SIP Help - New to Asterisk

Asterisk Asterisk General
1

Hello all, I am very new to Asterisk so please forgive my ignorance. I have 2 SIP software clients at different locations (NAT’d) which I would like to be able to talk to one another with video. They will be using H.263 for RTC and G.711 for audio. I have successfully compiled my Asterisk server and have it setup enough that each client can log in but I cannot ring either client. Furthermore I am unable to figure out how to get each client to login like xxxxxxxxxx@xxx.xxx.xxx.xxx(ip address) in order to see any traffic get generated at the other(ringing) client end. Even though I can see some traffic there is no ringing (more on that later.) Regardless, I don’t want to have to specify the endpoint IP when I dial because presumably both ends will be dynamic. My asterisk server has a static IP and is also NAT’d. I’m using TCP as I’ve had less success with UDP for authentication. Also my Asterisk is version 1.8.1.

I am a firewall guy so doing port forward for 5060 is no problem, of coarse there are some ephemeral ports in use as well and right now I have everything open to my Asterisk box, so nothing of that nature is being blocked (nothing is blocked here, I’ve checked my logs.) But, what I am seeing is that my client does not listen on port 5060. My client will pick a high port at random and listen on that, which it will tell to the SIP server when it registers. Whats strange is that when I dial that client, Asterisk always calls it on 5060 which results in an ICMP Port unreachable being sent back from my client. I believe this is because Asterisk does not know it is dialing a client which has registered because as I mentioned earlier, I have to dial like xxxxxxxxxx@domain.com which will result in nothing happening or an error on the console that extension does not exist or something.

My dial plan is very basic:
[default]
exten => 1111111111,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}@${SIPDOMAIN})
exten => 1111111112,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}@${SIPDOMAIN})
that is all that really gets used right now and there is a MYCURRENTDOMAIN=mydomain.com under the globals.

My sip.conf looks like this(IP’s/Domain obscured):
[general]
bindport=5060
bindaddr=192.168.0.30
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=g711
allow=h263p
callerid=unknown
context=default ; Default context for incoming calls
realm=mydomain.com
localnet=192.168.0.0/255.255.0.0
externaddr=xxx.xxx.xxx.xxx
externip=xxx.xxx.xxx.xxx
nat=yes
tcpenable=yes
transport=tcp
allowsubscribe=yes
notifyhold=yes
notifyringing=yes
limitonpeer=yes
videosupport=yes
t38pt_udptl=yes ; Default false
srvlookup=yes
domain=mydomain.com
autodomain=no
subscribecontext=default
canreinvite=no
fromdomain=mydomain.com
allowexternaldomains=yes
allowguest=no

[1111111111]
qualify=no
nat=yes
callerid=1111111111@mydomain.com <1111111111>
context=default
canreinvite=no
fromuser=1111111111@mydomain.com
fromdomain=mydomain.com
secret=xxxxxxxxxx
host=dynamic
username=1111111111@mydomain.com
dtmfmode=inband
type=friend
mailbox=1112
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
allow=g711

[1111111112]
qualify=no
nat=yes
callerid= 1111111112 <1111111112>
context=default
canreinvite=no
fromuser=1111111112
fromdomain=mydomain.com
secret=xxxxxxxxxx
host=dynamic
username=1111111112
dtmfmode=inband
type=friend
mailbox=1113
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
allow=g711

Basic Output:

[Dec 13 14:39:48] WARNING[4217]: chan_sip.c:8836 process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
– Executing [1111111112@default:1] Macro(“SIP/1111111111-00000008”, “stdexten,1111111112,SIP/1111111112@[WAN IP=xxx.xxx.xxx.xxx]”) in new stack
– Executing [s@macro-stdexten:1] Dial(“SIP/1111111111-00000008”, “SIP/1111111112@[WAN IP=xxx.xxx.xxx.xxx],10,tTr”) in new stack
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
– Called 1111111112@[WAN IP=xxx.xxx.xxx.xxx]
– Nobody picked up in 10000 ms

Can any help me? I’m sure I’ve got those conf files right buggered, but I’ve tried so many variations of each setting I’ve lost track now.

Thanks in advance for any assistance.

2

Howdy,

So are both of the clients registering successfully?

“sip show peers registered”

3

They can be depending on what iteration I’m currently in… haha…

proxy*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
1111111111/1111111111@dom xxx.xxx.xxx.xxx D 12254 Unmonitored
1111111112/1111111112@dom xxx.xxx.xxx.xxx D N 33504 Unmonitored
domain.com xxx.xxx.xxx.xxx N 5060 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]

I’ve made some changes since yesterday, still no luck though.

