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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 14 2007 at 4:54am | IP Logged
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Hello!
I still can not log in to one of our mediagateway.
When I observe log files i notice that your application did not recognize request for user id and password form my gateway. Do you have any idea whats wrong?
Maybe your application use some different SIP signaling?
Below I attached logs.
My ip is: 192.168.44.28
Gateway ip is: 192.168.2.34
LOGIN: 12345678900
Code:
************* Log Opened (Mar 14 10:43:34) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.2.34:5060) >>>>
REGISTER sip:192.168.2.34 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.28:5060;rport;branch=z9hG4bK008d0f05
From: <sip:12345678900@192.168.2.34>;tag=8d3ad8
To: <sip:12345678900@192.168.2.34>
Call-Id: e1a9e8e0-5071-49be-92e4-335e8b3859d9-00000ccc@192.168.44.28
CSeq: 9249446 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:12345678900@192.168.44.28:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.3.14 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.2.34:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.44.28:5060;rport;branch=z9hG4bK008d0f05
From: <sip:12345678900@192.168.2.34>;tag=8d3ad8
To: <sip:12345678900@192.168.2.34>
Call-ID: e1a9e8e0-5071-49be-92e4-335e8b3859d9-00000ccc@192.168.44.28
CSeq: 9249446 REGISTER
Expires: 3600
Max-Forwards: 69
Contact: <sip:12345678900@192.168.44.28:5060>;user=phone
User-agent: LanScape VOIP Media Engine/5.12.3.14 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
Proxy-Authenticate: Digest realm="SysMaster", nonce="5fa99def38833ad0d80760e64c72b3b1", opaque="6f07ebcda0ce372a71ca99bde8843090", uri="sip:192.168.2.34"
.
.
.
************* Log Closed (Mar 14 10:44:21) *************
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 14 2007 at 6:59am | IP Logged
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Hi tomach,
Looks like your VOIP application that uses the media engine has not been set up with authentication credentials.
See the AddAuthorizationCredentials() API procedure in the developer’s reference for more information.
For example, somewhere after the media engine gets initialized, call the proc as follows:
Code:
status = AddAuthorizationCredentials(
hStateMachine,
"12345678900",
"your_password",
"SysMaster"
);
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If you are using the multiline phone example app, you can specify authentication credentials in the settings dialog. Try and set the following values:
Realms: SysMaster
Login Names: 12345678900
Passwords: set this to whatever your password is for user 12345678900.
Repost with your results,
Support
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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 14 2007 at 9:07am | IP Logged
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Hello,
It did not help. I even tried with different sip proxy (with which all the previous softphones worked fine) and after your application send registration sip message my proxy asks for registration (login, password), then Your application sends again registaration sip message without password....and all over again...
Different proxy sends sip message: 401 Unauthorized as a respond to your appliation request registration.
I have no idea what is wrong with your applicaoitn, is it possible that you have different sip siggnaling implemented?
I tried all the possible combination with authentication settings in your application (multiline phone) and it has never worked.
I guess better solution would be to wait for your .NET support that I can implement more my own code, and understad better how your application works.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 14 2007 at 10:06am | IP Logged
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tomach,
Authentication is a basic operation that should work. We have not seen any registration + authentication inter-op related issues for quite some time. Maybe you have found a bug or incompatibility. We really need to take a look at this further to figure out what is happening.
If you can, we need you to generate 2 log files for us:
1)
Use the software example dual line phone. Try to have the app register. Save the log file data.
2)
Use one of your other SIP phones. Have it register. Save the SIP log file data.
Please email the 2 logs above to support ‘at’ lanscapecorp.com
We will look closer for the problem.
We would really like to figure this out. If it is possible for us to access your “server” for registration tests, that would be very helpful too.
We will wait for your email.
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 19 2007 at 8:13am | IP Logged
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tomach,
We looked at your ethereal logs you sent to us on March 14th, 2007. You have a configuration problem. Use the dual line soft phone example application for your testing.
Assuming you are still using the SipX server, do this:
1)
Open the dual line soft phone example app configuration dialog.
2)
Enable authentication. To do this you must “check” the “Use Authentication” checkbox.
3)
Set your Realm to test.com
4)
Set the login name to 201.
5)
Set the password for user 201 to whatever is configured at the SipX server.
6)
Restart the dual line soft phone. You should register with no errors.
We tested this here so we know that we authentication Ok with SipX servers.
Support
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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 20 2007 at 7:37am | IP Logged
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Hello!
1. OK
2. OK
3. OK
4. OK
5. OK
6. OK
And again I can not log in.
What I noticed. If I only checked Use Authentication it even didnt send SIP message to register. If I fill additionaly "Use SIP domain" and "Use Registration Server" then it send but didnt resend when SIPX asks for authentication, (even I have everything set).
It is really weird cos with all other softphones I didnt have probelms with registering.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 20 2007 at 9:03am | IP Logged
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tomach,
Ok. We are missing something here. Please generate and email to us a SIP log from the media engine. We would like the SIP log that the media engine writes to a file.
This has got to be something simple that we are not seeing.
