[asterisk-bugs] [Asterisk 0013482]: After installing PBX, unable to make SIP to PSTN calls most of the times
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Sep 15 12:45:31 CDT 2008
The following issue has been UPDATED.
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http://bugs.digium.com/view.php?id=13482
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Reported By: sunilgorle
Assigned To:
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Project: Asterisk
Issue ID: 13482
Category: Channels/chan_zap
Reproducibility: always
Severity: trivial
Priority: normal
Status: closed
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:
Resolution: no change required
Fixed in Version:
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Date Submitted: 2008-09-15 03:36 CDT
Last Modified: 2008-09-15 12:45 CDT
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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.
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Issue History
Date Modified Username Field Change
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2008-09-15 12:45 blitzrage Status new => closed
2008-09-15 12:45 blitzrage Resolution open => no change
required
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