Outgoing calls - unable to create channel of type dahdi

Dear List,

we are facing the problem, when we try to dial out a number from my sip softphone. Also could not receive any incoming calls

error:
unable to create channel of type dahdi

Asterisk - Version : Asterisk - 1.6.1
Dahdi - Version : 2.2-rc1

But, I am not facing any problem with Asterisknow 1.2 version of Installation. I could make calls / answer calls without any problem

I had given below the config file details,

/etc/dahdi/system.conf


span = 1,1,0,ccs,hdb3
bchan = 1-15,17-31
dchan = 16
loadzone = in
defaultzone = in

/etc/asterisk/chan_dahdi.conf


[trunkgroups]

[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
callgroup=1
pickupgroup=1
busydetect=yes
faxdetect=both
group=1
context=local
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31

/etc/asterisk/extensions.conf


[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no

[globals]
span_1 = Zap/g1

[default]
exten => 6050,1,VoiceMailMain
exten = 7000,1,Goto(voicemenu-custom-1|s|1)

[voicemenu-custom-1]
include = default
comment = Welcome
alias_exten = 7000
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Background(thank-you-for-calling)
exten = s,4,Background(if-u-know-ext-dial)
exten = s,5,Background(otherwise)
exten = s,6,Background(to-reach-operator)
exten = s,7,Background(pls-hold-while-try)
exten = s,8,WaitExten(6)

[numberplan-custom-1]
plancomment = DialPlan1
include = default
exten = _9XXXXXXXXX!,1,Macro(trunkdial,${span_1}/${EXTEN:0},${span_1_cid})
comment = _9XXXXXXXXX!,1,dlit,standard

[macro-trunkdial]
exten = s,1,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" > “6”]?${CALLERID(all)}:${ARG2})})
exten = s,n,Dial(${ARG1})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Hangup
exten = _s-.,1,NoOp

[DID_span_1]
include = default
exten = _X.,1,Goto(voicemenu-custom-1|s|1)
exten = s,1,ExecIf($[ “${CALLERID(num)}”="" ],SetCallerPres,unavailable)
exten = s,2,ExecIf($[ “${CALLERID(num)}”="" ],Set,CALLERID(all)=unknown <0000000>)
exten = s,3,Goto(voicemenu-custom-1|s|1)

[dlit_out]
exten => _X.,1,NoOp(The user have dialled ${EXTEN})
exten => _X.,n,Dial(ZAP/g1/${EXTEN})
exten => _X.,n,HangUp()

[macro-stdexten]
exten => s,1,Dial(${ARG2},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
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})

but, the FUN here is, that we could make calls happily by call files at /var/spool/asterisk/outgoing/

PLEASE GUIDE US TO RESOLVE THE PROBLEM

thanks / sithi

And the debug of a failed call look like ???

Ian

Dear Ian,

These are logs generated while the outgoing call is attempted from Sip Extension (softphone)

Connected to Asterisk SVN-branch-1.6.1-r173135 currently running on asterisk (pid = 4282)
Verbosity is at least 3
Core debug is at least 3
– Remote UNIX connection
– Unregistered SIP ‘6001’
– Registered SIP ‘6001’ at 192.168.1.218 port 11442
== Using SIP RTP CoS mark 5
– Executing [9940684780@DLPN_DialPlan1:1] Macro(“SIP/6001-0a141848”, “trunkdial,Dahdi/g1/9940684780,”) in new stack
– Executing [s@macro-trunkdial:1] Set(“SIP/6001-0a141848”, “CALLERID(all)=”) in new stack
– Executing [s@macro-trunkdial:2] Dial(“SIP/6001-0a141848”, “Dahdi/g1/9940684780”) in new stack
[Feb 6 07:50:08] WARNING[4641]: app_dial.c:1495 dial_exec_full: Unable to create channel of type ‘Dahdi’ (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-trunkdial:3] Goto(“SIP/6001-0a141848”, “s-CHANUNAVAIL,1”) in new stack
– Goto (macro-trunkdial,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp(“SIP/6001-0a141848”, “”) in new stack
– Auto fallthrough, channel ‘SIP/6001-0a141848’ status is 'CHANUNAVAIL’
asterisk*CLI>

Thanks for your help,

Sithi

Try this: forums.digium.com/viewtopic.php?t=66766

Ian.

Dear Ian,

thanks for your input

I tried as given in your link. But no results

FYI, we could make outgoing calls when calls are placed in spool folder

at the same time, we could not even receive any calls through this e1 line as well as make any calls from softphone

But the same system with different hard disk installed with AsteriskNow 1.2 could do perfectly with the same hardware and e1 line (both incoming and outgoing is fine)

As a newbie, i kindly request you to check my system.conf, chan_dahdi.conf and extensions.conf

Am is missing any thing in these or apart from these files

thanks / sithi