keywords: ip pbx voip gateway gsm gateway

×

Notice

The forum is in read only mode.
× Questions about A400/800/1200 Analog Interface Card

Lost digits when calling

12 years 10 months ago #7006 by carlo
Dear all,
I've set-up an Asterisk/Freepbx server with an A800P card, four internal lines and four voip lines to connect to the world.
I've had no particular problem in compiling the dahdi module and get everithing working but I have a problem while making calls:
from two of the four phones, when I compose a number, some digits are lost and of course I hear the asterisk message saying the the call cannot be completed (eg I dial 0661663045 but in the logs I see the number was received as 066163045, thus missing one digit).
Could anyone suggest how to solve this problem? I've seen there are three setting related to the gain (rxgain, txgain, cid_rxgain) but I'm unsure how to set them.

Thanks and best regards,
Carlo
12 years 10 months ago #7007 by jun
hello:
to solve the callerid problem, you have to check few things:
1) set this in chan_dahdi.conf:
cidstart=ring
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
; (see doc/India-CID.txt)
;

cidsignalling=dtmf
; Type of caller ID signalling in use
; bell = bell202 as used in US (default)
; v23 = v23 as used in the UK
; v23_jp = v23 as used in Japan
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
;
relaxdtmf=yes
usecallerid=yes

2) load driver with your country code : modprobe wctdm opermode=YOUR COUNTRY

3) check the indication.conf, set it to your country.

4) change echo_can oslec to echo_can mg2 in the "etc/dadhi/genconf_parameters

If the problem still not solved, pls provide SSH and your telephone number.


Regards,

Jun.Liu



MSN: This email address is being protected from spambots. You need JavaScript enabled to view it.
G-talk: This email address is being protected from spambots. You need JavaScript enabled to view it.
QQ : 1049787481

Quick Support: http://wiki.openvox.cn/index.php/OpenVox_Quick_Support
12 years 10 months ago #7008 by miaolin
which version asterisk you are using? maybe have to modify dsp.c
12 years 10 months ago #7010 by carlo
I'm using version 1.8.2.3.
Will try suggestions above and will let you know.

Thanks for now,
Carlo
12 years 10 months ago #7011 by jun
Time to create page: 0.037 seconds
Powered by Kunena Forum