Summary:ASTERISK-12702: In SIP to PSTN call, CDR disposition is ANSWERED always
Reporter:Ramaseshi Reddy Kolli (ramaseshi)Labels:
Date Opened:2008-09-09 08:26:43Date Closed:2011-06-07 14:02:44
Versions:Frequency of
Description:I have installed a TDM 400 card in a system that has installed Asterisk sucessfully.

I connected one PSTN line to TDM 400 card. Now, from a sip phone, i can call any outside pstn landline or mobile number.

Problem is, from a sip phone when I dial a mobile/landline number and I hangs up before other party answers the call, in CDR I got disposition as ANSWERED, instead   of N0-ANSWER or BUSY.

Please let me know, Is there any way to get proper disposition.
Comments:By: David Woolley (davidw) 2008-09-10 06:20:19

This looks like a configuration issue (or, less likely, feature request), which you should follow up in the support forum, or using IRC.  The key phrase is "answer supervision".

By: Ramaseshi Reddy Kolli (ramaseshi) 2008-09-15 03:08:40

my zapata.conf is, as follows.

Am I missing anything here..?  Please suggest me .

; Zapata telephony interface
; Configuration file

; 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

; Default language



; don't wait two seconds before answering calls
; Support Caller*ID on Call Waiting
; Support three-way calling
; Support flash-hook call transfer (requires three way calling)
; Support call forward variable
; 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.

; 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

; Logical groups can be assigned to allow outgoing rollover.  Groups
; range from 0 to 31, and multiple groups can be specified.
; 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

; Specify whether the channel should be answered immediately or
; if the simple switch should provide dialtone, read digits, etc.
; For fax detection, uncomment one of the following lines.  The default is *OFF*

#include "zapata-channels.conf"

By: Leif Madsen (lmadsen) 2008-09-15 12:47:53

I agree with davidw here. This appears to be a configuration issue, and thus 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!