SIP Configuration in Asterisk and SIP Phone

Hi,

I’m running Asterisk 1.8.4.4. on Ubuntu 11.10. I am trying to get my Nexus 4 to sink with my Asterisk box via SIP. I can’t seem to figure out where the problem is. My phone keeps saying “Account Registration is unsuccessful”.

Here are my details. Any suggestions or directions would be appreciated! Thanks for your time!

sip.conf
[nx4]
type = friend
allowguest = yes
context = incoming
host = dynamic
nat = yes
udpbindaddr = 0.0.0.0
username = nx4
secret = XXXXXX
bindport = 5060
canreinvite = no
insecure = invite

dialplan.conf*
[incoming]
exten => 4242, 1, Goto(google, start, 1)

[google]
exten => start, 1, Answer()
same => n, Playback(tt-monkeysintro)
same => n, Playback(hello-world)
same => n, Playback(vm-goodbye)
same => n, Hangup()

*Sip settings on phone

Username: nx4
password: XXXXXX
Server: xxx.xx.xxx.xxx ;ip for my asterisk box
Outbound proxy address: xxx.xx.xxx.xxx ;ip for my asterisk box
Port Number: 5060
Transportation type: UDP
Send keep-alive: Automatic

CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
1001/1001 (Unspecified) D N 0 UNKNOWN
kbeezy/0000000070 xx.xx.xxx.xx N 5060 OK (55 ms)
nx4/nx4 (Unspecified) D N 0 Unmonitored
office-phone (Unspecified) D N 0 UNKNOWN
zoiper/zoiper xx.xx.xx.xxx D 5060 Unmonitored
5 sip peers [Monitored: 1 online, 2 offline Unmonitored: 1 online, 1 offline]

******* sip debug*********

<— SIP read from UDP:xx.xx.77.xxx:5060 —>

<------------->
Reliably Transmitting (NAT) to xx.x4.xx0.xx:5060:
OPTIONS sip:xx.x4.xx0.xx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK3f47219a;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@1xxx.xx.xxx.xxx;tag=as7313814a
To: sip:xx.x4.xx0.xx
Contact: sip:asterisk@xxx.xx.xxx.xxx:5060
Call-ID: 4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Tue, 23 Jul 2013 18:03:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:xx.x4.xx0.xx:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK3f47219a;received=xxx.xx.xxx.xxx;rport=5060
From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as7313814a
To: sip:xx.x4.xx0.xx;tag=as6793089d
Call-ID: 4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: IPKall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '4d7bb95b6b6904094c4eda5510a8c7ac@xxx.xx.xxx.xxx:5060’ Method: OPTIONS
Really destroying SIP dialog ‘4eba1837cab8596cf822587f2b8f46b0@:xx.xx.77.xxx’ Method: REGISTER

<— SIP read from UDP::xx.xx.77.xxx:57120 —>
REGISTER sip:xxx.xx.xxx.xxx SIP/2.0
Call-ID: 2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx
CSeq: 7516 REGISTER
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=1453620334
To: “nx4” sip:nx4@xxx.xx.xxx.xxx
Via: SIP/2.0/UDP:xx.xx.77.xxx:57120;branch=z9hG4bK015b3fd1093f830d5c3963d7449cda6c323939;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Contact: “nx4” sip:nx4@:57120:xx.xx.77.xxx;transport=udp
Expires: 3600
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to :angry:x.xx.77.xxx:57120 (no NAT)

<— Transmitting (NAT) to :angry:x.xx.77.xxx:57120 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 24.90.77.182:57120;branch=z9hG4bK015b3fd1093f830d5c3963d7449cda6c323939;received=:xx.xx.77.xxx;rport=57120
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=1453620334
To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as7a11e87e
Call-ID: 2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx
CSeq: 7516 REGISTER
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="21f9bff1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘9795944343362fbb2ff976c7b5838f86@:xx.xx.77.xxx’ Method: REGISTER

<— SIP read from UDP::xx.xx.77.xxx:5060 —>

<------------->
Really destroying SIP dialog ‘2ed25e42a5dbecde0244eba5c6c1c637@:xx.xx.77.xxx’ Method: REGISTER

<— SIP read from UDP:24.90.77.182:5060 —>

<------------->

EDIT******

I made the following changes, but still cannot get the sip phone to register. When I take the secret off, the IP shows up, but the device is still unreachable.

It looks like the ports are different, despite what I set them up to port 5060. I added this to my sip and reloaded (not shown here) and still have trouble with registering. help!

