SIP Trunks on Digital Sets? 4

[2dfx]

So I have a newly upgraded BCM50a that went from R1 to R6 to take advantage of SIP trunking. I have said trunk registered, and I can dial incoming and dial sets. I've scoured the forum but how the hell do I assign a SIP trunk as a button? Every time I do in BEM it says "Cannot assign SIP/PRI trunks to digital sets".

Dialing "9" doesn't seem to work even though I have it setup in the dial plan.

 

[curlycord]

You do not assign PRI or SIP trunks to sets.
Target Lines is what you can use but it's for outgoing only.

PRI/SIP is not Key setup like POT's lines.
For outbound you need to use Dest Codes and Routes.

Dest Codes point to Routes
Routes point to Trunks (Pool or Bloc) in your case Bloc

Go to FAQ's above then scroll down to the bottom for more info on SIP Trunk programming.

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Toronto, Canada

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[2dfx]

I am having issues with step 10. Configuration -> Resources -> IP Trunks -> SIP Trunking -> Public -> Routing Table.

When I press "add" BEM gives me this error:

Error in setting configuration data. BCM_FEPSRemoteGatewaySettingData.GwRouteType is a required field.

I'm not sure how to get past this. The SIP account is registered and is taking incoming calls.

 

[curlycord]

Looks like your on the old BEM, I sent you a newer version just now.

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Toronto, Canada

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[2dfx]

Upgraded to the new BEM version successfully and was able to complete step 10. I don't seem to be able to dial out still.

EDIT: Made a stupid and realized that the dialing plan was set for pool A and not pool BlocA. So I changed it, now when I dial 9 I get the dead air but when I enter a number, all I get now is reorder tone.

 

[Firebird Scrambler]

In your SIP account settings, I think you might need to absorb one digit. There is a box where you enter it in.

Firebird Scrambler

Nortel & Avaya Meridian 1 / Succession & BCM / Norstar Programmer

Website =

http://somertel.net

linkedin

 

[curlycord]

Go to FAQ's above then scroll down to the bottom for more info on SIP Trunk programming."

Did you follow those steps?

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http://www.curlycord.com

Toronto, Canada

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[2dfx]

The steps have been followed with the same result.


Step 1:

Step 2:

Step 3:

Step 4:

Step 5:

Step 6:

Step 7:

Step 8:

Step 9:

Step 10:

 

[curlycord]

Do you have credits with Call Centric to make an outbound call?

I think your issues might be in your User Accounts/Call Centric
It's not allowing the call because there is no CLID entered (correctly as they want it)
Click "Modify"

Try this setup:

CLID Override = 177XXXXXX
Display name =
PAI CLID Override = 177XXXXXX
PAI Display name Override =
Contact Override = 177XXXXXX

Everything else seems to look in order.

How settings can be different at times...
Only change one by one then test if you have other issues.
In some case's having different settings will still work.

The differences I have in other places...

Accounts/Basic/CallCentric
SIP Domain
Remote: callcentric.com
Local: callcentric.com
Registrar
Address: blank
Port: 0

Accounts/Advanced/CallCentric
These are also checked:
Enable local NAT compensation
Enable media relay
Enable Connect Identity

I also use STUN server
System/IP Subsystem/Public Network/
Modify, check Address Discovery
Stun address: stun.ekiga.net

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http://www.curlycord.com

Toronto, Canada

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[2dfx]

Alright. This will be long winded, but joyous.

I've been hard at work at trying to get this thing to work. After hitting a brick wall with this config I engaged CallCentric's support, sent them screenshots, and asked for a SIP trace to see what was going on. As you know, I was getting reorder tone when dialing 9. Here's what I changed:

-Removed IP address from 'Modify account' and changed to callcentric.com

No dice. Still the same result. Good incoming, no outgoing. The whole reason I used an IP first was because the BCM didn't like sip.callcentric.net. At least it was registering with the callcentric.com domain. This is where I asked them for a SIP trace.

They replied saying my FROM call setup looked like this:

“From: <sip:1777XXXXXXX@192.168.0.5>" which needs to be @callcentric.com. For some reason it was transmitting my local IP for the BCM in the call setup. So 1st change to make to the FAQ for CallCentric was Configuration --> Resources --> IP Trunks --> SIP Trunking --> Global Settings here:

The FAQ says "local domain = leave blank". This does not work with CallCentric and I would probably argue most other providers. I made the change to 'callcentric.com' and this progressed me along as my CallCentric dashboard was finally seeing that I was attempting to dial out. It saw the digits I dialed and everything! I was able to successfully dial their test number 17771234567, HOWEVER any other number I dialed sent me back to reorder tone. The odd thing was that when I dialed a cell phone, the call came in for a fraction of a second.

Asked for another SIP trace. They sent me back the following:

"Note that there seems to be an additional header:

x-nt-corr-id: 10ab65f0-c0a80005-13c4-55013-e91a5ad8-215443e9-e91a5ad8

Please look for any reference of this within your settings, and disable this header."

I started combing through the SIP Trunking settings trying to find ANYTHING that could relate to this. I felt so close...and I finally found the setting that saved the day.

