Really destroying SIP dialog'..' , Method: Register

smile_9_2 · February 3, 2014, 10:30am

I have a problem in the configuration of the file sip.conf so the registration failed on twinkle .

I am trying on 2 PCs, First act as sip server & client (has asterisk & twinkle installed), with ip 192.168.1.3 & with device name "100 "and the other as a client only(has twinkle only installed) with ip=192.168.1.4 and with device name “101” - as written in the sip.conf .

;Sip.conf

[general]
bindport=5062
udpbindaddr=192.168.1.3
allowguest=no
disallow=all
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=ilbc
delayreject=yes
nochecksums=no
pedantic=no
srvlookup=yes
autodomain=yes
sipdebug = yes
domain=192.168.1.3
nat=no
insecure=invite
register => 100:xxxxxx@192.168.1.3/internal-phones
register => 101:zzzzzz@192.168.1.3/internal-phones
peer auth=100:xxxxxx@192.168.1.3
peer auth=101:zzzzzz@192.168.1.3
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[192.168.1.3] ;;;sip server
type=friend
usereqphone = yes
nat=no
host=192.168.1.3
fromdomain=192.168.1.3
fromuser=100
secret=xxxxxx
username=100
context=internal-phones
authname=100
dtmfmode = rfc2833
canreinvite=yes
insecure=invite

[100]
type=friend
context=internal-phones
secret=xxxxxx
nat=no
host=192.168.1.3
dtmfmode = rfc2833
insecure=invite

[101]
type=friend
context=internal-phones
secret=zzzzzz
host=192.168.1.4
nat=no
dtmfmode = rfc2833
insecure=invite

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;extensions.conf

[globals]
[general]
autofallthrough=yes

[internal-phones]
include => outgoing
exten => 100,1,Dial(SIP/100,20)
exten => 101,1,Dial(SIP/101,20)
exten => s,1,Dial(SIP/100,20)
exten => s,2,hangup

[100]
exten => s,1,Dial(SIP/100,20)

[outgoing]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@192.168.1.3,30)
exten => _NXXNXXXXXX, 1, dial(SIP/1${EXTEN}@192.168.1.3,30)

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

I opened asterisk from terminal in server pc with command “sudo asterisk -rvvvvvv” & I wrote “sip reload” I got some sip messages saying : “Really destroying SIP dialog ‘738799af1696365e5d6e15ed1526edaf@192.168.1.3’ Method: REGISTER”

Here is the output of sip reload:

Retransmitting #3 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK656bb11f
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #4 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK1a6f1d67
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #4 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK656bb11f
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #5 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK1a6f1d67
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #5 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK656bb11f
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #6 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK1a6f1d67
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #6 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK656bb11f
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #7 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK1a6f1d67
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #7 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK656bb11f
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK70a5bb25
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Really destroying SIP dialog ‘738799af1696365e5d6e15ed1526edaf@192.168.1.3’ Method: REGISTER
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK02409c23
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Really destroying SIP dialog ‘386371d26b4c4dcf756e2b432ded3b13@192.168.1.3’ Method: REGISTER

Retransmitting #1 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK70a5bb25
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #1 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK02409c23
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #2 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK70a5bb25
Max-Forwards: 70
From: sip:101@192.168.1.3;tag=as6f313b2f
To: sip:101@192.168.1.3
Call-ID: 738799af1696365e5d6e15ed1526edaf@192.168.1.3
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

Retransmitting #2 (no NAT) to 192.168.1.3:5060:
REGISTER sip:192.168.1.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5062;branch=z9hG4bK02409c23
Max-Forwards: 70
From: sip:100@192.168.1.3;tag=as0d536ee1
To: sip:100@192.168.1.3
Call-ID: 386371d26b4c4dcf756e2b432ded3b13@192.168.1.3
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.3:5062
Content-Length: 0

– >I tried 3 commands “sip show peers”,“sip show registry”& “sip show domains” and here is the output:
ubuntu*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100 192.168.1.3 5060 Unmonitored
101 192.168.1.4 5060 Unmonitored
192.168.1.3/100 192.168.1.3 5060 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]

ubuntu*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
192.168.1.3:5060 N 101 120 Request Sent
192.168.1.3:5060 N 100 120 Request Sent
2 SIP registrations.

