Summary:	ASTERISK-12732: After installing PBX, unable to make SIP to PSTN calls most of the times
Reporter:	sunilgorle (sunilgorle)	Labels:
Date Opened:	2008-09-15 03:36:26	Date Closed:	2011-06-07 14:10:01
Priority:	Trivial	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_zap
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
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.
Comments:	By: Leif Madsen (lmadsen) 2008-09-15 12:45:29 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!