[asterisk-users] calls are not going thru e1 line

[Albert]

Hi everyone,

I have fresh installation of e1 line with 15 B channels, but
unfortunately calls are not going thru.

Do you know what could be wrong with config at my end? Or maybe this is
something with telco? Any hint will do :)


I am getting following debug:
-------------------------------------------------------------------
*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [00256312261627 at phones:1] NoOp("SIP/5000-00000000",
"exten->00256312261627") in new stack
    -- Executing [00256312261627 at phones:2] Dial("SIP/5000-00000000",
"DAHDI/g1/00256312261627") in new stack
[Feb 17 11:22:23] DEBUG[8115]: sig_pri.c:917 sig_pri_request:
sig_pri_request 1
[Feb 17 11:22:23] DEBUG[8115]: sig_pri.c:6010 sig_pri_call: CALLER NAME:
5000 NUM: 5000
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/00256312261627
    -- DAHDI/i1/00256312261627-1 is proceeding passing it to
SIP/5000-00000000
    -- Channel 0/1, span 1 got hangup request, cause 31
[Feb 17 11:22:23] DEBUG[8115]: sig_pri.c:5809 sig_pri_hangup:
sig_pri_hangup 1
[Feb 17 11:22:23] DEBUG[8115]: sig_pri.c:5855 sig_pri_hangup: Not yet
hungup...  Calling hangup once with icause, and clearing call
    -- Hungup 'DAHDI/i1/00256312261627-1'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/5000-00000000' status is 'CHANUNAVAIL'
    -- Registered SIP '5000' at 83.238.75.120:25912
       > Saved useragent "Sipura/SPA941-4.1.8" for peer 5000

-------------------------------------------------------------------

My config looks following:

voice:/etc/asterisk# cat /etc/dahdi/system.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
loadzone	= uk
defaultzone	= uk


voice:/etc/asterisk# cat chan_dahdi.conf
[trunkgroups]

[channels]

; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
pridialplan=unknown
prilocaldialplan=unknown
language=en
resetinterval=never
usecallingpres=yes
callwaitingcallerid=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
overlapdial=no
callprogress=no
busydetec=yes
pulsedial=yes

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

voice:/etc/asterisk# cat /etc/asterisk/dahdi-channels.conf

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=1
callgroup=1
pickupgroup=1
context=incoming_calls		; incloming calls go to [incloming] in
extensions.conf
switchtype=euroisdn		; european standard E1
dtmfmode=rfc2833
signalling=pri_cpe		; signaling pri_cpe
channel => 1-15,17-31

extensions.conf

[outgoing_calls]
exten => _X.,1,NoOp(exten->${EXTEN})
exten => _X.,n,Dial(DAHDI/g1/${EXTEN})





Regards,
Albert