ubuntu*CLI> sip show domains
Our local SIP domains: Context Set by
192.168.1.3 (default) [Configured]
192.168.1.3 (default) [Automatic]
ubuntu

–>I made 2 accounts on twinkle, the 1st on a pc with ip 192.168.1.3(which act as server & client) with
username , name, auth name =100
password =xxxxxx
domain=192.168.1.3
sip proxy = 192.168.1.3

the 2nd account on a pc which has ip 192.168.1.4 with
username , name , auth name=101
password = zzzzzz
domain=192.168.1.3
sip proxy=192.168.1.3

In twinkle, I made SIP port : 5065 .
But in the 2 accounts it is written :" fetching registrations failed,503 service unavailable."

Please , help me if you can . Thanks

david55 · February 3, 2014, 10:59am

host is the address of the peer, not your own address.

Really destroying is not an error. The error is the lack of responses, which are happening because you are sending to a port, on the same machine, that is not running SIP.

smile_9_2 · February 3, 2014, 2:27pm

[quote=“david55”]host is the address of the peer, not your own address.

Really destroying is not an error. The error is the lack of responses, which are happening because you are sending to a port, on the same machine, that is not running SIP.[/quote]

I changed in the file but I got also some sip messages saying : “Really destroying SIP dialog ‘003786476527255f6c25a5122aeedb51@192.168.1.2’ Method: REGISTER”
&&
“Scheduling destruction of SIP dialog ‘003786476527255f6c25a5122aeedb51@192.168.1.2’ in 32000 ms (Method: REGISTER)”

When I try “sip show registry” it shows that I am registred … but it is written in twinkle account "100, fetching registrations failed : 503 service unavailable"
what should I change in the sip.conf?

—>ubuntu*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
192.168.1.2:5060 N 100 105 Registered Mon, 03 Feb 2014 16:15:21
1 SIP registrations.

—>ubuntu*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 192.168.1.2 D 5060 Unmonitored
101/101 (Unspecified) D 0 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline]

—>ubuntu*CLI> sip show domains
Our local SIP domains: Context Set by
192.168.1.2 (default) [Automatic]
ubuntu (default) [Automatic]

;;;;;;;;sip.conf;;;;;;;;;;;;;;;;

[general]
bindport=5060
udpbindaddr=192.168.1.2:5060
allowguest=no
disallow=all
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=ilbc
delayreject=yes
nochecksums=no
pedantic=no
srvlookup=yes
autodomain=yes
sipdebug = yes
nat=no
;;;;;domain=192.168.1.2
;;;insecure=invite,port

register => 100:xxxxxx@192.168.1.2/internal-phones
;;;;register => 101:zzzzzz@192.168.1.2/internal-phones
;;;;;peer auth=100:xxxxxx@192.168.1.2
;;;;;;peer auth=101:zzzzzz@192.168.1.2
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[192.168.1.2] ;;;sip server

;;;;type=peer
;;;;;type=friend
;;;user=phone
usereqphone = yes
nat=no
fromdomain=192.168.1.2
fromuser=100
secret=xxxxxx
username=100
context=internal-phones
authname=100
dtmfmode = rfc2833
canreinvite=yes
insecure=invite,port

[100]
type=friend
context=internal-phones
secret=xxxxxx
nat=no
;;;;;qualify=yes
host=dynamic
dtmfmode = rfc2833
;;;;;;;insecure=invite,port

[101]
type=friend
context=internal-phones
secret=zzzzzz
;;qualify=yes
host=dynamic
nat=no
dtmfmode = rfc2833
;;;;insecure=invite,port

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

Here is the output of sip reload command :

