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LanScape VOIP Media Engine™ - Pre-Sales Technical Support
 LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
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micro
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Joined: May 19 2006
Posts: 20
Posted: May 23 2006 at 6:14am | IP Logged Quote micro

We are getting following error on sip registration..

("A fatal error was detected while attempting to contact the\n"
                                                 "configured SIP registration server. Registration functions will\n"
                                                 "not be available.",
                                                 "Sip Registration Server Error",)
>> Can't you give GUI for us to check, since in this case we are getting error at every point ?

thanks
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support
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Posted: May 23 2006 at 9:48am | IP Logged Quote support

Hi micro,

The software examples are meant to be used as your "Test GUI". Getting the examples to register with a SIP Proxy/registrar is easy. We suggest you do the following:

1) Look up the error return code you get in the developer's reference. It will explain the error. If you need help understanding the error code, post the error return code to this forum thread and we will elaborate.

2) Start using the SIP log capability that is supported by the VOIP Media Engine. Looking at a SIP log 99.9% of the time will show you the problem.

3) Make sure your SIP registrar is on-line and configured to use the proper IP address and UDP port.

Support
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micro
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Posted: May 23 2006 at 10:32am | IP Logged Quote micro

can you provide a GUI based settings for us to test the trial version of voip media engine, so that we can test faster and buy the paid version ?

awaiting positive reply

thanks
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support
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Posted: May 23 2006 at 11:17am | IP Logged Quote support

micro,

Sorry for the confusion. We are not sure what you mean when you say:

"can you provide a GUI based settings for us to test the trial version of voip media engine".

Please try to elaborate for us specifically what you want and we will respond immediately.

Support
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micro
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Posted: May 23 2006 at 12:24pm | IP Logged Quote micro

hi,

I mean, is there any way that the configurations can be done in a GUI ,rather than in the coding ?

thanks
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support
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Posted: May 23 2006 at 2:36pm | IP Logged Quote support

micro,

Now I understand. You want a "different" ready made "canned" test application in binary form.

Unfortunately, there are no other test applications that come with the product other than the software examples. I have logged a change ticket to include SIP proxy and registrar GUI support in a future releases of the example applications. This will make things easier in the future. I know that we have had requests for this in the past. It will probable be included in an upcoming release.

Right now however, the software examples are what you will have to use to perform your evaluation.

Are you having a specific problem that is causing you to not be able to register and initiate and receive calls???

Here is additional info to make things clearer:

To make performing an evaluation easier, install the Media Engine trial on 2 development machines.

To enable SIP proxy and registrar support in the samples on machine #1, do the following:

1) Go to the PhoneBase.cpp module. Locate the ENABLE_SIP_REGISTRAR_AND_PROXY_SUPPORT macro. Change its value from 0 to 1.

2) Locate these other macros in the PhoneBase.cpp module:

SIP_PROXY_SERVER_ADDRESS - Set this to your SIP proxy server host name, FQDN or IP address.

SIP_PROXY_SERVER_PORT - set to your SIP proxy server port. Most likely 5060.

SIP_DOMAIN_NAME - Set this to your SIP domain name. The SIP domain name is configured at your SIP proxy (or Asterisk PBX).

REGISTRAR_SERVER_ADDRESS - Set this to your SIP registrar server host name, FQDN or IP address. Typically the same as the SIP proxy specified above.

REGISTRAR_SERVER_PORT - set to your SIP registrar server port. Most likely 5060. Generally the same as the SIP proxy port specified above.

REGISTRATION_INTERVAL_SECONDS - Set this to whatever you want. Try a value of 20 to 60 seconds.

3) Locate the DEFAULT_PHONE_NAME macro and change it to something like “111”. This is the “phone number” of the application.

4) Go to machine #2 and repeat all the simple steps above. Change the DEFAULT_PHONE_NAME macro on machine #2 to “222”.

5) Rebuild all examples on machine #1 and #2.

6) On Machine #1, start the single line phone. It should register OK with your server.

7) On Machine #2, start another single line phone. It should register OK with your server.

8) Make a call from single line phone “111” on machine #1 to the single line phone on machine #2. The “address book” for the sample applications is simple enough. You can manually add entries so that you can call between “111” and “222”. The address book file for the examples is called AddressBook.txt.

9) You should now be in a VOIP call.

Support
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micro
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Joined: May 19 2006
Posts: 20
Posted: May 24 2006 at 3:32am | IP Logged Quote micro

This is the error , we are getting..while trying to register..

