thr3ads.net - asterisk users - [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS [May 2011]

If this information is useful, please help other people find it:
Share via:

GNUbie

2011-May-08 00:59 UTC

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

Hello all,

I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still getting a 401 Unauthorized SIP message. So as of this
writing, I still cannot successfully REGISTER to my Asterisk box.

Below are the snippets of my Asterisk and SNOM 300 configurations
including the logs for your reference.

I hope anyone from this community can help me solve this problem. A
HOWTO of a similar scenario will help a lot.

Thank you in advance.

Regards,

GNUbie

- - - ASTERISK v1.8.3.3 - - -

[ /etc/asterisk/sip.conf ]

[general]
...
...
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/pbx.domain.com.pem
tlscipher=ALL
tlsclientmethod=tlsv1
tlsbindport=5061
externtlsport=5061
externtcpport=5061
tcpbindaddr=0.0.0.0
tcpbindport=5061
tcpenable=yes
srvlookup=yes

[361]
username=361
secret=*******
callerid="361-tls"<361>
mailbox=361 at family
context=family
transport=tls
port=5061
type=friend
host=dynamic
dtmfmode=rfc2833
canreinvite=no
nat=yes
qualify=yes
autoframing=yes
encryption=yes

*CLI> core show version
Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a
x86_64 running Linux on 2011-04-22 17:50:44 UTC

*CLI> sip show settings

Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
TLS SIP Bindaddress: 0.0.0.0:5061
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: Yes
Realm. auth: No
Our auth realm pbx.domain.com
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: "Asterisk rocks!"
SDP Session Name: Asterisk PBX 1.8.3.3-1digium1~squeeze
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No

Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 200
Jitterbuffer resync: 1200
Jitterbuffer impl: fixed
Jitterbuffer log: No

Network Settings:
---------------------------
SIP address remapping: Enabled using externhost
Externhost: pbx.domain.com
externaddr: 11.22.33.44:0
Externrefresh: 10
Localnet: 192.168.101.0/255.255.255.0

Global Signalling Settings:
---------------------------
Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc)
Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 15
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 1800 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Refuse
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 3000
Timer T1 minimum: 100
Timer B: 192000
No premature media: Yes
Max forwards: 70

Default Settings:
-----------------
Allowed transports: UDP
Outbound transport:	 UDP
Context: default
Force rport: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

*CLI> sip show peer 361

* Name : 361
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : family
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 361 at family
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "361-tls" <361>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 3000
Timer B : 192000
ToHost :
Addr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : TLS
Allowed.Trsp : TLS
Def. Username: 361
SIP Options : (none)
Codecs : 0x60e (gsm|ulaw|alaw|speex|ilbc)
Codec Order : (ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30)
Auto-Framing : Yes
100 on REG : No
Status : UNKNOWN
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Refuse
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : Yes


<--- SIP read from TLS:192.168.101.102:2061 --->
REGISTER sip:pbx.domain.com SIP/2.0
Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;rport
From: "361" <sip:361 at pbx.domain.com>;tag=6ulxay5gxm
To: "361" <sip:361 at pbx.domain.com>
Call-ID: 3c26701f2ede-afeuhg58c60m
CSeq: 7 REGISTER
Max-Forwards: 70
Contact: <sip:361 at
192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.101.102
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.101.102:2061 (no NAT)

<--- Transmitting (NAT) to 192.168.101.102:2061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS
192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;received=192.168.101.102;rport=2061
From: "361" <sip:361 at pbx.domain.com>;tag=6ulxay5gxm
To: "361" <sip:361 at pbx.domain.com>;tag=as16189b66
Call-ID: 3c26701f2ede-afeuhg58c60m
CSeq: 7 REGISTER
Server: "Asterisk rocks!"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.domain.com",
nonce="6408e8c3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c26701f2ede-afeuhg58c60m' in
192000 ms (Method: REGISTER)

<--- SIP read from TLS:192.168.101.102:2061 --->
REGISTER sip:pbx.domain.com SIP/2.0
Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-9cuvn4fglawu;rport
From: "361" <sip:361 at pbx.domain.com>;tag=hr7nz4nopk
To: "361" <sip:361 at pbx.domain.com>
Call-ID: 3c26701f2ede-afeuhg58c60m
CSeq: 8 REGISTER
Max-Forwards: 70
Contact: <sip:361 at
192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.101.102
Supported: path, gruu
Expires: 3600
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.101.102:2061 (no NAT)

<--- Transmitting (NAT) to 192.168.101.102:2061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS
192.168.101.102:2061;branch=z9hG4bK-9cuvn4fglawu;received=192.168.101.102;rport=2061
From: "361" <sip:361 at pbx.domain.com>;tag=hr7nz4nopk
To: "361" <sip:361 at pbx.domain.com>;tag=as6231d59a
Call-ID: 3c26701f2ede-afeuhg58c60m
CSeq: 8 REGISTER
Server: "Asterisk rocks!"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.domain.com",
nonce="6ea5895a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c26701f2ede-afeuhg58c60m' in
192000 ms (Method: REGISTER)

- - - SNOM 300 - - -

[ Setup > Identity 1 > Login ]

Displayname: 361
Account: 361
Password: ********
Registrar: pbx.domain.com
Outbound Proxy: sips:pbx.domain.com:5061
Authentication Username: 361

- - -

[ Setup > Certificates > Server Certificates ]

Country: ; State: ; Locality ; Organization: ; Common Name:
pbx.domain.com; eMail:
Version:	2
Serial Number:	00b6b63eb67ed2111345253c228264d093
Signature Algorithm:	1.2.840.113549.1.1.5 (sha1WithRSAEncryption)
Signature:
28ce574c9715e1e59dfc90829287ab31fdbf0e0212dc488b106e71ffaaa339610492dc091d440772...
Issuer:	Country: GB; State: Greater Manchester; Locality Salford;
Organization: Comodo CA Limited; Common Name: PositiveSSL CA; eMail:
Validity:	27/04/11 - 26/04/12
SHA1-Fingerprint:	38d13c709ab1cc9b434c2f05e927239fe4ae6f19
MD5-Fingerprint:	a9b62e186465055f34a04153ad7898de
PK Algorithm:	1.2.840.113549.1.1.1 (rsaEncryption)
RSA modulus:
00b90412744fd50459d807a04d007a9fd7d667189f1394f11ecd46e8556bd861526eb9be582a2631...
RSA exponent:	010001
Filename on FS:	f6700ff3f3059f4c629df2bff8678aeacb291ddb.DER

