opensipstack-osbcdevel — Discussions on OpenSBC

Re: [OpenSBC] Digit Stripping

[ehernaez <ope...@op...>]

try this in the latest build:

[[sip:123#*@1.2.3.4|http://www.opensourcesip.org:8080/clearspacex/sip:123#*@1.2.3.4]] [sip:(123#)*@5.6.7.8:5060|http://www.opensourcesip.org:8080/clearspacex/sip:%28123#%29*@5.6.7.8:5060]

Re: [OpenSBC] Digit Stripping

[eric h. <er...@so...>]

try this in the latest build:

[sip:123#*@1.2.3.4] sip:(123#)*@5.6.7.8:5060





Chris Veazey wrote:
> Should this B2BUA route expression strip 123# ?
>
>
> [sip:123#*@1.2.3.4] sip:*@5.6.7.8:5060
>
>
>  
>
>
>
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[OpenSBC] Digit Stripping

[Chris V. <chr...@gm...>]

Should this B2BUA route expression strip 123# ?


[sip:123#*@1.2.3.4] sip:*@5.6.7.8:5060


 

Re: [OpenSBC] OpenSBC 1.1.6, Summary of Feature Requests and Bug Reports

[Joegen E. B. <joe...@gm...>]

Dangi, Rajpal wrote:
> What about the feature to host OpenSBC behind NAT/firewall?
>   

I have tested OpenSBC in amazon EC2 cloud.  OpenSBC is installed in a 
virtual machine with a private IP address and Amazon exposes it through 
a public interface with one to one port binding.    I then set the 
static media address parameter  of OpenSBC to point to the public 
interface.   I was able to test this setup to be working.   The 
assumption here is that all traffic going to OpenSBC is coming from the 
public side of the network.   In this scenario, no feature request is 
needed to fulfill it.  It's already there.

The other scenario, which I think what you have in mind, is OpenSBC 
acting as a bridge between a private network and a public network while 
having itself bound by the same NAT firewall of the private network.   
Given that your NAT/Firewall is also a one to one setup, calls coming 
from the public space goinf to the public space would still work the way 
it worked in the amazon EC2 setup.  However the problem will arise if 
the call is coming from the public side towards the private network.  
OpenSBC would then need to determine when to rewrite the SDP using the 
static media address and not.   Given that we are able to do this, 
OpenSBC should be able to bridge the NAT properly.   This is always a 
tricky setup because no two systems behave similarly in such cases.    
There are also a lot of system specific behavior that are needed to be 
addressed not to mention screwed up routing tables.   Thus, this is into 
feature that would be solved with a snap of a finger.   I am sorry this 
cannot make it to the final release of 1.1.5 or even promise that 1.1.6 
will have this feature.    

If you have this kind of setup... then go for an NAT devices with a 
built-in ALG.

Joegen




> Rajpal
>
> -----Original Message-----
> From: ope...@li...
> [mailto:ope...@li...] On Behalf
> Of Joegen E. Baclor
> Sent: Tuesday, December 11, 2007 5:59 PM
> To: ope...@li...; listaopenSBC
> Subject: [OpenSBC] OpenSBC 1.1.6, Summary of Feature Requests and Bug
> Reports
>
> Hi Everyone,
>
> I am intending to finalize 1.1.5 release before the year ends so we have
>
> a clean slate for 1.1.6.  If you have feature requests or bug fixes 
> which you think should be part of this release, please document them and
>
> post them to the list ASAP.  We are going to implement  CVS branches 
> beginning 1.1.5 to allow bug fixes specific to a certain branch.   There
>
> will be some major architectural changes that will be implemented in 
> version 1.1.6 which might not be anymore compatible with the 1.1.5 
> branch.   I will enumerate the pending feature requests I summarized 
> from previous post.  Let me know if I missed smething
>
> Pending Feature Requests:
>
> 1.  Support for redirect in relay Routes - sam...@ve...
> 2.  Creation of folders (logs, registry, calea etc ) in places where 
> user has permission - H.Kropf
> 3.  Rewriting of From URI - Warren Kreckler
> 4.  Routing based on  Packet  Source - chr...@gm...
> 5.  Routing based on recipient NIC - Warren Kreckler
>
> If I  missed something feel free to reply inline.
>
> Planned Features for 1.1.6
>
> 1.  I have already announced the plan for a separate configuration 
> module on top of the current HTTP Admin.  
> 2.  We are also going to implement a separate  media proxy application 
> to allow HA/Load Balancing of streams on top of the internal media proxy
>
> implementation.  This change would allow an SBC to use multiple 
> instances of external media proxies giving way to a higher volume 
> deployment.  
> 3.  We are also going to extract the Media Server Trunk and promote it 
> as a fully featured VXML Media Server allowing PBX developers to create 
> PBX applications using the OpenSIPStack Framework. 
> 4.  The plan to implement Diameter as the AAA implementation in OpenSBC.
> 5.  Finally, the long awaited resurrection of the TCP and TLS
> transports.
>
> I have also Opened a  new development effort in SourceForge  called  
> Plugin++ (  http://sourceforge.net/projects/pluginpp ).  This new 
> project will be the entry point of a planned Plugin Framework/API for 
> OpenSBC and other OpenSIPStack applications.
>
> We are aiming for June 2008 as the ETC for the 1.1.6 release of 
> OpenSBC.  We will come up with milestone releases for each
> functionality.
>
>
> ------------------------------------------------------------------------
> -
> SF.Net email is sponsored by:
> Check out the new SourceForge.net Marketplace.
> It's the best place to buy or sell services for
> just about anything Open Source.
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> _______________________________________________
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> Ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>
>
>   