4

Howdy,

What kind of clients are these? Maybe someone has run into this before?

5

It is called “Vista Media Center Video Phone for Windows”, it’s a small app made by a company called OABSoftware.nl. It is free and as a result there is no way to contact them for support. It’s basically a video phone client that can integrate into Media Center. I’ve seen it work, sort of, with a free SIP proxy (sip2sip.info) but it’s so strange I have to open/close it dozens of times for it to work, which is why I thought I would try and make my own SIP Server.

Here is a copy of the packet which I do receive on the receiving end (no ring though.)

IP: SRC=WAN of Asterisk, DST=WAN of Receiver
TCP Port: SRC=Random, DST=5060 (client not listening on 5060, client listens on random and indicates this to Asterisk on upon registration so why does Asterisk dial him on 5060?)

INVITE sip:1111111111@[WAN IP of receiver] SIP/2.0
Via: SIP/2.0/UDP [Asterisk WAN IP HERE]:5060;branch=z9hG4bK287eed7e;rport
Max-Forwards: 70
From: "1111111112@domain.com" sip:1111111112@domain.com;tag=as5bcdfadf
To: <sip:1111111111@[WAN IP of receiver]>
Contact: <sip:1111111112@[Asterisk WAN IP HERE]:5060>
Call-ID: 5f6ca01717098e3550d6eb7e1ce83e1b@domain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1
Date: Tue, 14 Dec 2010 22:20:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 386

v=0
o=root 2103608930 2103608930 IN IP4 [Asterisk WAN IP HERE] s=Asterisk PBX 1.8.1 c=IN IP4 [Asterisk WAN IP HERE]
b=CT:384
t=0 0
m=audio 10070 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10052 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

Keep in mind I had to dial 1111111111@[WAN IP of receiver] to get that.

6

Interesting, I’ve simplified my sip.conf file and I’m getting slightly better results, but still no ring.

Now my clients look like this:

[1111111111]
secret=xxxxxx
host=dynamic
dtmfmode=inband ;tried info and rfc2833 too
type=friend

[1111111112]
secret=xxxxxx
host=dynamic
dtmfmode=inband
type=friend

and I can actually dial 1111111112@domain.com and see some traffic come to that device. Also the SIP INVITE packet which gets there has the proper destination port which is the random one used by the client and registered to asterisk. So I’m getting closer now, but still no ring.

7

If you’re dialing from one phone to another using the direct IP of the receiving phone, then you’re not dialing through Asterisk in the normal B2BUA sense. What if you simply dial the extension, without the domain? (I’m assuming the phone is forwarding the request to Asterisk as though it’s the proxy).

What does the SIP debug from Asterisk look like when it receives the initial invite from the first peer?
Does Asterisk attempt to invite the second peer (leg of the call)?

8

I can now dial using the domain name, not the ip and that part is working fine, I know it is dialing correctly now because it is using the correct ephemeral port ranges that the clients have registered with, and in addition I can see the packets show up at the Callee and they look good. Still no ringing, and all of my tests are done 3 times using inband, info, and rfc2833 for dtmfmode. For reference to these logs I should mention my Callee is currently at a static IP address, and is not being NAT’d (in the future this will not be the case and I have nat=no for that user=1111111111.) After viewing all the packets on both sides I can confirm that there is no NAT issues here, because neither client ever tries to establish a connection directly with any RFC1918 addresses, they always communicate back to the * Server.

Here is the sip debug output from a dial:

[code]<— SIP read from TCP:[CALLER PUBLIC IP]:33076 —>
INVITE sip:1111111111@domain.com SIP/2.0
Via: SIP/2.0/TCP [CALLER PRIVATE IP]:10282
Max-Forwards: 70
From: “1111111112” sip:1111111112@domain.com;tag=17a15ad7f338434998291725630b6d43;epid=fbe17d53ab
To: sip:1111111111@domain.com
Call-ID: b021e8876c7c4215b2fcff0301557df6
CSeq: 2 INVITE
Contact: <sip:1111111112@domain.com:10282;maddr=[CALLER PRIVATE IP];transport=tcp>;proxy=replace
User-Agent: RTC/1.3
Authorization: Digest username=“1111111112”, realm=“domain.com”, algorithm=MD5, uri="sip:1111111111@domain.com", nonce=“7da87ca7”, response="0c2805441c16f51d1eb6913f70438e01"
Content-Type: application/sdp
Content-Length: 686

