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. I register just fine to the remote host.
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
<------------->
<— 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
<------------->
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
<------------->
<— 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
<------------->
– 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’
asteriskCLI> 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
asteriskCLI> 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
<------------->