SIP****

[general]

context = unauthenticated
language = en
allowguest = no
srvlookup = yes
udpbindaddr = 0.0.0.0
tcpenable = no
pedantic = no
host = dynamic
registertimeout = 20 ;
allowoverlap = no ;
bindport = 5060 ;
qualify = yes ;
nat = force_rport, comedia
disallow= all
allow = ulaw
allow = alaw
allow = gsm
allow = g729

[nx4] ; call out

deny = 0.0.0.0/0.0.0.0
type = friend
host = dynamic
context = incoming
dtmfmode = inband ; changed from rfc… and worked
username = nx4 ; second part of sip peers --> channel/username
fromuser = nx4
user = nx4
outboundproxy = xxx.x1.xx8.x7x
secret = xxxxxxxx
canreinvite = no
nat = yes
transport = udp
permit = 0.0.0.0/0.0.0.0
callcounter = yes
faxdetect = no
disallow = all
allow = ulaw

Sip show peers*
Really destroying SIP dialog ‘70c72cc928c9254a313579c64ed53d04@ xxx.x1.xx8.x7x:5060’ Method: OPTIONS
asterisk*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
1001/1001 (Unspecified) D N 0 UNKNOWN
kbeezy/x0xxxx000 xx.xx.xx0.xx N 5060 OK (55 ms)
nx4/nx4 (Unspecified) D N A 0 UNKNOWN
office-phone (Unspecified) D N 0 UNKNOWN
zoiper/zoiper xx.xx.x7.xxx D 5060 OK (40 ms)
5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0 offline]
Really destroying SIP dialog ‘a0bbbefba59005b27993cb0f4a957ee6@24.90.77.182’ Method: REGISTER

Sip debug****
Retransmitting #3 (NAT) to xx.xx.xx.xxx:58078:
OPTIONS sip:nx4@xx.xx.xx.xxx:58078;transport=udp SIP/2.0
Via: SIP/2.0/UDP xxx.x1.xx8.x7x:5060;branch=z9hG4bK2185d521;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.x1.xx8.x7x;tag=as032e6299
To: sip:nx4@xx.xx.xx.xxx:58078;transport=udp
Contact: sip:asterisk@xxx.x1.xx8.x7x:5060
Call-ID: 68192523592fa2f0161ad7037f02a62a@xxx.x1.xx8.x7x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Tue, 23 Jul 2013 22:32:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #4 (NAT) to xx.xx.xx.xxx:58078:
OPTIONS sip:nx4@xx.xx.xx.xxx:58078;transport=udp SIP/2.0
Via: SIP/2.0/UDP xxx.x1.xx8.x7x:5060;branch=z9hG4bK2185d521;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.x1.xx8.x7x;tag=as032e6299
To: sip:nx4@xx.xx.xx.xxx:58078;transport=udp
Contact: sip:asterisk@xxx.x1.xx8.x7x:5060
Call-ID: 68192523592fa2f0161ad7037f02a62a@xxx.x1.xx8.x7x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Tue, 23 Jul 2013 22:32:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Really destroying SIP dialog '68192523592fa2f0161ad7037f02a62a@1xxx.x1.xx8.x7x:5060’ Method: OPTIONS
Really destroying SIP dialog '1a7a3d03c36ec8bbf28723c76d86e901@xx.xx.xx.xxx’ Method: REGISTER
Reliably Transmitting (NAT) to xx.xx.xx.xxx:58078:
OPTIONS sip:nx4@xx.xx.xx.xxx:58078;transport=udp SIP/2.0
Via: SIP/2.0/UDP xxx.x1.xx8.x7x:5060;branch=z9hG4bK43c75f32;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.x1.xx8.x7x;tag=as3263bd55
To: sip:nx4@xx.xx.xx.xxx:58078;transport=udp
Contact: sip:asterisk@xxx.x1.xx8.x7x:5060
Call-ID: 2f8c231e0c7271603f3f93332fdb50db@xxx.x1.xx8.x7x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Tue, 23 Jul 2013 22:32:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Never set insecure=invite on devices that may be given toll calling rights.

As well as normally being bad practice, allowguest does not work in device sections.

The phone should resend the register with authentication data. If it doesn’t, it suggests it doesn’t believe it has a password set (or cannot do basic authentication).

Hi David55, thanks for responding…

I took out allowguest and insecure, but I still can’t get the phone to register. I can see that the phone is trying to register (due to the feedback on its interface). I’m still not sure what the problem is. :frowning:

Can you take another look? Could it be something else that isn’t set correctly? Or is set incorrectly? I’d appreciate it!