- - -

[ Status > System Information ]

System Information:
Phone Type:	snom300-SIP
MAC-Address:	0004132F08DC
IP-Address:	192.168.101.102
Firmware-Version:	snom300-SIP 8.4.31
Firmware-URL:	http://provisioning.....4.31-SIP-f.bin
Production Information:	Mac:0004132F08DC;Version:Standard;Hardware:snom300
(H: R2A);Date:15/05/08;Copyright? snom technology AG
Uptime:	0 days, 1 hours, 27 minutes
LCS:	0 days, 0 hours, 53 minutes (0)
Memfree:	772 K
CPU:	0.04 0.02 0.03 1/10 96
Bootloader-Version:	1.1.3-u

SIP Identity Status:
Identity 1 Status:	361 at pbx.domain.com: Network Failure

- - -

[ Status > SIP Trace ]

Sent to tls:11.22.33.44:5061 at 24/12/2001 08:00:32:192 (729 bytes):
REGISTER sip:pbx.domain.com SIP/2.0
Via: SIP/2.0/TLS 192.168.101.102:2055;branch=z9hG4bK-9i3rt6llzqd1;rport
From: "361" <sip:361 at pbx.domain.com>;tag=hpleutmwxu
To: "361" <sip:361 at pbx.domain.com>
Call-ID: 3c26701f3456-58is2wtgld05
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: <sip:361 at
192.168.101.102:2055;transport=tls>;q=1.0;reg-id=1;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="
INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.101.102
Supported: path, gruu
Expires: 3600
Content-Length: 0
Sent to tls:11.22.33.44:5061 at 8/5/2011 00:24:03:610 (729 bytes):

REGISTER sip:pbx.domain.com SIP/2.0
Via: SIP/2.0/TLS 192.168.101.102:2056;branch=z9hG4bK-lriexp5iqoio;rport
From: "361" <sip:361 at pbx.domain.com>;tag=b11o8j7lk4
To: "361" <sip:361 at pbx.domain.com>
Call-ID: 3c26701f3456-58is2wtgld05
CSeq: 2 REGISTER
Max-Forwards: 70
Contact: <sip:361 at
192.168.101.102:2056;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="
INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/8.4.31
Allow-Events: dialog
X-Real-IP: 192.168.101.102
Supported: path, gruu
Expires: 3600
Content-Length: 0

- - -

[ Status > Log ]

[0] 24/12/2001 00:00:27: Phone::uboot_version:1.1.3-u
[1] 24/12/2001 00:00:29: Conf setup: code: 500, host: 127.0.0.1:80,
file: /dummy.htm
[0] 24/12/2001 08:00:31: TaskMon: LCS 21/0 recv LPCP took 1271 msecs
[0] 24/12/2001 08:00:31: LoopMon: LCS 21 took 1271 (290/0) msecs, read
1, 3/1 tasks
[1] 24/12/2001 08:00:32: TLS: Warning: Certificate with subject
Country: ; State: ; Locality ; Organization: ; Common Name:
pbx.domain.com; eMail: has expired according to the local time of the
phone.
[0] 24/12/2001 08:00:33: TaskMon: LCS 30/0 recv LPCP took 934 msecs
[0] 24/12/2001 08:00:33: LoopMon: LCS 30 took 968 (42/32) msecs, read
1, 3/1 tasks
[0] 8/5/2011 00:22:49: TaskMon: LCS 93/0 recv LPCP took 434 msecs
[0] 8/5/2011 00:22:49: TaskMon: LCS 94/0 recv LPCP took 461 msecs
[0] 8/5/2011 00:22:50: TaskMon: LCS 96/0 recv LPCP took 576 msecs
[0] 8/5/2011 00:23:03: TaskMon: LCS 148/0 recv LPCP took 238 msecs
[2] 8/5/2011 00:23:03: Transport Error: Pending packet 1000000: generating fake
[2] 8/5/2011 00:23:03: Registrar 361 at pbx.domain.com timed out
[0] 8/5/2011 00:23:05: TaskMon: LCS 157/0 recv LPCP took 372 msecs
[0] 8/5/2011 00:23:05: LoopMon: LCS 157 took 850 (499/478) msecs, read
1, 4/1 tasks
[0] 8/5/2011 00:24:04: TaskMon: LCS 359/0 recv LPCP took 872 msecs
[0] 8/5/2011 00:24:04: LoopMon: LCS 359 took 872 (306/0) msecs, read
1, 3/1 tasks
[2] 8/5/2011 00:24:34: Transport Error: Pending packet 1000002: generating fake
[2] 8/5/2011 00:24:34: Registrar 361 at pbx.domain.com timed out
[0] 8/5/2011 00:24:48: TaskMon: LCS 508/0 recv LPCP took 443 msecs
[0] 8/5/2011 00:24:48: LoopMon: LCS 508 took 444 (16/0) msecs, read 1, 3/1 tasks
[0] 8/5/2011 00:24:48: TaskMon: LCS 509/0 recv LPCP took 506 msecs
[0] 8/5/2011 00:24:48: LoopMon: LCS 509 took 507 (72/0) msecs, read 1, 4/1 tasks
[0] 8/5/2011 00:24:49: TaskMon: LCS 510/0 recv LPCP took 1293 msecs
[0] 8/5/2011 00:24:49: LoopMon: LCS 510 took 1337 (500/0) msecs, read
1, 5/1 tasks
[0] 8/5/2011 00:25:35: TaskMon: LCS 673/0 recv LPCP took 871 msecs
[0] 8/5/2011 00:25:35: LoopMon: LCS 673 took 871 (118/0) msecs, read
1, 3/1 tasks
[2] 8/5/2011 00:26:05: Transport Error: Pending packet 1000004: generating fake
[2] 8/5/2011 00:26:05: Registrar 361 at pbx.domain.com timed out
[0] 8/5/2011 00:27:06: TaskMon: LCS 986/0 recv LPCP took 871 msecs
[0] 8/5/2011 00:27:06: LoopMon: LCS 986 took 871 (419/0) msecs, read
1, 3/1 tasks
[2] 8/5/2011 00:27:36: Transport Error: Pending packet 1000006: generating fake
[2] 8/5/2011 00:27:36: Registrar 361 at pbx.domain.com timed out
[0] 8/5/2011 00:28:37: TaskMon: LCS 1296/0 recv LPCP took 869 msecs
[0] 8/5/2011 00:28:37: LoopMon: LCS 1296 took 870 (387/0) msecs, read
1, 3/1 tasks
[2] 8/5/2011 00:29:07: Transport Error: Pending packet 1000008: generating fake
[2] 8/5/2011 00:29:07: Registrar 361 at pbx.domain.com timed out
[0] 8/5/2011 00:30:08: TaskMon: LCS 1605/0 recv LPCP took 870 msecs
[0] 8/5/2011 00:30:08: LoopMon: LCS 1605 took 871 (458/0) msecs, read
1, 3/1 tasks
[2] 8/5/2011 00:30:38: Transport Error: Pending packet 1000010: generating fake
[2] 8/5/2011 00:30:38: Registrar 361 at pbx.domain.com timed out
[0] 8/5/2011 00:31:39: TaskMon: LCS 1918/0 recv LPCP took 874 msecs
[0] 8/5/2011 00:31:39: LoopMon: LCS 1918 took 875 (346/0) msecs, read
1, 3/1 tasks
[0] 8/5/2011 00:32:03: TaskMon: LCS 1996/0 recv LPCP took 424 msecs
[0] 8/5/2011 00:32:03: LoopMon: LCS 1996 took 430 (24/4) msecs, read
1, 3/1 tasks

