the Firewall/Router is not performing some sort of Transformation on your SIP messages.

Our firewall is a FortiGate 100E and I just noticed that the VoIP feature is disabled... maybe the issue is that the firewall should be swapping out our private IP address with the public IP address, but the whole feature in general is disabled?  It's a managed firewall... managed by the same company that's providing the SIP Trunks and I don't have the rights needed to enable the feature myself.  I'll mention it to them tomorrow.

The other thing I would try, like you stated, is making sure the 44 is in front of the dialed numbered. I noticed that you didn't put the 0 when you showed it with the 44 in front of it so my guess is that a leading 0 is only allowed when the country code isn't on there?

If you want to test that out then just setup a new Facility Group that removes the 07 and puts in the 447 into the dialed digits. Then use that Facility Group in a new Route Group that looks for your test number to be dialed out. For testing purposes you can set this to Route 1 since it should only work for a specific number being dialed. Don't leave this new Route Group at 1 though when you are done testing as that is the route that is used for Emergency calls. If you know which Route Group it would normally take then put it right before that one instead of Route 1.

I don't understand how Facility/Route Groups work and find a good explanation online, so I'm not going to touch it!  If the SIP Trunk provider tells me the issue is due to the format of the number being dialled, I'll ask the phone system maintainer to get involved again.

Thanks again for your input! 

[Tech Electronics]

Mattybrownuk,

Well since you seem to be a non-certified Mitel technician it would probably be best to contact the vendor who works on your equipment and have them work with your SIP provider to solve the problem.

I have run into many issues with the documents from the SIP Center of Excellence not being accurate.

If you don’t want to do that you can ask the SIP provider what they find wrong with your Outbound Calls and we can help with that.

Thanks,

TE


Sent from my iPhone using Tapatalk

[mattybrownuk]

SIP Trunk Groups\82002\Configuration\IP Address had been set to the provider's IP address.  Once this had been changed back to the default value, 255.255.255.255 INVITE's started to succeed.

Our SIP Trunk provider's instructions said "IP Address: Configure the IP address.", which should have probably read, "IP Address: Leave as default, 255.255.255.255".  I went through the configuration guide over and over before realising that the INVITE messages had the SIP Trunk provider's IP address listed, but not our FQDN.

I now have incoming and outgoing calls working.  Thanks for your help, TE.