Summary: | ASTERISK-12702: In SIP to PSTN call, CDR disposition is ANSWERED always | ||
Reporter: | Ramaseshi Reddy Kolli (ramaseshi) | Labels: | |
Date Opened: | 2008-09-09 08:26:43 | Date Closed: | 2011-06-07 14:02:44 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_zap |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
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 [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 answeronpolarityswitch=yes ; 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 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! |