Paul Hayes

2011-May-09 18:14 UTC

head link

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

Hi,

It looks to me that the 401 unauth packets aren't getting back to the
phones. Which suggests a network/router/nat issue rather than anything wrong
with the asterisk or phone configuration.

Cheers,
Paul.



On 8 May 2011, at 01:59, GNUbie <gnubie at gmail.com> wrote:
> Hello all,
> 
> I have installed the .deb packages of the Asterisk v1.8.3.3 from the
> upstream project on my Debian GNU/Linux Squeeze server and bought the
> Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
> exercise. After setting up everything and trying to fix this problem,
> I am still getting a 401 Unauthorized SIP message. So as of this
> writing, I still cannot successfully REGISTER to my Asterisk box.
> 
> Below are the snippets of my Asterisk and SNOM 300 configurations
> including the logs for your reference.
> 
> I hope anyone from this community can help me solve this problem. A
> HOWTO of a similar scenario will help a lot.
> 
> Thank you in advance.
> 
> Regards,
> 
> GNUbie
> 
> - - - ASTERISK v1.8.3.3 - - -
> 
> [ /etc/asterisk/sip.conf ]
> 
> [general]
> ...
> ...
> tlsenable=yes
> tlsbindaddr=0.0.0.0
> tlscertfile=/etc/asterisk/keys/pbx.domain.com.pem
> tlscipher=ALL
> tlsclientmethod=tlsv1
> tlsbindport=5061
> externtlsport=5061
> externtcpport=5061
> tcpbindaddr=0.0.0.0
> tcpbindport=5061
> tcpenable=yes
> srvlookup=yes
> 
> [361]
> username=361
> secret=*******
> callerid="361-tls"<361>
> mailbox=361 at family
> context=family
> transport=tls
> port=5061
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> canreinvite=no
> nat=yes
> qualify=yes
> autoframing=yes
> encryption=yes
> 
> *CLI> core show version
> Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a
> x86_64 running Linux on 2011-04-22 17:50:44 UTC
> 
> *CLI> sip show settings
> 
> Global Settings:
> ----------------
> UDP Bindaddress: 0.0.0.0:5060
> TCP SIP Bindaddress: 0.0.0.0:5060
> TLS SIP Bindaddress: 0.0.0.0:5061
> Videosupport: No
> Textsupport: No
> Ignore SDP sess. ver.: No
> AutoCreate Peer: No
> Match Auth Username: No
> Allow unknown access: No
> Allow subscriptions: Yes
> Allow overlap dialing: Yes
> Allow promsic. redir: No
> Enable call counters: No
> SIP domain support: Yes
> Realm. auth: No
> Our auth realm pbx.domain.com
> Use domains as realms: No
> Call to non-local dom.: Yes
> URI user is phone no: No
> Always auth rejects: Yes
> Direct RTP setup: No
> User Agent: "Asterisk rocks!"
> SDP Session Name: Asterisk PBX 1.8.3.3-1digium1~squeeze
> SDP Owner Name: root
> Reg. context: (not set)
> Regexten on Qualify: No
> Caller ID: asterisk
> From: Domain:
> Record SIP history: Off
> Call Events: Off
> Auth. Failure Events: Off
> T.38 support: No
> T.38 EC mode: Unknown
> T.38 MaxDtgrm: -1
> SIP realtime: Disabled
> Qualify Freq : 60000 ms
> Q.850 Reason header: No
> 
> Network QoS Settings:
> ---------------------------
> IP ToS SIP: CS0
> IP ToS RTP audio: CS0
> IP ToS RTP video: CS0
> IP ToS RTP text: CS0
> 802.1p CoS SIP: 4
> 802.1p CoS RTP audio: 5
> 802.1p CoS RTP video: 6
> 802.1p CoS RTP text: 5
> Jitterbuffer enabled: Yes
> Jitterbuffer forced: No
> Jitterbuffer max size: 200
> Jitterbuffer resync: 1200
> Jitterbuffer impl: fixed
> Jitterbuffer log: No
> 
> Network Settings:
> ---------------------------
> SIP address remapping: Enabled using externhost
> Externhost: pbx.domain.com
> externaddr: 11.22.33.44:0
> Externrefresh: 10
> Localnet: 192.168.101.0/255.255.255.0
> 
> Global Signalling Settings:
> ---------------------------
> Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc)
> Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30
> Relax DTMF: No
> RFC2833 Compensation: No
> Symmetric RTP: No
> Compact SIP headers: No
> RTP Keepalive: 0 (Disabled)
> RTP Timeout: 15
> RTP Hold Timeout: 0 (Disabled)
> MWI NOTIFY mime type: application/simple-message-summary
> DNS SRV lookup: Yes
> Pedantic SIP support: Yes
> Reg. min duration 1800 secs
> Reg. max duration: 3600 secs
> Reg. default duration: 120 secs
> Outbound reg. timeout: 20 secs
> Outbound reg. attempts: 0
> Notify ringing state: Yes
> Include CID: No
> Notify hold state: No
> SIP Transfer mode: open
> Max Call Bitrate: 384 kbps
> Auto-Framing: No
> Outb. proxy: <not set>
> Session Timers: Refuse
> Session Refresher: uas
> Session Expires: 1800 secs
> Session Min-SE: 90 secs
> Timer T1: 3000
> Timer T1 minimum: 100
> Timer B: 192000
> No premature media: Yes
> Max forwards: 70
> 
> Default Settings:
> -----------------
> Allowed transports: UDP
> Outbound transport:     UDP
> Context: default
> Force rport: No
> DTMF: rfc2833
> Qualify: 0
> Use ClientCode: No
> Progress inband: Never
> Language:
> MOH Interpret: default
> MOH Suggest:
> Voice Mail Extension: asterisk
> 
> *CLI> sip show peer 361
> 
> * Name : 361
> Secret : <Set>
> MD5Secret : <Not set>
> Remote Secret: <Not set>
> Context : family
> Subscr.Cont. : <Not set>
> Language :
> AMA flags : Unknown
> Transfer mode: open
> CallingPres : Presentation Allowed, Not Screened
> Callgroup :
> Pickupgroup :
> MOH Suggest :
> Mailbox : 361 at family
> VM Extension : asterisk
> LastMsgsSent : 32767/65535
> Call limit : 0
> Max forwards : 0
> Dynamic : Yes
> Callerid : "361-tls" <361>
> MaxCallBR : 384 kbps
> Expire : -1
> Insecure : no
> Force rport : Yes
> ACL : No
> DirectMedACL : No
> T.38 support : No
> T.38 EC mode : Unknown
> T.38 MaxDtgrm: -1
> DirectMedia : No
> PromiscRedir : No
> User=Phone : No
> Video Support: No
> Text Support : No
> Ign SDP ver : No
> Trust RPID : No
> Send RPID : No
> Subscriptions: Yes
> Overlap dial : Yes
> DTMFmode : rfc2833
> Timer T1 : 3000
> Timer B : 192000
> ToHost :
> Addr->IP : (null)
> Defaddr->IP : (null)
> Prim.Transp. : TLS
> Allowed.Trsp : TLS
> Def. Username: 361
> SIP Options : (none)
> Codecs : 0x60e (gsm|ulaw|alaw|speex|ilbc)
> Codec Order : (ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30)
> Auto-Framing : Yes
> 100 on REG : No
> Status : UNKNOWN
> Useragent :
> Reg. Contact :
> Qualify Freq : 60000 ms
> Sess-Timers : Refuse
> Sess-Refresh : uas
> Sess-Expires : 1800 secs
> Min-Sess : 90 secs
> RTP Engine : asterisk
> Parkinglot :
> Use Reason : No
> Encryption : Yes
> 
> 
> <--- SIP read from TLS:192.168.101.102:2061 --->
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=6ulxay5gxm
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 7 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
> 
> <------------->
> --- (14 headers 0 lines) ---
> Sending to 192.168.101.102:2061 (no NAT)
> 
> <--- Transmitting (NAT) to 192.168.101.102:2061 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TLS
>
192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;received=192.168.101.102;rport=2061
> From: "361" <sip:361 at pbx.domain.com>;tag=6ulxay5gxm
> To: "361" <sip:361 at pbx.domain.com>;tag=as16189b66
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 7 REGISTER
> Server: "Asterisk rocks!"
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="pbx.domain.com",
nonce="6408e8c3"
> Content-Length: 0
> 
> 
> <------------>
> Scheduling destruction of SIP dialog '3c26701f2ede-afeuhg58c60m' in
> 192000 ms (Method: REGISTER)
> 
> <--- SIP read from TLS:192.168.101.102:2061 --->
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-9cuvn4fglawu;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=hr7nz4nopk