("One or more of the servers you are attempting to register with requires\n"        & nbsp;         & nbsp;      "authentication credentials. Please configure your authentication settings\n"        & nbsp;         & nbsp; "and try reregistering with the server again.",)

>> I think there is no configuration for password of the sip registration..so its not getting registered. Kindly reply on this error immidiately.
>>suppose our sip settings are as follows.
1. sip.sixtel.net
>>username-micromann
>>password-password ( not the right password)

can you send all the configurations with these sip settings inserted ?

thanks in advance.
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micro
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Joined: May 19 2006
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Posted: May 24 2006 at 4:16am | IP Logged Quote micro

here is the log file for the error

************* Log Opened (May 24 14:39:20) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.1.50:5060) >>>>
REGISTER sip:server SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21:5060
From: <sip:102@server>;tag=f8dc29
To: <sip:102@server>
Call-Id: b9b95b85-4a4b-4018-9a3d-8c55511b6fed-0000019c@192.168.1.21
CSeq: 16297181 REGISTER
Expires: 20
Max-Forwards: 70
Contact: <sip:102@192.168.1.21:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.50:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.21:5060
To: <sip:102@server>
From: <sip:102@server>;tag=f8dc29
Call-ID: b9b95b85-4a4b-4018-9a3d-8c55511b6fed-0000019c@192.168.1.21
CSeq: 16297181 REGISTER
User-Agent: Axon 1.04
WWW-Authenticate: Digest realm="axon@server",nonce="v69359qaq61593w",opaque="",stale= FALSE,algorithm=MD5
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (31 Ms, To: 192.168.1.50:5060) >>>>
REGISTER sip:server SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21:5060
From: <sip:102@server>;tag=f90679
To: <sip:102@server>
Call-Id: b9b95b85-4a4b-4018-9a3d-8c55511b6fed-0000019c@192.168.1.21
CSeq: 16297182 REGISTER
Expires: 20
Max-Forwards: 70
Contact: <sip:102@192.168.1.21:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (31 Ms, From: 192.168.1.50:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.21:5060
To: <sip:102@server>
From: <sip:102@server>;tag=f90679
Call-ID: b9b95b85-4a4b-4018-9a3d-8c55511b6fed-0000019c@192.168.1.21
CSeq: 16297182 REGISTER
User-Agent: Axon 1.04
WWW-Authenticate: Digest realm="axon@server",nonce="v69359qaq61593w",opaque="",stale= FALSE,algorithm=MD5
Content-Length: 0




************* Log Closed (May 24 14:39:30) *************
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support
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Posted: May 24 2006 at 7:38am | IP Logged Quote support

Micro,

We are glad to see that you have posted a SIP log. We now know that your sample applications are trying to talk to the AXON server.

To add authentication credentials to your example software is easy. You must add a call to the AddAuthorizationCredentials() API procedure in your PhoneBase.cpp module. It is just a few lines of additional code.

1)
We assume the following:
Your authentication Domain name (Realm): sip.sixtel.net
User name: micromann
Password: password

2)
In the PhoneBase.cpp modulem locate the line of code that reads:

// enable the VOIP Media Engine's capability to receive network
// data.
SipTelephonyEnable(pCPhoneBase->hSipEngine);

Its around line #1083.


3)
Add the following line of code just before the call to the SipTelephonyEnable() API procedure:

status = AddAuthorizationCredentials(
           pCPhoneBase->hSipEngine,
           “micromann”,
           “password”,
           “sip.sixtel.net”
           );

4)
Your code should now look something like:

// call into the parent class just in case they want to do some extra
// work before we enable the engine.
pCPhoneBase->InitializationComplete(pCPhoneBase->hSipE ngine);


// give the Media Engine challenge authentication credentials so if
// any far end device or server challenges us, the Media Engine will
// handle it automatically.
//
// Note:     You can call this API proc as many times as you want in
//         &nbs p;order to add many (username,password,realm) triplets.
//
status = AddAuthorizationCredentials(
           pCPhoneBase->hSipEngine,
           "your_user_name",
           "your_password",
           "your_authentication_realm"
           );

// enable the VOIP Media Engine's capability to receive network
// data.
SipTelephonyEnable(pCPhoneBase->hSipEngine);



5)
Rebuld the examples and try it again. You should register ok.


6)
You should become as familiar with the software developer’s reference as possible. All of this information is contained in that document. Please give it a quick review. There is a lot of helpful information in “The Telephony API” section that will benefit you.


We will be updating the example apps to include configuring proxy, registrar and authentication information via the GUI. The updates will be included in a later product release.

Support
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micro
Intermediate
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Joined: May 19 2006
Posts: 20
Posted: May 24 2006 at 8:55am | IP Logged Quote micro

we are trying this out, but still not getting registered

can you send all the configurations with the file & with these sip settings inserted ?

