Hi ,
I am trying to dial in from two sip phones on one end, through
digium card to E1 card running application on another end.
with following configuration
/etc/asterisk/zapata.conf
group=1
context=default
euroisdn=EuroISDN
signalling= pri_net
context=incoming
channel=1-15,17-31
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
/etc/asterisk/sip.conf
[phone1]
type=friend
host=192.168.1.67
dtmfmode=rfc2833
context=sip
port=5060
nat=yes
[phone2]
type=friend
host=192.168.1.53
dtmfmode=rfc2833
context=sip
port=5060
nat=yes
/etc/asterisk/extension.conf
[sip]
exten=>112,1,Dial(SIP/phone2,20,tr)
; Dialing from sip phone1 at one system (192.168.1.67)through
; through soft switch to sip Phone2 (192.168.1.53) running at
; at other system having IP 192.168.1.53
exten=>113,1,Dial(ZAP/1,16)
; Dialing from sip phone1 at one system (192.168.1.67) through
; asterisk PBX having digium card to other E1
; card running application
exten=>115,1,Dial(ZAP/1,16)
[incoming]
exten=>114,1,Dial(SIP/phone1,20,tr)
; Making call from E1 card running application
; to soft switch through digium card and
; diverting to sip phone1 rinning on system
; 192.168.1.67
I am able to dial from phone1 to E1 card running application
successfully
but when I dial from phone2 to Ei card running application it gives error
message.
app_dial.c:1076dial_exec_full:unable to create channel of type
ZAP(cause 0
unknown)
Everyone is busy/conjusted at this time (1:0/0/1)
auto fall through channel 'SIP/192.168.1.53/081c63b8' Status is "
CHANUNAVAILABLE".
Can anybody help me to solve this problem.
thanks & regards
Sanchal Singh
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
