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The Mitel 5000 and Bandwidth.com SIP Trunks

sfitzgerald — Fri, 03 Sep 2010 14:47:51 +0000

I recently had the opportunity to configure 3 Bandwidth.com SIP Trunks on our Mitel 5000.

The Mitel 5000 comes with a template for creating Bandwidth.com SIP trunk groups.
In DB Programming, it's under System\Devices and Feature Codes\SIP Peers\Sip Trunk Groups.
Right click, and choose "Create SIP Trunk Group from Template".
Among the list presented is a file named "Bandwidth.stg"
This template includes not only the IP addresses of 216.82.224.202 and 216.82.225.202 that Bandwidth.com gives in its setup information, but also 4.79.212.236, 4.68.250.148, and 209.247.16.221, which I assume are additional SIP gateways that Bandwidth.com operates.
Once these additional IP addresses were added, inbound calling began to work immediately.

I am using a Mitel 5000 running call processing 4.0.3.76. We have an expansion card, making it a CS 5400. We are not using an HX chassis.

The expansion card requires an IP on our LAN, in addition to the IP the 5000 is already using. On the WAN side, we have a /28 from our ISP, of which I have allocated 2 IP addresses to the 5000 -- 1 for the 5000 primary interface, and 1 for the expansion card.

We are using a SonicWall TZ 190. I have NAT mapped any/all traffic to the 2 designated WAN IP addresses to the 2 LAN IP addresses assigned to our 5000. I also created reflexive NAT rules so that outbound traffic from either of the LAN IP addresses would appear as coming from their respective WAN IP addresses. This is important as Bandwidth.com is looking for traffic from our PBX to come from either of these IP addresses. For systems without the expansion card, only the 1 WAN and LAN IP is necessary. In the firewall portion, I have allowed all traffic from the WAN to these 2 WAN IP addresses; counter-intuitively, this goes in the WAN -> LAN section of the firewall.

I also had to make sure that the WAN IP addresses were programmed in System\IP Setting\System NAT IP Address and System\IP Connections\

\NAT IP Address. The primary WAN IP for the 5000 went in both System\IP Setting\System NAT IP Address and System\IP Connections\P6000\NAT IP Address. The WAN IP address for the expansion card went in System\IP Connections\P6001\NAT IP Address.

We have 3 SIP trunks and 3 DID's from Bandwidth.com. The 3 DIDs are sent to a call routing table, but can be sent to a single extension if desired.

It is important to note: you must have static WAN IP addresses with which to work. Also, while Bandwidth.com's sipstation SIP trunks use authentication and registration, Bandwidth.com's regular and supported SIP trunks use neither.

Open Source is a good thing

grazman — Fri, 05 Mar 2010 09:49:50 +0000

Our firm has been installing and using this product for over 5 years.

With recent software changes and updates we can now do the following:

1. Install an open source firewall which will pass your voip traffic.
2. Allow remote users with hardware or software based phones to connect, including automatically provisioned devices or software (no end user fiddling needed).
3. Setup siptrunking to bandwidth.com

All of this on open source with no licensing fees. Pretty cool!

International Long Distance Dial Rules?

escott — Wed, 24 Feb 2010 20:28:21 +0000

Hello,

I've got the local and long distance calls working fine but I'm having a problem getting the correct pattern for international long distance.

I've tried:
011.
011
+011
(also tried leaving it off completely and dialing without the 011)

But haven't been able to get any of them to work. I have a + in the Outbound Prefix box but I've also tried without it and get the same results.

We are using Trixbox CE 2.6.1

A little help?

Regards,
Erik

Caller ID

smaloney@fdeleon.com — Fri, 25 Sep 2009 19:58:41 +0000

I need to be able to set the caller ID NAME I'm sending on my TrixBox Pro. We have two smaller companies with-in our organization, the parent company name "Alpha" sends out no problem, the sub companies "Bravo" and "Charlie" have their own numbers, which display properly, but are stuck with the name "Acme" being displayed. i was told that no matter what i send out the name will be changed to display the billing name on file. How can i get this to work for us?

Configure a Nortel SCS with Bandwidth.com SIP Trunks

srivers — Thu, 20 Aug 2009 21:46:36 +0000

See the latest Application Note attached.

