[asterisk-users] Probelm with incoming calls to my DID-Please help
me
Crazy Boy
crazymoonboy at yahoo.com
Fri Sep 1 04:37:24 MST 2006
Hi friends,
Thank you to all for your response and cooperation to me. I have a doubt.
We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error message on Asterisk console.
Contents in IAX.CONF file:
disallow=all
allow = ulaw
[general]
register => teliaxusername:teliaxpassword at voip-co1.teliax.com
[teliax]
context=telincoming
type=friend
host=voip-co1.teliax.com
auth=md5
secret=teliaxpassword
disallow=all
allow=ulaw
allow=alaw
allow=gsm
Contents in Sip.conf file:
[105]
type=friend
username=105
secret=ravi
callerid="RaviKanth"
host=dynamic
context=leader
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
mailbox=605 at vmail
[107]
type=friend
username=107
secret=suresh
callerid="Suresh"
host=dynamic
context=administration
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all
mailbox=607 at vmail
Contents in Extensions.conf file:
[telincoming]
exten => 303xxxxxxx, 1, Answer()
exten => 303xxxxxxx, n, Wait,2
exten => 303xxxxxxx, n, Goto(incoming,s,1)
include => internal
include => incoming
[incoming]
exten => s,1,Wait(3)
exten => s,n,Answer
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(/tmp/virg2)
exten => s,n,Goto(s,1)
exten => s,n,Hangup()
include => internal
[internal]
exten => 105,1,SetMusicOnHold(default)
exten => 105,2,Dial(SIP/105,7,t,m,T)
exten => 1605,1,VoiceMailMain(605 at vmail)
exten => 105,3,VoiceMail(605 at vmail)
exten => 105,4,Hangup
exten => 107,1,SetMusicOnHold(default)
exten => 107,2,Dial(SIP/107,7,t,m,T)
exten => 1607,1,VoiceMailMain(607 at vmail)
exten => 107,3,VoiceMail(607 at vmail)
exten => 107,4,Hangup
[uscall]
exten => _1XXXXXXXXXX,1,DIAL(IAX2/teliaxusername at teliax/${EXTEN},30,tr)
[manager]
include => uscall
include => internal
The error message on Asterisk console:
*CLI> -- Executing Dial("SIP/105-007951e0", "IAX2/teliaxusername at teliax/1303xxxxxxx|30|tr") in new stack
-- Called teliaxusername at teliax/1303xxxxxxx
-- Call accepted by 207.174.202.2 (format ulaw)
-- Format for call is ulaw
-- IAX2/teliax-1 is ringing
-- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0
-- IAX2/teliax-1 is ringing
-- IAX2/teliax-1 is busy
-- Hungup 'IAX2/teliax-1'
== Everyone is busy/congested at this time (1:1/0/0)
== Auto fallthrough, channel 'SIP/105-007951e0' status is 'BUSY'
What is the problem? Can you please tell me the solution. Looking forward to your response. Thank you.
Regards,
Chandra.
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