[asterisk-gui] Can't Receive a call

Thu Jul 16 12:05:05 CDT 2009

Here is System and Users:

/etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Mon Jul 13 17:53:33 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4

# Span 2: WCTDM/4 "Wildcard TDM400P REV E/F Board 5"
fxsks=5
echocanceller=mg2,5
fxsks=6
echocanceller=mg2,6
fxsks=7
echocanceller=mg2,7
fxsks=8
echocanceller=mg2,8

# Span 3: WCTDM/4 "Wildcard TDM400P REV E/F Board 5"
fxoks=9
echocanceller=mg2,9
fxoks=10
echocanceller=mg2,10
fxoks=11
echocanceller=mg2,11
fxoks=12
echocanceller=mg2,12

# Global data

loadzone        = us
defaultzone     = us
/etc/asterisk/users.conf
;!
;! Automatically generated configuration file
;! Filename: users.conf (/etc/asterisk/users.conf)
;! Generator: Manager
;! Creation Date: Tue Jul 14 16:53:55 2009
;!
;
; User configuration
;
; Creating entries in users.conf is a "shorthand" for creating individual
; entries in each configuration file.  Using users.conf is not intended to
; provide you with as much flexibility as using the separate configuration
; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the
; simple task of adding users.  Note that creating individual items (e.g.
; custom SIP peers, IAX friends, etc.) will allow you to override specific
; parameters within this file.  Parameter names here are the same as they
; appear in the other configuration files.  There is no way to change the
; value of a parameter here for just one subsystem.
;

[general]
;
; Full name of a user
;
fullname = New User
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Set voicemail mailbox 6000 password to 1234
;
vmsecret = 1234
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = yes
;
; Create H.323 friend
;
;hash323 = yes
; Create manager entry
;
hasmanager = no
;
; Set permissions for manager entry (see manager.conf.sample for documentation)
; (defaults to *all* permissions)
;managerread = system,call,log,verbose,command,agent,user,config
;managerwrite = system,call,log,verbose,command,agent,user,config
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
vmexten = 7000

;[6000]
;fullname = Joe User
;email = joe at foo.bar
;secret = 1234
;zapchan = 1
;hasvoicemail = yes
;vmsecret = 1234
;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
;callwaiting = no
;context = international

[trunk_1]
trunkname = CBTrunk  ; GUI metadata
busydetect = yes
busycount = 3
busypattern = 500,500
ringtimeout = 8000
answeronpolarityswitch = no
hanguponpolarityswitch = no
callprogress = no
progzone = us
usecallerid = yes
cidstart = ring
pulsedial = nodisallow = all
allow = all
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 1
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 2
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 3
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 4
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 5
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 6
gui_volume = 3  ; GUI metadata
signalling = fxs_ks
gui_fxooffset = 0  ; GUI metadata
rxgain = 2
txgain = 0.0
channel = 7

[6001]
username = 6001
transfer = yes
mailbox = 6001
call-limit = 100
type = peer
fullname = Margie Mogle
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6001
hasvoicemail = yes
vmsecret =
email =
threewaycalling = no
hasdirectory = yes
callwaiting = no
hasmanager = yes
hasagent = yes
hassip = yes
hasiax = no
secret =
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
managerread = system,call,log,verbose,command,agent,user,config,originate
managerwrite = system,call,log,verbose,command,agent,user,config,originate
disallow = all
allow = ulaw,gsm,alaw,speex,g726

[6002]
username = 6002
transfer = yes
mailbox = 6002
call-limit = 100
type = peer
fullname = Bob Crandell
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6002
hasvoicemail = yes
vmsecret =
email =
threewaycalling = yes
hasdirectory = yes
callwaiting = yes
hasmanager = yes
hasagent = yes
hassip = yes
hasiax = no
secret =
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
signalling = fxo_ks
flash = 750
rxflash = 1250
autoprov = no
label =
macaddress =
linenumber = 2
LINEKEYS = 1
managerread = system,call,log,verbose,command,agent,user,config,originate
managerwrite = system,call,log,verbose,command,agent,user,config,originate
dahdichan = 9
disallow = all
allow = ulaw,gsm,alaw,speex,g726

[6003]
username = 6003
transfer = yes
mailbox = 6003
call-limit = 100
type = peer
fullname = Brett Ansite
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6003
hasvoicemail = yes
vmsecret =
email =
threewaycalling = no
hasdirectory = yes
callwaiting = no
hasmanager = yes
hasagent = yes
hassip = yes
hasiax = no
secret =
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 3
LINEKEYS = 1
managerread = system,call,log,verbose,command,agent,user,config,originate
managerwrite = system,call,log,verbose,command,agent,user,config,originate
disallow = all
allow = ulaw,gsm,alaw,speex,g726

