Sip trunk with another pbx

Hello everyone,

i am brand new in freepbx and im still trying to figure out most things about it. I have created 4 pjsip extensions and internal calls works fine. I want to connect the freepbx with another pbx we have in our company, so i created a 4 sip trunks that connects the two PBX’s. The sip trunks from asterisk CLI seams to be available but not in use, so i guess the 2 PBX’s are now connected. My purpose at this stage is that one pjsip extension from freepbx, can make a successful call through the sip trunk i created to an another extension in our PBX. I have created an inbound route and an outbound route. Can anyone help me with an idea what i have been missing?

Thank you.

Why do you need more than one trunk?

You need to provide screenshots of your configuration and/or the full log for a failed call. The former may be difficult for a new user, but you can use, for the logs and either paste the fully URI marked up as pre-formatted text, or the random filename part of it.

As @david55 suggested, but please use instead. I feel that it’s important that future readers, who search for their problem and find an old thread, can follow along, including viewing logs, etc. Unfortunately, a recent change was made to that forcibly deletes pastes after no more than 30 days. So, I no longer recommend .

Thank you for your answers. I searched a lot in forums and wiki and i found a solution. I can now make calls from one pbx to another. But i have no sound, no voice at all. Any idea about it? Also what screenshots do you need to provide it to you? I am currently using udp ports 18.000- 18.100.

Try to add the other PBX IP by navigating to settings → Asterisk SIP settings and add it under
“NAT Settings → Local networks”

Ι have already done that. Otherwise, one PBX won’t “see” the other. I can make successfull calls both ways, but no speech at all.

At the Asterisk command prompt, type
pjsip set logger on
sip set debug on
according to channel drivers used.
Make a failing call, paste the relevant section of the Asterisk log at and post the link here
Also post a description of the networking between the machines (VPN, NAT, firewalls, etc.)

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Here, you’ve got a good start to do it, even if this document is done for Elastix. but Elastisk used FreePBX 2.11 and the schema is the same.
You can replace IAX with SIP.

Here from-pstn or from-trunk is not used, just keep your mind both FreePBX server are in the same context = from-internal.
With it, if you are calling 200 (freepbx B) from 100 (FreePBX A) , it should work.

Here is the pastebin link.

Also, no firewall, the built at that time is only internal, no vpn.

I take it this is a paste from the PBX receiving the call? All I see is the PBX re-transmitting a 200 OK back to the far side for the INVITE. The other PBX should be sending back an ACK to acknowledge the 200 OK and since there is never an ACK and the 200 OK keeps getting re-transmitted the call is never fully setup and audio never starts to flow causing a disconnect after 30 seconds.

You need to see why the other PBX is either A) not responding to the 200 OK or B) responding to the 200 OK but the ACK never makes it to this PBX.

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Thank you, i will check it and i will come back here. What about this message?

“res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/1231-00000013’ for lack of audio RTP activity in 37 seconds”

As I said, call isn’t getting a complete setup and audio isn’t flowing because of it.

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