Hi I have the following setup sip phone --- Kamailio for Registration and Routing - Freeswitch for billing and media handling - VOIP Provider Calls are sent from Kamailio to FS and Authentication works as it should, but I cant get the calls to be forwarded to the VOIP provider when they arrive from Kamailio, if I register directly on FS the calls are delivered to the VOIP provider and I can make a call, I did try to create an inbound route but that did not work. Any hints on what I have missed please ? Thanks
On Wed, Apr 4, 2012 at 5:36 AM, Ali Jawad <ali.jawad@splendor.net> wrote:
Any hints on what I have missed please ?
We're going to need more info. Logs from FreeSWITCH would be a good place to start. There are nearly an infinite number of options when configuring FreeSWITCH, SIP proxies and gateways. We can help unless you give us more of an idea about what you're seeing. Also, a look at your dialplan and possibly some SIP traces would go a long way toward helping. Best, Gabe
participants (2)
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Ali Jawad
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Gabriel Gunderson