Configuration --> Resources --> IP Trunks --> SIP Trunking --> Media Parameters

I looked at the tooltip for "Enable Voice Activity Detection" which says "If selected uses bandwidth conservation capabilities of some codecs. Coordinate this setting with the system at the other end of the trunk."

I said to myself "there's NO way there's another Nortel/Avaya system on CallCentric's end". And sure enough, after unchecking that box, I CAN PLACE CALLS!!!!!!!!!

2 amendments to that SIP FAQ I would think. And a tremendous thank you to all of you for the replies. I hope this helps someone else along the way.

My only remaining issue now is, I had to change my dialing plan DN length to 12 as any number not sent to them as 1416xxxxxxx is rejected. Would be nice if there was a way I could 10 digit dial here and have it sent as 11 digits.

 

[Firebird Scrambler]

A star from me for all of your efforts.

Have you tried increasing the number of received digits in the Public DN area as I had this problem for UK trunks where our numbers start with a "0".

It was originally set to 11 digits but I had to increase it to 14 or 15.

Firebird Scrambler

Nortel & Avaya Meridian 1 / Succession & BCM / Norstar Programmer

Website =

http://somertel.net

linkedin

 

[2dfx]

Thought I was out of the woods...not yet.

I said I had no issues with incoming. I was wrong. Incoming calls will automatically cut off after 30 seconds. When it rings in and I answer:

-I cannot hear the incoming party
-The incoming party hears me fine

Yet outgoing calls I place have absolutely no issues.

Ideas?

 

[exsmogger]

I don't have a SIP trunk running through a BCM50 anymore (now using Asterisk), but I had this same problem when I was using a BCM50.

As I recall, the solution was adjusting the registration expiry interval.

As you can see, the expiry period on a dummy account I set up is 3600. If yours is 1800, then try setting it to 3600 and see if that fixes it.

Brian Cox
Georgia Telephone

http://Georgia-Telephone.com

http://www.linkedin.com/in/briancox1952

https://brian-cox.square.site/

 

[2dfx]

Tried setting expiry to 3600 with no effect.

 

[2dfx]

Here's another oddity - I have 2 sets on this system - an i2004 and a T7208. Calling inbound to the T7208 I can get 2 way calling, inbound hears me, I hear inbound etc. But the i2004 set gives me the 1 way issue.

The call STILL cuts off after 30 seconds on the T7208. SIP/ALG is disabled in my router.

 

[curlycord]

"Enable Voice Activity Detection"
Some do not have this setting enabled, however mine is enabled.
This I believe helps audio issues and also act like Disconnect Supervision, I forget!

As for Global Setting/ Local Domain, that may effect other voip carrier settings.
Mine is blank as I have 3 different carriers/servers.
Are you saying this helped your dial out issues? "any other number I dialed sent me back to reorder tone"

For testing purposes, blank it out again and setup the STUN settings instead.

Dialing Plan / DN Length
Assuming your trying to have the BCM dial 1 for when using Call Centric would it work if you tried putting 1 in the external number in the route?

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[curlycord]

Again about Global Setting/ Local Domain

Instead of trying the STUN method, I see your Local Domain is blank under UsersAccounts/Basic/SIP Settings "Local Domain" put the entry there instead? and blank out the Global Setting/ Local Domain.

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http://www.curlycord.com

Toronto, Canada

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[2dfx]

Good news, all is well again. I had to change "ITSP association method" Configuration --> Resources --> IP Trunks --> SIP Trunking --> Public --> Accounts --> Advanced from the default of "From header domain match" to "R-URI Called Number Username match" and now all my incoming works perfectly. No dropped calls, can hear both parties. I think I'm just about to declare this adventure done. Here's screenshots and thoughts of my working settings:

The username MUST be inserted into CLID override or outgoing will not work.

Yes, if you are using more than 1 VoIP carrier, then this should be blank. For CallCentric specifically as per HERE the RFC2833 payload should be set to 101.

For me, if "Enable Voice Activity Detection" is checked then I cannot place outgoing calls. I have had no issues so far with disconnect supervision. No matter which way I order the codecs, the BCM seems to default to G.711-uLaw. I can remove the other codecs to force it to use G.729 but I have more than enough upload bandwidth to let the BCM choose G.711.

The magical ITSP setting that made incoming work perfectly.

Just need to figure out now how to add a leading 1 when dialing 10 digit. Right now, I need to dial 11-digit for everything even if it's a local call as that's how CallCentric interprets calls.

By the way, no STUN server is set also as per CallCentric's setup guide. They specifically want it disabled.

 

[curlycord]

You can have STUN setup (a global setting) for accounts that can, in Account Advanced/ uncheck Enable local NAT compensation for accounts that do not use STUN.

For the Global / Local Domain, you can use the account settings instead as mentioned above.
UsersAccounts/Basic/SIP Settings "Local Domain" put callcentric.com and blank out the Global Setting/ Local Domain.

Here is all my settings, and working:

STUN

Call Centric does not appear to allow the BCM to change (override) the CLID Name.

"leading 1 when dialing 10 digit"
Easy with POT's (Pools) by inserting 1 in the External # in Routes, but not with SIP (Bloc's), best effort is an External Autodial button programmed as 91.

FYI - Voip.MS does not require the 1

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http://www.curlycord.com

Toronto, Canada

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