Outbound "480 Temporarily unavailable"

kevnhowe
Joined: Mon 21 of Jul, 2008

Outbound "480 Temporarily unavailable"

Posted:Mon 21 of Jul, 2008 (03:34 UTC)
Can anyone take a look at my configs and my sip debug log for an outgoing call and tell me if anything is obviously wrong. I have spent hours researching online, trying different configs, beating my brains in, and I still can't get outbound calls to work. Any help would be greatly appreciated.

SIP.CONF:

[general]
context=default
allowguest=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
externhost=*******.dyndns.org
localnet=192.168.1.0/255.255.255.0
register=267055****:*******@sip.talkinip.net/267055****
nat=yes

[incoming_DID]
type=friend
host=64.154.41.100
context=incoming_calls
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
nat=yes

[267055****]
username=267055****
type=peer
fromuser=267055****
secret=*******
nat=yes
insecure=very
host=sip.talkinip.net
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw

[100]
type=friend
username=100
secret=****
context=phones
mailbox=100
disallow=all
allow=ulaw
host=dynamic

[101]
type=friend
username=101
secret=****
context=phones
mailbox=101
disallow=all
allow=ulaw
host=dynamic


EXTENSIONS.CONF:

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]

[default]
include => incoming_calls

[incoming_calls]
exten => 9496821530,1,Answer()
exten => 9496821530,n,Wait(1)
exten => 9496821530,n,Playback(tt-weasels)
exten => 9496821530,n,Hangup()

[outbound_calls]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/267055****/${EXTEN})

[internal]
exten => _XXX,1,Dial(SIP/${EXTEN},10)
exten => _XXX,n,Voicemail(${EXTEN})
exten => _XXX,n,Hangup()

[phones]
include => internal
include => outbound_calls


SIP SET DEBUG FOR OUTBOUND CALL:

<------------>
-- Executing [949226****@phones:1] NoOp("SIP/100-00de0420", "") in new stack

-- Executing [949226****@phones:2] Dial("SIP/100-00de0420",
"SIP/267055****/949226****") in new stack
Audio is at 70.181.**.** port 11648
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.154.41.200:5060:
INVITE sip:949226****@sip.talkinip.net SIP/2.0
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK6a5a411a;rport
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
To: <sip:949226****@sip.talkinip.net>
Contact: <sip:267055****@70.181.**.**>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.**.**
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 20 Jul 2008 19:33:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 12475 12475 IN IP4 70.181.**.**
s=session
c=IN IP4 70.181.**.**
t=0 0
m=audio 11648 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---

-- Called 267055****/949226****
asterisk*CLI> sip set debug
<--- SIP read from 64.154.41.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK6a5a411a;rport
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
To: <sip:949226****@sip.talkinip.net>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.**.**
CSeq: 102 INVITE
Content-Length: 0
<------------->

--- (7 headers 0 lines) ---

<--- SIP read from 64.154.41.200:5060 --->
SIP/2.0 407 Proxy Authentication Required
Proxy-Authenticate: Digest
nonce="12861056002:337760d429debb0fd7c37d08ff6335cd",algorithm=MD5,realm="64
.154.41.110"
To: <sip:949226****@sip.talkinip.net>;tag=3425571201-608654
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
Contact: <sip:949226****@64.154.41.200:5060>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.**.**
CSeq: 102 INVITE
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK6a5a411a;rport
Content-Length: 0
<------------->

--- (9 headers 0 lines) ---
Transmitting (NAT) to 64.154.41.200:5060:
File: Unsaved Document 1 Page 2 of 3
ACK sip:949226****@sip.talkinip.net SIP/2.0
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK6a5a411a;rport
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
To: <sip:949226****@sip.talkinip.net>;tag=3425571201-608654
Contact: <sip:267055****@70.181.**.**>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.76.17
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 70.181.**.** port 11648
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.154.41.200:5060:
INVITE sip:949226****@sip.talkinip.net SIP/2.0
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK7bb2d33c;rport
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
To: <sip:949226****@sip.talkinip.net>
Contact: <sip:267055****@70.181.**.**>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.**.**
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="267055****", realm="64.154.41.110",
algorithm=MD5,
uri="sip:949226****@sip.talkinip.net",
nonce="12861056002:337760d429debb0fd7c37d08ff6335cd",
response="661201838b5c020c7528fcc9f04f68aa"
Date: Sun, 20 Jul 2008 19:33:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 12475 12476 IN IP4 70.181.**.**
s=session
c=IN IP4 70.181.**.**
t=0 0
m=audio 11648 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
asterisk*CLI> sip set debug

<--- SIP read from 64.154.41.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK7bb2d33c;rport
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
To: <sip:949226****@sip.talkinip.net>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.**.**
CSeq: 103 INVITE
Content-Length: 0
<------------->

--- (7 headers 0 lines) ---

<--- SIP read from 64.154.41.200:5060 --->
SIP/2.0 480 Temporarily Unavailable
To: <sip:949226****@sip.talkinip.net>;tag=3425571202-13099
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
File: Unsaved Document 1 Page 3 of 3
Contact: <sip:949226****@64.154.41.200:5060>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.76.17
CSeq: 103 INVITE
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK7bb2d33c;rport
Content-Length: 0
<------------->

--- (8 headers 0 lines) ---

-- Got SIP response 480 "Temporarily Unavailable" back from 64.154.41.200
Transmitting (NAT) to 64.154.41.200:5060:
ACK sip:949226****@sip.talkinip.net SIP/2.0
Via: SIP/2.0/UDP 70.181.**.**:5060;branch=z9hG4bK7bb2d33c;rport
From: "Reception" <sip:267055****@70.181.**.**>;tag=as48059dd0
To: <sip:949226****@sip.talkinip.net>;tag=3425571202-13099
Contact: <sip:267055****@70.181.**.**>
Call-ID: 2a62ca96071a08e0502180ae308c95f2@70.181.**.**
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---

-- SIP/267055****-00de4760 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/100-00de0420' status is 'CONGESTION'
asterisk*CLI> sip set debug
<--- Transmitting (NAT) to 192.168.1.11:33470 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
192.168.1.11:33470;branch=z9hG4bK-d8754z-df0985165752ef19-1---
d8754z-;received=192.168.1.11;rport=33470
From: "Reception"<sip:100@192.168.1.12:5060>;tag=c5629552
To: "949226****"<sip:949226****@192.168.1.12:5060>;tag=as1931c301
Call-ID: ZTNlNGZiZTI5NzQzMDdlOGQ2N2ZkYzY5NmNiMWI0ZWE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:949226****@192.168.1.12>
Content-Length: 0
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
<------------>

Really destroying SIP dialog '2a62ca96071a08e0502180ae308c95f2@70.181.**.**'
Method: INVITE
Really destroying SIP dialog 'ZTNlNGZiZTI5NzQzMDdlOGQ2N2ZkYzY5NmNiMWI0ZWE.'
Method: INVITE
asterisk*CLI> sip set debug
<--- SIP read from 192.168.1.11:33470 --->
ACK sip:949226****@192.168.1.12:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.11:33470;branch=z9hG4bK-d8754z-df0985165752ef19-1---d8754z-;rport
To: "949226****"<sip:949226****@192.168.1.12:5060>;tag=as1931c301
From: "Reception"<sip:100@192.168.1.12:5060>;tag=c5629552
Call-ID: ZTNlNGZiZTI5NzQzMDdlOGQ2N2ZkYzY5NmNiMWI0ZWE.
CSeq: 2 ACK
Content-Length: 0
<------------->