v=0
o=- 0 0 IN IP4 [CALLER PRIVATE IP]
s=session
c=IN IP4 [CALLER PRIVATE IP]
b=CT:303
t=0 0
m=audio 16308 RTP/AVP 97 111 112 6 0 8 4 5 3 101
k=base64:Tx00aMZxSQN19mtlKeMAqHe/TRmaWJX9d9Dt7j6oTBs
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:optional
m=video 41010 RTP/AVP 34 31
k=base64:0mMiX4mCEIMrDBZviwl/AAoYX7AuW7a2kfn1HFa6tME
a=recvonly
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=encryption:optional
<------------->
— (12 headers 28 lines) —
Sending to [CALLER PUBLIC IP]:33076 (NAT)
Using INVITE request as basis request - b021e8876c7c4215b2fcff0301557df6
Found peer ‘1111111112’ for ‘1111111112’ from [CALLER PUBLIC IP]:33076
== Using SIP RTP CoS mark 5
Found RTP audio format 97
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 3
Found RTP audio format 101
Found audio description format red for ID 97
Found audio description format SIREN for ID 111
Found audio description format G7221 for ID 112
[Dec 16 10:34:37] WARNING[14141]: chan_sip.c:8836 process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
Found audio description format DVI4 for ID 6
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format DVI4 for ID 5
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found RTP video format 31
Found video description format H263 for ID 34
Found video description format H261 for ID 31
Capabilities: us - 0x18000e (gsm|ulaw|alaw|h263|h263p), peer - audio=0xc2f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc)/video=0xc0000 (h261|h263)/text=0x0 (nothing), combined - 0x8000e (gsm|ulaw|alaw|h263)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port [CALLER PRIVATE IP]:16308
Peer video RTP is at port [CALLER PRIVATE IP]:41010
Looking for 1111111111 in default (domain domain.com)
list_route: hop: <sip:1111111112@domain.com:10282;maddr=[CALLER PRIVATE IP];transport=tcp>

<— Transmitting (NAT) to [CALLER PUBLIC IP]:33076 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP [CALLER PRIVATE IP]:10282;received=[CALLER PUBLIC IP];rport=33076
From: “1111111112” sip:1111111112@domain.com;tag=17a15ad7f338434998291725630b6d43;epid=fbe17d53ab
To: sip:1111111111@domain.com
Call-ID: b021e8876c7c4215b2fcff0301557df6
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1111111111@[* PUBLIC IP]:5060;transport=TCP>
Content-Length: 0

<------------>
– Executing [1111111111@default:1] Dial(“SIP/1111111112-00000000”, “SIP/1111111111,20”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 5060
Video is at [* PUBLIC IP]:5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding video codec 0x80000 (h263) to SDP
Reliably Transmitting (no NAT) to [CALLEE PUBLIC IP]:13037:
INVITE sip:[CALLEE PUBLIC IP]:13037;transport=tcp SIP/2.0
Via: SIP/2.0/TCP [* PUBLIC IP]:5060;branch=z9hG4bK75404463
Max-Forwards: 70
From: “1111111112” sip:1111111112@domain.com;tag=as6ebd62a1
To: <sip:[CALLEE PUBLIC IP]:13037;transport=tcp>
Contact: <sip:1111111112@[* PUBLIC IP]:5060;transport=TCP>
Call-ID: 1bb1203d6e335fe760b42fc34fe5cb92@domain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1
Date: Thu, 16 Dec 2010 17:34:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 43335454 43335454 IN IP4 [* PUBLIC IP]
s=Asterisk PBX 1.8.1
c=IN IP4 [* PUBLIC IP]
b=CT:384
t=0 0
m=audio 10060 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10072 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

-- Called 1111111111
-- Nobody picked up in 20000 ms[/code]

This is my 1111111111 client registering:

[code]<— SIP read from TCP:[1111111111 PUBLIC IP]:54812 —>
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185
Max-Forwards: 70
From: sip:1111111111@domain.com;tag=000c1f0e0cce4efdb90f229c7042a5de;epid=e8c96fe4b6
To: sip:1111111111@domain.com
Call-ID: 7bcad561ec0648baa7450571d16bd6fa
CSeq: 1 REGISTER
Contact: <sip:[1111111111 PUBLIC IP]:9185;transport=tcp>;methods=“INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BENOTIFY”;proxy=replace
User-Agent: RTC/1.3.5369 (Vista Media Center SIP Phone 2.4.0.0)
ms-keep-alive: UAC;hop-hop=yes
Event: registration
Allow-Events: presence
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to [1111111111 PUBLIC IP]:54812 (NAT)