To enable SIP file logging, go to the PhoneBase.cpp file and look for the 2 lines:
Code:
StartupParams.LogSipMessages = FALSE;
StartupParams.pSipLogFileName = "SipMessageLog.log";
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... change them to:
Code:
StartupParams.LogSipMessages = TRUE;
StartupParams.pSipLogFileName = "SipMessageLog.log";
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... rebuild the multi line phone. Run your test just like in you latest post and email us the SIP log file called "SipMessageLog.log".
This complete SIP log should tell us exactly what is happening. This is very weird....
Support
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tomach
Intermediate
Joined: February 23 2007
Location: Poland
Posts: 22
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| Posted: March 20 2007 at 9:36am | IP Logged
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log file:
************* Log Opened (Mar 20 15:39:37) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.2.192:5060) >>>>
REGISTER sip:sipx.test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bK001a7bf3
From: <sip:tomek@sipx.test.com>;tag=1a8400
To: <sip:tomek@sipx.test.com>
Call-Id: 804a8526-f66b-4c17-8fae-35445d38cec7-00000c6c@192.168.2.133
CSeq: 1736827 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:tomek@192.168.2.133:5060>
User-Agent: LanScape VOIP Media Engine/5.12.3.14 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.2.192:32815) <<<<
SIP/2.0 401 Unauthorized
From: <sip:tomek@sipx.test.com>;tag=1a8400
To: <sip:tomek@sipx.test.com>
Call-Id: 804a8526-f66b-4c17-8fae-35445d38cec7-00000c6c@192.168.2.133
Cseq: 1736827 REGISTER
Via: SIP/2.0/UDP 192.168.2.133:5060;rport=5060;branch=z9hG4bK001a7bf3
Www-Authenticate: Digest realm="test.com", nonce="a53f76f2e1cfec5aa05e0d0e4c1f1e101171138435"
Date: Sat, 10 Feb 2007 20:13:55 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
User-Agent: sipX/3.2.0 (Linux)
Accept-Language: en
Supported: gruu
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (2328 Ms, To: 192.168.2.192:5060) >>>>
REGISTER sip:sipx.test.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.133:5060;rport;branch=z9hG4bK001aac3d
From: <sip:tomek@sipx.test.com>;tag=1a6443
To: <sip:tomek@sipx.test.com>
Call-Id: 804a8526-f66b-4c17-8fae-35445d38cec7-00000c6c@192.168.2.133
CSeq: 1736828 REGISTER
Expires: 0
Max-Forwards: 70
Contact: <sip:tomek@192.168.2.133:5060>
User-Agent: LanScape VOIP Media Engine/5.12.3.14 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (2234 Ms, From: 192.168.2.192:32815) <<<<
SIP/2.0 401 Unauthorized
From: <sip:tomek@sipx.test.com>;tag=1a6443
To: <sip:tomek@sipx.test.com>
Call-Id: 804a8526-f66b-4c17-8fae-35445d38cec7-00000c6c@192.168.2.133
Cseq: 1736828 REGISTER
Via: SIP/2.0/UDP 192.168.2.133:5060;rport=5060;branch=z9hG4bK001aac3d
Www-Authenticate: Digest realm="test.com", nonce="3e1111e89425ba2812f6a81eac876e961171138437"
Date: Sat, 10 Feb 2007 20:13:57 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
User-Agent: sipX/3.2.0 (Linux)
Accept-Language: en
Supported: gruu
Content-Length: 0
************* Log Closed (Mar 20 15:39:45) *************
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 20 2007 at 12:23pm | IP Logged
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tomach,
We are looking right now. Hope to have an answer today...
Support.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 22 2007 at 9:12am | IP Logged
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Hi tomach,
We had to reconfigure a new sipX server here and perform some inter-op testing. We are using sipX PBX v3.7.5-8790 on Fedora Core 5.
We have found some inter-op issues with this version of sipX.
If the authentication credentials on the multi lime soft phone example are properly configured, the authentication responses computed by the media engine that are sent to the sipX server are correct.
We notice that if the soft phone is not using SIP port 5060, then sipX will never authenticate the registration and continually responds with “401 Unauthorized”.