Re: [OpenSBC] [OpenSIPStack] B2BUA how to route

[Joegen E. B. <joe...@gm...>]

sales@ER wrote:
> Hi Joegen
>
>   
>> I see what you mean.   I am not really familiar with the use of the
>> Remote-Party-Id.   We have implemented P-Asserted-Identity for this
>> instead.   Can you point me to the RFC that discusses the use cases for
>> Remote-Party-Id?
>>     
>
> Yes the P-Asserted-Identity replaced the Remote-Party-Id in the RFC but it
> is still in used in older SBC models and my ITSP has not updated the SBC i
> am accessing.  I need to modifry the xml code to and and elseif to test for
> this possiblity.   Please direct me to the xml that is managing this
> identity.
>   

For you be able to rewrite any header before it gets sent to the UAS you 
need to override SBCBackDoorCallHandler::OnOutgoingCall();

Look for the declaration of class SBCBackDoorCallHandler in 
SBCBackDoorTrunk.cxx.  Add a new member function

virtual void OnOutgoingCall(
    B2BUAConnection & connection,
    B2BUACall & call,
    SIPMessage & invite
  );


This function will be called whenever there is a new INVITE that will be 
sent out by the backdoor trunk.   Implement this function right after 
BOOL SBCBackDoorCallHandler::OnReceivedMergedInvite() methid in 
SBCBackDoorTrunk.cxx


You may add special headers to invite using this code

SIPHeader myHeader(  "Remote-Party-Id", "Whatever the value is" );
invite.AddCustomHeader( myHeader );