<— Transmitting (no NAT) to [1111111111 PUBLIC IP]:9185 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185;received=[1111111111 PUBLIC IP]
From: sip:1111111111@domain.com;tag=000c1f0e0cce4efdb90f229c7042a5de;epid=e8c96fe4b6
To: sip:1111111111@domain.com;tag=as79b79a76
Call-ID: 7bcad561ec0648baa7450571d16bd6fa
CSeq: 1 REGISTER
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“domain.com”, nonce="354de1e7"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7bcad561ec0648baa7450571d16bd6fa’ in 32000 ms (Method: REGISTER)

<— SIP read from TCP:[1111111111 PUBLIC IP]:54812 —>
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185
Max-Forwards: 70
From: sip:1111111111@domain.com;tag=000c1f0e0cce4efdb90f229c7042a5de;epid=e8c96fe4b6
To: sip:1111111111@domain.com
Call-ID: 7bcad561ec0648baa7450571d16bd6fa
CSeq: 2 REGISTER
Contact: <sip:[1111111111 PUBLIC IP]:9185;transport=tcp>;methods=“INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BENOTIFY”;proxy=replace
User-Agent: RTC/1.3.5369 (Vista Media Center SIP Phone 2.4.0.0)
Authorization: Digest username=“1111111111”, realm=“domain.com”, algorithm=MD5, uri=“sip:domain.com”, nonce=“354de1e7”, response="74dcfb2f8d5f7f03171c64e8ed27fc09"
ms-keep-alive: UAC;hop-hop=yes
Event: registration
Allow-Events: presence
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to [1111111111 PUBLIC IP]:9185 (no NAT)
– Registered SIP ‘1111111111’ at [1111111111 PUBLIC IP]:9185

<— Transmitting (no NAT) to [1111111111 PUBLIC IP]:9185 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185;received=[1111111111 PUBLIC IP]
From: sip:1111111111@domain.com;tag=000c1f0e0cce4efdb90f229c7042a5de;epid=e8c96fe4b6
To: sip:1111111111@domain.com;tag=as79b79a76
Call-ID: 7bcad561ec0648baa7450571d16bd6fa
CSeq: 2 REGISTER
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:[1111111111 PUBLIC IP]:9185;transport=tcp>;expires=120
Date: Thu, 16 Dec 2010 18:32:56 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7bcad561ec0648baa7450571d16bd6fa’ in 32000 ms (Method: REGISTER)

<— SIP read from TCP:[1111111111 PUBLIC IP]:54812 —>
SUBSCRIBE sip:1111111112@domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185
Max-Forwards: 70
From: “1111111111” sip:1111111111@domain.com;tag=ad5143c368ff4bc981cea057c7348007;epid=e8c96fe4b6
To: sip:1111111112@domain.com
Call-ID: a6f5692c313d4af090a6c31e718f96bc
CSeq: 1 SUBSCRIBE
Contact: <sip:1111111111@domain.com:9185;maddr=[1111111111 PUBLIC IP];transport=tcp>;proxy=replace
User-Agent: RTC/1.3
Event: presence
Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml
Supported: com.microsoft.autoextend
Supported: ms-benotify
Proxy-Require: ms-benotify
Supported: ms-piggyback-first-notify
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to [1111111111 PUBLIC IP]:54812 (NAT)
list_route: hop: <sip:1111111111@domain.com:9185;maddr=[1111111111 PUBLIC IP];transport=tcp>
Found peer ‘1111111111’ for ‘1111111111’ from [1111111111 PUBLIC IP]:54812