-----> ubuntu*CLI> sip reload

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK26bdaaf3
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as20eb1fb4
To: sip:100@192.168.1.2
Call-ID: 59d06b604bb63a2e3449731715284f3e@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“43de2aaa”, response="a9fee15b93ecf62ca4db3b4681190817"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK26bdaaf3;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as20eb1fb4
To: sip:100@192.168.1.2;tag=as20eb1fb4
Call-ID: 59d06b604bb63a2e3449731715284f3e@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:14:49 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘59d06b604bb63a2e3449731715284f3e@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK26bdaaf3;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as20eb1fb4
To: sip:100@192.168.1.2;tag=as20eb1fb4
Call-ID: 59d06b604bb63a2e3449731715284f3e@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:14:49 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘59d06b604bb63a2e3449731715284f3e@192.168.1.2’ Method: REGISTER
ubuntu*CLI> sip reload
Reloading SIP
== Parsing ‘/etc/asterisk/sip.conf’: == Found
== Using SIP CoS mark 4
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK425203e6
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK425203e6
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK425203e6;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2;tag=as467677ee
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 102 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7a5961e8"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘73c0000e323b5eb35613a7f966705074@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK425203e6;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2;tag=as467677ee
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 102 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7a5961e8"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 192.168.1.2
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK76256274
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“7a5961e8”, response="29a2d9fadf919818a2485e7907fb4b75"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK76256274
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“7a5961e8”, response="29a2d9fadf919818a2485e7907fb4b75"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK76256274;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2;tag=as467677ee
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:15:17 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘73c0000e323b5eb35613a7f966705074@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK76256274;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as467677ee
To: sip:100@192.168.1.2;tag=as467677ee
Call-ID: 73c0000e323b5eb35613a7f966705074@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:15:17 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘73c0000e323b5eb35613a7f966705074@192.168.1.2’ Method: REGISTER
ubuntu*CLI> sip reload
Reloading SIP
== Parsing ‘/etc/asterisk/sip.conf’: == Found
== Using SIP CoS mark 4
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48abc689
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48abc689
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48abc689;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2;tag=as557c276a
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 102 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="36610199"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘15115c0635147827443078481ab9a4e1@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48abc689;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2;tag=as557c276a
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 102 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="36610199"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 192.168.1.2
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK50d8799a
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“36610199”, response="d99c96cbf2401af8bf2297dc25a5e07e"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK50d8799a
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“36610199”, response="d99c96cbf2401af8bf2297dc25a5e07e"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK50d8799a;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2;tag=as557c276a
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:15:19 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘15115c0635147827443078481ab9a4e1@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK50d8799a;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as557c276a
To: sip:100@192.168.1.2;tag=as557c276a
Call-ID: 15115c0635147827443078481ab9a4e1@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:15:19 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘15115c0635147827443078481ab9a4e1@192.168.1.2’ Method: REGISTER
ubuntuCLI>
ubuntuCLI> sip reload
Reloading SIP
== Parsing ‘/etc/asterisk/sip.conf’: == Found
== Using SIP CoS mark 4
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK77a057c8
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK77a057c8
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 102 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK77a057c8;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2;tag=as7d371056
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 102 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4a6519d1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘003786476527255f6c25a5122aeedb51@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK77a057c8;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2;tag=as7d371056
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 102 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4a6519d1"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 192.168.1.2
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1190afcd
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“4a6519d1”, response="6759f45f9dd65a723b71a4658c01f300"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1190afcd
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 103 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“4a6519d1”, response="6759f45f9dd65a723b71a4658c01f300"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1190afcd;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2;tag=as7d371056
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:15:21 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘003786476527255f6c25a5122aeedb51@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1190afcd;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as7d371056
To: sip:100@192.168.1.2;tag=as7d371056
Call-ID: 003786476527255f6c25a5122aeedb51@192.168.1.2
CSeq: 103 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Mon, 03 Feb 2014 14:15:21 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘003786476527255f6c25a5122aeedb51@192.168.1.2’ Method: REGISTER

david55 · February 3, 2014, 6:04pm

The message is normal. It is not an error. Always look for the first symptom (which was the retransmissions).

On a very quick skim, the register seems to be working.

smile_9_2 · February 4, 2014, 1:22am

[quote=“david55”]The message is normal. It is not an error. Always look for the first symptom (which was the retransmissions).

