Eureka!
I have managed to get a BCM50 Release 2 to communicate with an authenticating public SIP trunk provider (Voiceflex, in case you are wondering.) Now, neither Voiceflex, nor Nortel support this. According to our suppliers, nobody else is doing this. But, by putting Asterisk in between them (which Voiceflex seem to be using too, judging from their voice prompts), I have the BCM50 working fine on Voiceflex’s SIP trunks.
Now all I need to do is get it up and running on an NSLU2 or similar.
Watch this space to find out how it goes at a client site.
UPDATE – Voiceflex now do support this – if you are connecting from a static WAN address, they will allow connections from this IP without authentication if you request it. If you are using the BCM50 (this includes release 3), this will work for you – unless you have a dynamic WAN address, in which case the technique mentioned here is still useful.
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Hi Scott
Well done on getting that to work. Please note that we do allow IP based authentication (instead of username / password) on request, so you can get this working without the box in the middle.
We are working hard with Nortel at the moment to achieve full interoperability. The good news is that the BCM50 is exactly the device we are currently testing with. We hope to have this complete by the end of this quarter.
Happy SIP trunking!
Regards,
Richard
–
Richard Brady
VoIP Engineer
Voiceflex
t: +44 (0)207 4202 757
f: +44 (0)207 4401 817
w: http://www.voiceflex.com
Hi Richard
Thank you for the interest, it is good to know that you can use IP based authentication, this would save us the time and expense of the box in the middle. The box-in-the middle solution is now set up and running “out in the wild” too, not just in our lab.
It is excellent to hear that your BCM50 interoperability work is progressing well, since this is such a popular PBX for us. We will certainly be configuring many BCM50’s to use your SIP trunks in the future.
Regards,
Scott Donnelly
Dear all,
I’m very interested on this. So now, I’m have a problem about sip trunking when configure BCM50 to register to SIP server (broadsoft).
Could you please help me to resolve this problem?
Thanks in advance.
Hi Scott,
I recently started developing around and administering a bcm50 we currently have 2 isdn2e lines but i was hoping to remove these and connect the bcm50 to an existing asterisk server elsewhere over the internet just wondering if this would be possible and what configuration i would need on the bcm eg. if i need to buy the voip or sip addons for bcm ?
Hi afterm4th,
You certainly can connect the BCM50 to an asterisk server. Either IP Trunk licenses or SIP Trunk licenses would be required. If you have a BCM50 release 3 or higher, I would go for the SIP trunks – this would give you the ability to interoperate with a SIP trunk provider or other softswitches in addition to asterisk. If you have a BCM at an earlier software level, it may be better to use the IP Gateway licenses, which enable H323 based VoIP trunking, as the SIP functionality on release 1 and 2 is less mature.
I am out of my office at the moment so do not have access to the asterisk config files and notes I made when setting this up – but if you still need some help with the BCM I can post some details of the config when I am back in my office on Monday.
Scott, i have a lot of pain trying to trunking my asterisk box with nortel bcm50 via sip trunk, i have sip trunk gateway lisenses on the bcm. Any help will be appriacate
Hi Lisander,
It has been a while since I have done this so I will post the notes that I made last time I did this. Here you go:
BCM50 Release 3 - SIP Compatibility settings=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
Telephony resources - IP Trunks
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
routing table
=============
asterisk 9 asterisk ast.ip.addr 0 BCM none SIP off 0.0
SIP Settings
============
local domain - asterisk
SIP proxy
=========
* domain - asterisk
route all via proxy- yes
mcdn - none
outbound proxy table - asterisk, ast.ip.addr, 5060, 0, none
SIP URI map
===========
private/subscriber - asterisk
SIP authentication
==================
local auth - off
no localor remote accts
-=-=-=-=
routes
-=-=-=-=
dn type - public(unknown)
-=-=-=-=-=-=-=-=-
dialplan - public network
-=-=-=-=-=-=-=-=
dn length - 90 - 12
=-=-=-=--==-====-=-
add a targetline
=-=-=-=-=-=-=-=-==
If you have any issues then get back in touch.