|Summary:||ASTERISK-12732: After installing PBX, unable to make SIP to PSTN calls most of the times|
|Date Opened:||2008-09-15 03:36:26||Date Closed:||2011-06-07 14:10:01|
|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
; 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*
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!