SIP Trunk to Anveo Direct 3

[QueBall780]

So I'm doing a bit of testing to see what SIP services I can get to work on a BCM 6.0 system (with system update 21)

I had a bit of trouble figuring out what combination of settings were required to get 2 way audio working on an Anveo Direct SIP trunking setup with BCM version 6 (Have system update 21 which includes IPTEL 129 latest version of SIP software for BCM)

We are assuming you have appropriate license on your BCM to enable SIP trunking. Either the SIP Trunk or VOIP Gateway trunk(allows both SIP and h323 trunk types)

Settings that are working for me.
Anveodirect trunk settings
1. On Anveo Direct I configure Outbound Services/Call Termination with the Static Public IP for this site. Anveodirect does not support dynamic IP.
2. On Anveo Direct I setup my Inbound Destination SIP trunks with $[E164]$@198.51.100.154 so calls will be sent with the DID number in the user part of the SIP invite and I point my DID number to use this SIP trunk for inbound calls.

Port Forwarding on my Firewall
The RTP Port ranges are set to the default. You can check them in element manager under Configuration->Resources->Port Ranges RTP over UDP 30000-30999 has been the Port range my BCM has been sending for the inbound audio. You must forward this port to the BCM internal IP from any external source. Anveo direct does not proxy their audio so there is no list of IP address where inbound audio is coming from that you can restrict this to, you must accept all sources of RTP traffic coming to your accepted range.
Signalling traffic will only come from the Anveodirect servers so you should only forward inbound traffic to port 5060 on the BCM from those servers to avoid SIP scanners trying to send you fake or telemarketing calls and probing your system for weak passwords.
The IP's are on the anveodirect.com homepage in the FAQ section. Review it for changes. They should email everyone with any updates to this but your spam filter might not like those messages so check it yourself now and then.
Currently:
What IP addresses should I open on my firewall?
You need to allow the following IP addresses to reach your network:
67.212.84.21 - SIP Signaling
176.9.39.206 - SIP Signaling
50.22.102.242 - SIP Signaling
50.22.101.14 - SIP Signaling *
72.9.149.25 - SIP Signaling

On my firewall I actually made a rule to temp block any IP's trying to register that I have not explicitly allowed.
In Configuration->Resources->IP Trunks->General | IP Trunk Settings
Forward redirected OLI: First redirect
Remote capability MWI: on
Send name display: on
Ignore in-band DTF in RTP: off
In Configuration->Resources->IP Trunks->SIP Trunking | Global Settings
Local Domain: leave blank
Call signalling port: 5060
RTP Keepalives: None
In Configuration->Resources->IP Trunks->SIP Trunking | Media Parameters
Make sure G711-aLaw is in your Selected list, you can have them all selected if you like, and order them top to bottom in order of preference.
Enable voice activity detection: off
Jitter buffer: auto
G.729 payload size(ms): 30
G.711 payload size(ms): 30
Fax Transport: T.38
Force G.711 for 3.1k audio: off
Provide in-band ringback: off
In Configuration->Resources->IP Trunks->SIP Trunking | Public | Accounts
Select Add -> no template
Name: give it a name, Saw a bug before when I put funny character in name it screwed up my config and I had to reset. Keep it simple.
Description: blank or anything you want
SIP domain: sbc.anveo.com
Registration required: off
SIP username: blank
Password: blank
In Configuration->Resources->IP Trunks->SIP Trunking | Public | Accounts -> anveo
Basic tab:
Local: I put the domain name that points to my static IP. (do not think it uses it anywhere though)
Proxy: leave blank
Registrar: leave blank
Outbound proxy table: leave empty
Advanced tab:
Enable local NAT compensation: on
Enable media relay: on
Use maddr in R-URI: off
Use maddr in Contact: off
Support 100rel: on (not sure if they honor it or not, couldn't confirm)
Allow UPDATE: on
Use Null IP to hold: on
Use user=phone: off
Force E164 international dialing: off
Enable SDP OPTIONS query: on
Allow REFER: on
Support Replaces: on
Enable Connected Identity: off
Standard SIP Caps Exchange: off

NAT Pinhole maintenance Signaling method: None Interval: 30
Session timer Session refresh method: Disable
Active call limit: 0 (MAX)

ITSP association method: From header address DNS match
Outbound Called characters to absorb: 1 (Removes your access code for outside line (dial 9 or 8 or whatever you choose)
Inbound Called prefix to prepend: blank
Authentication realm: blank
User Accounts
I left everything blank. You can possibly use this to tweak your outbound call display

In Configuration->Resources->IP Trunks->SIP Trunking | Public | Routing Table
Set the digits used to select this trunk when dialling Eg dial 9 for outside line put 9 in destination digits.

Lines
In Configuration->Telephony->Lines->Active VoIP Lines
You should have a line for the number of trunk licenses you have.
Assign them a line type Pool eg: Pool:BlocB
If you wanted to split your VOIP licenses between internal private trunks and public external trunks you could put some in separate pools, but usually they are just all in the same pool. Some Pools may already but used by other trunk interfaces like PRI.

Dialing Plan
In Configuration->Telephony->Dialing Plan->Routing | Routes
Add a new route. give it a number up to 3 digits This is not the number you have to hit but you might as well make it the same if you can
eg: route number 9. Use the pool you assigned your VOIP lines to and set DN type to Public (Unknown) I'm not sure what the difference are between the different DN types or why you would choose one over another.
In Configuration->Telephony->Dialing Plan->Routing | Destination Codes
Add a new destination code for the digit you need to dial to access your Voip trunk to make outbound call. eg: 9
Set the Normal Route number to be the Route number you created.
Set absorbed length to 0. We remove the extra dial 9 digit in the SIP trunking Advanced tab already and this preserves the digit so the sip trunking routing tab still sees it. If you select All it will remove the destination code before passing to the SIP trunking module to process.