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 8 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
> 
> <------------->
> --- (14 headers 0 lines) ---
> Sending to 192.168.101.102:2061 (no NAT)
> 
> <--- Transmitting (NAT) to 192.168.101.102:2061 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TLS
>
192.168.101.102:2061;branch=z9hG4bK-9cuvn4fglawu;received=192.168.101.102;rport=2061
> From: "361" <sip:361 at pbx.domain.com>;tag=hr7nz4nopk
> To: "361" <sip:361 at pbx.domain.com>;tag=as6231d59a
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 8 REGISTER
> Server: "Asterisk rocks!"
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="pbx.domain.com",
nonce="6ea5895a"
> Content-Length: 0
> 
> 
> <------------>
> Scheduling destruction of SIP dialog '3c26701f2ede-afeuhg58c60m' in
> 192000 ms (Method: REGISTER)
> 
> - - - SNOM 300 - - -
> 
> [ Setup > Identity 1 > Login ]
> 
> Displayname: 361
> Account: 361
> Password: ********
> Registrar: pbx.domain.com
> Outbound Proxy: sips:pbx.domain.com:5061
> Authentication Username: 361
> 
> - - -
> 
> [ Setup > Certificates > Server Certificates ]
> 
> Country: ; State: ; Locality ; Organization: ; Common Name:
> pbx.domain.com; eMail:
> Version:    2
> Serial Number:    00b6b63eb67ed2111345253c228264d093
> Signature Algorithm:    1.2.840.113549.1.1.5 (sha1WithRSAEncryption)
> Signature:   
28ce574c9715e1e59dfc90829287ab31fdbf0e0212dc488b106e71ffaaa339610492dc091d440772...
> Issuer:    Country: GB; State: Greater Manchester; Locality Salford;
> Organization: Comodo CA Limited; Common Name: PositiveSSL CA; eMail:
> Validity:    27/04/11 - 26/04/12
> SHA1-Fingerprint:    38d13c709ab1cc9b434c2f05e927239fe4ae6f19
> MD5-Fingerprint:    a9b62e186465055f34a04153ad7898de
> PK Algorithm:    1.2.840.113549.1.1.1 (rsaEncryption)
> RSA modulus:   
00b90412744fd50459d807a04d007a9fd7d667189f1394f11ecd46e8556bd861526eb9be582a2631...
> RSA exponent:    010001
> Filename on FS:    f6700ff3f3059f4c629df2bff8678aeacb291ddb.DER
> 
> - - -
> 
> [ Status > System Information ]
> 
> System Information:
> Phone Type:    snom300-SIP
> MAC-Address:    0004132F08DC
> IP-Address:    192.168.101.102
> Firmware-Version:    snom300-SIP 8.4.31
> Firmware-URL:    http://provisioning.....4.31-SIP-f.bin
> Production Information:   
Mac:0004132F08DC;Version:Standard;Hardware:snom300
> (H: R2A);Date:15/05/08;Copyright? snom technology AG
> Uptime:    0 days, 1 hours, 27 minutes
> LCS:    0 days, 0 hours, 53 minutes (0)
> Memfree:    772 K
> CPU:    0.04 0.02 0.03 1/10 96
> Bootloader-Version:    1.1.3-u
> 
> SIP Identity Status:
> Identity 1 Status:    361 at pbx.domain.com: Network Failure
> 
> - - -
> 
> [ Status > SIP Trace ]
> 
> Sent to tls:11.22.33.44:5061 at 24/12/2001 08:00:32:192 (729 bytes):
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2055;branch=z9hG4bK-9i3rt6llzqd1;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=hpleutmwxu
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f3456-58is2wtgld05
> CSeq: 1 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2055;transport=tls>;q=1.0;reg-id=1;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="
>
INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
> Sent to tls:11.22.33.44:5061 at 8/5/2011 00:24:03:610 (729 bytes):
> 
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2056;branch=z9hG4bK-lriexp5iqoio;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=b11o8j7lk4
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f3456-58is2wtgld05
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2056;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="
>
INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
> 
> - - -
> 
> [ Status > Log ]
> 
> [0] 24/12/2001 00:00:27: Phone::uboot_version:1.1.3-u
> [1] 24/12/2001 00:00:29: Conf setup: code: 500, host: 127.0.0.1:80,
> file: /dummy.htm
> [0] 24/12/2001 08:00:31: TaskMon: LCS 21/0 recv LPCP took 1271 msecs
> [0] 24/12/2001 08:00:31: LoopMon: LCS 21 took 1271 (290/0) msecs, read
> 1, 3/1 tasks
> [1] 24/12/2001 08:00:32: TLS: Warning: Certificate with subject
> Country: ; State: ; Locality ; Organization: ; Common Name:
> pbx.domain.com; eMail: has expired according to the local time of the
> phone.
> [0] 24/12/2001 08:00:33: TaskMon: LCS 30/0 recv LPCP took 934 msecs
> [0] 24/12/2001 08:00:33: LoopMon: LCS 30 took 968 (42/32) msecs, read
> 1, 3/1 tasks
> [0] 8/5/2011 00:22:49: TaskMon: LCS 93/0 recv LPCP took 434 msecs
> [0] 8/5/2011 00:22:49: TaskMon: LCS 94/0 recv LPCP took 461 msecs
> [0] 8/5/2011 00:22:50: TaskMon: LCS 96/0 recv LPCP took 576 msecs
> [0] 8/5/2011 00:23:03: TaskMon: LCS 148/0 recv LPCP took 238 msecs
> [2] 8/5/2011 00:23:03: Transport Error: Pending packet 1000000: generating
fake
> [2] 8/5/2011 00:23:03: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:23:05: TaskMon: LCS 157/0 recv LPCP took 372 msecs
> [0] 8/5/2011 00:23:05: LoopMon: LCS 157 took 850 (499/478) msecs, read
> 1, 4/1 tasks
> [0] 8/5/2011 00:24:04: TaskMon: LCS 359/0 recv LPCP took 872 msecs
> [0] 8/5/2011 00:24:04: LoopMon: LCS 359 took 872 (306/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:24:34: Transport Error: Pending packet 1000002: generating
fake
> [2] 8/5/2011 00:24:34: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:24:48: TaskMon: LCS 508/0 recv LPCP took 443 msecs
> [0] 8/5/2011 00:24:48: LoopMon: LCS 508 took 444 (16/0) msecs, read 1, 3/1
tasks
> [0] 8/5/2011 00:24:48: TaskMon: LCS 509/0 recv LPCP took 506 msecs
> [0] 8/5/2011 00:24:48: LoopMon: LCS 509 took 507 (72/0) msecs, read 1, 4/1
tasks
> [0] 8/5/2011 00:24:49: TaskMon: LCS 510/0 recv LPCP took 1293 msecs
> [0] 8/5/2011 00:24:49: LoopMon: LCS 510 took 1337 (500/0) msecs, read
> 1, 5/1 tasks
> [0] 8/5/2011 00:25:35: TaskMon: LCS 673/0 recv LPCP took 871 msecs
> [0] 8/5/2011 00:25:35: LoopMon: LCS 673 took 871 (118/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:26:05: Transport Error: Pending packet 1000004: generating
fake
> [2] 8/5/2011 00:26:05: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:27:06: TaskMon: LCS 986/0 recv LPCP took 871 msecs
> [0] 8/5/2011 00:27:06: LoopMon: LCS 986 took 871 (419/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:27:36: Transport Error: Pending packet 1000006: generating
fake
> [2] 8/5/2011 00:27:36: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:28:37: TaskMon: LCS 1296/0 recv LPCP took 869 msecs
> [0] 8/5/2011 00:28:37: LoopMon: LCS 1296 took 870 (387/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:29:07: Transport Error: Pending packet 1000008: generating
fake
> [2] 8/5/2011 00:29:07: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:30:08: TaskMon: LCS 1605/0 recv LPCP took 870 msecs
> [0] 8/5/2011 00:30:08: LoopMon: LCS 1605 took 871 (458/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:30:38: Transport Error: Pending packet 1000010: generating
fake
> [2] 8/5/2011 00:30:38: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:31:39: TaskMon: LCS 1918/0 recv LPCP took 874 msecs
> [0] 8/5/2011 00:31:39: LoopMon: LCS 1918 took 875 (346/0) msecs, read
> 1, 3/1 tasks
> [0] 8/5/2011 00:32:03: TaskMon: LCS 1996/0 recv LPCP took 424 msecs
> [0] 8/5/2011 00:32:03: LoopMon: LCS 1996 took 430 (24/4) msecs, read
> 1, 3/1 tasks
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

