In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesnt working (Im using Digium AEX800B PCI Express card). I can hear my self talking on the phone. ..
Guys, is there any way to disable all Asterisk Features? We are having false dtmf detections and randon calls being put on-hold and suspect that dtmf features is the cause. Changing features.conf aparently keeps the default options. Since we dont ..
Jonas Kellens wrote: > Will ODBC become the default then ? As far as Im aware, yes. You need to make sure that unix-odbc development libraries are installed. Under Mandriva, Mageia: urpmi unixodbc-devel Ubuntu/Debian: apt-get install unixodbc-dev ..
I hope for a hint on this issue. I had a voicemail running on ast release 1.6.2 latest which i upgraded to 1.8.11 latest release. during this process I did add a couple of fields like minsecs and maxsecs. I do now get empty emails where the attac..
This is a very strange problem (at least for me). I just realized that started from April 20th 2012 every inbound call is from unknown. Prior that, asterisk succesfully displayed the caller callers ID for SOME of the calls (30-50% success rate). I..
This setting specifies how to handle connections with peers. By default Asterisk will authenticate all connections (this is the same as insecure=no). The parameters invite and port were added in v1.2.x (and are to be used when you trust the IP..
Hi I was asked by our development departement to setup asterisk in a manner that if someone calls an extension in the department that was was only configured, but a handset was never attached to it to fall back to a default extension. For example: Some..
all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version – 184.108.40.206 DAHDI Version: 220.127.116.11 Echo Canceller(s): Configuration ====================== SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Chan..
Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? By default the sound file is vm-login.. Is there a way to specify which sound file is played per context or some other way to play a differ..
> I am facing an issue with Peer registration in my asterisk server . > > I am using asterisk version 18.104.22.168 and using SIP real-time > architecture.when i am doing registration it registered fine on asterisk > as peer is available in Database. > > ..