calling to sip uri

I have to move abroad and taking out my laptop with asterisk pbx installed at it.
I have at office a DID number, for local city, a geographoc number.
Due of law regulations, i cannot use pbxes.com to access it, will not work, since foreign ip address.
I have cisco ata186 at office and all well working, just tested 1 minutes ago.
So, when i will take my asterisk out, people will be able still make and receive local calls at ata186, but!

i want to route sip calls from my account at pbxes.com there too.

since ata186 do not support registration at two gatekeepers at the same time - i can use only one feature

i can make calls to number@ciscoataip

it is ok for me if i will have only inbound calls

can someone explain me, is it possible to build some trunk, or something, to let call out sip uri, without registration at the pbxes.com

i see only one way right now, it is to use enum routing.

If you have a fixed IP address you can do this with
account@IPaddress.

Account should be whatever account the ATA is registered to.

You will probably need also port forwarding on your router or STUN, on the device, but not both. If the ATA registers with an outbound proxy you will only be able to use Port forwarding on the router. As for a STUN server, you can use any. Freeworlddialup or SIPPhone should work fine for most of the Americas. Faktortel has one in Australia. STUN server should ping less than 200ms or use a different one.

In the PBXES make a SIP extension. Now go back and edit that extension. down towards the bottom you will see SIP/ambervoip-XXX where XXX is the number of the extension.

change that line to read SIP/account@IPAddress

I emphasize again account in the previous line refers to the actual SIP Auth or User account that the Cisco is registering to.

if your ATA lisytens on a port other than 5060 you should use SIP/account@IPAddress:PORT

This post has been edited 3 time(s), it was last edited by sup on 26.05.2006 at 12:55.