HTH

Joegen
> Warren Kreckler
>
>
> ----- Original Message ----- 
> From: "Joegen E. Baclor" <joe...@gm...>
> To: "sales@ER" <sa...@el...>
> Sent: Tuesday, December 11, 2007 8:32 PM
> Subject: Re: [OpenSIPStack] B2BUA how to route
>
>
>   
>> sales@ER wrote:
>>     
>>> Hi Joegen
>>>
>>> Thank you very much for your replies.
>>>
>>> 1.  I'm using the lastest version.
>>>
>>>       
>> Then your ITSP must be seeing just a single via.  If you think the
>> contrary, send me a packet capture from sipx->OpenSBC and OpenSBC->ITSP
>>
>>
>>
>>     
>>> 3.  sipX does not re-write  header as far as I know.  Are you asking for
>>> sipX header(s) dealing with Caller-ID?
>>>
>>>  Remote-Party-ID to determine the Calling ID.  This is not an element
>>> created
>>>  by Sipx.  The SBC will need to extract the user part of the From URI
>>>       
> and
>   
>>>  create a Remote-Party-ID. I did not see this capability with OpenSBC.
>>>  Without this, the called party on the PSTN will either see "Private
>>>  Caller"or "Anonymous" on their phone instead of the DID.
>>>
>>>
>>>       
>> I see what you mean.   I am not really familiar with the use of the
>> Remote-Party-Id.   We have implemented P-Asserted-Identity for this
>> instead.   Can you point me to the RFC that discusses the use cases for
>> Remote-Party-Id?
>>
>>
>>     
>>> Warren Kreckler
>>>
>>>
>>>
>>>
>>>
>>> ----- Original Message ----- 
>>> From: "Joegen E. Baclor" <joe...@gm...>
>>> To: "sales@ER" <sa...@el...>
>>> Cc: <ope...@li...>
>>> Sent: Sunday, December 09, 2007 7:21 PM
>>> Subject: Re: [OpenSIPStack] B2BUA how to route
>>>
>>>
>>>
>>>       
>>>> inline...
>>>>
>>>> sales@ER wrote:
>>>>
>>>>         
>>>>> Yes  They call it peer to peer.  By that they meam
>>>>>
>>>>>
>>>>> 1. Via Headers: ITSP has stated that they can accept only 1 Via
>>>>> statement in an INVITE.  As background, each device will add a Via
>>>>>
>>>>>           
>>> statement
>>>
>>>       
>>>>> to the INVITE to if it has processed the INVITE.  Only the last or top
>>>>>
>>>>>           
>>> entry
>>>
>>>       
>>>>> is really of interest to the party that next handles the INVITE.  In
>>>>>
>>>>>           
>>> order
>>>
>>>       
>>>>> for ITSP to accept the INVITE of an outbound call, OpenSBC will
>>>>> need to strip off all previous Via statements from the INVITE and add
>>>>>
>>>>>           
>>> its'
>>>
>>>       
>>>>> own.  I have not found any capability to remove the previously
>>>>>           
> inserted
>   
>>> Via
>>>
>>>       
>>>>> statements.
>>>>>
>>>>>
>>>>>           
>>>> What version are you using?   There was a bug introduced when we got
>>>> back from sipIT 21 due to the changes made there that had the vias not
>>>> getting stripped.  Please use the latest CVS.   OpenSBC should be
>>>> stripping the via before the B2BUA sends the INVITE out to the UAS.
>>>>
>>>>
>>>>
>>>>         
>>>>> 2.  Lock IP Address and port to first sender: This option comes into
>>>>>
>>>>>           
>>> play
>>>
>>>       
>>>>> when a call has been answered either by a person or system component
>>>>>
>>>>>           
>>> (i.e.
>>>
>>>       
>>>>> Auto Attendant) and a transfer is attempted.  When the transferred
>>>>>           
> call
>   
>>> is
>>>
>>>       
>>>>> answered by a new phone or component, it will negotiate use of a new
>>>>>           
> RTP
>   
>>>>> port for the media stream.  Some service providers, ITSP included,
>>>>> do not allow the RTP port to change once the initial call is
>>>>>
>>>>>           
>>> established.
>>>
>>>       
>>>>> They do this to protect against the "hijacking" of a call by Hackers.
>>>>>
>>>>>           
>>> Since
>>>
>>>       
>>>>> the media is flowing through a SBC, the SBC then needs to manage which
>>>>>
>>>>>           
>>> ports
>>>
>>>       
>>>>> are used to exchange media (voice).   If the original port is not
>>>>>
>>>>>           
>>> utilized
>>>
>>>       
>>>>> for the media back to the carrier, the PSTN will not hear any audio
>>>>>           
> once
>   
>>> the
>>>
>>>       
>>>>> call is transferred.  I do not see this capability with OpenSBC.
>>>>>
>>>>>
>>>>>
>>>>>           
>>>> In media proxy mode (Always Proxy Media = true), OpenSBC does not
>>>>         
> change
>   
>>>> the port of RTP even during reInvites.
>>>>
>>>>
>>>>
>>>>
>>>>         
>>>>> 3.  Calling ID:  SIPxchange utilizes the From: element to provide the
>>>>> Calling ID (DID).  It normally inserts the userID in the user part of
>>>>>
>>>>>           
>>> the
>>>
>>>       
>>>>> >From URI.  ITSP uses the INVITE element
>>>>> Remote-Party-ID to determine the Calling ID.  This is not an element
>>>>>
>>>>>           
>>> created
>>>
>>>       
>>>>> by Sipx.  The SBC will need to extract the user part of the From URI
>>>>>           
> and
>   
>>>>> create a Remote-Party-ID. I did not see this capability with OpenSBC.
>>>>> Without this, the called party on the PSTN will either see "Private
>>>>> Caller"or "Anonymous" on their phone instead of the DID.
>>>>>
>>>>>
>>>>>
>>>>>           
>>>> Can you send a sample of this from header that is rewritten by sipX?
>>>>
>>>>
>>>>
>>>>
>>>>         
>>>>> Warren Kreckler
>>>>>
>>>>> ----- Original Message ----- 
>>>>> From: "Joegen E. Baclor" <joe...@gm...>
>>>>> To: <ope...@li...>
>>>>> Cc: <jo...@op...>
>>>>> Sent: Friday, December 07, 2007 12:08 AM
>>>>> Subject: Re: [OpenSIPStack] B2BUA how to route
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>           
>>>>>> You need to use the SIP Trunking capability of OpenSBC for this.   Do
>>>>>> you need to authenticate calls with your ITSP?
>>>>>>
>>>>>>
>>>>>> sales@ER wrote:
>>>>>>
>>>>>>
>>>>>>             
>>>>>>> Hi
>>>>>>>
>>>>>>> Almost have this puppy working.
>>>>>>>
>>>>>>> Sipx and opensbc generally well understood.
>>>>>>>
>>>>>>> Problem:
>>>>>>>
>>>>>>> When OSBC receives INVITE from sipX =>  ITSP,
>>>>>>> OSBC route the INVITE back to sipX.
>>>>>>>
>>>>>>> We have two rules in the B2Bua route
>>>>>>> [sip:202.100.2.23:5060] sip: 202.100.2.23  this goes to our ITSP
>>>>>>> [sip:sipx.sip.net:5060] sip:sipx.sip.net            this point to
>>>>>>>               
> our
>   
>>>>>>>               
>>>>> sipx
>>>>>
>>>>>
>>>>>           
>>>>>>> the missing rule/route?
>>>>>>>
>>>>>>> Where do you put the rule and what should the rule say to route
>>>>>>>               
> INVITE
>   
>>>>>>>               
>>>>> out
>>>>>
>>>>>
>>>>>           
>>>>>>> to our ITSP?
>>>>>>>
>>>>>>> Warren Kreckler
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>> ------------------------------------------------------------------------
>>>>         
> -
>   
>>>>>>> SF.Net email is sponsored by: The Future of Linux Business White
>>>>>>>               
> Paper
>   
>>>>>>> from Novell.  From the desktop to the data center, Linux is going
>>>>>>> mainstream.  Let it simplify your IT future.
>>>>>>> http://altfarm.mediaplex.com/ad/ck/8857-50307-18918-4
>>>>>>> _______________________________________________
>>>>>>> opensipstack-devel mailing list
>>>>>>> ope...@li...
>>>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>> ------------------------------------------------------------------------
>>>>         
> -
>   
>>>>>> SF.Net email is sponsored by:
>>>>>> Check out the new SourceForge.net Marketplace.
>>>>>> It's the best place to buy or sell services for
>>>>>> just about anything Open Source.
>>>>>> http://sourceforge.net/services/buy/index.php
>>>>>> _______________________________________________
>>>>>> opensipstack-devel mailing list
>>>>>> ope...@li...
>>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>>>>
>>>>>>
>>>>>>
>>>>>>             
>>>>>
>>>>>
>>>>>           
>>>
>>>
>>>
>>>       
>>     
>
>
>
>
>   

