ASL3 node stopped working and SIP connection not working also

AllStarLink
1

Hi All,

For some reason my ASL3 node # 60694 stopped working recently on my Debian 12 system. Also, my SIP phone connection to my Cisco SP525G2 has stopped working also.

I have tried a purge of the ASL files, did a re-install and used backup config files but it is still not working and I have tried the troubleshooting steps as listed at link below also. My node is registering ok though on allstar website.

I have followed the steps listed here: Setting up a SIP Phone - AllStarLink Manual

asterisk -rvvv

pjsip set logger

but there is no output on the Asterisk CLI

I have checked the following also and it is all configured correctly.

port forwarding to 4569 is ok

port 5060 is open

firewall is ok

  • rpt.conf: is ok

  • extensions.conf: is ok

  • iax.conf is ok

  • asl-node-lookup is ok

  • Searchable Node List -ok

My SIP phone has an orange color for allstar connection and its configuration page says not registered)

Does anyone else have any other recommendations/troubleshooting steps to get my setup working again ?

And which logs I can view for any error messages ?

Thanks and 73,

KE4QCM

Thomas

AllstarLink node # 60694

2

Hi All,

I am still having issues with my Allstar node. Does anyone have any troubleshooting tips ?

Thanks,

Tom

KE4QCM

3

Tom, perhaps give everyone some kind of starting point…

If while try running it in the foreground (asterisk -rvvv)

Is asterisk running ? If not you may have an error in your config.

I am not sure how you determined your files are OK. Obviously some typo in them.

Try restarting asterisk from the system prompt and see error messages ?

A good place to start is where you last made changes in editing before last restart.

4

Hi Mike,

yes Asterisk is running. I have reviewed my config files and there are no typos in them.

Can I try a re-install of Asterisk? best steps to do it ?

I have tried a purge of the files and install backup of config but it didn’t resolve it.

Thanks,

Tom

KE4QCM

5

We have a new tool, asl-node-auth-check, currently in "beta" that may help troubleshoot your issue. Have a look at Error connecting to other nodes - #2 by N8EI for more info.

6

Hi Allan,

Thanks,

running asl-node-auth-check I got an error message: Duplicated HTTP and IAX registrations for 60694.

Tom

KE4QCM

7

That means you are duplicate registering for both IAX and HTTP. You only need one. Remove the register line out of iax.conf and restart asterisk.

8

Hi Jason,

Thanks. I removed the IAX registration line in iax.conf but my SIP phone is still not connecting to Allstar. Since I didn’t get any more error messages I think it might be more of a SIP phone configuration issue since my SIP phone (Cisco SP525G2) still won’t connect to Allstar though I have double checked my settings on the SIP phone config page and they appear to be correct.

Tom

KE4QCM

9

So, i guess, is it just a SIP connection that isn't working? You said "ASL3 ... stopped working".

10

Hi Jason,

Here is the output of ‘asl-node-auth-check’

Checking configuration:
Info: rpt.conf has configuration for: 60694, 1999
Info: /etc/asterisk/iax.conf contains no registration lines.
Info: /etc/asterisk/rpt_http_registrations.conf contains 1 registration line(s)
Info: Registrations present for configured node(s): 60694
Error: No registration present for configured node(s): 1999
Error: Cannot continue; reconcile configuration before proceeding

My SIP phone has a green light now but still it can’t connect to Allstar from it. I am not sure if Allstar is working though either ? I have checked my settings on Allstar per Setting up a SIP Phone - AllStarLink Manual and everything looks correct. My node # is 60694 and my ext to dial from SIP phone into Allstar is 1001. 1999 is the node # for an Echolink connection that I had previously configured.

Tom

KE4QCM

11

Anything using the PJSIP system is independent of AllStarLink/app_rpt. I would suggest making sure everything the app_rpt is working using conventional means such as a USB audio interface. There's no reason to create a private node # for EchoLink unless you're trying to segment it off somewhere. Currently the tools don't know how to deal with private nodes - probably something to fix.

12

From my point of view, I can’t distinguish the different problems exactly.

But I will assume your asl is working and just your sip routing is off.

So review in extentions.conf the stanza for the node connection you are dialing.

a stanza ending in something like this

exten => 29285,n,Rpt,29285|P

That is the part that makes the connection.
If by itself, the 'n' should be a number 1 (first instruction)
There must be a first (1) instruction, later instructions can be wildcard n (next)
Many times folks change these/add/remove lines and mess up the stanza.

You will have to watch asterisk in the foreground while dialing and report the error to get better help. But if you look at the dialing stanza, you might figure it out on your own.

You should spend a little time understanding the asterisk dial plan for better understanding when using sip/iax phones. Probably a little tough to grasp at first but once you get it, you will enjoy what you can build out.

You can only connect to local nodes with it or what you have provided routing for.