I might also be missing something, but the SIP proxy is very different. Where do I enable transparent proxy and setup a SIP routing table for different domains? I tried a simple non-transparent configuration, but that also doesnt appear to function.
I'm trying to use a fresh ASG 6.993 for SIP VoIP test.
To do that, I configured my ASG with a NAT masquerading rule and I enabled the "VoIP Security" -> "SIP":
SIP server networks: Any
SIP client networks:
I'm not using a SIP server behind my ASG so I believe there is not necessary to define a DNAT rule...
However, when I call from my SIP-phone, I can establish a connection but there is no voice traffic.
Here a packet filter log line:
i have the same problem. I have an astrisk server on a different subnet that i can't talk to.. Our phones use ports 10000 and up.. but i can't figure out how to get it working yet...and it looks like astaro isn't helping much on this issue either..[:S]
i have the same problem. I have an astrisk server on a different subnet that i can't talk to.. Our phones use ports 10000 and up.. but i can't figure out how to get it working yet...and it looks like astaro isn't helping much on this issue either..[:S]
I have the same problem with V7.001! When I disable VoIP Security I can receive calls and dial but I have voice traffic!!! But when I configure VoIP Security the phone is ringing but NO voice traffic is possible! Also when I use my voip-phone I can dial but still NO voice traffic! I configured at the server option my sip-provider and the client is my voip-phone. Is this a known bug????
Same issue here, running the latest version - have sip proxy configured with any for internal and any for external - we have a trixbox server and multiple voip hardphones internally that function fine iwth calls to an external sip wholesaler externally - calls from remote phones to external numbers (dials the server which connects the external provider and remote phone) function fine - calls to internal extensions have no audio.