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 doesn't working (I'm using Digium AEX800B PCI Express card). I
can hear my self talking on the phone. How to solve this? I think I need to
recompile dahdi 2.6.1 with OSLEC support? how? [root@callcenter ~]# dahdi_cfg -vvv
DAHDI Tools Version - 2.6.1 DAHDI Version: 2.6.1
Echo Canceller(s): HWEC
Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01)
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
use it, is there any way to disable it?
Jonas Kellens wrote:
> Will ODBC become the default then ? As far as I'm 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 And here is a good link to get you started: http://nerdvittles.com/?p=604 Doug
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 attached voicefile only contains the
the message length written in the email is ok.
If I go to the voicemailbox during the recording then I can se the files
grow to the filesize i would expect, looks like everything is ok until then.…
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 caller's ID for SOME
of the calls (30-50% success rate). I am using PBX | monitoring menu to see
this report. As far as I remember, I dont modify any settings that related to caller ID,
but few days ago (I dont remember the exact date), I modify the rxgain and
txgain value in chan_dahdi.conf.
The inbound caller ID doesn't display…
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 of the caller), yes and very were removed in v1.6.x. Note: yes/very are deprecated in 1.8
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: Someone calls extension
2408, but there's no phone attached to 2408 it should fall back and
ring at 2400.. How do I setup asterisk to find out if there's a phone attached to an
internal number if not ring another extension?
I have problems starting dahdi.
dahdi_cfg -vvv allwasy comes back with:
DAHDI Tools Version - 18.104.22.168 DAHDI Version: 22.214.171.124
====================== SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01)
Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02)
Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I searched the internet but could not yet find…
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.
other way to play a different sound file in place of vm-login? I have already replaced the default file and named it the same vm-login.x
but still I am only able to play one file, not a different file depending
on the VM context I send the caller to. I am sure someone has…
> I am facing an issue with Peer registration in my asterisk server .
> I am using asterisk version 126.96.36.199 and using SIP real-time
> architecture.when i am doing registration it registered fine on asterisk
> as peer is available in Database.
> But now i am doing 'sip reload' or 'reload' due to some reason my peer
> registration is going out and i cannot able to call that peer even though
> in SIP client it shows me 'registered'.
> Can any body elaborate…