[6004]
username = 6004
transfer = yes
mailbox = 6004
call-limit = 100
type = peer
fullname = Mike Owen
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6004
hasvoicemail = yes
vmsecret =
email =
threewaycalling = yes
hasdirectory = yes
callwaiting = yes
hasmanager = yes
hasagent = yes
hassip = yes
hasiax = no
secret =
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
signalling = fxo_ks
flash = 750
rxflash = 1250
autoprov = no
label =
macaddress =
linenumber = 4
LINEKEYS = 1
managerread = system,call,log,verbose,command,agent,user,config,originate
managerwrite = system,call,log,verbose,command,agent,user,config,originate
dahdichan = 10
disallow = all
allow = ulaw,gsm,alaw,speex,g726

[6005]
username = 6005
transfer = yes
mailbox = 6005
call-limit = 100
type = peer
fullname = Tim Hughes
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6005
hasvoicemail = yes
vmsecret =
email =
threewaycalling = yes
hasdirectory = yes
callwaiting = yes
hasmanager = yes
hasagent = yes
hassip = yes
hasiax = no
secret =
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
signalling = fxo_ks
flash = 750
rxflash = 1250
autoprov = no
label =
macaddress =
linenumber = 5
LINEKEYS = 1
managerread = system,call,log,verbose,command,agent,user,config,originate
managerwrite = system,call,log,verbose,command,agent,user,config,originate
dahdichan = 11
disallow = all
allow = ulaw,gsm,alaw,speex,g726

[6006]
username = 6006
transfer = yes
mailbox = 6006
call-limit = 100
type = peer
fullname = FAX
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6006
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = yes
hassip = yes
hasiax = no
secret =
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 6
signalling = fxo_ks
flash = 750
rxflash = 1250
autoprov = no
label =
macaddress =
linenumber = 6
LINEKEYS = 1
dahdichan = 12
disallow = all
allow = ulaw,gsm,alaw,speex,g726

[trunk_2]
trunkname = FAXTrunk
busydetect = yes
busycount = 3
busypattern = 500,500
ringtimeout = 8000
answeronpolarityswitch = no
hanguponpolarityswitch = no
callprogress = no
progzone = us
usecallerid = yes
cidstart = ring
pulsedial = no
cidsignalling = bell
flash = 750
rxflash = 1250
mailbox =
callerid = asreceived
dahdichan = 8
context = DID_trunk_2
group = 2
hasexten = no
hasiax = no
hassip = no
registeriax = no
registersip = no
trunkstyle = analog
disallow = all
allow = all
signalling = fxs_ks
channel = 8
cidsignalling = bell
flash = 750
rxflash = 1250
mailbox = 6002
callerid = asreceived
dahdichan = 1,2,3,4,5,6,7
context = DID_trunk_1
group = 1
hasexten = no
hasiax = no
hassip = no
registeriax = no
registersip = no
trunkstyle = analog

>>> Matt Sales <msales at gmail.com> 7/14/2009 05:47 PM >>>
Bob, can you paste your /etc/dahdi/system.conf, users.conf, and extensions.conf? Also if you are connected to analog lines have you run fxotune?

On Tue, Jul 14, 2009 at 11:45 AM, Bob Crandell <bob at assuredcomp.com> wrote:

Hi All,

Thanks to the help I'm getting here I can now make calls and the voice quality sounds pretty good.

An incoming call does not get answered. This is the log:
Connected to Asterisk 1.6.0.9 currently running on PBX (pid = 5068)
Verbosity is at least 6
-- Starting simple switch on 'DAHDI/1-1'
[Jul 14 08:33:01] NOTICE[5692]: chan_dahdi.c:7490 ss_thread: Got event 18 (Ring Begin)...
[Jul 14 08:33:03] NOTICE[5692]: chan_dahdi.c:7490 ss_thread: Got event 2 (Ring/Answered)...
-- Executing [s at DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack
-- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/1-1'
-- Starting simple switch on 'DAHDI/1-1'
[Jul 14 08:33:13] NOTICE[5697]: chan_dahdi.c:7490 ss_thread: Got event 18 (Ring Begin)...
[Jul 14 08:33:15] NOTICE[5697]: chan_dahdi.c:7490 ss_thread: Got event 2 (Ring/Answered)...
-- Executing [s at DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack
-- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/1-1'
-- Starting simple switch on 'DAHDI/1-1'
[Jul 14 08:33:29] WARNING[5704]: chan_dahdi.c:7634 ss_thread: CallerID returned with error on channel 'DAHDI/1-1'
-- Executing [s at DID_trunk_1:1] ExecIf("DAHDI/1-1", "1?SetCallerPres(unavailable)") in new stack
-- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/1-1'

It's says (Ring/Answered)... but it doesn't go to an extension or voice mail.

I have incoming calling rules defined. I've tried directing the calls to a ring group, a voice mail and an extension. No joy.
This is running on openSuSE 11.1 fully patched and running asterisk16-1.6.0.9-73.18.
Asterisk GUI-version : SVN-branch-2.0-r4970

Will you help me track this down?
Thanks

Bob Crandell
Assured Computing, Inc.
541-868-0331
ComputerBase USA
541-349-0404

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