Hello everyone,

Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi
2.6.? (and 2.5.?).

When you specify any cadence in an app (Dial, Queue) then caller id does
not work.

For instance with the default cadences (everything commented out in
chan_dahdi.conf) :

Dial(DAHDI/54) caller id works

Dial(DAHDI/54r1) caller id does not work (even for r1)

I just found this issue did not have time to investigate further. Can
anyone else verify that this is true for tonezones other than 13 (gr) which
I am using?

Cheers,

Panos

Hi,

I’d like a single queue to announce the caller’s position, etc., in more than one language without user interaction. ie. announce position in English then in French then in Spanish

Is this possible (without ivr)?

Can anyone please give me a Queue cmd example with ‘announceoverride’?

Thanks,

Vieri

Hi

I’m not agree problem could be cause from IRQ setting, I think in that way problem should be more unstable, moreover no voice communication problem with DAHDI service start up.
Best regards

Mc GRATH Ricardo
E-Mail mcgrathr@mail2web.com

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 on the log and on the phone.

I’m running asterisk 1.8.7.0
FreePBX 2.8.1
FXO card using Digium AEX800B

The caller ID is display well on the phone If I connect the phone directly
without connecting to asterisk (just for testing purpose)

This is my chan_dahdi.conf

[trunkgroups]

[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=8.0
txgain=8.0
callgroup=1
pickupgroup=1

;Uncomment these lines if you have problems with the disconection of your
analog lines
;busydetect=yes
;busycount=3

immediate=no

#include dahdi-channels.conf
#include chan_dahdi_additional.conf
chan_dahdi.conf (END)

This is my dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Fri Mar 30 22:32:16 2012
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;

; Span 1: WCTDM/0 “Wildcard AEX800 Board 1″ (MASTER)
;;; line=”1 WCTDM/0/0 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line=”2 WCTDM/0/1 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line=”3 WCTDM/0/2 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
context=default

;;; line=”4 WCTDM/0/3 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

;;; line=”5 WCTDM/0/4 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 5
callerid=
group=
context=default

;;; line=”6 WCTDM/0/5 FXSKS”
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 6
callerid=
group=
context=default

Any help would be very appeciated.

Thanks and best regards,
Anam.

What is directmedia?

To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting is YES.

When to use it?

If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off.

If you want to allow media path redirection (reinvite) only when the peer where the media is being  sent is known to not be behind a NAT (as the RTP core can determine it based on the apparent IP address the media arrives from), set this to nonat.