vicidial.org

Hello,

We have a bit of a unique scenario. We have a bunch of inbound DIDs hosted at a number of different providers that we have to keep active for various reasons. Currently, we just forward all of these numbers to a single toll free number at our main carrier, which is integrated into Vici.

This is however very costly. We've been told that we only have two options for routing these calls to ourselves:

1) Forward the calls to an 8xx number (which we're doing).
2) Forward the calls directly to our gateway via "SIP AWARE".

#2 would save us a lot of money, but is this possible with a Vici setup? Is there a way to give them our IP address, forward the call directly to VICI without a carrier in the middle? Do we need to setup something like OpenSIPS to create a gateway?

Any advice is much appreciated as always.

-Brian

Vicibox 6.0.4 from Vicibox_v.6.0.x86_64-6.0.4.iso | ViciDial SVN Trunk v.2.10-443a build 140617-2017 revision 2130 | 1.8.32.3-vici | Multiple Servers: 6 Dial - 1DB - 1 Web| Amfletec VoiceSync PCI Chips | No Extra Software After Installation | Intel MultiCore Processors

Instead of using those bunch of DID's. You can use only the TFN and have your carrier of the TFN forward the call to your vici IP via sip. With this setup, you only need to pay for the TFN and the calls coming IN to this number and also the MRC.

Thanks for the suggestion. We actually need to keep all of the local DIDs and can't swap them out for a TFN.

For anyone else interested in this, we figured out our problem. Asterisk rejected the calls as 401 SIP Unauthorized, even with the DID in VICI. We just has to create a carrier entry with the forwarding switch's IP address as the fromdomain for asterisk to allow the call to connect. (Even though the company forwarding the call to us isn't really a carrier).

csbe wrote:... (Even though the company forwarding the call to us isn't really a carrier).

Because of course your concept of "Carrier" is a Social decision. Vicidial/Asterisk consider anyone sending a call to be a carrier. If they are not listed as a SIP account, and the Guest account is off ... rejection ensues. Because there really is no difference between a Carrier and a SIP Phone. A SIP account is a SIP account. Carriers, phones, servers, they are all really the same thing in the eyes of the Asterisk server. Configuration options after they "connect" may differ (limited lines, codecs, DTMF handling), but those are all SIP options that CAN be applied to any SIP connection.