Typically only ULAW and OPUS unless you are outside the US. I recommend that you never leave a PBX default to all the codecs like that. How does the SIP negotiation work for selecting codec? Is the client or the server the master when determining which one to use? When I moved all but G.711 a/mu over to the available but not selected then all of my issues cleared up. Yesterday when I was having the issue, all of the codecs from Zoiper were in the selected codec list. The two left columns are from Zoiper and the rightmost from FreePBX. What was it set to choose first before? And what is your server set to accept? ![]() I didn't think the STUN server would play into this particular call test since it was extension to extension on the same network segment and never had to exit the WAN interface on the said in FreePBX / Random loss of audio said in FreePBX / Random loss of audio said in FreePBX / Random loss of audio. The Zoiper client was running on a laptop connected wirelessly to a Ubiquiti AC Pro which itself feeds into the netgear switch. They run through an unmanaged netgear switch that sits behind a Cisco ASA 5505. The PBX and the phones are all on the same network, 192.168.2.0/24 ![]() I'm going to do some more testing tomorrow and verify. I changed Zoiper to only allow G.711a/u and that seemed to fix the issue. It did seem to be the codec, in this particular instance at least. If SIP-ALG is disabled, make sure your STUN settings are correct. The most common of those is having SIP-ALG turned on (it's normally on my default.) SIP-ALG more or less seems to exist to create these kinds of headaches (to help vendors sell more services.) That's normal, but that's the kind of starting point we need for diagnosis.īy far these issues tend to come from router / firewall issues. ![]() Looks like your phones are on a private network separate from the PBX. I notice also that if I capture then the softphone is constantly sending RTP packets to the server but the server is not sending any RTP packets back to the softphone, which I presume is why I'm not hearing the audio of the voicemail greeting from the said in FreePBX / Random loss of audio. Is it normal to receive receive 401 Unauthorized before a successful registration?. One thing I noticed is that when my Zoiper softphone first registers with the server it seems to receive a 401 Unauthorized from the server before successfully registering. I've broke out wireshark to see if I could determine anything. I seem to be able to make a call from the softphone to the virtual extension with no problem in that it seems to connect each time but the audio to hear the voicemail greeting is not received most of the time. Right now I have two extensions, one a SIP extension using Zoiper softphone and the other a virtual extension with voicemail enabled, just for something to test with. I'm setting up a FreePBX instance and doing some testing.
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