<— Transmitting (no NAT) to [1111111111 PUBLIC IP]:9185 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185;received=[1111111111 PUBLIC IP]
From: “1111111111” sip:1111111111@domain.com;tag=ad5143c368ff4bc981cea057c7348007;epid=e8c96fe4b6
To: sip:1111111112@domain.com;tag=as1ae98532
Call-ID: a6f5692c313d4af090a6c31e718f96bc
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“domain.com”, nonce="331b7624"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘a6f5692c313d4af090a6c31e718f96bc’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from TCP:[1111111111 PUBLIC IP]:54812 —>
SUBSCRIBE sip:1111111112@domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185
Max-Forwards: 70
From: “1111111111” sip:1111111111@domain.com;tag=ad5143c368ff4bc981cea057c7348007;epid=e8c96fe4b6
To: sip:1111111112@domain.com
Call-ID: a6f5692c313d4af090a6c31e718f96bc
CSeq: 2 SUBSCRIBE
Contact: <sip:1111111111@domain.com:9185;maddr=[1111111111 PUBLIC IP];transport=tcp>;proxy=replace
User-Agent: RTC/1.3
Authorization: Digest username=“1111111111”, realm=“domain.com”, algorithm=MD5, uri="sip:1111111112@domain.com", nonce=“331b7624”, response="009940edde37a4fad0d9c1dcb21eeb88"
Event: presence
Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml
Supported: com.microsoft.autoextend
Supported: ms-benotify
Proxy-Require: ms-benotify
Supported: ms-piggyback-first-notify
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to [1111111111 PUBLIC IP]:9185 (no NAT)
Found peer ‘1111111111’ for ‘1111111111’ from [1111111111 PUBLIC IP]:54812
Looking for 1111111112 in default (domain domain.com)

<— Transmitting (no NAT) to [1111111111 PUBLIC IP]:9185 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP [1111111111 PUBLIC IP]:9185;received=[1111111111 PUBLIC IP]
From: “1111111111” sip:1111111111@domain.com;tag=ad5143c368ff4bc981cea057c7348007;epid=e8c96fe4b6
To: sip:1111111112@domain.com;tag=as1ae98532
Call-ID: a6f5692c313d4af090a6c31e718f96bc
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘a6f5692c313d4af090a6c31e718f96bc’ Method: SUBSCRIBE
[/code]

This is my 1111111112 client registering:

[code]<— SIP read from TCP:[1111111112 PUBLIC IP]:32900 —>
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796
Max-Forwards: 70
From: sip:1111111112@domain.com;tag=6e5af4d44ea24e6ba810d91c22c38981;epid=468f5f2371
To: sip:1111111112@domain.com
Call-ID: 03e5765f566640a08a32e33b6e698246
CSeq: 1 REGISTER
Contact: <sip:[1111111112 PRIVATE IP]:12796;transport=tcp>;methods=“INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BENOTIFY”;proxy=replace
User-Agent: RTC/1.3.5369 (Vista Media Center SIP Phone 2.4.0.0)
ms-keep-alive: UAC;hop-hop=yes
Event: registration
Allow-Events: presence
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to [1111111112 PUBLIC IP]:32900 (NAT)

<— Transmitting (NAT) to [1111111112 PUBLIC IP]:32900 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796;received=[1111111112 PUBLIC IP];rport=32900
From: sip:1111111112@domain.com;tag=6e5af4d44ea24e6ba810d91c22c38981;epid=468f5f2371
To: sip:1111111112@domain.com;tag=as14ac5358
Call-ID: 03e5765f566640a08a32e33b6e698246
CSeq: 1 REGISTER
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“domain.com”, nonce="279205a2"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘03e5765f566640a08a32e33b6e698246’ in 32000 ms (Method: REGISTER)

<— SIP read from TCP:[1111111112 PUBLIC IP]:32900 —>
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796
Max-Forwards: 70
From: sip:1111111112@domain.com;tag=6e5af4d44ea24e6ba810d91c22c38981;epid=468f5f2371
To: sip:1111111112@domain.com
Call-ID: 03e5765f566640a08a32e33b6e698246
CSeq: 2 REGISTER
Contact: <sip:[1111111112 PRIVATE IP]:12796;transport=tcp>;methods=“INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER, BENOTIFY”;proxy=replace
User-Agent: RTC/1.3.5369 (Vista Media Center SIP Phone 2.4.0.0)
Authorization: Digest username=“1111111112”, realm=“domain.com”, algorithm=MD5, uri=“sip:domain.com”, nonce=“279205a2”, response="4e604f508376cb3872d4ef1958ffcdf4"
ms-keep-alive: UAC;hop-hop=yes
Event: registration
Allow-Events: presence
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to [1111111112 PUBLIC IP]:32900 (NAT)
– Registered SIP ‘1111111112’ at [1111111112 PUBLIC IP]:32900

