[Asterisk-Users] T1 E&M false busy after dial

jltaylor jltaylor at metrotel.net
Sun Apr 24 15:43:09 MST 2005


If Feature Group B signaling is working properly (and you have Feature Group
B trunks), then
to reach your Asterisk box you would dial from the PSTN (1)+950+WXXX {W is 1
or 0 based on the number assigned to you}.

If you are dialing "out" {terminating where you look like the carrier} on
FGB then it depends on if you are connected to an Equal Access End Office or
a Access Tandem.

Are you sure about the Feature Group B thing or do you have trunks that just
require MF signaling?

If you want MF, you might try the "featdmf" setting, however, the telco
needs to know that you want FGD.
AND...
If you are connecting to an Access Tandem instead of and End Office, then
the "featdmf" in Asterisk will not work.
I have submitted a request for a quote to Digium to modify the code to make
this work properly.

Likewise, true FGB "terminating" (where it looks like you are the carrier)
works through an Access Tandem and the additional code is missing for that
also.

Take out the featb and add:
em_w

This will let you see if just plain old DTMF works.

James Taylor
903-793-1956



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of John Ackley
Sent: Sunday, April 24, 2005 4:44 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] T1 E&M false busy after dial


TE101P card T1 E&M trunk to telco

on a SIP->PSTN call, after dial
SIP phone hears two seconds busy tone (1) then ring tone

how do we get rid of busy tone?


(1) two second busy
(480+620/500 0/500 480+620/500 0/500)
---------------------------------------------------

extensions.conf:
;
; dial-out to the PSTN with 7 digits
;
exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXXXXX,n,Hangup()

zaptel.conf:
span=1,1,0,esf,b8zs
e&m=1-24
loadzone = us
defaultzone=us

zapata.conf:
[trunkgroups]
[channels]
language=en
context=default
signalling=featb
usecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=8
channel => 1-24



--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.18 - Release Date: 4/19/2005

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




More information about the asterisk-users mailing list