On a very quick skim, the register seems to be working.[/quote]

Thanks very much David55 , the registration succeeded on twinkle aslo [ it wasn’t successful before because I was writting my previous IP not the new one , I changed it in the files but forgot to change it on twinkle ! ]

Note : I installed only asterisk 1.8 & twinkle on ubuntu 12.04 LTS . I understood from searching that asterisk is a sip server so we don’t need to search for a sip server if asterisk is installed. Tell me if there is something else shoubld be installed to be able to make a call (As I am new to ubuntu & asterisk)

But I have 2 problems now if you can help me :
1- when the other pc which has only twinkle installed with device name 101 try to register on twinkle , it is written : “fetching registrations failed, 408 request timeout” !! although the registration succeded on the pc that acts as server & client with device name 100.
why?!!!

2- also on pc with device name 100 , I try to call “100” on twinkle , it is written in twinkle : " line 1:call failed , 503 service unavailable" .

[I think there is no problem to call myself , as I tried this before using account on twinkle & an account on a sip server called “sip2sip.info” before using asterisk and twinkle was ringing for an incoming call.]
[ I think also if there is a problem to call myself(call 100 when i am 100) then if 101 registration succeded on twinkle , i won’t be able to call 101 from 100.]

SIP port on twinkle : 5080

I got some sip messages saying on terminal , when i called 100 from device 100:

" – SIP/100-0000004a is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/100-00000049’ status is ‘CONGESTION’"
&& “Really destroying SIP dialog ‘580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060’ Method: INVITE”
&& “Got SIP response 482 “Loop Detected””

I searched on google , someone had the same problem & the solution was to check for sip trunks or firewalls
-------> I tried 2 commands in terminal & here is the output
--------->dell@ubuntu:~$ sudo iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source destination

Chain FORWARD (policy ACCEPT)
target prot opt source destination

Chain OUTPUT (policy ACCEPT)
target prot opt source destination

--------->dell@ubuntu:~$ sudo ufw status
Status: inactive

Here is sip.conf & extensions.conf: (I didn’t make the server a friend or peer , is it important to make it friend or peer? … I tried to make it friend , but also I have the same 2 problems!! )

;;;sip.conf;;;;

[general]
bindport=5060
;;;;;;;;bindaddr = 0.0.0.0
;;;;;;;;;;;udpbindaddr=0.0.0.0
udpbindaddr=192.168.1.2:5060
allowguest=yes
disallow=all
allow=ulaw,alaw,gsm,g729,ilbc
delayreject=yes
nochecksums=no
pedantic=no
srvlookup=yes
autodomain=yes
sipdebug = yes
domain=192.168.1.2
nat=no
notifyringing=yes
notifyhold=yes

;;;;;;;;;;;;;;insecure=very
insecure=invite,port

register => 100:xxxxxx@192.168.1.2/internal-phones
register => 101:zzzzzz@192.168.1.2/internal-phones
peer auth=100:xxxxxx@192.168.1.2
peer auth=101:zzzzzz@192.168.1.2
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[192.168.1.2] ;;;sip server

;;;;;;;type=peer
;;;;;;;;;;type=friend
;;;;;;user=phone
;;;;;;type=user
usereqphone = yes
nat=no
;;;;host=192.168.1.4
;;;;;host=dynamic
fromdomain=192.168.1.2
fromuser=100
secret=xxxxxx
username=100
context=internal-phones
authname=100
dtmfmode = rfc2833
;;;;;canreinvite=no
canreinvite=yes
notifyringing=yes
notifyhold=yes

;;;;;;;;;;;;;insecure=very
insecure=invite,port
;;;;;;insecure=invite

peer auth=100:xxxxxx@192.168.1.2
peer auth=101:zzzzzz@192.168.1.2

disallow=all
allow=ulaw,alaw,gsm,g729,ilbc

[100]
type=friend
context=internal-phones
secret=xxxxxx
nat=no
;;;;;nat=yes
qualify=no
host=dynamic
;;;;;;;;;;;;dtmfmode=inband
dtmfmode = rfc2833
;;;;;;;;insecure=invite
;;;;;;;;;;;;;;;;insecure=invite,port

