Add a real HSS connection to BEA sip server 3.1

Hi, I have a real HSS configed alrdy and wanna make the sip server can connect to that HSS? How do i do it?

As the Product Manager for the Oracle Communications Converged Application server (the former BEA WebLogic SIP server) and the former product manager for the Sailfin (I was PM for the Sailfin product before I left Ericsson to work for BEA) I must say that the Oracle product is absolutely superior, both in terms of quality and design, to any other competing product. I can also say that Oracle is fully behind this technology. Remeber that in the long run the issue is not "cost", but "value". Open source software is generally free for a reason.

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Where is JainSipApi.jar in bea sip server?

hi,
to my understanding JSR-116 reuse the sip stack from JSR-32
but I do not find any JainSipApi.jar in bea sip server.
i am a bit confused. anyone can help?

Hi,
The WebLogic SIP Server does not provide a simple ava langunage binding to the SIP protocol, as you would get with the JAIN SIP API. Our SIP message parser/assembler and transaction layer is fully integrated with our Servlet container and is not visible as a .jar file.
The SIP Servlet API is part of the JAIN initiative within the JCP but it does not have any direct dependency on, or relationship with, the JAIN SIP specification.
BR,
-Mike

Bea Sip Server 3.1

Hi all,
I developed few sip services for Oracle OCMS. Now I'm trying to migrate the services to Bea sip server. I would like to know if Bea Sip server has proxy/registrar, presence functionalities like OCMS has. In OCMS there were out-of-the-box, but for what I've seen Bea doesnt have them. Are they going to be available?
Many tks
JF

As the Product Manager for the Oracle Communications Converged Application server (the former BEA WebLogic SIP server) and the former product manager for the Sailfin (I was PM for the Sailfin product before I left Ericsson to work for BEA) I must say that the Oracle product is absolutely superior, both in terms of quality and design, to any other competing product. I can also say that Oracle is fully behind this technology. Remeber that in the long run the issue is not "cost", but "value". Open source software is generally free for a reason.

How to connect j2me to sip server

Hi friends ...
I am new to this topic and i just want to know how to connect a real sip server with the j2me application .
I tried using the GoSIP demo given in wtk2.5 . its working fine with local SIP proxy and registrar given in that .
But i want to connect to a real SIP Server. Even i have a registered sip number.
Can any one please help me how to sort out this problem as i am very new to this topic. Is there any source given for this ??? If so please tell me ...
Thanks in advance ......

is it possible to make
voice calls from the GoSIP example given in the
wtk2.5 demo examples ?
Probably not. Have you tried?
I have a sip account and i just want to know how to
register my sip account with the sip server using a
j2me application .
Change the SipHeader and / or SipAddress
What are the changes to be made in the GoSIP example
to connect to a real sip server and not to the sip
proxy server
See above
and what are the syntax to be changed
in it ????
Syntax is a property of the programming language, not the application.
And one question mark is enough to pose a question, 4 is overkill and perceived as rudeness.
Please help me in solving out this problem ....
What have you done to help yourself?
Read the javadoc for jsr-180
http://www.forum.nokia.com/document/Java_ME_Developers_Library_v2/GUID-2508C2ED-C0BE-4512-9302-6805AB7ACB0E/index.html
Introduction to the SIP API for Java ME
http://dev2dev.bea.com/lpt/a/565
A presentation on the architecture and capabilites of SIP
http://phoenix.labri.fr/documentation/sip/Documentation/Papers/Programming_SIP/Presentation/Jain/SIP_for_J2ME.pdf
Thanks in advance ....
If you're looking to solve a need by cut-n-paste programming sans understanding, you're unlikely to get it here.
Increase your level of understanding, experiment with your codes and post back when you have a specific issue.
Good luck, Darryl

BEA Sip Server 3.1 Socketexception

Hi All,
I am getting a SocketException on the multicast address specified for the Data and Engine Cluster while starting the managed servers in Weblogic SIP Server 3.1.
Does anyone know how to fix this?
Thanks !

As the Product Manager for the Oracle Communications Converged Application server (the former BEA WebLogic SIP server) and the former product manager for the Sailfin (I was PM for the Sailfin product before I left Ericsson to work for BEA) I must say that the Oracle product is absolutely superior, both in terms of quality and design, to any other competing product. I can also say that Oracle is fully behind this technology. Remeber that in the long run the issue is not "cost", but "value". Open source software is generally free for a reason.

Bea sip server 3.1 xml error:(

hi.i am a new this subject.i develop a instant messaging with SIP using java.But i have given error xml.When i start the sip server and enter the console.all xml files cannot process.why?i dont know configure the sip server? do i miss someone? pls help me:(
Error message is admin console:The template you provided at /WEB-INF/templates/form.xml could not be processed
The template you provided at /WEB-INF/templates/.xml could not be processed
The template you provided at /WEB-INF/templates/summarypage.xml could not be processed. and all xmls can not process.
Error message is start server prompt.
Error: 'Instruction unknown:loadinstruction' (but i is not seen.there small 2 instead of i)
Fatal Error: 'Could not compile stylesheet'
Can I load only web logic sip server 3.1 or can i load with other web logic server for example Oracle Web Logic Server 10.3?????????do they work with together??????????
Edited by: user13107827 on May 24, 2010 5:50 AM

As the Product Manager for the Oracle Communications Converged Application server (the former BEA WebLogic SIP server) and the former product manager for the Sailfin (I was PM for the Sailfin product before I left Ericsson to work for BEA) I must say that the Oracle product is absolutely superior, both in terms of quality and design, to any other competing product. I can also say that Oracle is fully behind this technology. Remeber that in the long run the issue is not "cost", but "value". Open source software is generally free for a reason.