Here is what I have on the phone itself, the sip.conf, debug and sip show peers:

******************** settings on actual phone************

Username: nx4
password: mypass
server: my.asterisk.box
outbound proxy address: my.asterisk.box
port number: 5060
transportation type: UDP
Send keep-alive: automatic
display name:
authentication username:

sip conf***

[general]

context = unauthenticated
language = en
allowguest = no
srvlookup = yes
udpbindaddr = 0.0.0.0
tcpenable = no
pedantic = no
host = dynamic
registertimeout = 20 ;
allowoverlap = no ;
bindport = 5060 ;
qualify = yes ;
nat = force_rport, comedia
disallow= all
allow = ulaw
allow = alaw
allow = gsm
allow = g729

[nx4]

deny = 0.0.0.0/0.0.0.0
type = friend
host = dynamic
context = incoming
dtmfmode = inband
username = nx4 ; second part of sip peers --> channel/username
fromuser = nx4
user = nx4
outboundproxy = xxx.xx.xxx.xxx ; asterisk box
secret =mypass
nat = yes
transport = udp
permit = 0.0.0.0/0.0.0.0
callcounter = yes
faxdetect = no
disallow = all
allow = ulaw
bindport =5060

ship show peers
Really destroying SIP dialog ‘1edaf14b7bd613b16bb168941aa7891a@198.61.168.179:5060’ Method: OPTIONS
asteriskCLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
1001/1001 (Unspecified) D N 0 UNKNOWN
kbeezy/xxxxxxxxx xx.xx.xxx.xx N 5060 OK (56 ms)
nx4/nx4 (Unspecified) D N A 0 UNKNOWN
office-phone (Unspecified) D N 0 UNKNOWN
zoiper/zoiper xx.xx.xx.xxx D 5060 OK (46 ms)
5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0 offline]
asterisk
CLI>

sip debug output****

Reliably Transmitting (NAT) to xx.xx.xxx.xx :5060:
OPTIONS sip:xx.xx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK20f3ad4a;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as35b1d57f
To: sip:xx.xx.xxx.xx
Contact: sip:asterisk@xxx.xx.xxx.xxx:5060
Call-ID: 12652ac42817c7205206be18601df5ce@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Thu, 25 Jul 2013 15:20:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '12652ac42817c7205206be18601df5ce@xxx.xx.xxx.xxx:5060’ Method: OPTIONS
Really destroying SIP dialog '305cf6cf6ea54ff642d864c5ebbf9bb3@xx.xx.xxx.xx’ Method: REGISTER

There is no register attempt in the current debug output. The previous one showed an abandoned one. It should go REGISTER, 401, REGISTER, 200, but stopped after the 401. That means that either you provided incomplete logs, or the phone realised that it was unable to authenticate, and gave up.

My apologies! Zoiper and IPKall work fine, so far. Nx4 is the one I’m having issues with.

Here’s a full one:

<— SIP read from UDP:xx.xx.xx.xxx:57753 —>
REGISTER sip:xxx.xx.xxx.xxx SIP/2.0
Call-ID: 50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx
CSeq: 6386 REGISTER
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=2778056207
To: “nx4” sip:nx4@xxx.xx.xxx.xxx
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bK225c148413dd14575ca6993ad7a5484d313631;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Contact: *
Expires: 0
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to xx.xx.xx.xxx:57753 (no NAT)

<— Transmitting (NAT) to xx.xx.xx.xxx:57753 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bK225c148413dd14575ca6993ad7a5484d313631;received=xx.xx.xx.xxx;rport=57753
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=2778056207
To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as02009c6f
Call-ID: 50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx
CSeq: 6386 REGISTER
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="227edcda"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:xx.xx.xx.xxx:5060 —>

<------------->
Reliably Transmitting (NAT) toxx.xx.xxx.xx:5060:
OPTIONS sip:xx.xx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK192b5810;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as701fca91
To: sip:xx.xx.xxx.xx
Contact: sip:asterisk@xxx.xx.xxx.xxx:5060
Call-ID: 77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Thu, 25 Jul 2013 17:15:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Reliably Transmitting (no NAT) to xx.xx.xx.xxx:5060:
OPTIONS sip:zoiper@xx.xx.xx.xxx:5060;rinstance=7bc586e32d632a74;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK21403b8c
Max-Forwards: 70
From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as56d43c49
To: sip:zoiper@xx.xx.xx.xxx:5060;rinstance=7bc586e32d632a74;transport=UDP
Contact: sip:asterisk@xxx.xx.xxx.xxx:5060
Call-ID: 2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Date: Thu, 25 Jul 2013 17:15:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:xx.xx.xxx.xx:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK192b5810;received=xxx.xx.xxx.xxx;rport=5060
From: “asterisk” sip:asterisk@xxx.xx.xxx.xxx;tag=as701fca91
To: sip:xx.xx.xxx.xx;tag=as3505656d
Call-ID: 77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
User-Agent: IPKall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '77208fab2f871c5d04e1382768d71bda@xxx.xx.xxx.xxx:5060’ Method: OPTIONS