Re: [OpenSBC] OpenSBC Project file reference to OSStest

[Joegen E. B. <joe...@gm...>]

Bill Stoner wrote:
> Hi all,
>
>
>  I am just getting my feet wet with OpenSBC (Windows deployment) and have a few questions.  I am not a software developer, but do dable with Visual Studio 2005.  
>
>
> First, the OpenSBC.sln file includes a project reference to OSSTest.vcproj.  This project doesn't appear in the zip file.  What is it and where can I get it?
>   

OSSTest project file is just an internal project we use to test blocks 
of code every now and then.   You don't need it to actually compile 
OpenSBC so you can safely remove it from the solution.




>
>  The service has an app that runs in the tray, but the icon is blank (space is provided, but icon not visible).  Placing the cusor over the area will activate the app.  Where is the ion?  If you select "Properties" from the dropdown, the app crashes.  Any ideas what is going on here?
>
>   


That's the tray icon for the OpenSBC service which shouold not be doing 
anything.   I remember I removed that thing in the default build.   It 
might have been accidentally reactivated.   I will check on it.




>  
>
>
> Thanks for your support
>
>
>  
>
>
> Bill
>
>
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by:
> Check out the new SourceForge.net Marketplace.
> It's the best place to buy or sell services
> for just about anything Open Source.
> http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace
> _______________________________________________
> Opensipstack-osbcdevel mailing list
> Ope...@li...
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>
>
>   

Re: [OpenSBC] OpenSBC 1.1.6, Summary of Feature Requests and Bug Reports

[Dangi, R. <raj...@as...>]

What about the feature to host OpenSBC behind NAT/firewall?

