No sound outbound forwarded calls

FreePBX Configuration
1

Hi all,

We have been using FreePBX for over a year now but all of a sudden we have an issue with sound. When users setup an unconditional forward towards their own personal phone number they have no sound on both sides of the isle when an external call comes in meaning :

External call → FreePBX → External personal phone number

But if we call from internal towards external it works, external towards internal works but an external forward towards another external number, that does not work.

I’ve got no clue where to look for, do any of you have in idea by any chance?

Thanks in advance!

2

Which channel technology and driver?

Which version?

Please provide the full log, at verbosity 5, and the appropriate protocol log, for the channel driver in use, as plain text, and redacted, but so as not to hide the difference between different addresses and different types of address.

3

Hi David,

The technology we’re using is PJSIP (i’ve been trying to google about a driver but i’m not quite sure what to tell you here honestly)

System version : 12.7.8-2306-1.sng7
Asterisk version : 18.19.0
FPBX-16.0.40.11(18.19.0)

Could you help me out what the best way would be to provide said logg at verbosity 5?

4

For chan_pjsip, you will need to use the CLI command “pjsip set logger on”

By driver I meant chan_pjsip, as against, say cham_dahdi, or chan_sip. Technology was probably confusing as it is means the same as the driver, in Asterisk terms, but I really meant, analogue, ISDN, SIP, H.323, SCCP, mobile air interface voice, etc.

5

Is the PBX behind NAT?

7

Hi David, we are using chan_pjsip with a SIP trunk from the ISP

I’ve looked into those logs and this is everything under verbose 5, i’ve kinda blurred some of the personal information. Now that file/log is quite humongous, what would be the best way to share it here? (480 lines of information)

8

Yes it is

9

Have you made proper firewall rules for the RTP traffic?

10

The standard way of sharing is via pastebin. There is a pastebin.freepbx.org, but some prefer the original pastebin.

11

I sort of assumed that it worked without the forwarding, but I suppose one should explicitly ask that question. If this were plain Asterisk, and it worked without forwarding, but not with forwarding, I’d suggest direct media should be disabled, but for FreePBX, you normally have to go out of your way for the preconditions for attempting direct media to be met, as FreePBX generally wants to see DTMF, and that is most often done with RFC 4833, which requires the media flow through Asterisk.

12

To be fair, i said “yes” rather blindly since i did enable to when i was configuring everything when we launched it but now i’ve seen in our firewall that some rules & NAT seem to be disabled. Basic calls still work (as in we can still help customers atm) so i’ll have to wait to make some changes & tests outside of the office hours

This is the pastebin url : Call Issue - FreePBX Pastebin

13

This has nothing to do with direct media at this point. FreePBX uses local channels to initiate Call Forwarding:

[s@macro-dial-one:57]Dial("PJSIP/Telenet-SIP-000***47","Local/00470****55@from-internal/n,10,HhtrIb(func-apply-sipheaders^s^1)")innewstack

When that happens, unless media is being pushed over that local channel from the PBX (which it is not) then no RTP ports are selected or used at this point. Meaning, there’s nothing to open the NAT/firewall for when the call is sent out. Media doesn’t start to flow until the call is answered and since this is a local channel, when answered the media flows into the PBX first.

If there isn’t proper firewall/NAT rules to allow new incoming requests for RTP to come into the PBX, no audio stream is really made.