I need to do the following, how?
If my extension is 500 and I need to call the extension 501, so when dialing 501, then I need to be able to see the name of the 501 (for example, the name was: Mike, so I need to see at my IP Phone that I am calling Mike which is the name of the destination).
* You are viewing the archive for August 6th, 2012
I discover that I have to place the wave files in the /var/lib/asterisk/sounds/custom/
So, can I understand that the only solution I have is to copy the files that are existed in the path /var/lib/asterisk/sounds/en/ to the path /var/lib/asterisk/sounds/custom? Or there is any other solution?
I am using FreePBX and the asterisk version is: Asterisk 1.8.11-cert1
Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below:
I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable versions of the two projects, respectively 3.3.1 and 10.7.0. You can find it at:
The tutorial focuses on how to use Asterisk’s database structure to perform authentication in Kamailio SIP server, along with user location, nat traversal, instant messaging, presence, a.s.o., offloading processing from Asterisk. Asterisk will still handle all the calls, enabling rich telephony such as MoH, transcoding, ring back, IVR, etc.
Reusing as much as possible the Asterisk database makes the architecture presented in the tutorial easy to be applied to existing installations, without losing management interfaces or other admin tools.
Hope it is useful for many folks out there.
question about register refresh time.
One of our supplier had a maintenance work on sat 4 Aug which was replacing the production server for an Asterisk 1.4 running version.
We have few Asterisk connected to them (version 1.6, 1.8 and 1.10) with register Username and Passwd. After the new server came up, no one of our Asterisks get registered back, which means no calls -incoming and outgoing- at all since this date
A sip show registry this morning (Mon 6 Aug) show us following:
Host dnsmgr Username Refresh State Reg.Time sip.domain.com:5060 N MyUser 105 No Authentication Sat,04
Aug 2012 16:55:37
A simple sip reload made thinks working again.
Why our Asterisks didn’t get back for registration, refresh register time being the standard 120 seconds?
Thanks for your explanation