>> We suggest that there has to be user friendliness in the settings that have to be done..Put up a system like a dll/activex component with thorough help file for each function..this will speed up matters..

goodluck
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micro
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Posts: 20
Posted: May 24 2006 at 8:56am | IP Logged Quote micro

>>suppose our sip settings are as follows.
1. sip.sixtel.net
>>username-micromann
>>password-password ( not the right password)
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support
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Posts: 1666
Posted: May 24 2006 at 10:49am | IP Logged Quote support

micro,

Ok, We see you are using AXON free PBX server. We decided to download the latest version and perform a test. Everything works as expected. You are having a configuration problem. Here is what we have done.

1)
Installed AXON 1.06 on an old Celeron development PC here in our lab and performed basic configuration. This is PC #1. It has a host name of Celeron400.

2)
AXON has some of pre configured extensions. We went into AXON and configured extension 102 and 103 to use the password "guest".

3)
On a second PC, we configured a VOIP utility soft phone we have here that uses the VOIP Media Engine. We configured it as extension 102 and specified the authentication credentials of:
user name: 102
password: guest
realm: axon@Celeron400

4)
On a third PC, we configured a VOIP utility soft phone as in the previous step but used extension 103. We specified the authentication credentials of:
user name: 103
password: guest
realm: axon@Celeron400

5)
Both utility soft phones registered OK.

6)
Using soft phone 102, we placed a call to extension 103. the call connected with no problems. We then hung up. Here is a SIP log file for the transaction:


************* Log Opened (May 24 10:30:44) *************

>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.1.3:5060) >>>>
REGISTER sip:Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK0643eb17
From: <sip:102@Celeron400:5064>;tag=643b39b
To: <sip:102@Celeron400:5064>
Call-Id: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455558 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:102@192.168.1.2:5064>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK0643eb17
To: <sip:102@Celeron400:5064>
From: <sip:102@Celeron400:5064>;tag=643b39b
Call-ID: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455558 REGISTER
User-Agent: Axon 1.06
WWW-Authenticate: Digest realm="axon@celeron400",nonce="v5285qaq84603w",opaque="",sta le=FALSE,algorithm=MD5
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (15 Ms, To: 192.168.1.3:5060) >>>>
REGISTER sip:Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK064418d3
From: <sip:102@Celeron400:5064>;tag=643f6fa
To: <sip:102@Celeron400:5064>
Call-Id: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455559 REGISTER
Authorization: Digest algorithm=md5, nonce="v5285qaq84603w", realm="axon@celeron400", response="b8a39b6777d38478a65d6083adc13a96", uri="sip:Celeron400", username="102"
Expires: 3600
Max-Forwards: 70
Contact: <sip:102@192.168.1.2:5064>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK064418d3
To: <sip:102@Celeron400:5064>
From: <sip:102@Celeron400:5064>;tag=643f6fa
Call-ID: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455559 REGISTER
User-Agent: Axon 1.06
Contact: <sip:102@192.168.1.2:5064>;expires=3600;q=0.5
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (9469 Ms, To: 192.168.1.3:5060) >>>>
INVITE sip:103@Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442660
From: "Ext 102" <sip:102@Celeron400>;tag=644056a;x-UaId=xxxxx-yyyy-zzz zzz
To: <sip:103@Celeron400>
Contact: <sip:102@192.168.1.2:5064>;x-inst="VGVzdCBDYWxsIERhdGE gZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: 257ed806-d106-4336-9c7e-6a1ef56fe4c8-00000c34@192.168.1.2
CSeq: 4462304 INVITE
Max-Forwards: 70
Organization: 1D05B88F-516D-483E-B5A1-E22654099440
Content-Length: 222
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=102 105103234 105103234 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8902 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442660
To: <sip:103@Celeron400>
From: "Ext 102" <sip:102@Celeron400>;tag=644056a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: 257ed806-d106-4336-9c7e-6a1ef56fe4c8-00000c34@192.168.1.2
CSeq: 4462304 INVITE
User-Agent: Axon 1.06
Proxy-Authenticate: Digest realm="axon@celeron400",nonce="v5285qaq84603w",opaque="",sta le=FALSE,algorithm=MD5
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (16 Ms, To: 192.168.1.3:5060) >>>>
ACK sip:103@Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442660
From: "Ext 102" <sip:102@Celeron400>;tag=644056a;x-UaId=xxxxx-yyyy-zzz zzz
To: <sip:103@Celeron400>
Call-Id: 257ed806-d106-4336-9c7e-6a1ef56fe4c8-00000c34@192.168.1.2
CSeq: 4462304 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.1.3:5060) >>>>
INVITE sip:103@Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
To: <sip:103@Celeron400>
Contact: <sip:102@192.168.1.2:5064>;x-inst="VGVzdCBDYWxsIERhdGE gZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
Max-Forwards: 70
Organization: 1D05B88F-516D-483E-B5A1-E22654099440
Proxy-Authorization: Digest algorithm=md5, nonce="v5285qaq84603w", realm="axon@celeron400", response="b96d0495d90cad153b516b87fbf2c395", uri="sip:103@Celeron400", username="102"
Content-Length: 222
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=102 105103250 105103250 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8902 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
User-Agent: Axon 1.06
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
User-Agent: Axon 1.06
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
User-Agent: Axon 1.06
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
User-Agent: Axon 1.06
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
User-Agent: Axon 1.06
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 INVITE
User-Agent: Axon 1.06
Contact: <sip:103@192.168.1.3:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Accept: application/sdp
Supported: replaces
Content-Type: application/sdp
Content-Length: 171