[101]
type=friend
context=internal-phones
secret=zzzzzz
qualify=no
host=dynamic
;;;;;;;;;nat=yes
nat=no
;;;;;;;;;;;;dtmfmode=inband
dtmfmode = rfc2833
;;;;;;;;;insecure=invite
;;;;;;;;;;;;insecure=invite,port

;;;;;;;;;;;;;;;;extensions.conf;;;;;;;;;;;;;;;;;;

[globals]
[general]

;;;;;;;;;;;;;;;;;;;;autofallthrough=yes
exten => 100,1,Dial(SIP/100,60)
;;;;;;;;;;;exten => 101,1,Dial(SIP/101,60)
exten => s,1,Dial(SIP/100,60)
exten => s,2,hangup

[internal-phones]
exten => 100,1,Dial(SIP/100,60)
;;;;;;;exten => 101,1,Dial(SIP/101,60)
exten => s,1,Dial(SIP/100,60)
exten => s,2,hangup

----------> ubuntu*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 192.168.1.2 D 5060 Unmonitored
101/101 192.168.1.2 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

--------->ubuntuCLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
192.168.1.2:5060 N 101 105 Registered Tue, 04 Feb 2014 02:23:22
192.168.1.2:5060 N 100 105 Registered Tue, 04 Feb 2014 02:23:22
2 SIP registrations.
-----> ubuntuCLI> sip show domains
Our local SIP domains: Context Set by
192.168.1.2 (default) [Configured]
192.168.1.2 (default) [Automatic]
ubuntu (default) [Automatic]

here is the full output on the terminal ,when I try to call 100 from twinkle :

<------------>
– Executing [100@internal-phones:1] Dial(“SIP/100-00000049”, “SIP/100,60”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 8842
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
INVITE sip:100@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK14d0b903
Max-Forwards: 70
From: “100” sip:100@192.168.1.2;tag=as3399b912
To: sip:100@192.168.1.2:5060
Contact: sip:100@192.168.1.2:5060
Call-ID: 580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Date: Tue, 04 Feb 2014 00:25:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 338

v=0
o=root 554310590 554310590 IN IP4 192.168.1.2
s=Asterisk PBX SVN-branch-1.8-r406721
c=IN IP4 192.168.1.2
t=0 0
m=audio 8842 RTP/AVP 0 8 3 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/100

<— SIP read from UDP:192.168.1.2:5060 —>
INVITE sip:100@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK14d0b903
Max-Forwards: 70
From: “100” sip:100@192.168.1.2;tag=as3399b912
To: sip:100@192.168.1.2:5060
Contact: sip:100@192.168.1.2:5060
Call-ID: 580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Date: Tue, 04 Feb 2014 00:25:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 338

v=0
o=root 554310590 554310590 IN IP4 192.168.1.2
s=Asterisk PBX SVN-branch-1.8-r406721
c=IN IP4 192.168.1.2
t=0 0
m=audio 8842 RTP/AVP 0 8 3 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
— (14 headers 15 lines) —

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK14d0b903;received=192.168.1.2
From: “100” sip:100@192.168.1.2;tag=as3399b912
To: sip:100@192.168.1.2:5060;tag=as3399b912
Call-ID: 580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060
CSeq: 102 INVITE
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK14d0b903;received=192.168.1.2
From: “100” sip:100@192.168.1.2;tag=as3399b912
To: sip:100@192.168.1.2:5060;tag=as3399b912
Call-ID: 580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060
CSeq: 102 INVITE
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Got SIP response 482 “Loop Detected” back from 192.168.1.2:5060
Transmitting (no NAT) to 192.168.1.2:5060:
ACK sip:100@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK14d0b903
Max-Forwards: 70
From: “100” sip:100@192.168.1.2;tag=as3399b912
To: sip:100@192.168.1.2:5060;tag=as3399b912
Contact: sip:100@192.168.1.2:5060
Call-ID: 580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
ACK sip:100@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK14d0b903
Max-Forwards: 70
From: “100” sip:100@192.168.1.2;tag=as3399b912
To: sip:100@192.168.1.2:5060;tag=as3399b912
Contact: sip:100@192.168.1.2:5060
Call-ID: 580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– SIP/100-0000004a is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/100-00000049’ status is ‘CONGESTION’