For example, this is the soft phone using SIP port 5090:
Code:
************* Log Opened (Mar 22 09:04:00) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#70, 2000 Ms, To: 192.168.1.121:5060) >>>>
REGISTER sip:lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5090;rport;branch=z9hG4bK00ae300f
From: <sip:200@lslab.com:5090>;tag=ae1d60
To: <sip:200@lslab.com:5090>
Call-Id: f69b0011-5490-4234-8550-2166f5bee102-000018ec@192.168.1.2
CSeq: 11374228 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:200@192.168.1.2:5090>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.3.17 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#70, 2016 Ms, From: 192.168.1.121:32781) <<<<
SIP/2.0 401 Unauthorized
From: <sip:200@lslab.com:5090>;tag=ae1d60
To: <sip:200@lslab.com:5090>
Call-Id: f69b0011-5490-4234-8550-2166f5bee102-000018ec@192.168.1.2
Cseq: 11374228 REGISTER
Via: SIP/2.0/UDP 192.168.1.2:5090;rport=5090;branch=z9hG4bK00ae300f
Www-Authenticate: Digest realm="lslab.com", nonce="45a90473048f448ec291483fcf93bbd61159772658"
User-Agent: sipX/3.7.0 sipX/registry (Linux)
Date: Mon, 02 Oct 2006 07:04:18 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
Accept-Language: en
Supported: gruu
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#71, 16 Ms, To: 192.168.1.121:5060) >>>>
REGISTER sip:lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5090;rport;branch=z9hG4bK00ade8a9
From: <sip:200@lslab.com:5090>;tag=ae11d1
To: <sip:200@lslab.com:5090>
Call-Id: f69b0011-5490-4234-8550-2166f5bee102-000018ec@192.168.1.2
CSeq: 11374229 REGISTER
Authorization: Digest algorithm=md5, nonce="45a90473048f448ec291483fcf93bbd61159772658", realm="lslab.com", response="e145ac23c4cb133d8d55fadde03cd927", uri="sip:lslab.com", username="200"
Expires: 3600
Max-Forwards: 70
Contact: <sip:200@192.168.1.2:5090>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.3.17 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#71, 0 Ms, From: 192.168.1.121:32781) <<<<
SIP/2.0 401 Unauthorized
From: <sip:200@lslab.com:5090>;tag=ae11d1
To: <sip:200@lslab.com:5090>
Call-Id: f69b0011-5490-4234-8550-2166f5bee102-000018ec@192.168.1.2
Cseq: 11374229 REGISTER
Via: SIP/2.0/UDP 192.168.1.2:5090;rport=5090;branch=z9hG4bK00ade8a9
Www-Authenticate: Digest realm="lslab.com", nonce="45a90473048f448ec291483fcf93bbd61159772658"
User-Agent: sipX/3.7.0 sipX/registry (Linux)
Date: Mon, 02 Oct 2006 07:04:18 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
Accept-Language: en
Supported: gruu
Content-Length: 0
************* Log Closed (Mar 22 09:04:04) *************
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And this is the soft phone using SIP port 5060:
Code:
************* Log Opened (Mar 22 09:07:47) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#3, 3172 Ms, To: 192.168.1.121:5060) >>>>
REGISTER sip:lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK00b19d6a
From: <sip:200@lslab.com>;tag=b15570
To: <sip:200@lslab.com>
Call-Id: fbf893f1-4d37-4975-a939-eebbfe4327e9-000018ec@192.168.1.2
CSeq: 11646083 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:200@192.168.1.2:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.3.17 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#3, 3125 Ms, From: 192.168.1.121:32782) <<<<
SIP/2.0 401 Unauthorized
From: <sip:200@lslab.com>;tag=b15570
To: <sip:200@lslab.com>
Call-Id: fbf893f1-4d37-4975-a939-eebbfe4327e9-000018ec@192.168.1.2
Cseq: 11646083 REGISTER
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK00b19d6a
Www-Authenticate: Digest realm="lslab.com", nonce="3f2673d31bf6ba5678aea8e2cae25b901159772885"
User-Agent: sipX/3.7.0 sipX/registry (Linux)
Date: Mon, 02 Oct 2006 07:08:05 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
Accept-Language: en
Supported: gruu
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTx (#4, 0 Ms, To: 192.168.1.121:5060) >>>>
REGISTER sip:lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK00b1bcda
From: <sip:200@lslab.com>;tag=b16e14
To: <sip:200@lslab.com>
Call-Id: fbf893f1-4d37-4975-a939-eebbfe4327e9-000018ec@192.168.1.2
CSeq: 11646084 REGISTER
Authorization: Digest algorithm=md5, nonce="3f2673d31bf6ba5678aea8e2cae25b901159772885", realm="lslab.com", response="c6aa0eadebb1527094a9367f4c92f8d5", uri="sip:lslab.com", username="200"
Expires: 3600
Max-Forwards: 70
Contact: <sip:200@192.168.1.2:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.3.17 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRx (#4, 16 Ms, From: 192.168.1.121:32782) <<<<
SIP/2.0 200 OK
From: <sip:200@lslab.com>;tag=b16e14
To: <sip:200@lslab.com>;tag=1189641421
Call-Id: fbf893f1-4d37-4975-a939-eebbfe4327e9-000018ec@192.168.1.2
Cseq: 11646084 REGISTER
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;branch=z9hG4bK00b1bcda
Contact: <sip:200@192.168.1.2:5060>;EXPIRES=962
User-Agent: sipX/3.7.0 sipX/registry (Linux)
Date: Mon, 02 Oct 2006 07:08:05 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
Accept-Language: en
Supported: gruu
Content-Length: 0
************* Log Closed (Mar 22 09:07:51) *************
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We will take a look into why this is occurring. So in the meantime, make sure your media engine apps all use port 5060 when communicating with sipX servers.
We will also retest proxy authentication and basic call capabilities and repost later.
Question:
From your original post, what manufacturer is your original media gateway??? Is it sipX?
If it is not sipX, you could help us by posting a lanscape SIP log for that too.
Thanks,
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
Topic: Need help with authentication
