Thanks Ron. I have had my chan_dahdi.conf file set as follows with the same result.

[trunkgroups]
[channels]
switchtype=national
usecallerid=yes
callerid=asreceived
cidsignalling=smdi
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
usesmdi=yes
smdiport=/dev/ttyS0
signalling = em_w
immediate = no
group = 1
channel => 1-3



Bill Dunn



----- Original Message ----- From: Ron Bergin
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, July 06, 2012 12:34 PM
Subject: Re: [asterisk-users] DAHDI DTMF problem?


Bill Dunn - VCI Internet Services wrote:
I have an Asterisk server configured to run as voicemail with a T1 and
SMDI.
It has 1.6.1.6 (dahdi 2.1.0.4) and Centos 5.6 and has worked great for a
few
years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on
Centos 5.8

The problem I am having appears to be related to DTMF detection. When the
test phone number is called (2704083000) Asterisk only receives a portion
of
the dialed number. It varies as to what numbers are detected. Sometimes it
sees a single digit, sometimes 3 or 4 of the digits of the dialed number.

When I compare this to the old server the debug below is similar but there
isn't any mention of the "sig_analog.c" lines shown below.

I am told the T1's on the old server and the new server are configured the
same. I can make outgoing calls on the T1 from Asterisk.

Can someone give me a clue as to what could be causing this?


Bill Dunn


Try setting:
relaxdtmf=yes

We used to have that same problem on most of our servers.  Setting
relaxdtmf to yes solved the problem for us.

--
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.




--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to