<— Reliably Transmitting (no NAT) to 192.168.1.2:5080 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.2:5080;branch=z9hG4bKqumbyktt;received=192.168.1.2;rport=5080
From: “100” sip:100@192.168.1.2;tag=rzadr
To: sip:100@192.168.1.2;tag=as7cf756ed
Call-ID: dldwqtpdalapebi@ubuntu.ubuntu-domain
CSeq: 919 INVITE
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0

<------------>
Really destroying SIP dialog ‘580f712d31757a7462064b6c1fba48dd@192.168.1.2:5060’ Method: INVITE

<— SIP read from UDP:192.168.1.2:5080 —>
ACK sip:100@192.168.1.2 SIP/2.0
v: SIP/2.0/UDP 192.168.1.2:5080;rport;branch=z9hG4bKqumbyktt
Max-Forwards: 70
t: sip:100@192.168.1.2;tag=as7cf756ed
f: “100” sip:100@192.168.1.2;tag=rzadr
i: dldwqtpdalapebi@ubuntu.ubuntu-domain
CSeq: 919 ACK
User-Agent: Twinkle/1.4.2
l: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog 'dldwqtpdalapebi@ubuntu.ubuntu-domain’ Method: ACK
Really destroying SIP dialog 'ptljmbcrtmbdwyd@ubuntu.ubuntu-domain’ Method: REGISTER
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48f9158c
Max-Forwards: 70
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 104 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“6e026f11”, response="60b7efc90c4365bc54bf1343672ebb35"
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48f9158c
Max-Forwards: 70
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 104 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“6e026f11”, response="60b7efc90c4365bc54bf1343672ebb35"
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48f9158c;received=192.168.1.2
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2;tag=as5ff4199f
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 104 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="714ac407"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘20b11fa95a3e283e610087267f064ba7@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK48f9158c;received=192.168.1.2
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2;tag=as5ff4199f
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 104 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="714ac407"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 192.168.1.2
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6fbbcbc1
Max-Forwards: 70
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 105 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“714ac407”, response="3e60ecc75a265390f7d4d02c78221c2a"
Expires: 120
Contact: sip:101@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6fbbcbc1
Max-Forwards: 70
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 105 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“714ac407”, response="3e60ecc75a265390f7d4d02c78221c2a"
Expires: 120
Contact: sip:101@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6fbbcbc1;received=192.168.1.2
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2;tag=as5ff4199f
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 105 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:101@192.168.1.2:5060;expires=120
Date: Tue, 04 Feb 2014 00:26:16 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘20b11fa95a3e283e610087267f064ba7@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6fbbcbc1;received=192.168.1.2
From: sip:101@192.168.1.2;tag=as5ff4199f
To: sip:101@192.168.1.2;tag=as5ff4199f
Call-ID: 20b11fa95a3e283e610087267f064ba7@192.168.1.2
CSeq: 105 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:101@192.168.1.2:5060;expires=120
Date: Tue, 04 Feb 2014 00:26:16 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘20b11fa95a3e283e610087267f064ba7@192.168.1.2’ Method: REGISTER
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3d17dca0
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 104 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“5cb86253”, response="405eb728ba9fc587fd20b2e6112a61e9"
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3d17dca0
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 104 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“5cb86253”, response="405eb728ba9fc587fd20b2e6112a61e9"
Expires: 120
Contact: sip:internal-phones@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3d17dca0;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2;tag=as552fe160
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 104 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0499f6b6"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘244a9d0b07ab133f75abf38c5f34edef@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3d17dca0;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2;tag=as552fe160
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 104 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0499f6b6"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 192.168.1.2
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1ac89b29
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 105 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“0499f6b6”, response="1f54c0f9fd4f89e7b84976b06504a8c7"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<— SIP read from UDP:192.168.1.2:5060 —>
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1ac89b29
Max-Forwards: 70
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 105 REGISTER
User-Agent: Asterisk PBX SVN-branch-1.8-r406721
Authorization: Digest username=“100”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.1.2”, nonce=“0499f6b6”, response="1f54c0f9fd4f89e7b84976b06504a8c7"
Expires: 120
Contact: sip:100@192.168.1.2:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.2:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1ac89b29;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2;tag=as552fe160
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 105 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Tue, 04 Feb 2014 00:26:16 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘244a9d0b07ab133f75abf38c5f34edef@192.168.1.2’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK1ac89b29;received=192.168.1.2
From: sip:100@192.168.1.2;tag=as552fe160
To: sip:100@192.168.1.2;tag=as552fe160
Call-ID: 244a9d0b07ab133f75abf38c5f34edef@192.168.1.2
CSeq: 105 REGISTER
Server: Asterisk PBX SVN-branch-1.8-r406721
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:100@192.168.1.2:5060;expires=120
Date: Tue, 04 Feb 2014 00:26:16 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘244a9d0b07ab133f75abf38c5f34edef@192.168.1.2’ Method: REGISTER

