HI,
I tried to set up two friends of mine to connect to my asterisk and they're
landing into "friends" context.
We tried using 5002 and 5003 for them to use as Extentions and do a RTP
direct in Asterisk. So their media doesn't go through my OpenWRT box and It
works fine with "native bridging".
then, I changed the exten to match their home phone number(so they avoid
using extra number) , got the following errors when they call each other. I
don't rember there's major change on my asterisk except changing the number.
They both don't have a NAT issue and one is using SPA3K and the other is
using SPA3102(orginally with an old IPbox at 5003).
----------->
-- Executing [9057630...@friends:1]
SetCallerPres("SIP/9058861300-005a8748", "allowed") in new stack
== Auto fallthrough, channel 'SIP/9058861300-005a8748' status is 'UNKNOWN'
Scheduling destruction of SIP dialog '[email protected]' in
32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 174.112.54.36:8060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 174.112.54.36:8060
;branch=z9hG4bK-64f4bc6c;received=174.112.54.36
From: Chenqiuping <sip:[email protected]:8060>;tag=93a1888a4fb5ef70o0
To: <sip:[email protected]:8060>;tag=as1c850347
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]:8060>
Content-Length: 0
-- Executing [9057630...@friends:1]
SetCallerPres("SIP/9058861310-00599ac8", "allowed") in new stack
== Auto fallthrough, channel 'SIP/9058861300-00599ac8' status is 'UNKNOWN'
Both party has identical problem, but I can talk to each of them just fine.
Here's partial dial plan and sip.conf
sip.conf
;canreinvite=yes ;default
directrtpsetup=yes
[9058861300]
host=dynamic
type=friend
context=friends
username=9058861300
secret=abc134
callerid=Cba <9058861300>
[9057630200]
host=dynamic
type=friend
context=friends
username=9057630200
secret=abc233
callerid=cbaa <9057630200>
exten.conf
exten => 9057630200,1,SetCallerPres(allowed)
exten => 9057630200/_20XX,n,SetSet(CALLERID(all)=P LI <6477226900>)
exten => 9057630200,n,Dial(SIP/${EXTEN},${RT})
exten => 9057630200,n,Goto(out6-${DIALSTATUS},1)
exten =>
9057630200,n,Hangup
exten => out6-BUSY,1,Hangup(17)
exten => out6-CONGESTION,1,Hangup(3)
exten => out6-CHANUNAVAIL,1,Hangup(3)
exten => out6-NOANSWER,1,Hangup(16)
exten => _out6-.,1,Hangup(16)
exten => 9058861300,1,SetCallerPres(allowed)
exten => 9058861300/_20XX,n,SetSet(CALLERID(all)=P aI <6477226900>)
exten => 9058861300,n,Dial(SIP/${EXTEN},${RT})
exten => 9058861300,n,Goto(out7-${DIALSTATUS},1)
exten => 9058861300,n,Hangup
exten => out7-BUSY,1,Hangup(17)
exten => out7-CONGESTION,1,Hangup(3)
exten => out7-CHANUNAVAIL,1,Hangup(3)
exten => out7-NOANSWER,1,Hangup(16)
exten => _out7-.,1,Hangup(16)
I was wondering what the best debug I can use and what the problem might be.
why asterisk is decling? I was trying to have them inviting or direct rtping
each other.
Thanks in advance.
peng
