Incoming calls are fine, outgoing calls send no audio

[dmyze]

So I am using v16 and the windows softphone.

I have run the firewall checker and all tests are passed. I have a pfSense Firewall. my 3cx is on a windows server 2019.

I just finished migrating from windows server 2012 and so this is a new server, but 3cx was restored from a backup.

The other changed I made was the 3cx dashboard was complaining about me using an unsupported SIP (flow route) so I deleted the SIP truck, and re-added it. I think I must have been using "other" before and it was just manually configured with the right settings. I'm not sure when it broke because the issue only happens with outgoing calls and I might not have tested that along the way.

So everything works just fine if I receive a phone call. However if I make an outgoing call the call goes though, I can hear the other person, but they cannot hear me. I thought it was a microphone issue, but I tired several settings and different apps and it works fine, not to mention it works fine when I receive a phone call.

I also have a Panasonic IP phone on another extension and it has the same issue, so I don't think it's a problem with the soft phone.

 

[JohnS_3CX]

Hello,

Firstly I would recommend to dial *777 and talk to hear if the echo test works well (just to verify two way audio between client and PBX). Use the latest version of our client, or use the webclient for your test.

Then to see why there is no audio when you make outgoing calls specifically, you can run a capture on the PBX (Activity Log -> Capture) and see whether there is audio in the leg between the client and the PBX (* 777 should already answer that) and if there is audio from the leg going from PBX->Provider.

Also show us a screenshot of the trunk's Options tab if possible

 

[dmyze]

*777 works fine from the softphone. A little choppy but works from the IP phone.

Here is a shot of the options page from ths SIP provider:

https://imgur.com/6LZuSs9

I'm not sure what you want from the log, here is the connected record on outgoing call:

11/12/2019 8:43:17 AM - L:4.1[Extn:11] got Connected.UAS Send 200/INVITE from 0.0.0.0:0 tid=Pj5227bab8e45145d6a05765d269855f97 Call-ID=2193e506e3ea4a71aec0e799206c8022:
SIP/2.0 200 OK
Via: SIP/2.0/TCP xxx:32862;rport=32862;branch=z9hG4bKPj5227bab8e45145d6a05765d269855f97;alias
Require: timer
Contact: <sip:xxx@xxx:5060>
To: <sip:xxx@xxx;tag3cx=02afbe4aebc9455cb14a0ff9bf697a0e>;tag=3738a667
From: "xxx"<sip:11@xxx>;tag=566ca61e1a5148ecb9d29bbd59bb1419
Call-ID: 2193e506e3ea4a71aec0e799206c8022
CSeq: 25314 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 16.0.3.676 (676)
Content-Length: 378

v=0
o=3cxPS 37497077760 117557952513 IN IP4 10.0.107.12
s=3cxPS Audio call
c=IN IP4 xxx
t=0 0
m=audio 7014 RTP/AVP 3 18 0 8 9 96
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1

 

[dmyze]

There is an article about this on flowroute's website:

https://blog.flowroute.com/2015/06/15/top-4-ways-to-fix-one-way-audio/

Nothing really stands out to me as an issue though, I don't think I have SIP ALG, not sure about the Advertising the correct public IP address, I do have a dynamic IP so this could come into play. Offering unsupported codecs, could be the issue. I tired paring down the list to just G.711u-Law on both the server and my client but it didn't seem to help. not sure if I needed a reboot after making changes.

I did restore the old version of my server with the old flowroute settings I had before but the issue was still there. Feels like a firewall issue to me, your documentation on setting up pfSense is a bit dated. (none of my config pages looked like what you show) I did my best though to still configure it as you suggest. So maybe it's a firewall issue?

 

[JohnS_3CX]

It could be a firewall issue but as you said the incoming calls work fine, and in addition the firewall tester passed (no port remapping or ALG detected).

The above log shows the internal leg (Extension to PBX) so it wont tell us the whole story. I think you should run a capture at this point and then make an outgoing call that someone will answer. After you hang up, end the capture and download the file.

We want to see the call leg from the PBX to the Provider and see if we are sending out an audio stream to them at the correct IP and port. If this is confirmed, then they should also be able to confirm whether they receive it. If they don't then you can focus on the firewall specifically.