GNUbie

2011-May-19 07:19 UTC

head link

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

Anyone? Please advice. Thank you.

On Sun, May 8, 2011 at 8:59 AM, GNUbie <gnubie at gmail.com>
wrote:> Hello all,
>
> I have installed the .deb packages of the Asterisk v1.8.3.3 from the
> upstream project on my Debian GNU/Linux Squeeze server and bought the
> Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
> exercise. After setting up everything and trying to fix this problem,
> I am still getting a 401 Unauthorized SIP message. So as of this
> writing, I still cannot successfully REGISTER to my Asterisk box.
>
> Below are the snippets of my Asterisk and SNOM 300 configurations
> including the logs for your reference.
>
> I hope anyone from this community can help me solve this problem. A
> HOWTO of a similar scenario will help a lot.
>
> Thank you in advance.
>
> Regards,
>
> GNUbie
>
> - - - ASTERISK v1.8.3.3 - - -
>
> [ /etc/asterisk/sip.conf ]
>
> [general]
> ...
> ...
> tlsenable=yes
> tlsbindaddr=0.0.0.0
> tlscertfile=/etc/asterisk/keys/pbx.domain.com.pem
> tlscipher=ALL
> tlsclientmethod=tlsv1
> tlsbindport=5061
> externtlsport=5061
> externtcpport=5061
> tcpbindaddr=0.0.0.0
> tcpbindport=5061
> tcpenable=yes
> srvlookup=yes
>
> [361]
> username=361
> secret=*******
> callerid="361-tls"<361>
> mailbox=361 at family
> context=family
> transport=tls
> port=5061
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> canreinvite=no
> nat=yes
> qualify=yes
> autoframing=yes
> encryption=yes
>
> *CLI> core show version
> Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a
> x86_64 running Linux on 2011-04-22 17:50:44 UTC
>
> *CLI> sip show settings
>
> Global Settings:
> ----------------
> UDP Bindaddress: 0.0.0.0:5060
> TCP SIP Bindaddress: 0.0.0.0:5060
> TLS SIP Bindaddress: 0.0.0.0:5061
> Videosupport: No
> Textsupport: No
> Ignore SDP sess. ver.: No
> AutoCreate Peer: No
> Match Auth Username: No
> Allow unknown access: No
> Allow subscriptions: Yes
> Allow overlap dialing: Yes
> Allow promsic. redir: No
> Enable call counters: No
> SIP domain support: Yes
> Realm. auth: No
> Our auth realm pbx.domain.com
> Use domains as realms: No
> Call to non-local dom.: Yes
> URI user is phone no: No
> Always auth rejects: Yes
> Direct RTP setup: No
> User Agent: "Asterisk rocks!"
> SDP Session Name: Asterisk PBX 1.8.3.3-1digium1~squeeze
> SDP Owner Name: root
> Reg. context: (not set)
> Regexten on Qualify: No
> Caller ID: asterisk
> From: Domain:
> Record SIP history: Off
> Call Events: Off
> Auth. Failure Events: Off
> T.38 support: No
> T.38 EC mode: Unknown
> T.38 MaxDtgrm: -1
> SIP realtime: Disabled
> Qualify Freq : 60000 ms
> Q.850 Reason header: No
>
> Network QoS Settings:
> ---------------------------
> IP ToS SIP: CS0
> IP ToS RTP audio: CS0
> IP ToS RTP video: CS0
> IP ToS RTP text: CS0
> 802.1p CoS SIP: 4
> 802.1p CoS RTP audio: 5
> 802.1p CoS RTP video: 6
> 802.1p CoS RTP text: 5
> Jitterbuffer enabled: Yes
> Jitterbuffer forced: No
> Jitterbuffer max size: 200
> Jitterbuffer resync: 1200
> Jitterbuffer impl: fixed
> Jitterbuffer log: No
>
> Network Settings:
> ---------------------------
> SIP address remapping: Enabled using externhost
> Externhost: pbx.domain.com
> externaddr: 11.22.33.44:0
> Externrefresh: 10
> Localnet: 192.168.101.0/255.255.255.0
>
> Global Signalling Settings:
> ---------------------------
> Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc)
> Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30
> Relax DTMF: No
> RFC2833 Compensation: No
> Symmetric RTP: No
> Compact SIP headers: No
> RTP Keepalive: 0 (Disabled)
> RTP Timeout: 15
> RTP Hold Timeout: 0 (Disabled)
> MWI NOTIFY mime type: application/simple-message-summary
> DNS SRV lookup: Yes
> Pedantic SIP support: Yes
> Reg. min duration 1800 secs
> Reg. max duration: 3600 secs
> Reg. default duration: 120 secs
> Outbound reg. timeout: 20 secs
> Outbound reg. attempts: 0
> Notify ringing state: Yes
> Include CID: No
> Notify hold state: No
> SIP Transfer mode: open
> Max Call Bitrate: 384 kbps
> Auto-Framing: No
> Outb. proxy: <not set>
> Session Timers: Refuse
> Session Refresher: uas
> Session Expires: 1800 secs
> Session Min-SE: 90 secs
> Timer T1: 3000
> Timer T1 minimum: 100
> Timer B: 192000
> No premature media: Yes
> Max forwards: 70
>
> Default Settings:
> -----------------
> Allowed transports: UDP
> Outbound transport: ? ? ?UDP
> Context: default
> Force rport: No
> DTMF: rfc2833
> Qualify: 0
> Use ClientCode: No
> Progress inband: Never
> Language:
> MOH Interpret: default
> MOH Suggest:
> Voice Mail Extension: asterisk
>
> *CLI> sip show peer 361
>
> * Name : 361
> Secret : <Set>
> MD5Secret : <Not set>
> Remote Secret: <Not set>
> Context : family
> Subscr.Cont. : <Not set>
> Language :
> AMA flags : Unknown
> Transfer mode: open
> CallingPres : Presentation Allowed, Not Screened
> Callgroup :
> Pickupgroup :
