[asterisk-bugs] [Asterisk 0013482]: After installing PBX, unable to make SIP to PSTN calls most of the times

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13482 
====================================================================== 
Reported By:                sunilgorle
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13482
Category:                   Channels/chan_zap
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-15 03:36 CDT
Last Modified:              2008-09-15 12:45 CDT
====================================================================== 
Summary:                    After installing PBX, unable to make SIP to PSTN
calls most of the times
Description: 
I have a TDM 400 card installed in a system that has asterisk-1.4.
I connected one PBX line to TDM 400. 
one PSTN line connected PBX.

Now, when I call a mobile/land line number from SIP, most of the times
calls are not going.

Asterisk console shows that its sending DTMF digits. but calls are
notgoing.

my zapata.conf is ::

;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
;        group => <trunkgroup>,<dchannel>[,<backup1>...]
;
;        trunkgroup  is the numerical trunk group to create
;        dchannel    is the zap channel which will have the
;                    d-channel for the trunk.
;        backup1     is an optional list of backup d-channels.
;
;trunkgroup => 1,24,48
;
; Spanmap: Associates a span with a trunk group
;        spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
;
;        zapspan     is the zap span number to associate
;        trunkgroup  is the trunkgroup (specified above) for the mapping
;        logicalspan is the logical span number within the trunk group to
use.
;                    if unspecified, no logical span number is used.
;
;spanmap => 1,1,1
;spanmap => 2,1,2
;spanmap => 3,1,3
;spanmap => 4,1,4

[channels]
;
; Default language
;
;language=en

busydetect=1
busycount=7

relaxdtmf=yes
callwaiting=yes

; don't wait two seconds before answering calls
;usecallerid=no
;
; Support Caller*ID on Call Waiting
;
;callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
;
; Support call forward variable
;
cancallforward=yes
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel
; when the circuit path is entirely TDM.  You may, however, reverse this
; behavior by enabling the echo cancel during pure TDM bridging below.
;
echocancelwhenbridged=yes

; In some countries, a polarity reversal is used to signal the disconnect
; of a phone line. If the hanguponpolarityswitch option is selected, the
; call will be considered "hung up" on a polarity reversal
;
hanguponpolarityswitch



; Logical groups can be assigned to allow outgoing rollover.  Groups
; range from 0 to 31, and multiple groups can be specified.
;
group=1
; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is
ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#.  For simple offices,
just
; make these both the same
;
callgroup=1
pickupgroup=1

;
; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
;
immediate=no
; For fax detection, uncomment one of the following lines.  The default is
*OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

#include "zapata-channels.conf"
usecallerid=yes
cidstart=ring
cidsignalling=dtmf


Please help me out to solve the proble.
====================================================================== 

---------------------------------------------------------------------- 
 (0092513) blitzrage (administrator) - 2008-09-15 12:45
 http://bugs.digium.com/view.php?id=13482#c92513 
---------------------------------------------------------------------- 
This issue appears to be a configuration issue, and I would encourage you
to utilize the help channels available to you. Please see the
asterisk-users mailing list, or the #asterisk channel on the Freenode IRC
network at irc.freenode.net

Thanks for using Asterisk! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-15 12:45 blitzrage      Note Added: 0092513                          
======================================================================