No analog devices in peers list

I installed Asterisk and configured SIP phones, they works correctly, however i got issues configuring my analog card Digium Wildcard TDM410P.
My ‘chan_dahdi.conf’ :

[trunkgroups]   
[channels]
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes

#include /etc/asterisk/dahdi-channels.conf

my ‘dahdi-channels.conf

; Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=incoming
channel => 1
callerid=
group=
context=default

;;; line="2 WCTDM/0/1 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=incoming
channel => 2
callerid=
group=
context=default

;;; line="3 WCTDM/0/2 FXSKS"
signalling=fxs_ks
callerid="Channel3" <702>
group=0
context=incoming
channel => 3
callerid=
group=
context=default

;;; line="4 WCTDM/0/3 FXOKS  (In use) (EC: MG2 - INACTIVE)"
signalling=fxo_ks
callerid="Channel4" <502>
mailbox=
group=5
context=call-out
channel => 4
callerid=
mailbox=
group=
context=

asterisk see dahdi configuration

pbx*CLI> dahdi show channels
   Chan Extension       Context         Language   MOH Interpret        Blocked    In Service Description                     
      1                 incoming                   default                         Yes                                        
      2                 incoming                   default                         Yes                                        
      3                 incoming                   default                         Yes                                        
      4                 call-out                   default                         Yes                   

however i cannot call to phone in 4th port:

Executing [502@call-out:1] Dial("SIP/402-00000001", "SIP/502") in new stack
[Feb  9 10:07:13] WARNING[1019][C-00000002]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/402-00000001' status is 'CHANUNAVAIL'

According to :

 Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

analog devices is not connected to asterisk.

Please, suggest me what to do!

Regards,
Nufay.

You are trying to call a SIP device, but your line 4 is a DAHDI device!!!

(Your address part is also 502, when you actually want port 4, not port 502.)

how can i call from a sip device to dahdi device?

Dial the DAHDI device. Don’t dial a SIP device.

I am very reluctant to provide specifics, for free, for something that you should be able to do for yourself, if you have obtained the minimum knowledge needed to manage Asterisk.

my extensions.conf

;outbound calls
[call-out]
;call internal number
exten => s,1,Log(NOTICE, Incoming call from ${CALLERID(all)} to ${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
include => handup-sip

; Call POTS numbers through Foo Provider (any number longer than 5 digits starting with 9)
exten => _7XXXX.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} GoIp)
exten => _7XXXX.,n,Dial(SIP/701/${EXTEN:1})
exten => _7XXXX.,n,Hangup()

;hang up phone
[handup-sip]
exten => _X!,1,HangUp()

As i understand, i need to write something like

exten => _5XX,1,Dial(DAHDI/4)

in extensions.conf ?

Now my extensions.conf looks like:

 ;outbound calls
[call-out]
;call dahdi devices
exten => s,1,Log(NOTICE, Incoming call from ${CALLERID(all)} to ${EXTEN})
exten => _5XX,1,Dial(DAHDI/4)
include => handup-sip

;call internal number
exten => s,1,Log(NOTICE, Incoming call from ${CALLERID(all)} to ${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
include => handup-sip

; Call POTS numbers through Foo Provider (any number longer than 5 digits starting with 9)
exten => _7XXXX.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} GoIp)
exten => _7XXXX.,n,Dial(SIP/701/${EXTEN:1})
exten => _7XXXX.,n,Hangup()

;hang up phone
[handup-sip]
exten => _X!,1,HangUp()

All works now!
Thank you, David551.

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