<— Transmitting (NAT) to [1111111112 PUBLIC IP]:32900 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796;received=[1111111112 PUBLIC IP];rport=32900
From: sip:1111111112@domain.com;tag=6e5af4d44ea24e6ba810d91c22c38981;epid=468f5f2371
To: sip:1111111112@domain.com;tag=as14ac5358
Call-ID: 03e5765f566640a08a32e33b6e698246
CSeq: 2 REGISTER
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:[1111111112 PRIVATE IP]:12796;transport=tcp>;expires=120
Date: Thu, 16 Dec 2010 18:35:41 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘03e5765f566640a08a32e33b6e698246’ in 32000 ms (Method: REGISTER)

<— SIP read from TCP:[1111111112 PUBLIC IP]:32900 —>
SUBSCRIBE sip:1111111111@domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796
Max-Forwards: 70
From: “1111111112” sip:1111111112@domain.com;tag=71f71077d1104118ba0168e247016bf1;epid=468f5f2371
To: sip:1111111111@domain.com
Call-ID: 69ce7af753d645089dd5236bfc70ce22
CSeq: 1 SUBSCRIBE
Contact: <sip:1111111112@domain.com:12796;maddr=[1111111112 PRIVATE IP];transport=tcp>;proxy=replace
User-Agent: RTC/1.3
Event: presence
Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml
Supported: com.microsoft.autoextend
Supported: ms-benotify
Proxy-Require: ms-benotify
Supported: ms-piggyback-first-notify
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to [1111111112 PUBLIC IP]:32900 (NAT)
list_route: hop: <sip:1111111112@domain.com:12796;maddr=[1111111112 PRIVATE IP];transport=tcp>
Found peer ‘1111111112’ for ‘1111111112’ from [1111111112 PUBLIC IP]:32900

<— Transmitting (NAT) to [1111111112 PUBLIC IP]:32900 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796;received=[1111111112 PUBLIC IP];rport=32900
From: “1111111112” sip:1111111112@domain.com;tag=71f71077d1104118ba0168e247016bf1;epid=468f5f2371
To: sip:1111111111@domain.com;tag=as4b20d05b
Call-ID: 69ce7af753d645089dd5236bfc70ce22
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“domain.com”, nonce="408d2627"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘69ce7af753d645089dd5236bfc70ce22’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from TCP:[1111111112 PUBLIC IP]:32900 —>
SUBSCRIBE sip:1111111111@domain.com SIP/2.0
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796
Max-Forwards: 70
From: “1111111112” sip:1111111112@domain.com;tag=71f71077d1104118ba0168e247016bf1;epid=468f5f2371
To: sip:1111111111@domain.com
Call-ID: 69ce7af753d645089dd5236bfc70ce22
CSeq: 2 SUBSCRIBE
Contact: <sip:1111111112@domain.com:12796;maddr=[1111111112 PRIVATE IP];transport=tcp>;proxy=replace
User-Agent: RTC/1.3
Authorization: Digest username=“1111111112”, realm=“domain.com”, algorithm=MD5, uri="sip:1111111111@domain.com", nonce=“408d2627”, response="4268620615546d79d1fb37aad9a4ed81"
Event: presence
Accept: application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml
Supported: com.microsoft.autoextend
Supported: ms-benotify
Proxy-Require: ms-benotify
Supported: ms-piggyback-first-notify
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to [1111111112 PUBLIC IP]:32900 (NAT)
Found peer ‘1111111112’ for ‘1111111112’ from [1111111112 PUBLIC IP]:32900
Looking for 1111111111 in default (domain domain.com)

<— Transmitting (NAT) to [1111111112 PUBLIC IP]:32900 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP [1111111112 PRIVATE IP]:12796;received=[1111111112 PUBLIC IP];rport=32900
From: “1111111112” sip:1111111112@domain.com;tag=71f71077d1104118ba0168e247016bf1;epid=468f5f2371
To: sip:1111111111@domain.com;tag=as4b20d05b
Call-ID: 69ce7af753d645089dd5236bfc70ce22
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog ‘69ce7af753d645089dd5236bfc70ce22’ Method: SUBSCRIBE
[/code]

9

Sorry, someone more skilled than I is going to have to tell you why the phone isn’t liking the invite from Asterisk as it tries to setup the second leg of the call.

:frowning:

10

Thanks for trying anyway.

One other thing, this software I use has the ability to check on contacts status, so if it knows a contact is online it will show in green otherwise it’s red. In my case my contacts always show in red, which tells me that this software is trying to do some other kind of communication with the * Server and it is failing that. When I login to sip2sip.info instead of my * Server, my online contacts go green. There must be something else there as well.