TDM410P no dial tone on fxs

Just installed a TDM410P with two fxo and one fxs ports. The card is properly recognized with all ports, but whatever I do I cannot get a dial tone on the fxs analog port (4) using a standard phone.
I can receive inbound calls via the fxo ports (1&2).

I am using the latest DAHDI and Asterisk:

  • asterisk-1.6.0.6
  • dahdi-linux-complete-2.1.0.4+2.1.0.2
  • asterisk-gui 2.0 svn (20090301)

dahdi_cfg -vv

DAHDI Tools Version - 2.1.0.2
DAHDI Version: 2.1.0.4
Echo Canceller(s): MG2
Configuration

Channel map:
Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 04: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
3 channels to configure.
Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
Setting echocan for channel 4 to mg2

After loading the dahdi module the phone has power and I can hear the dtmf tones when pushing the buttons. But no matter what … I do not get a dial tone.

When I try to call the analog phone from a SIP, the SIP gets a busy tone and asterisk reports this:
– Executing [10@DLPN_DialPlan1:1] Dial(“SIP/11-08241078”, “DAHDI/4”) in new stack
[Mar 2 10:04:09] WARNING[12428]: app_dial.c:1470 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/11-08241078’ status is ‘CHANUNAVAIL’

Since I am brand-new to Asterisk, maybe I did something wrong.
I was reading lots of postings about ‘no dial tone’ and I tried several things, but not change.

I am also having a few difficulties with the Asterisk-gui as I seem to be unable to set certain parameters, e.g. echocanceller=mg2,1-4 for Analog Hardware.

Here are some configuration files snippets:
/etc/dahdi/modules: wctdm24xxp
/etc/dahdi/system.conf :
fxsks = 1,2
fxoks = 4
loadzone = us
defaultzone = us
echocanceller=mg2,1-4

/etc/asterisk/dahdi-channels.conf :
; Span 1: WCTDM/0 “Wildcard TDM410P Board 1” (MASTER)
;;; line="1 WCTDM/0/0"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line="2 WCTDM/0/1"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line="4 WCTDM/0/3"
signalling=fxo_ks
callerid=“Channel 4” <4004>
mailbox=4004
group=5
context=from-internal
channel => 4
callerid=
mailbox=
group=
context=default

/etc/asterisk/dahdi_scan.conf :
[1]
active=yes
alarms=OK
description=Wildcard TDM410P Board 1
name=WCTDM/0
manufacturer=Digium
devicetype=Wildcard TDM410P
location=PCI Bus 07 Slot 06
basechan=1
totchans=4
irq=22
type=analog
port=1,FXO
port=2,FXO
port=3,none
port=4,FXS

/etc/asterisk/gui_confighw.conf :

[PCI Bus 07 Slot 06]
device = Wildcard TDM410P
basechan = 1
type = analog
[ANALOGPORTS]
FXS = 4
FXO = 1,2

BTW, I initially tried to use a Xen DomU with pciback, but when loading the dahdi dummy driver, the system always crashed and it seems that dahdi did not recognize the card. While a lspci showed the card correctly.
Anyhow, Asterisk is now in Dom0.
… and still not dial tone

Anybody with some ideas what could be wrong?

" Executing [10@DLPN_DialPlan1:1] Dial(“SIP/11-08241078”, “DAHDI/4”) in new stack
[Mar 2 10:04:09] WARNING[12428]: app_dial.c:1470 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 0 - Unknown) "

That tells you the problem. DAHDI/4 is your fxo port. you are trying to dial out of your fxo port and nothing is there. You need to look at your internal extensions line or whatever it is. Or you can paste your extensions.conf in here so we can see what is going on. I think I saw a mention of trixbox or something, you may want to look at your calling rules or whatever it is called in trixbox.

[ANALOGPORTS]
FXS = 4
I have my analog phone directly connected to port 4 of the TDM410P (5’ cable).

I have currently two extensions:

  • 10 analog phone on port 4 of TDM410P
  • 11 SIP on my PC

I used initially the sample extensions.conf, but started a new extensions.conf and then using the GUI. Still no change … no dial tone.

When would I get the dial-tone?

  1. After just loading the DAHDI module?
  2. After starting Asterisk?

I presume after starting Asterisk … (beginners question)

extensions.conf by GUI:

;!
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Mon Mar 2 16:52:24 2009
;!
[DID_trunk_1]
include = DID_trunk_1_default
[DID_trunk_1_default]
exten = s,1,ExecIf($[ “${CALLERID(num)}”="" ],SetCallerPres,unavailable)
exten = s,2,ExecIf($[ “${CALLERID(num)}”="" ],Set,CALLERID(all)=unknown <0000000>)
exten = s,3,Goto(default,11,1)

[CallingRule_MyLine1]
exten = _[678]XX.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},trunk_1,trunk_1)

[DLPN_DialPlan1]
include = CallingRule_MyLine1
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
[general]
static = yes
writeprotect = no
clearglobalvars = yes
[globals]
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
CID_10 = 10
[default]
[macro-stdexten]
exten = s,1,Set(_DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($["${FOLLOWME
${ARG1}}" = “1”]?5:3)
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-stdexten-followme]
exten = s,1,Answer
exten = s,2,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,3,Set(__FMCIDNUM=${CALLERID(num)})
exten = s,4,Set(__FMCIDNAME=${CALLERID(name)})
exten = s,5,Followme(${ARG1},${FOLLOWMEOPTIONS})
exten = s,6,Voicemail(${ARG1},u)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-pagingintercom]
exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER})
exten = s,2,Page(${ARG1}|${ARG2})
exten = s,3,Hangup
[conferences]
[ringgroups]
[queues]
[voicemenus]

Well, after adding the first trunk line (I used the second one alone) and manually editing some configuration files, e.g. users.conf, extensions.conf instead of completely relying on the gui … I got my dial tone and I can now make calls between the SIP and the analog phone. Huh …
Now I am going to learn how to configure and make outbound calls …