No audio on VOIP calls after calls transferring
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@stephenw10 It looks like it is a screen capture. Its a png image file actually.
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Oh. Ha!
Yeah, OK. The actual pcap file here would be much more useful.
And we need to know what the various IP addresses are here and where they are.
What and where is 192.168.0.103?
One of those things is probably incorrectly sending it's internal IP to connect to for RTP because that's almost what's broken with VoIP.
You can see in the screenshot that it appears the phone at 192.168.202.99 starts sending RTP traffic to the PBX before the PBX replies with the port unreachable error.
Steve
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@wintok Can you please check and configure as may be required a few basic yet often critical settings so we can eliminate them having any influence on the issue.
1/ VoIP sessions are timing sensitive. Can y0u please ensure that your phones and FreePBX are all syncing to the same time server (NTP). This is so that all network and SIP/RTP traffic timestamps are in sync.
2/ Please configure the SIP session timers and registration timer settings on your phones to a period short than the pfSense default lifetime of dormant states. Usually a setting of 20-30 seconds is adequate. ( I mentioned this earlier in the thread)
Here are the pfSense dormant state timeouts for your reference, found at...........
https://docs.netgate.com/pfsense/en/latest/config/advanced-firewall-nat.html#config-advanced-firewall-optimization3/ Re-check and be sure that the SIP and RTP ports in use by your PBX and phones are correctly configured to match as required between each phone and the PBX.
4/ Ensure there is no outbound traffic firewall configuration on your pfSense appliance that could restrict outbound traffic on the ports required in item "3/".
Let us know the results.
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@stephenw10 Yep, I agree the actual pcap file would be been much better and more informative.
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@stephenw10 said in No audio on VOIP calls after calls transferring:
You can see in the screenshot that it appears the phone at 192.168.202.99 starts sending RTP traffic to the PBX before the PBX replies with the port unreachable error.
Could that just be a UDP thing? You know, let's send it and let the other end complain if it all goes wrong.
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Usually in this sort of setup you get one way audio because, with only one device behind pfSense, only one way is trying to send to an invalid IP.
Something else is in play here. Perhaps in addition. -
@stephenw10 said in No audio on VOIP calls after calls transferring:
Usually in this sort of setup you get one way audio
So re-clarifying the problem as I understand it from what wintok has said earlier in the thread......
a) calls to and from an external third party phone work properly without issue, No audio problems.
b) The problem only exists with transferred calls via the remote PBX, between phones on his LAN behind pfSense. SIP signalling works as in the call is supposedly transferred, but no audio is present.So if I have that correct, to me it suggests the problem has it's source at the point where the remote PBX attempts to communcate with the phone behind pfSense that is receiving the transferred call. As we have suggested, without more detailed info (pcap file) to go on, the initial thought would be that a pfSense state does not exist at that time between the PBX and the phone receiving the transferred call. My focus, given wintoks setup, has been primarily on the configuration of the phones at this point, to try and maintain states in pfSense. Do we have a compatibility issue with the phones given the overall system set up? That is also a possibility.
Also I hope wintok has checked and made the RTP related configuration requirements to FreePBX that I made available ealier in the thread, that are relevant to and apparently required for the type of set up wintok is using. Here it is again......
FreePBX server's sip.conf file important settings.
nat=force_rport,comedia
directmedia=noIn the end it may be that wintok will need to fall back to bingo600's STUN solution earlier in the thread, which will of course assist with the NAT aspect for the phones behind pfSense. Still wintok should ensure all the fundamentals are set up correctly first even if going with the STUN solution.
If using the STUN solution, I am not so sure the above FrePBX sip.conf file configuration perameters would remain relevant. I believe a STUN server sets up a pont to point type connection for RTP. Sort of like a Teamviewer session would work for example if i understand it correctly. -
The initial problem description seems unrelated to call transfers:
@wintok said in No audio on VOIP calls after calls transferring:
Behind my pfsense firewall I can only ring each other but no audio between them.
But then:
@wintok said in No audio on VOIP calls after calls transferring:
The problem is with transferring calls only.
If both or one them behind pfsense that when the call seems cannot transfer between them. You can hear the ringing from each other only when you pick you can't hear the sound.But I'm not sure 'transfer' is the correct term here.
We need clarification. Just speculating without it.
Steve
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@stephenw10 said in No audio on VOIP calls after calls transferring:
But I'm not sure 'transfer' is the correct term here.
I have been treating the term "transfer" and "transferring" soley as a formal telephony term, in this instance within the context of a common PBX feature. As in a call has comes in on a specific phone (extension) and then is required to be "transferred" to a different phone (extension).
In the end we are at a pfSense forum dealing with a VoIP system issue, which is fine, but not really specific to pfSense. So I wonder if wintok has posted his problem on the FreePBX forum? I am sure he is not the first to be dealing with this issue and other FreePBX users will have better and specific qualified advice to assist him, relating to the set up of the VoIP system. That is where the better advice with VoIP systems will reside for his problem in my opinion.
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@wintok Hi wintok,
I have another option for you that may assist you with overcoming your problem.
I have just noticed that FreePBX has OpenVPN available.
pfSense also has OpenVPN available.Setup a VPN tunnel between FreePBX and pfsense then let your phones communcate with FreePBX over the VPN tunnel. It should then be a matter of traffic rules to get it all working.
Here is a link with some instructions.
https://wiki.freepbx.org/display/FDT/%5BHow-to%5D+Setup+VPN+between+pfsense+and+FreePBXCheers