v=0
o=LanScape 3357473870 3357473870 IN IP4 192.168.1.3
s=LanScape
c=IN IP4 192.168.1.3
t=0 0
m=audio 8030 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (19968 Ms, To: 192.168.1.3:5060) >>>>
ACK sip:103@Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442670
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
To: <sip:103@Celeron400>;tag=5805
Call-Id: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462320 ACK
Max-Forwards: 70
Route: <sip:103@192.168.1.3>
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (4782 Ms, To: 192.168.1.3:5060) >>>>
BYE sip:103@Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442d71
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
To: <sip:103@Celeron400>;tag=5805
Call-Id: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462321 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06442d71
To: <sip:103@Celeron400>;tag=5805
From: "Ext 102" <sip:102@Celeron400>;tag=644057a;x-UaId=xxxxx-yyyy-zzz zzz
Call-ID: cce6a904-6dd9-47d6-ad5f-1fcbcadda33d-00000c34@192.168.1.2
CSeq: 4462321 BYE
User-Agent: Axon 1.06
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (1515 Ms, To: 192.168.1.3:5060) >>>>
REGISTER sip:Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06446e4e
From: <sip:102@Celeron400:5064>;tag=6442654
To: <sip:102@Celeron400:5064>
Call-Id: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455560 REGISTER
Expires: 0
Max-Forwards: 70
Contact: <sip:102@192.168.1.2:5064>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06446e4e
To: <sip:102@Celeron400:5064>
From: <sip:102@Celeron400:5064>;tag=6442654
Call-ID: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455560 REGISTER
User-Agent: Axon 1.06
WWW-Authenticate: Digest realm="axon@celeron400",nonce="v5285qaq84603w",opaque="",sta le=FALSE,algorithm=MD5
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.1.3:5060) >>>>
REGISTER sip:Celeron400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06448daf
From: <sip:102@Celeron400:5064>;tag=6443ee9
To: <sip:102@Celeron400:5064>
Call-Id: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455561 REGISTER
Authorization: Digest algorithm=md5, nonce="v5285qaq84603w", realm="axon@celeron400", response="b8a39b6777d38478a65d6083adc13a96", uri="sip:Celeron400", username="102"
Expires: 0
Max-Forwards: 70
Contact: <sip:102@192.168.1.2:5064>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.3:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5064;rport;branch=z9hG4bK06448daf
To: <sip:102@Celeron400:5064>
From: <sip:102@Celeron400:5064>;tag=6443ee9
Call-ID: 5af06c59-a4ff-4264-abc9-5bace4062068-00000c34@192.168.1.2
CSeq: 4455561 REGISTER
User-Agent: Axon 1.06
Content-Length: 0

************* Log Closed (May 24 10:31:28) *************


We now see that due to your AXON setup, you must use the authentication domain (also called the Realm) of: “axon@server”

So, the code change we wanted you to make for a previous post would look like:

status = AddAuthorizationCredentials(
            pCPhoneBase->hSipEngine,
            “micromann”,
            “password”,
            “axon@server”
            );

This assumed that you have configure an extension in AXON called “micromann” having the password “password”.


It all works as expected.

Support
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micro
Intermediate
Intermediate


Joined: May 19 2006
Posts: 20
Posted: May 24 2006 at 11:37am | IP Logged Quote micro

hi,

finally the registration is done..thanks for the support.
>> there is error in the transfer function...the call gets disconnected..
>> you are requested to guide us along with settings for the following functions immidiately .we are waiting for your answer on this.

1. call transfer
2. conference
3. voice activity detection
4. multiple line simultaneous calling

We can finalize immidiately once we get help in checking these functions.we are waiting for your answer on this.

goodluck & thanks again
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support
Administrator
Administrator


Joined: January 26 2005
Location: United States
Posts: 1666
Posted: May 24 2006 at 12:10pm | IP Logged Quote support


So we can assist you further, you must post specific questions to this forum. Your last questions are not specific enough.

If you have additional specific question, post them to a new thread.

Thanks for your help.
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