When authentication times out that is one thing, but when it just fails like below Asterisk by default will not re-register until you the admin reload the sip or asterisk server:
voipserver*CLI> sip show registry
Host dnsmgr Username Refresh State&........
Downloading and compiling from source to get the latest version of Asterisk is really simple with this guide.
apt install gcc make g++ libedit-dev uuid-dev libjansson-dev apt install libxml2-dev sqlite3 libsqlite3-dev
tar -zxvf asterisk-16-current.tar.gz
If you get this error change y........
Normally an ls will just produce the actual contents of the current or target directory eg.
But what if you need to find the full or relative path to another program that cares whether that be zip or etc.?
You need the "-d" switch and the asterisk inside the actual directory.
ls -d mydir/*
This problem seemed to happen recently but was likely causing issues before where the phone(s) do not ring.
Now there are a few reasons why this can happen especially if your adapter has DND mode enabled (disable it).
However that wasn't my issue and Ionly figured it out the other day when by fluke if you're on the phone (making a call) then calls will come in.
That's when Ifigured out the solution:
This likely app........
yum -y install fail2ban
enabled = true
filter = asterisk
action = iptables-multiport[name=asterisk-tcp, port="5060,5061", protocol=tcp]
sendmail-whois[name=Asterisk, email@example.com, firstname.lastname@example.org]
logpath = /var/log/asterisk/messages
This example involves an Aterisk message log of about 26GB, but with any server it usually does not get deleted until the server is stopped/restarted:
asterisk 13729 root 6w REG 0,41 27277943090 59097971 (deleted) /var/log/asterisk/messages
So if you've deleted a bunch of large logs, make sure you restart the server for them to regain your space.
It's really that simple, though some say "extensions reload" but it doesn't work for me, perhaps it depends on the version of Asterisk.........
This happened to a customer Asterisk server and it somehow found the ID of the registration account to the upstream SIP server and was railing connection attempts (it filled up the console and there were literally thousands per second). Basically this caused all incoming and outgoing calls to fail.
It was a temporary fix but the solution was to block that specific IP, it's hard to stop it 100% because the customer needs the default SIP port.........
NOTICE chan_sip.c: Registration from 'user ' failed for '192.168.5.25' - Peer is not supposed to register
You have to setup as a "friend" and not "peer".
NOTICE: rtp.c:1808 ast_rtp_read: Unknown RTP codec 100 received from
I've found that trying several times may work, but I also read changing your Sipura VOIP adapter settings as follows helps (but it still fails for me sometimes):
In adapter change the following under SIP/Advanced
Codec to G711a from G711u
Passthru Method from: NSE to ReINVITE
FAX CED Detect Enable: Yes
FAX CNG Detect........
It's basically free bash shell script available from: http://wpkg.org/email2fax/index.php/Main_Page
Make sure you have the required tools:
Where you can e-mail your Asterisk box and it will fax it to the phone number in the subject line. The good news ends there, it is fairly undocumented and buggy.
Take for example how the documentation mentions you can invoke from the com........
Asterisk FreeBSD compile problemsI couldn't get it to compile without using the following options at compile time:
make install WITHOUT_ZAPTEL=YES WITHOUT_MYSQL=YES WITHOUT_FAX=YES WITHOUT_ODBC=YES WITHOUT_H323=YES[/b:b7d672ee28]Some problems you might have with copying over your Linux config files to FreeBSD is different paths.
[b:f044931f41]For example the etc path for Asterisk on BSD will now be:[/b:f044931f41]
Telus + 2Wire 2700 Router HorribleWell first of all let me say this is the only router/switch that sometimes seems to crash/disconnect computers on the local network.
This device also thought it would be smart to block VOIP packets coming from my Sipura ATA VOIP adapters so I disabled the [quote:cb89ba7bff]"Invalid TCP Flag Attacks (NULL/XMAS/Other)"[/quote:cb89ba7bff] option
Then all of a sudden I couldn't get onto any web pages, the wireless........
Asterisk Agent Login ProblemUsing the AgentCallbackLogin function I couldn't login
I knew I had my agents setup properly in agents.conf so why couldn't I login?
The reason is because agents.conf was missing the [b:994d7a34af][agents][/b:994d7a34af] context!
I just added [agents] above the existing agent declarations and then I was able to login as expected.
It took a lot of figuring out and reading though!........
Asterisk Queue Context ExplainedThis was never explained in voip-info or any other site I read.
It is understood you can escape to a context from a queue and how to specify it.
What is NOT mentioned is that the context= you specify within the queues.conf refers to a [i:882f1e0aee]context that exists in extensions.conf[/i:882f1e0aee][/b:882f1e0aee]
This will save you headaches if you need to escape from the queue :)........