[asterisk-users] Query

sanchal.singh at alliance-infotech.com sanchal.singh at alliance-infotech.com
Mon Aug 6 01:41:43 CDT 2007


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




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