Asterisk Debug Extension Troubleshooting Tips

Table of Contents

Sometimes your system may display a message indicating debug extension asterisks. There can be several reasons for this problem.

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When a specific person needs to debug a problem or gather additional information about various PBXware issues, Asterisk’s own CLI can come in handy. It prints many entries that are not visible in PBXware CLIR releases for every call made to the console and a few more situations.

Before you can see any of the messages in the command line interface, you need to SSH into the system with an asterisk using the ssh command (if you are using Linux on your computer) or putty or similar software if you are using PC / MAC. …After that, you can enter the Asterisk CLI, which will display the following Asterisk command:

-rvvvv

, where V is the verbosity level of the CLI.

Once inside, you will see a ton of useful information about any movement in the system, but related to Asterisk. You’ll see:

Phone calls
Peer Notification
Update registrations
Subscribing to full body components (internal lines, connectorse lines, IVR, etc.)

If, for some unexplained reason, you have an unexplained problem, such as when Asterisk won’t start, you can try running various switches with the cli, which should provide application-specific debug information, which also means boot sequence. Connecting to the database, attempting a new request, etc.

 asterisk -rdddd

where the nintendo ds lite count determines the verbosity of these debug messages.

If the command in front of you tells you that Asterisk isn’t just running, try starting it like this:

 asterisk -cdddd

If you don’t need to be in the CLI or just need to run a command without worrying about the result, you can do it using the Asterisk command with the following parameters:

 asterisk -rx 'restart now'

The first important commands you should be aware of is the SIP debug suite, including commands useful when you need to view the SIP data stream installed with Asterisk. Simple commandsa – enable SIP debugging for a SIP phone using:

 INSTALL PEER PHONE_EXT DEBUGGING

where PHONE_EXT is an extension / telephone number in the system. PHONE_EXT must be the name of the trunk so that your company can see all SIP traffic on that particular trunk.If for some reason most extensions or channels are not registered with a peer IP, which is usually unknown to the star, the above command usually does not work and the CLI does not display any SIP messages. In such a case, in cases where you know the IP address from the intended traffic, it is better to reverse the direction while debugging for that particular IP address than if it is:

 SIP SET DEBUG IP PEER_IP

where PEER_IP is the IP address of the specialist who should send data traffic to the specified extension / trunk.

When you’re done debugging, the sip stream should disable SIP debugging, as starting it will overload the CLI results and you might miss other important system information. To disable Debug-Sip, run this command:

debug extension asterisk

 SIP SET DEBUG OFF

If for some pBecause you are experiencing sound problems, some statements may indicate incompatibility of codecs on the device. In such cases, with the correct CORE command, you can see some possible translation paths in Asterisk:

SEE TRANSLATION

This command trains an array of possible transcoding / streaming codecs channels that can be monitored on the device. If you see the – sign, it means that transcoding between the mentioned codecs is often not possible. In most cases, this problem is caused by a missing codec.

Besides the fact that you probably manually made changes to the Asterisk dial plan, you need to reload the complete configuration of a specific Asterisk subsystem, which can be done with a simple command:

CHARGING

This will reload some configurations related to the Asterisk telephony core.

If restarting Asterisk was not enough for the changes you made, or for most other reasons, you can continue Asterisk entirely by doing the following:

 RESTORE NOW

ImplementationAsterisk core PBXware uses AGI to manage call routing recommendations through Asterisk, but for a variety of reasons, you may tend to change certain aspects of call routing. Uses the default dialing plan star to forward calls to various second locations. Dialplan information can be found in some of the configuration files (see Asterisk White Papers).

For example, if you change the dial plan from the extensions.conf file, reinstall the Asterisk configuration. You will then need to review the recruitment plan to see which experts claim your changes have been applied for this.

 DIALPLAN SHOW

 RECRUITING PLAN SHOW YOUR RECRUITING PLAN

This command only offers the YOUR_DIALPLAN portion of this list.

If you have verified the availability of telephones / trunks, you can print a list of these peers on the system:

 SIP SHOW STAFF Where PER SHOW IAX2

The first command creates a list of SIP peers connected to the system, with additional information such as statusnetwork and the IP address from which clients connect. The second command does the same for IAX peers.

If calls through PRI cards don’t work or your business is having problems, you can enable PRI debugging in a specific Asterisk CLI:

debug extension asterisk