How to Configure a SUTUS BC200 with Bandwidth.com SIP Trunks

srivers — Wed, 08 Apr 2009 22:19:14 +0000

Bandwidth.com will require the BC200 to reside behind an Edgemarc router and will require that we deliver 10-digits to the SUTUS, not our default E.164. Also we will need to setup outbound calling for 10-digit.

See attached Application note for more information.

How to configure an Aastralink Pro 160 with Bandwidth.com SIP Trunks

srivers — Thu, 02 Apr 2009 15:25:04 +0000

I have attached an application note on configuring Bandwidth.com SIP trunks with Aastralink Pro 160.

How to Configure an Aastralink RP with Bandwidth.com SIP Trunks

srivers — Thu, 02 Apr 2009 15:23:25 +0000

I have attached an application note for how to configure the Aastralink Response Point.

new to bandwidth.com new to trixbox

comnet — Thu, 02 Apr 2009 03:52:56 +0000

Hello, I'm new to both products and looking for some assistance. I have a basic guide from bandwidth that somewhat matches the trixbox version that I am using. I hope to set all of this up behind my Astaro firewall. Can someone point me in the right direction? I am starting with 3 SIP trunks from bandwidth, with 4 phone numbers and 1 - toll free number assigned. Will also attempt to use a fax machine with one of the 4 numbers.

Thanks!

David

Application Note: Configure NEC Univerge SV8100/UX5000 w/ Bandwidth.com SIP Trunking

srivers — Mon, 23 Mar 2009 13:02:54 +0000

The document attached is designed to help technicians, with a functional knowledge of the Univerge SV8100/UX5000, configure SIP Trunking from Bandwidth.com. For more information about NEC please visit http://www.necinfrontia.com .

The NEC Univerge 8100/UX5000 is designed to work behind the Bandwidth.com provided EdgeMarc Router/ALG device.

Remember Bandwidth.com does not support the PBX only the service. Please visit your authorized representative for support.

Using pfSense firewall with trixbox - fix for one way audio

c0mputernick — Thu, 05 Mar 2009 22:14:35 +0000

I just wanted let everyone know that i figured out how to fix the one way audio problem when using trixbox behind a pfsense firewall.

The one way audio problem didnt happen when using an IVR, only when using DID.

Besides opening the ports with NAT, and i do mean ALL ports, bandwidth.com doesnt conform to the standard media ports you have to open ALL ports to your trixbox from 10000 on. thanks bandwidth.com that will be a nice security nightmare. but thats another topic.
(they say 1024 - 64000, but it works with 10000 - 64000)

The setting that fixed my one way audio problem was under Firewall -> NAT -> Outbound
You have to switch the setting from automatic to manual.
It will then create a rule for you that you have to edit.
Check the box for "static port" and apply the settings.
That will change the rule to "static port=yes"

This will fix the one way audio problem, since pfsense uses symmetric nat instead of full coned nat, the returning packets get lost if the ports arent static outbound. aka one way audio.

if you are using another firewall and are having one way audio problem you might try checking if it has an option like that or not.

I hope that this helps someone out there.

Using pfSense firewall with trixbox - fix for one way audio

c0mputernick — Thu, 05 Mar 2009 22:11:23 +0000

fax over sip

georgegti — Tue, 13 Jan 2009 16:57:58 +0000

has anyone had any experience with faxing over sip trunks. i know my device needs to be T.38 compatible to accept faxes. im trying to work with an altigen system and get them converted from PRI to SIP trunks to save some money. they have fax DIDs coming to a multitech faxfinder.
anyone have any ideas if this would work or not.

[HOW-TO] Configure Bandwidth SIP trunkings with asterisk + FreePBX

yqueret — Tue, 06 Jan 2009 11:48:23 +0000

Introduction :
You can find bellow, a bandwidth SIP Trunking configuration running with asterisk + FreePBX.