david55 · February 4, 2014, 8:01am

SIP/2.0 482 Loop Detected

You have another own address somewhere.

Also note that insecure is called insecure because it compromises security. In this configuration you should not need it unless you have other configuration problems. This is one of the biggest cut and paste errors in Asterisk configuration.

smile_9_2 · February 4, 2014, 5:31pm

[quote=“david55”]SIP/2.0 482 Loop Detected

You have another own address somewhere.

Also note that insecure is called insecure because it compromises security. In this configuration you should not need it unless you have other configuration problems. This is one of the biggest cut and paste errors in Asterisk configuration.[/quote]

Now , I removed the “insecure=invite, port” from the file but i think it didn’t change anything . I have the same problems.
Note: that the 2 PCs are not in the same place , 101 device which try to register on twinkle to server PC is owned by my friend which is in her house far away from me. Please tell me if the 2 PCs must be in the same place to be connected to same network for example!

sorry , I don’t understand what do you mean by "you have another own address somewhere"
here is what is written in my connection information :
IPv4
IP address: 192.168.1.2
Broadcast address:192.168.1.255
Subnet Mask:255.255.255.0
Default route:192.168.1.1
primary DNS:192.168.1.1

I tried to get my IP using whatismyip.com/ , It shows that my IP is 197.162.77.5 but when I tried to change in sip.conf all IP to that IP , when I reload sip , I found that I am not registered using this IP:
ubuntuCLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
197.162.77.5:5060 N 101 120 Request Sent
197.162.77.5:5060 N 100 120 Request Sent
2 SIP registrations.
ubuntuCLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 (Unspecified) D A 0 Unmonitored
101/101 (Unspecified) D A 0 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
ubuntu*CLI>

And here is the output of “ifconfig” command:

dell@ubuntu:~$ ifconfig
eth0 Link encap:Ethernet HWaddr 78:2b:cb:e7:72:3a
UP BROADCAST MULTICAST MTU:1500 Metric:1
RX packets:0 errors:0 dropped:0 overruns:0 frame:0
TX packets:0 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:0 (0.0 B) TX bytes:0 (0.0 B)

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:4675 errors:0 dropped:0 overruns:0 frame:0
TX packets:4675 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:558328 (558.3 KB) TX bytes:558328 (558.3 KB)

wlan0 Link encap:Ethernet HWaddr 8c:a9:82:60:85:22
inet addr:192.168.1.2 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::8ea9:82ff:fe60:8522/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:33618 errors:0 dropped:0 overruns:0 frame:0
TX packets:36880 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:21074465 (21.0 MB) TX bytes:7152362 (7.1 MB)

david55 · February 4, 2014, 10:44pm

Your first problem was due to having your own address, rather than the peer address. As you are getting a loop reported, you are sending to yourself, so there must be another place where you have your own address, rather than the peer’s.

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Really destroying SIP dialog'..' , Method: Register

Here is the output of sip reload:

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