Regarding INVITE retransmission (BEA sip server)

Hi all,
I have a question about the WebLogic SIP server (version 3.1) behavior.
I am testing a simple call scenario from UAC to UAS on UDP transport (please see the call flow below).
It seems that the WebLogic SIP server generated a retransmission of an INVITE even if the 100 response had already been sent to the server.
According to RFC3261 page 124,
"After
receiving a 1xx response, any retransmissions cease altogether, and
the client waits for further responses",
the server (acting as a UAC in this case) should stop the t1 timer and do not generate any retransmissions of the INVITE after the 100 response had been received.
I wonder if there is something wrong with my configuration. Has anyone come across this issue before ?
Any ideas would be greatly appreciated.
PS: I have verified that the Call-ID are the same in the 100 response and the initial INVITE.
Best Regards,
kirati
BEA WebLogic Server---------------OpenSIPS proxy---------------SIPP UAS
|---------INVITE-------------------------> | |
| | |
|<---------100------------------------------| |
| |---------INVITE----------------->|
| |<---------180--------------------- |
| |<---------200----------------------|
|--------re-INVITE---------------------->| |
| |-----------re-INVITE------------>|
|<------180-------------------------------- | |
|<------200-------------------------------- | |
|<------200-------------------------------- | |
|--------re-INVITE----------------------->| |
|<------200-------------------------------- | |
|--------re-INVITE----------------------->| |
|<------200-------------------------------- | |
|---------ACK-----------------------------> | |
| |---------ACK---------------------->|

Sorry the call flow was messed up. I will post it again:
BEA----INVITE--------->Open SIPS proxy----------------------------SIPP UAS
BEA<----100--------------Open SIPS proxy----------------------------SIPP UAS
BEA--------------------------Open SIPS proxy--INVITE-------------->SIPP UAS
BEA--------------------------Open SIPS proxy<----180-----------------SIPP UAS
BEA--------------------------Open SIPS proxy<-----200----------------SIPP UAS
BEA-----Re-INVITE-------Open SIPS proxy-------------------------->SIPP UAS
BEA--------------------------Open SIPS proxy-----Re-INVITE------->SIPP UAS
BEA<----180---------------Open SIPS proxy----------------------------SIPP UAS
BEA<----200---------------Open SIPS proxy----------------------------SIPP UAS
BEA<----re-200-----------Open SIPS proxy----------------------------SIPP UAS
BEA-----Re-INVITE-------Open SIPS proxy-------------------------->SIPP UAS
BEA<----re-200-----------Open SIPS proxy----------------------------SIPP UAS
BEA-----Re-INVITE-------Open SIPS proxy-------------------------->SIPP UAS
BEA<----re-200-----------Open SIPS proxy----------------------------SIPP UAS
BEA----ACK-------------->Open SIPS proxy----------------------------SIPP UAS
BEA--------------------------Open SIPS proxy--ACK------------------>SIPP UAS

Dial-up connection between Asterik SIP server and Cisco 5400

Hi,
We have a setup for dial-up. The incoming call lands on Cisco 5400 and is forward to PGW using mgcp intergration. PGW sends it to Asterik SIP server over SIP. This Asterik Server sends it to another Cisco 5400 over SIP for Internet.
The issue is for some dial-up user, it works fine. But for some it doesn't. The problem is when we see the log on the last Cisco 5400, the incoming call tone is detected as FAX tone rather then Modem tone.
Any idea why is it so??

Hi,
We have a setup for dial-up. The incoming call lands on Cisco 5400 and is forward to PGW using mgcp intergration. PGW sends it to Asterik SIP server over SIP. This Asterik Server sends it to another Cisco 5400 over SIP for Internet.
The issue is for some dial-up user, it works fine. But for some it doesn't. The problem is when we see the log on the last Cisco 5400, the incoming call tone is detected as FAX tone rather then Modem tone.
Any idea why is it so??

