here's a captured packet inside the home network adjacent to the phone
adapter.
No. Time Source Destination Protocol
Info
35 17.994223 75.119.224.90 192.168.10.188 SIP
Status: 603 Declined
Frame 35 (536 bytes on wire, 536 bytes captured)
Ethernet II, Src: D-Link_5d:43:31 (00:1b:11:5d:43:31), Dst:
CiscoLin_c0:b6:3d (00:0e:08:c0:b6:3d)
Internet Protocol, Src: 75.119.224.90 (75.119.224.90), Dst: 192.168.10.188
(192.168.10.188)
User Datagram Protocol, Src Port: 8060 (8060), Dst Port: 8060 (8060)
Session Initiation Protocol
Status-Line: SIP/2.0 603 Declined
Status-Code: 603
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.10.188:8060
;branch=z9hG4bK-ebd3fded;received=174.112.54.36
Transport: UDP
Sent-by Address: 192.168.10.188
Sent-by port: 8060
Branch: z9hG4bK-ebd3fded
Received: 174.112.54.36
From: Cba <sip:[email protected]:8060>;tag=1889c974909fef35o0
SIP Display info: Chenqiuping
SIP from address: sip:[email protected]:8060
SIP tag: 1889c974909fef35o0
To: <sip:[email protected]:8060>;tag=as4cfebc38
SIP to address: sip:[email protected]:8060
SIP tag: as4cfebc38
Call-ID: [email protected]
CSeq: 102 INVITE
Sequence Number: 102
Method: INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]:8060>
Contact Binding: <sip:[email protected]:8060>
URI: <sip:[email protected]:8060>
SIP contact address: sip:[email protected]:8060
Content-Length: 0
Thanks in advance.
peng
On Thu, Oct 1, 2009 at 1:15 PM, Peng Li <[email protected]> wrote:
> 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
>
>
