I appology for the wrong subject used in previous email. thanks 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 > > 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
