FreePBX 13 CHANSIP Port Error

I am a web developer who has setup a Asterisk / FreePBX phone system in my office. Some of the decisions I have made may not make sense.

I setup the system and it was working fine but decided to upgrade to Asterisk 13. That is when my problems started.

On my FreePBX dashboard I get the following error message:

“CHANSIP Port Moved”
“Chansip was assigned to the same port as pjsip for UDP traffic. The Chansip port has been changed to 5062”

See screenshot below:

The “Resolve” link URL is “http://192.168.1.221/admin/1” This link goes to a 404 page.

I have not been able to resolve this issue. I have having a few issues that I believe to be related to this error but I am not certain.

  1. I cannot update the timezone on my phones
  2. I cannot add new phones. They will not register.

The phones added to my system before the Asterisk update continue to function as before.

I suspect the “port” issue is the root of my problems. Attempts to debug or change the ports have been unsuccessful.

I would really appreciate some help and advice. Below are some relevant settings (I think):



Version from Dashboard:
FreePBX 13.0.190.2 ‘VOIP Server’



Settings → Advanced Settings → Dial Plan and Operational
SIP Channel Driver = both



Settings → SIP Settings → Chan SIP Settings → Advanced General Settings → Bind Port
Bind Port = 5160



Settings → SIP Settings → Chan PJSIP Settings → 0.0.0.0 (udp) → Port Listen On
Port to Listen On = 5060


Thank you,

Eddie

It sounds largely like coincidence to me.

Your resolve link points to a resolved issue. Clear the error and you should be fine.

If your phones connect, you are using the right port. I’m gonna guess you are using PJ-SIP for your phones, which makes the 5060 port selection make sense.

Adding new phones has nothing to do with your SIP/PJSIP settings. You will need to explain what error you are seeing when you try to add new phones.

Are using using one of the EPM modules? If so, you should be able to add the phones there. If not, you need to configure the phones to talk to the server. Username and password are obvious sticking points, but without more information (e.g., where does it hurt) we’re not going to be able to help you with this much.

the message is 5 months old

@cynjut and @bksales thank for responding so quickly.

As for the port error. Yes it was old and when I clear it, it just comes back. Since yesterday I changed:

Settings -> Advanced Settings -> Dial Plan and Operational -> SIP Channel Driver to SIP from BOTH

Then I changed the bind port 5060. After I did that I cleared the error and it’s gone now.

I am using Endpoint Manager. What I can’t get to work is provisioning via endpoint manager. It worked months ago and has not worked since. I think this is now my main problem.

I tried, for the first time to connect a phone by typing in the shared secret. That worked. If I enter the secret directly in the phone it will function.

When I point my phone to the provisioning server it tried to provision, reboots, and just says “no account registered.” I’m using Grandstream GXP2130 phones. Again if I enter the secret it works. If I do the provision it no longer works.

I am running “SIP SET DEBUG ON” asterisk console and I can see what appears the phone trying to register. I’m not sure what to look for so I apologize without giving you a more detailed error. I suspect that is what you need.

Please let me know what additional information will be helpful. Again, thanks ahead of time.

Eddie

check how you have the template set. grandstream by default uses http for provisioning.

@bksales It’s interesting. I’ve started to look at if TFTP is working anymore.

I’ve set the TFTP log to verbose and I’m tailing the log. When I ask the phone to provision and / or download firmware I can see the phone request the files. For examples I’ll see requests for:

ring1.dat
ring2.dat
ring3.dat
gxp2130fw.bin

It seems like it’s hitting the TFTP and requesting the files but I’m not sure it’s actually getting the files. I’ve read that TFTP connects on port 69 but then users other ephemeral ports to actually transfer data.

The thing that’s leading me to think this is a TFTP problem is that I started a TFTP server on my Mac and put the firmware file on there. I pointed my phone at my map and hit upgrade and it worked.

Any reason why TFTP work work for a while on my Asterisk server and then stop? Also I realize I’m making some logical leaps here. If you disagree with the path I’m on, I’m totally open to that.

@bksales I’m making progress. I’ve got TFTP working. The phones are picking up the config now. The problem now is that after the phones pick up the config and reboot they don’t have any accounts registered. Any thoughts?

i am assuming that you have the global setting correct and that the template settings for the grandstream is correct. in particular the registration server?

Bob,

Thanks for the response. I think I got everything fixed. There were a number of things wrong. I’m not sure how things were broken but it really had to do with the config files and how they were being built. First off, I didn’t realize that when you changed the template that you need to select “Save and Rebuild.” I realized also that sometimes when I add an extension the config was not building. I’d save it and then see that no cfg[macaddress] file existed in the tftpboot directory. For some reason sometimes when you fill out the extension -> other fields and press save they just go blank.

I feel like I’ve learned a lot about asterisk, freepbx, and linux this week. Thanks for your help.

Eddie