> MOH Suggest :
> Mailbox : 361 at family
> VM Extension : asterisk
> LastMsgsSent : 32767/65535
> Call limit : 0
> Max forwards : 0
> Dynamic : Yes
> Callerid : "361-tls" <361>
> MaxCallBR : 384 kbps
> Expire : -1
> Insecure : no
> Force rport : Yes
> ACL : No
> DirectMedACL : No
> T.38 support : No
> T.38 EC mode : Unknown
> T.38 MaxDtgrm: -1
> DirectMedia : No
> PromiscRedir : No
> User=Phone : No
> Video Support: No
> Text Support : No
> Ign SDP ver : No
> Trust RPID : No
> Send RPID : No
> Subscriptions: Yes
> Overlap dial : Yes
> DTMFmode : rfc2833
> Timer T1 : 3000
> Timer B : 192000
> ToHost :
> Addr->IP : (null)
> Defaddr->IP : (null)
> Prim.Transp. : TLS
> Allowed.Trsp : TLS
> Def. Username: 361
> SIP Options : (none)
> Codecs : 0x60e (gsm|ulaw|alaw|speex|ilbc)
> Codec Order : (ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30)
> Auto-Framing : Yes
> 100 on REG : No
> Status : UNKNOWN
> Useragent :
> Reg. Contact :
> Qualify Freq : 60000 ms
> Sess-Timers : Refuse
> Sess-Refresh : uas
> Sess-Expires : 1800 secs
> Min-Sess : 90 secs
> RTP Engine : asterisk
> Parkinglot :
> Use Reason : No
> Encryption : Yes
>
>
> <--- SIP read from TLS:192.168.101.102:2061 --->
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=6ulxay5gxm
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 7 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
>
> <------------->
> --- (14 headers 0 lines) ---
> Sending to 192.168.101.102:2061 (no NAT)
>
> <--- Transmitting (NAT) to 192.168.101.102:2061 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TLS
>
192.168.101.102:2061;branch=z9hG4bK-b6veg4r2tybi;received=192.168.101.102;rport=2061
> From: "361" <sip:361 at pbx.domain.com>;tag=6ulxay5gxm
> To: "361" <sip:361 at pbx.domain.com>;tag=as16189b66
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 7 REGISTER
> Server: "Asterisk rocks!"
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="pbx.domain.com",
nonce="6408e8c3"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '3c26701f2ede-afeuhg58c60m' in
> 192000 ms (Method: REGISTER)
>
> <--- SIP read from TLS:192.168.101.102:2061 --->
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2061;branch=z9hG4bK-9cuvn4fglawu;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=hr7nz4nopk
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 8 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2061;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
>
> <------------->
> --- (14 headers 0 lines) ---
> Sending to 192.168.101.102:2061 (no NAT)
>
> <--- Transmitting (NAT) to 192.168.101.102:2061 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TLS
>
192.168.101.102:2061;branch=z9hG4bK-9cuvn4fglawu;received=192.168.101.102;rport=2061
> From: "361" <sip:361 at pbx.domain.com>;tag=hr7nz4nopk
> To: "361" <sip:361 at pbx.domain.com>;tag=as6231d59a
> Call-ID: 3c26701f2ede-afeuhg58c60m
> CSeq: 8 REGISTER
> Server: "Asterisk rocks!"
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="pbx.domain.com",
nonce="6ea5895a"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '3c26701f2ede-afeuhg58c60m' in
> 192000 ms (Method: REGISTER)
>
> - - - SNOM 300 - - -
>
> [ Setup > Identity 1 > Login ]
>
> Displayname: 361
> Account: 361
> Password: ********
> Registrar: pbx.domain.com
> Outbound Proxy: sips:pbx.domain.com:5061
> Authentication Username: 361
>
> - - -
>
> [ Setup > Certificates > Server Certificates ]
>
> Country: ; State: ; Locality ; Organization: ; Common Name:
> pbx.domain.com; eMail:
> Version: ? ? ? ?2
> Serial Number: ?00b6b63eb67ed2111345253c228264d093
> Signature Algorithm: ? ?1.2.840.113549.1.1.5 (sha1WithRSAEncryption)
> Signature: ? ?
?28ce574c9715e1e59dfc90829287ab31fdbf0e0212dc488b106e71ffaaa339610492dc091d440772...
> Issuer: Country: GB; State: Greater Manchester; Locality Salford;
> Organization: Comodo CA Limited; Common Name: PositiveSSL CA; eMail:
> Validity: ? ? ? 27/04/11 - 26/04/12
> SHA1-Fingerprint: ? ? ? 38d13c709ab1cc9b434c2f05e927239fe4ae6f19
> MD5-Fingerprint: ? ? ? ?a9b62e186465055f34a04153ad7898de
> PK Algorithm: ? 1.2.840.113549.1.1.1 (rsaEncryption)
> RSA modulus: ?
?00b90412744fd50459d807a04d007a9fd7d667189f1394f11ecd46e8556bd861526eb9be582a2631...
> RSA exponent: ? 010001
> Filename on FS: f6700ff3f3059f4c629df2bff8678aeacb291ddb.DER
>
> - - -
>
> [ Status > System Information ]
>
> System Information:
> Phone Type: ? ? snom300-SIP
> MAC-Address: ? ?0004132F08DC
> IP-Address: ? ? 192.168.101.102
> Firmware-Version: ? ? ? snom300-SIP 8.4.31
> Firmware-URL: ? http://provisioning.....4.31-SIP-f.bin
> Production Information: Mac:0004132F08DC;Version:Standard;Hardware:snom300
> (H: R2A);Date:15/05/08;Copyright? snom technology AG
> Uptime: 0 days, 1 hours, 27 minutes
> LCS: ? ?0 days, 0 hours, 53 minutes (0)
> Memfree: ? ? ? ?772 K
> CPU: ? ?0.04 0.02 0.03 1/10 96
> Bootloader-Version: ? ? 1.1.3-u
>
> SIP Identity Status:
> Identity 1 Status: ? ? ?361 at pbx.domain.com: Network Failure
