Call Center


Hi All

Has anyone used Asterisk for a Call Center operation? What I mean is: given a list of phone numbers, can Asterisk dial each number, play a message and accept some DTMF? I ask because I am an employee of a non-profit company based in San Diego, CA. I already evaluated Voicent and Voxeo. The former has expensive licensing terms and the latter is not best suited for a call center. I would appreciate your kind comments.

Thanking you murthy

Asterisk Users 20 hours ago 1 Answer

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 3 days 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 3 days ago 0 Answers

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 4 days ago 7 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 5 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 6 days ago 0 Answers



Hi list

I'm using Asterisk - is there any way to apply (for example) a bandpass filter to Asterisk RTP audio in the realtime audio stream?

I'm looking for a way to (for example) filter out a 50Hz AC hum present in some calls I push through my asterisk.



Asterisk Users 6 days ago 1 Answer

Update Of Dialed Number On Sip Phones



I have a dialplan that search a phone from dialed code, i.e:

mysql table: code:1234 dest: +555133449966

query in odbc function: SELECT dest FROM my_table WHERE code = '${ARG1}'

dialplan: exten => _#7XXXX,1,Set(DESTNO=${ODBC_query_dest_in_table(${EXTEN:2})}) same => n,Dial(SIP/${DESTNO},30,tT) same => n,hangup

I need to update the dialed number in the screen of SIP phones, when query search the number, the Asterisk should update the #7xxxx for the found number. I know that this facility probably should to be supported by the phones. I tried change the parameters like pid, rpid, but no success. In Asterisk, is it possible? Which parameters need I change? And in the…

Asterisk Users 10 days ago 0 Answers

Recording INCOMING Calls


Hi list!

I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf:

automixmon => *3

then, in my dialplan:

exten => 1,n,Dial(SIP/00493511111111,20,RcxX)

Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too.

Could you suggest me what is wrong?

Thanks Luca Bertoncello (

Asterisk Users 17 days ago 3 Answers

How To Call A Group Of Peers All Registered With The Same Login?


Dear Asterisk-Users,

We have one Asterisk end N softphones that will be registered on it.

We need to configure the Asterisk to get the following scenario working well:

Some softphones (N -1) is to be registered using the same login 6001. So, there will be N-1 users 6001 registered in the SIP REGISTRAR (Asterisk). One softphone is to be registered with the login 6000.

The softphone 6000 must call the others N-1 peers. So, the sofphone 6000 will dial 6001 and all the others N-1 peers must ring.

For that, I suspect that some special configuration must be done in the…

Asterisk Users 18 days ago 0 Answers