<— SIP read from UDP:xx.xx.xx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xx.xxx.xxx:5060;branch=z9hG4bK21403b8c
Contact: sip:xx.xx.xx.xxx:5060
To: sip:zoiper@xx.xx.xx.xxx:5060;rinstance=7bc586e32d632a74;transport=UDP;tag=b7a22458
From: "asterisk"sip:asterisk@xxx.xx.xxx.xxx;tag=as56d43c49
Call-ID: 2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11619
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog '2bf05a3a6b9b931f5678e44d342d8dfd@xxx.xx.xxx.xxx:5060’ Method: OPTIONS

<— SIP read from UDP:xx.xx.xx.xxx:57753 —>
REGISTER sip:xxx.xx.xxx.xxx SIP/2.0
Call-ID: f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx
CSeq: 8575 REGISTER
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=96168691
To: “nx4” sip:nx4@xxx.xx.xxx.xxx
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bKb63f244a27efe1d3ff6996a361ffac24313631;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Contact: “nx4” sip:nx4@xx.xx.xx.xxx:57753;transport=udp
Expires: 3600
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to xx.xx.xx.xxx:57753 (no NAT)

<— Transmitting (NAT) to xx.xx.xx.xxx:57753 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bKb63f244a27efe1d3ff6996a361ffac24313631;received=xx.xx.xx.xxx;rport=57753
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=96168691
To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as6a01b7cd
Call-ID: f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx
CSeq: 8575 REGISTER
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5b605158"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'f1ad591c984e6c1fd7e79d09a9596748@xx.xx.xx.xxx’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '50d6618f51cfb430c1e047a0f470fd8f@xx.xx.xx.xxx’ Method: REGISTER

<— SIP read from UDP:xx.xx.xx.xxx:5060 —>

<------------->

<— SIP read from UDP:xx.xx.xx.xxx:57753 —>
REGISTER sip:xxx.xx.xxx.xxx SIP/2.0
Call-ID: 10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx
CSeq: 7849 REGISTER
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262
To: “nx4” sip:nx4@xxx.xx.xxx.xxx
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Contact: “nx4” sip:nx4@xx.xx.xx.xxx:57753;transport=udp
Expires: 3600
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to xx.xx.xx.xxx:57753 (no NAT)

<— Transmitting (NAT) to xx.xx.xx.xxx:57753 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;received=xx.xx.xx.xxx;rport=57753
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262
To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as245ea5e6
Call-ID: 10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx
CSeq: 7849 REGISTER
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6357a9e2"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:xx.xx.xx.xxx:57753 —>
REGISTER sip:xxx.xx.xxx.xxx SIP/2.0
Call-ID: 10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx
CSeq: 7849 REGISTER
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262
To: “nx4” sip:nx4@xxx.xx.xxx.xxx
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Contact: “nx4” sip:nx4@xx.xx.xx.xxx:57753;transport=udp
Expires: 3600
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to xx.xx.xx.xxx:57753 (NAT)

<— Transmitting (NAT) to xx.xx.xx.xxx:57753 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.xx.xxx:57753;branch=z9hG4bKd474bf3983b685f00d33eae6216a04c2313631;received=xx.xx.xx.xxx;rport=57753
From: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=4020576262
To: “nx4” sip:nx4@xxx.xx.xxx.xxx;tag=as245ea5e6
Call-ID: 10bf44298552e260bc84d1d276c99788@xx.xx.xx.xxx
CSeq: 7849 REGISTER
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6357a9e2"
Content-Length: 0

As well as the problems that I already mentioned, in relation to the phone, xx.xx.77.xxx:57120 may not be routable from the Asterisk side.

In all cases, the phone is not making an appropriate response to the 401.

Hi David55,

Thanks again for taking a look. Would a different SIP app on the phone help? I’m currently trying SIPDroid versus the native SIP settings via the phone.

Or, does this mean that the phone won’t work at all?

Can anyone suggest some work-arounds?

Again, thanks for the time and the suggestions! I appreciate it!

Well…

I changed “secret” to “remotesecret”, reloaded my sip settings and checked my peers. SipDroid only works with pbxes.org to tunnel VOIP calls. The Zoiper VoIP app works great!

On to the next step! Wooo-who! 8)

Using only remotesecret for a peer that is registering with you effectively disables authentication for that registration!