I willbe installing SIP trunks on my UCX20 this friday. AT&T is using IP Authentication and I am lost. I have done SIP trunks using a username and password with no problems any help or hints would be greatly appreciated.
Per the E-Metrotel documentation, the following SIP Trunk Providers are supported:
Allstream
VoIP.ms
Skype
Broadvox
Flowroute
SIP Affinity
Sipgate
SIPmly
SoTel
However, here is a link to the UCx Documentation that ddresses IP Authentication.
I understand that they have only tested it with those providers you listed, but the sales documentation claims they work with SIP Trunks. No constraints too specific carriers is ever mentioned.
I have alrady seen the document you referenced me too, but I am looking for some additional information. Hopefully someone who has already done this with AT&T.
The best way to deal with this kind of problem is to call the SIP provider and ask them if they have a configuration that supports Asterisk based systems. I have yet to come across on that doesn't. They may not have certified the emetrotel system yet nor has emetrotel done the interop testing with them but that doesn't mean that the carrier hasn't made it work for someone else in the past. If the have, they will have notes on the proper configuration and should be more than happy to share the info with you.
If they authenticate using username and password, try starting with the config for voip.ms. That's how they do it and AT&T is probably very similar. If you can't figure it out, call them and ask.
In general, for IP authenticated SIP trunks, you should add a new SIP trunk (using the PBX - PBX Configuration - Trunks page) with the following information entered:
Trunk Name -> a user friendly trunk name that is visible in the GUI (e.g., in the inbound or outbound route sections)
Outgoing Setting
Trunk Name -> the trunk name that will be visible in logs (do not use spaces - use underscores as separators if desired)
Peer Details
host=xxx.xxx.xxx.xxx -> the IP address of the SIP trunk provider's server
type=peer
trunk=yes
qualify=yes
insecure=port,invite
Incoming Settings
User Context -> blank
User Details -> blank
Registration
Registration String -> blank
With a little bit of luck, this is all you need to enter (for IP authenticated trunks). Other (optional) settings use defaults that are configured on the SIP Settings page. You could override some of these defaults if required by adding extra lines to the Peer Details section.
You could take a look at this thread for an example that was used for IP authenticated SIP trunk provided by Level3:
I have not turned these up yet. AT&T and the IT company is having some issues getting the VPN to work. That needs tgo be up to switch over to the fiber service.
Thanks for the help, I will keep everybody informed.
The signaling address is the IP address of their SIP signaling server - you should use this address in the host field (and also the permit field if you use it).
The media address is the IP address of their media gateway - that's the source/destination of RTP packets. You shouldn't need to specify this address anywhere - it is provided to UCx in call setup messages by the signaling server. You might need it only if you have a restrictive company firewall that blocks all traffic from unknown addresses or something like that.
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