>
> - - -
>
> [ Status > SIP Trace ]
>
> Sent to tls:11.22.33.44:5061 at 24/12/2001 08:00:32:192 (729 bytes):
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2055;branch=z9hG4bK-9i3rt6llzqd1;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=hpleutmwxu
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f3456-58is2wtgld05
> CSeq: 1 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2055;transport=tls>;q=1.0;reg-id=1;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="
>
INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
> Sent to tls:11.22.33.44:5061 at 8/5/2011 00:24:03:610 (729 bytes):
>
> REGISTER sip:pbx.domain.com SIP/2.0
> Via: SIP/2.0/TLS 192.168.101.102:2056;branch=z9hG4bK-lriexp5iqoio;rport
> From: "361" <sip:361 at pbx.domain.com>;tag=b11o8j7lk4
> To: "361" <sip:361 at pbx.domain.com>
> Call-ID: 3c26701f3456-58is2wtgld05
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Contact: <sip:361 at
192.168.101.102:2056;transport=tls>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a473ab2-1159-4286-9cdb-385c32d8003d>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="
>
INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
> User-Agent: snom300/8.4.31
> Allow-Events: dialog
> X-Real-IP: 192.168.101.102
> Supported: path, gruu
> Expires: 3600
> Content-Length: 0
>
> - - -
>
> [ Status > Log ]
>
> [0] 24/12/2001 00:00:27: Phone::uboot_version:1.1.3-u
> [1] 24/12/2001 00:00:29: Conf setup: code: 500, host: 127.0.0.1:80,
> file: /dummy.htm
> [0] 24/12/2001 08:00:31: TaskMon: LCS 21/0 recv LPCP took 1271 msecs
> [0] 24/12/2001 08:00:31: LoopMon: LCS 21 took 1271 (290/0) msecs, read
> 1, 3/1 tasks
> [1] 24/12/2001 08:00:32: TLS: Warning: Certificate with subject
> Country: ; State: ; Locality ; Organization: ; Common Name:
> pbx.domain.com; eMail: has expired according to the local time of the
> phone.
> [0] 24/12/2001 08:00:33: TaskMon: LCS 30/0 recv LPCP took 934 msecs
> [0] 24/12/2001 08:00:33: LoopMon: LCS 30 took 968 (42/32) msecs, read
> 1, 3/1 tasks
> [0] 8/5/2011 00:22:49: TaskMon: LCS 93/0 recv LPCP took 434 msecs
> [0] 8/5/2011 00:22:49: TaskMon: LCS 94/0 recv LPCP took 461 msecs
> [0] 8/5/2011 00:22:50: TaskMon: LCS 96/0 recv LPCP took 576 msecs
> [0] 8/5/2011 00:23:03: TaskMon: LCS 148/0 recv LPCP took 238 msecs
> [2] 8/5/2011 00:23:03: Transport Error: Pending packet 1000000: generating
fake
> [2] 8/5/2011 00:23:03: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:23:05: TaskMon: LCS 157/0 recv LPCP took 372 msecs
> [0] 8/5/2011 00:23:05: LoopMon: LCS 157 took 850 (499/478) msecs, read
> 1, 4/1 tasks
> [0] 8/5/2011 00:24:04: TaskMon: LCS 359/0 recv LPCP took 872 msecs
> [0] 8/5/2011 00:24:04: LoopMon: LCS 359 took 872 (306/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:24:34: Transport Error: Pending packet 1000002: generating
fake
> [2] 8/5/2011 00:24:34: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:24:48: TaskMon: LCS 508/0 recv LPCP took 443 msecs
> [0] 8/5/2011 00:24:48: LoopMon: LCS 508 took 444 (16/0) msecs, read 1, 3/1
tasks
> [0] 8/5/2011 00:24:48: TaskMon: LCS 509/0 recv LPCP took 506 msecs
> [0] 8/5/2011 00:24:48: LoopMon: LCS 509 took 507 (72/0) msecs, read 1, 4/1
tasks
> [0] 8/5/2011 00:24:49: TaskMon: LCS 510/0 recv LPCP took 1293 msecs
> [0] 8/5/2011 00:24:49: LoopMon: LCS 510 took 1337 (500/0) msecs, read
> 1, 5/1 tasks
> [0] 8/5/2011 00:25:35: TaskMon: LCS 673/0 recv LPCP took 871 msecs
> [0] 8/5/2011 00:25:35: LoopMon: LCS 673 took 871 (118/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:26:05: Transport Error: Pending packet 1000004: generating
fake
> [2] 8/5/2011 00:26:05: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:27:06: TaskMon: LCS 986/0 recv LPCP took 871 msecs
> [0] 8/5/2011 00:27:06: LoopMon: LCS 986 took 871 (419/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:27:36: Transport Error: Pending packet 1000006: generating
fake
> [2] 8/5/2011 00:27:36: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:28:37: TaskMon: LCS 1296/0 recv LPCP took 869 msecs
> [0] 8/5/2011 00:28:37: LoopMon: LCS 1296 took 870 (387/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:29:07: Transport Error: Pending packet 1000008: generating
fake
> [2] 8/5/2011 00:29:07: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:30:08: TaskMon: LCS 1605/0 recv LPCP took 870 msecs
> [0] 8/5/2011 00:30:08: LoopMon: LCS 1605 took 871 (458/0) msecs, read
> 1, 3/1 tasks
> [2] 8/5/2011 00:30:38: Transport Error: Pending packet 1000010: generating
fake
> [2] 8/5/2011 00:30:38: Registrar 361 at pbx.domain.com timed out
> [0] 8/5/2011 00:31:39: TaskMon: LCS 1918/0 recv LPCP took 874 msecs
> [0] 8/5/2011 00:31:39: LoopMon: LCS 1918 took 875 (346/0) msecs, read
> 1, 3/1 tasks
> [0] 8/5/2011 00:32:03: TaskMon: LCS 1996/0 recv LPCP took 424 msecs
> [0] 8/5/2011 00:32:03: LoopMon: LCS 1996 took 430 (24/4) msecs, read
> 1, 3/1 tasks

