Insecure Meaning.


Hi all,

When configuring an extension on Asterisk we use the Syntax "Insecure=very " or "Insecure=port" etc. I did some research on Internet and I found that this is used to authenticate the peers, based on their IP/port. But I couldn't understand what's the difference between them.

The following page gives a simple explanation :

insecure=port ; Allow matching of peer by IP address without matching port number insecure = no; Normal IP-based peers matching and authentication of incoming INVITE. insecure=very ; To allow registered hosts to call without re-authenticating insecure=port,invite ; (both).

Can someone provide more details about this, or any…

Asterisk Users 9 hours ago 0 Answers

CEL Eventtime Incorrect, But CDR Times Are Correct -


Hi list

I have a huge problem with a Asterisk instance not logging CEL events with the correct eventtimes.

I'm logging via ODBC to MariaDB 15.1 Distrib 10.0.20-MariaDB

I'm logging into a MyISAM table.

If I start the Asterisk instance, logged times are correct, but the longer the box runs the more the eventtime in the CEL rows created by Asterisk via ODBC drift backwards.

E. g. the clock on the server says 08:15 for example (I enter the date "command" in the terminal) and if I run a query and check the most recent CELs immediately and about ten minutes after startup, they…

Asterisk Users 11 hours ago 0 Answers

How To Configure Through The GUI 35 Cisco Ip Phones -spa502g



I am new to Asterisk and VOIP so I am trying to find a decent howto guide on setting up cisco ip spa502g VOIP phones.

I have found a interesting document on Cisco website but unable to access it. -- I have contacted them for access and waiting on their reply.

Can someone please suggest some other guides that will assist me.

Thanks for the help,


Asterisk Users 17 hours ago 1 Answer

Windows Asterisk Help


Hi All, Downloaded latest version of Asterisk from and installed on Windows 7. Here is my sip.conf [general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = ; IP address to bind to ( binds to all)srvlookup = yes ; Enable DNS SRV lookups on outbound callscontext=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip= localnet= nat=yes maxexpiry=15minexpiry=14;rtautoclear=no;autofallthrough=yes register =>16194077214:<@ [authentication][3000]type = friendcontext = defaultusername = 3000host = dynamicmailbox = 3000dtmfmode = rfc2833[3001]type = friendcontext = defaultusername = 3001host = dynamicmailbox = 3001dtmfmode = rfc2833 [3002]type =…

Asterisk Users 1 days ago 6 Answers

Queues Don't Follow Dialplan If No Members Are Registered



I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf: exten => s,1,Queue(myqueue,rtnC,18) same => n,Background(user_unavail) same => n,WaitExten(10) exten => 1,1,Voicemail(1111@my-vm,s)

This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a voicemail. This works well when at least 1 member is registered in the queue, however if no members are registered in the queue, the Queue() call never seems to return, and thus the remaining steps in the dialplan never execute. How can I correct this behavior…

Asterisk Users 2 days ago 4 Answers

Re-invite Update Dialog


I don't know if this is something asterisk can do at the moment but on my setup, it does not.

What I intend to do is, while a client is in a call, it will send an in-dialog re-invite to asterisk (after changes on the client i.e. IP address). Asterisk should handle this and update internal dialog. when the other party hangs up, BYE will be sent to the new IP.

in my setup, asterisk still sends BYE to the old IP.

Is this something we can already do? or possible to add?

Kelvin Chua

Asterisk Users 2 days ago 0 Answers

No Audio On SIP Over WebRTC


I'm following this tutorial ( to deploy WebRTC support but I'm having an issue with RTP when the WebRTC softphone is behind NAT.

In my scenario, the Asterisk server is running a public IPv4, and the softphone is behind NAT. I can register and make a call normally, but I don't get any audio in neither way (Asterisk/softphone and softphone/Asterisk). Using the very same config files but having the softphone and Asterisk on the same network it works fine.

Any tips on how to solve this? Here's my relevant files.

*;sip.conf:* [general] udpbindaddr= realm.201.0.106 ;replace with your Asterisk server public IP address…

Asterisk Users 3 days ago 0 Answers