This configuration working for us :-)

Configure SIP Trunking

In FreePBX web interface add 2 SIP trunk (Primary & Secondary)

First SIP Trunk

Outgoing Dial Rules
Outbound Dial Prefix : +

Outgoing Settings
Trunk Name : BW-SIP-A
PEER Details :
canreinvite=yes
dtmfmode=rfc2833
host=216.82.224.202
outboundproxy=216.82.224.202
progressinbound=yes
qualify=300
type=peer
disallow=all
allow=ulaw

Incoming Settings
USER Context : from-bandwidth-A
USER Details :
type=peer
reinvite=yes
port=5060
insecure=invite,port
host=216.82.224.202
fromdomain=216.82.224.202
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
qualify=300

Second SIP Trunk

Outgoing Dial Rules
Outbound Dial Prefix : +

Outgoing Settings
Trunk Name : BW-SIP-B
PEER Details :
canreinvite=yes
dtmfmode=rfc2833
host=216.82.225.202
outboundproxy=216.82.225.202
progressinbound=yes
qualify=300
type=peer
disallow=all
allow=ulaw

Incoming Settings
USER Context : from-bandwidth-B
USER Details :
type=peer
reinvite=yes
port=5060
insecure=invite,port
host=216.82.225.202
fromdomain=216.82.225.202
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
qualify=300

Configure Outbound routes

Now you need to add 1 Outbound Routes

Route Name : US-Canada
Dial patterns :
911
1NXXNXXXXXX
1NXXNXXXXXX
etc ..
Trunk sequence :
SIP/BW-SIP-A
SIP/BW-SIP-B

Configure Inbound routes

Description : John Doe
DID Number : +12025081234
Destination : Choose an extension, IVR, Queue, ... (what you want ..)

Problem using allworx with bandwith.com sip trunkin

Fernando — Tue, 30 Dec 2008 00:18:10 +0000

I have my main office using sip trunking provided by bandwith.com using working allworx 10x as pbx.
I also have five satelite locations conected to the same unit.
Bandwith.com ported each line from the diferente offices to one block of sip lines and all the lines was brought to my main location.
When somebody calls one of my satelite locations, The Allworx unit recognizes the number dialed, picks up the call and forwards to the extension chosen at the satelite location.
My bigest problem is: When any satelite locations is dialing out, they cannot choose their own number to call. I wanna dial out using the line that corresponds to the phone number of that location. For two reasons.
1 - Lots of people dials back by using call Id. If they use call id the phone may ring on a different location that has that number.
2 and most important.- How do I call 911 in case of emergency? Imagine calling 911 for my office in Plymouth, MA but the police shows up in my location in Potomac, MD.
I need to be able to choose the line I want, on a block of sip lines.

Exemple: my phone number in location A - is 508 444 3333 - When I call out, I want to pick up that line number to call.
Any sugestions? Has anybody dealt with similar situation?

Known DTMF Issue Asterisk ver. 1.4.22/23

srivers — Tue, 23 Dec 2008 21:59:27 +0000

Ther is a known DTMF issue with Asterisk Version 1.4.22/23 Digium has issued a patch find it and more information at http://bugs.digium.com/view.php?id=13209

How To Configure Response Point with Bandwidth.com SIP Trunks

mtindall — Thu, 23 Oct 2008 20:14:22 +0000

How To Configure Response Point with Bandwidth.com SIP Trunks

How To Configure Bandwidth.com SIP Trunks with Talk Switch

mtindall — Thu, 17 Apr 2008 18:23:36 +0000

How To Configure Bandwidth.com SIP Trunks with Talk Switch

How To Configure Cisco Call Manager 4.1 with Edgemarc / SIP Trunks

mtindall — Wed, 19 Mar 2008 14:35:17 +0000

The following AppNote explains how to configure Cisco Call Manager 4.1 or higher using an Edgemarc 4500 and Bandwidth.com SIP Trunks.

Application Note : Configure POTS 911 Redirect on 4500

mtindall — Mon, 07 Jan 2008 22:15:39 +0000

This document explains how to configure 911 Redirect to an FXO line plugged into an available line port on a 4500 EdgeMarc.

Note: Compatible with SIP Protocol ONLY will NOT work with MGCP

911 Redirect will work with SIP phones or Analog phones that are connected directly to one of the EdgeMarc's FXS ports.[/b]

If you are a Bandwidth.com customer please contact Bandwidth.com support if you are having any issues at 800-808-5150 or customercare@bandwidth.com