As usual with dialplan you can make things more complicated if needed. For example I had to jump through some hoops to add 1 if someone dials a local number with 10 digits when Anveo expects all calls to include country code (so 1 for north america)

Last inbound calls. You will need to add your DID to a target line Public Received # and attach that target line to an extension or setup callpilot to answer that line number if you want it going to an IVR.

I probably forgot something but that should work.

I took the example SIP template and made it work for Anveo direct. If you add the file to a ZIP archive it will allow you to replace an existing version if you tinker with a setting in the file otherwise it complains it already exists if you already imported it once.

Save this in plain text as filename: ANVEO_DIRECT_MEDIARELAY.sipt
##########################################################################
##
## Name: ANVEO_DIRECT_MEDIARELAY
##
## Template Builder Version: 1
## Date Created:09/28/2017
##
##
##
##
##
##
##
##
$START
Name:ANVEO_DIRECT_MEDIARELAY
Description:Anveodirect (Media Relay)
ItspDomainOrIpAddr:sbc.anveo.com
LocalDomainOrIpAddr:
ProxyFqdnOrIpAddr:
ProxyPort:5060
AllowRefer:0
RegistrarAddr:
RegistrarPort:5060
EnableMaddrInRUri:0
EnableMaddrInContact:0
EnableUserEqPhone:0
ForceE164Int:0
SessionRefreshMethod:0
SessionMinSE:90
SessionExpires:1800
SessionRefresherRemote:0
AuthRealm:
EnableNat:1
EnableMediaRelay:1
SipKeepaliveMethod:0
SipKeepaliveInterval:30
Support100rel:1
SdpAllowUpdate:1
SdpEnableOptionsQuery:1
SdpUseNullIpToHold:1
ItspAssocMethod:12
SupportReplaces:1
RegistrationRequired:0
Comment:Anveo Direct wholesale SIP service. Media Relay feature is enabled. \n\nBe sure to setup outbound routing with static IP on anveodirect portal.
SipStandardCapsEnabled:0
$END

 

[curlycord]

Here is a step by step too:

http://www.tek-tips.com/faqs.cfm?fid=7741

Have not heard of the need to allow IP address's yet.

Thanks for the sample.





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[exsmogger]

I haven't had to allow IP addresses in my system either. As long as my BCM can register with the SIP provider, calls in and out work fine. It looks like that's the difference in your setup as registration is set to off. Thanks for posting the template.

Brian Cox
Georgia Telephone

http://Georgia-Telephone.com

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

 

[Firebird Scrambler]

Nice post. A star from me.

Firebird Scrambler

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

Website =

http://somertel.com

linkedin

 

[QueBall780]

Anveodirect is wholesale pricing so crazy insane cheap, but you have to deal with their static IP requirement to make it work which means it does not fit with the classic settings where your BCM registers with the signalling server and they send all call audio from their proxy address.

When you have a typical voice over IP provider with registration and the audio is proxied from their SIP server you often do not need to make any firewall changes for a typical NAT firewall setup. Your outbound registration packets are enough to get calls through most NAT routers. Because registration has a refresh period it is constantly keeping that connection open for you and your typical NAT firewall needs no special setup.

But in this case, You are sending all your signalling info to anveodirect but they connect your audio directly to the carrier delivering the call on another IP address. That means all media goes directly to the carriers session border controllers IP and they have a huge list of carriers they use. One nice thing about that setup is you don't have to worry as much about your latency to anveo. Signalling traffic isn't very latency sensitive you just need to watch out for packetloss. The call media usually ends up on a SBC server in the same region as the number you are calling so latency is usually just your distance away from the number you are calling unless there is some particularly bad IP routing.

Anveo has a retail product which operates more traditionally and has all the fancy server side stuff that you can use to nearly replace your PBX if you wanted it hosted in the cloud. It competes with a service like voip.ms.

I expect the same settings will work with other providers who use a similar server setup like Flowroute and ThinQ. (Flowroute does allow server registration in addition to just IP authentication so it's not entirely the same though they still use direct media for it so you might need some firewall tricks on inbound RTP audio) ThinQ is a direct competitor to anveo direct and is also a wholesale aggregation specialist. These services come in at a very low price point. The only way to go cheaper is to sign up directly with the wholesale providers but they usually have minimum commitments and they usually cannot handle all of your traffic so you would need to setup your own least cost routing system to send calls out on the best routes on a call by call basis. (Probably have to be doing crazy high minutes every month to make that a viable option)

I just never came across a similar BCM setup example so it was mostly trial and error and wireshark. Most Avaya examples/solutions providers are either registration based with proxy audio or a managed service where the carrier provides you a dedicated router to send traffic through a private network route so no firewall involved. At least most of the ones with an official Avaya solution document you can follow.

 

[exsmogger]

I have a couple of SIP trunks from Flowroute and I'm using the registration option since I have a dynamic IP. I didn't have to do anything in the router firewall to make them work. I also use remote IP telephones with the NAT Traversal license and those do need to have ports forwarded to the BCM. Thanks for the additional info. Very helpful.

Brian Cox
Georgia Telephone

http://Georgia-Telephone.com

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

 

[curlycord]

Nice info.
I added a link to this thread in my FAQ.

Notha star

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