Accepting Anonymous Calls

Redhat Sangoma Linux release 7.5.1805 (Core)

I installed FreePBX on a VmWare server and registered our two Lifesize Icon 600 video conferencing units to it. I am able to make calls between the video conferencing units through the FreePBX server, but I am not able to receive anonymous, unauthenticated, calls.

I tested this by unregistering one of the units, removing the credentials, and trying to call the other unit. The call would fail without any ringing, and in the Asterisk logs I would see “authentication failed”.

I do have “Allow Anonymous Inbound SIP Calls” and “Allow SIP Guests” enabled.

I read some forum posts that indicated fixes involving a Chan SIP trunk, but I’m using Chan PJSIP, since my understanding is that’s the preferred protocol. I also read something about setting up trunks for each incoming number, is that what I need to do?

What is the right way to set these units up so that I can accept external calls from anonymous unauthenticated sources?

I feel like I must be missing something pretty simple, but I haven’t been able to solve my problem through documentation or old posts so far. Thanks in advance for the help.

Is the Lifesize unit still configured with the same username / caller ID information? If so, it is probably still matching the endpoint configured in FreePBX. You would have to remove that endpoint’s configuration from FreePBX (or change it in some way so that it cannot match) – not just unregister and remove the credentials from the endpoint.

Allow SIP Guests with PJSIP sets up the anonymous endpoint which should allow calls from anywhere, as long as they don’t match some other identifier (see above). Allow Anonymous SIP Calls directs those calls to the Inbound Routes section of FreePBX. It’s not documented clearly, but extensions can be direct-dialed in this way also, without setting up explicit Inbound Routes for them.

I should have stated that I also changed the username/caller ID to an extension that doesn’t exist; “Potato”.

Sounds like it should have worked. Maybe I needed to restart the Lifesize unit or something. I’ll give this another try and post the error if I get one.

I agree; it does not sound like a complicated setup. If you’re stuck, post Asterisk logs (/var/log/asterisk/full) and maybe SIP in a pastebin ( for help reviewing.

I’m not getting an authentication error, like I was seeing last time. This is the only thing that comes up in the asterisk logs when I try to dial from the “outside” video conferencing unit;

[2019-12-30 15:38:16] ERROR[17602] res_pjsip.c: Unable to retrieve PJSIP transport ‘udp,tcp,ws,wss’

Here’s the longer logs;

You’ll see a endpoint registered on “3”, that’s just my cell phone (voip) that I put on there for giggles. The two video conferencing devices are usually on “1” and “2”.

On the asterisk console (asterisk -r from an ssh session) you can get more verbosity real-time by using core set verbose 9 and you can get SIP traces real-time with pjsip set logger on. Try these to see if you can get more insight.

That’s a helpful tool for sure, unfortunately it’s still not really telling me what’s going on. Presumably I broke something while trying to get this to work so I suppose I’ll reinstall the server from scratch and give it another go.

[2019-12-30 16:39:45] ERROR[17602]: res_pjsip.c:3031 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport 'udp,tcp,ws,wss'
    <--- Transmitting SIP response (475 bytes) to TCP: --->
    SIP/2.0 500 Internal Server Error
    Via: SIP/2.0/TCP;rport=16396;received=;branch=z9hG4bK-5e0a2851-57ad8004-5a21ed70;alias
    Call-ID: 2c77db30-341fa8c0-13c4-45026-5e0a2851-18df7ab8-5e0a2851
    From: "Conference Room" <sip:[email protected]>;tag=2c7708e0-341fa8c0-13c4-45026-5e0a2851-573e3bd6-5e0a2851
    To: <sip:[email protected]>;tag=z9hG4bK-5e0a2851-57ad8004-5a21ed70
    CSeq: 1 INVITE
    Server: FPBX-
    Content-Length:  0

<--- Received SIP request (715 bytes) from TCP: --->
ACK sip:[email protected];transport=TCP SIP/2.0
From: "Conference Room"<sip:[email protected];transport=TCP>;tag=2c7708e0-341fa8c0-13c4-45026-5e0a2851-573e3bd6-5e0a2851
To: <sip:[email protected]>;tag=z9hG4bK-5e0a2851-57ad8004-5a21ed70
Call-ID: 2c77db30-341fa8c0-13c4-45026-5e0a2851-18df7ab8-5e0a2851
CSeq: 1 ACK
Via: SIP/2.0/TCP;alias;branch=z9hG4bK-5e0a2851-57ad8004-5a21ed70
Max-Forwards: 70
User-Agent: LifeSize Icon 600/LS_RM3_2.9.0 (1982)
Contact: "Conference Room"<sip:[email protected];transport=TCP>
Accept: application/sdp,application/media_control+xml,application/presentation_token_control+xml
Content-Length: 0

Before you tear it apart, I would like to call on expert @jcolp to comment on what that error might mean. That looks funny, as though it was a line from a chan_sip configuration that FreePBX attempted to apply to a pjsip configuration.

The “transport” option in chan_sip specifies the technologies that are allowed. The option in chan_pjsip specifies an explicitly configured transport. It looks like FreePBX logic isn’t correct in that scenario when writing the configuration and is treating it as a chan_sip one.

@SpiceBlend Do you have any reason not to use the latest distro? 7.5.1805 is old. This bug may have been fixed by now.

edit: indeed, the bug was reported and fixed:

Running updates on your machine should be enough (will take a while), as it was fixed in FreePBX 14.

I thought I just downloaded whatever image was on the site under “stable releases”. It was a few weeks ago, maybe I clicked on the wrong thing.
I’ll go ahead and run updates instead of reinstalling, thanks for getting back to me!

This button:

Gives you this link:

Which is FreePBX 15.

Thanks all. Should I update to 15?

I ran updates on the current version and that did fix the issue I was having, though FreePBX was just immediately hanging up on the caller. So instead of calling anonymously I added a trunk, since I didn’t really want to do anonymous calls anyway, I was just doing it for testing.

That worked, but now it’s always connecting with “G.711 (µ-Law)” instead H264, so I’m only getting audio, and I didn’t see any way to enable video codecs on a trunk. Is there some place to turn this on or am I approaching this the wrong way?

It’s an open issue since 2017:

As a workaround, you can add it in the config files manually.

Edit /etc/asterisk/pjsip.endpoint_custom_post.conf.

Add a section like this:


where “Zoom” (a trunk I have setup with that video provider) would be replaced with the name of your trunk (“Trunk Name” in the first tab of trunk settings).

Then run fwconsole r at the ssh command line to apply the setting.

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Great! I’m 99% sure I’m up and running. The web console on my video conferencing units are saying they’re connecting with H264 now, I’m just going to test with my eyes. :smile:
Thanks again for all the help!

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