Marcello Ceschia

2011-May-22 18:48 UTC

head link

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

see: https://issues.asterisk.org/view.php?id=19182

On May 22, 2011, at 7:00 PM, asterisk-users-request at lists.digium.com wrote:
> Date: Sun, 22 May 2011 19:44:21 +0800
> From: GNUbie <gnubie at gmail.com>
> Subject: Re: [asterisk-users] Unable to REGISTER to the Asterisk
> 	v1.8.3.3 server via SIP/TLS
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 	<asterisk-users at lists.digium.com>
> Message-ID: <BANLkTin9TROuRB4_-g7JHfGYa59YemsW=g at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
> 
> Hello,
> 
> On Thu, May 19, 2011 at 3:40 PM, A E [Gmail] <all.eforums at
gmail.com> wrote:
>> 
>> That's WAYY too much info for me to go through right now, and I
don't know
>> anything about TLS registration but what I would ask for is if you have
the
>> following lines in your sip.conf
>> domain=<IP/FQDN of your asterisk server>:<TLS port>
>> so in your case add the lines
>> domain=pbx.domain.com:5061
>> and then do a "sip reload"
>> So far, all problems I've had, have been solved because of this. At
the end
>> of your sip.conf add those lines and it should fix your problem.
> 
> Thanks, but it didn't worked. :-(
> 
> Any other possible solution for this problem?
> 
> Regards,
> 
> GNUbie

Seemingly Similar Threads

Search for more reasonably related threads

asterisk users - May 2011 - Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

[asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

Seemingly Similar Threads