Unable to connect to SIP server

Hi,
Need some assistance here.(using E71)
When i try to connect to a SIP Server(USING Sip Express Router), it failed to get registered. From the sniffing via the ethereal(WIFI), the Flow are follow :
1. UA request register request to sip server
2. Sip Server reply with : Status : 401 Unauthorised
In a actually environment, the UA should have the 3rd steps where it re-sent the reqgister request with the username and password.
But somehow, nokia E71 is not doing it.
any suggestion?
P/S : http://www.sipcenter.com/sip.nsf/html/SIP+Application+Server+Functionality
(it shows how the UA should connect to the Server)

The problem could be the router or its firewall, but it could also be that your Indian isp is blocking voip connections. There have been reports that this is not unknown in India. Sometimes this type of blocking is done by discarding packets destined to the standard voip sip signalling port 5060. You can easily change the sip signalling port in the SPA3102 for the packets coming back to the SPA and in fact routers often changes the port without your knowledge, but packets going to the voip provider must use the port number specified by the provider. Since you can access the internet with your pc, I would try to run some tests to see if this is the case.
I would run some ping tests and tests to see if I could register with other providers or services that use alternate or alternative sip signalling ports.

SIP Server FQDN missing

My phones will not connect to my Lync server. When I run the dhcputil - emulate my SIP Server FQDN is blank. I believe this entry comes from :
netsh dhcp server add optiondef 120 UCSipServer Binary 0 comment="Sip Server Fqdn"
But I have ran this with changing to my Lync server, but it still comes up blank.
Any ideas?
Thanks
Baze

Here's link for Configuring DHCP Options to Enable Sign-in for IP Phones that may be helpful as well
http://technet.microsoft.com/en-us/library/gg398088%28v=ocs.14%29.aspx
http://um.losrios.edu/network-config-to-support-the-lync-phones/
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question, please click "Mark As Answer"
Mai Ali | My blog: Technical | Twitter:
Mai Ali

CUCM starting connection with unknown media server.

Hi.
Lets say you have a SIP server. A CUCM is connected to this SIP server via a SIP trunk. A call is established from the external sip server to the CUCM. The calling phone press "on hold " function(the phone registered in the external SIP server). The external SIP server, sends an INVITE to the CUCM asking to initiate an RTP stream with an external media server(so the CUCM phone can receive music on hold). The CUCM has no knowing of this external media server.
It's possible to tell the CUCM to accept this session with the external media server?
I think the external SIP server, who knows about the media server existence, should proxy the connection between the media server and the CUCM(like an CUCM MTP).
SIP server---------------------------------------------------------------CUCM
                                  |
                                  |
                         Media server
Thanks.

The phone on CUCM that is being put on hold will be asked to connect to this "media server" and in theory as long as the phone has ip connectivity to the "media server", this should work fine.

Cannot add a real instrument track

When I try to add a real instrument track It adds a software instrument track. I can add a basic track however I cannot hear any monitoring of my mic at all. I have it connected to a very basic mixer then into the computers audio in port. I also tried to record with an audio interface. Garage band does recognize the interface but again i cant hear anything or see any levels. Any help would be greatly appreciated.

did you try the steps outlined in the link? those are the first things to try, especially with something that no one else is having a prob with.
as for the mic:
http://www.bulletsandbones.com/GB/GBFAQ.html#micline
and the interface (depending on its type, I'll list both possibilities):
http://www.bulletsandbones.com/GB/GBFAQ.html#recordlinein
http://www.bulletsandbones.com/GB/GBFAQ.html#selectinterface

Config Office Add-In - Connection to BI Office Server

Dears,
I've installed the office-add in..so far so good. But now when I want to login to the addin in excel, I've to set up a connection to the office server.
Can anyone help me with that...I'm always failing, even after following the installation guide.
Many thanks for your feedback.
Kr

Hej Remc0
thanks for the answer...but even after following all these steps, I still have test connection failed...
The office server name, is this the name you also have when you want to connect to your presentation service?
As it's all straight forwards, I don't have a clue to this...
Txs for your feedback.

How to add two different database connections on Model

Hi,
In my application, I need to create View Objects on different databases. So I need to add two different database connections on the Model.
But it seems like Model can be connected to only one database. I tried adding another project to add the second db connection. But after I created a read-only View Object in the second project, it didn't generate corresponding data control in the Data Controls panel.
Could anyone help me on this?
Thanks!

Do you mean adding the second project folder in ViewController->Project Properties->Libraries and Classpath?Yes, either that or: ViewController -> Project Properties -> Dependencies
And I didn't see the first Model project in the ViewController classpath. Otherwise it wouldn't work. The first model project IS in 'classpath'. Here's how:
ViewController -> Project Properties -> Dependencies
I tried this and recreated the View Object. Still didn't generate the datacontrol. No idea why you had to recreate the VO.
Is your VO in an application module inside the SECOND project?
After you are sure that everything is in the ViewController's classpath, just try restarting JDeveloper.
This is very basic. has to work.

TS1717 I have older version of itunes on Windows Vista, when i check for updates it states it cannot connect to itunes update server even though i am connected to the internet, i have an iphone that has songs in the cloud and i wan to add them to my libra

I have older version of itunes on Windows Vista, when i check for updates it states it cannot connect to itunes update server even though i am connected to the internet, i have an iphone that has songs in the cloud and i want to add them to my library?

Try updating your iTunes using an iTunesSetup.exe (or iTunes64Setup.exe) installer file downloaded from the Apple website:
http://www.apple.com/itunes/download/

Add a real HSS connection to BEA sip server 3.1

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