Rajpal

-----Original Message-----
From: ope...@li...
[mailto:ope...@li...] On Behalf
Of Joegen E. Baclor
Sent: Tuesday, December 11, 2007 5:59 PM
To: ope...@li...; listaopenSBC
Subject: [OpenSBC] OpenSBC 1.1.6, Summary of Feature Requests and Bug
Reports

Hi Everyone,

I am intending to finalize 1.1.5 release before the year ends so we have

a clean slate for 1.1.6.  If you have feature requests or bug fixes=20
which you think should be part of this release, please document them and

post them to the list ASAP.  We are going to implement  CVS branches=20
beginning 1.1.5 to allow bug fixes specific to a certain branch.   There

will be some major architectural changes that will be implemented in=20
version 1.1.6 which might not be anymore compatible with the 1.1.5=20
branch.   I will enumerate the pending feature requests I summarized=20
from previous post.  Let me know if I missed smething

Pending Feature Requests:

1.  Support for redirect in relay Routes - sam...@ve...
2.  Creation of folders (logs, registry, calea etc ) in places where=20
user has permission - H.Kropf
3.  Rewriting of From URI - Warren Kreckler
4.  Routing based on  Packet  Source - chr...@gm...
5.  Routing based on recipient NIC - Warren Kreckler

If I  missed something feel free to reply inline.

Planned Features for 1.1.6

1.  I have already announced the plan for a separate configuration=20
module on top of the current HTTP Admin. =20
2.  We are also going to implement a separate  media proxy application=20
to allow HA/Load Balancing of streams on top of the internal media proxy

implementation.  This change would allow an SBC to use multiple=20
instances of external media proxies giving way to a higher volume=20
deployment. =20
3.  We are also going to extract the Media Server Trunk and promote it=20
as a fully featured VXML Media Server allowing PBX developers to create=20
PBX applications using the OpenSIPStack Framework.=20
4.  The plan to implement Diameter as the AAA implementation in OpenSBC.
5.  Finally, the long awaited resurrection of the TCP and TLS
transports.

I have also Opened a  new development effort in SourceForge  called =20
Plugin++ (  http://sourceforge.net/projects/pluginpp ).  This new=20
project will be the entry point of a planned Plugin Framework/API for=20
OpenSBC and other OpenSIPStack applications.

We are aiming for June 2008 as the ETC for the 1.1.6 release of=20
OpenSBC.  We will come up with milestone releases for each
functionality.


------------------------------------------------------------------------
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Check out the new SourceForge.net Marketplace.
It's the best place to buy or sell services for
just about anything Open Source.
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_______________________________________________
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Ope...@li...
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[OpenSBC] OpenSBC Project file reference to OSStest

[Bill S. <b.s...@da...>]

Hi all,


 I am just getting my feet wet with OpenSBC (Windows deployment) and have a few questions.  I am not a software developer, but do dable with Visual Studio 2005.  


First, the OpenSBC.sln file includes a project reference to OSSTest.vcproj.  This project doesn't appear in the zip file.  What is it and where can I get it?


 The service has an app that runs in the tray, but the icon is blank (space is provided, but icon not visible).  Placing the cusor over the area will activate the app.  Where is the ion?  If you select "Properties" from the dropdown, the app crashes.  Any ideas what is going on here?


 


Thanks for your support


 


Bill

Re: [OpenSBC] Call through Sip Trunk if Enum fails

[Joegen E. B. <joe...@gm...>]

You would probably be able to reach at least up to enum, osbc...  But 
you might not be able to failover to a sip-trunk right now because the 
SIP-Trunk is handled differently than any other route.   I am not very 
sure.  Have you tried

[sip:*] enum:nu...@e1..., sip:numkber@osbc2, 
sip:number@your_itsp;sip-trunk=true

?

Joegen


Dinesh Dialani wrote:
> Hi Joegen,
>
> I am using Sip Trunk as well as Enum capabilities of OpenSBC. I have a 
> question.
>
> I want that whenever an outbound call is established, it should first 
> search enum records corresponding to phone number dialed and if found 
> make a call directly.
>
> But it may happen that the other party also has an OpenSBC installed 
> and it has put IP of my OpenSBC in List of Blocked IP's, so the other 
> party will send a 403 Forbidden message.
>
> Now two situation can arise:
> 1. The call should be disconnected.
> 2. The call should now be made using Sip Trunk facilities of OpenSBC.
>
> I want to know how can I handle the situation in which if a call is 
> rejected by other party through Enum then the call should be 
> established through Sip Trunk.
>
>
>
>                                           Do an Enum lookup    
>     UA----->OpenSBC 1        
> ---------------------------->                  Enum Server
>
>                                          Enum lookup is successful   
>                                          <----------------------------
>                                           return a Sip URI
>
>
>        
>
>                OpenSBC 1          Call directly to other party
>                                            
> ----------------------------->                  OpenSBC 2
>
>
>                                            Blocked IP of OpensBC 1
>                                            
> <-----------------------------           
>                                             Sends a 403 Forbidden message
>
>
>                                            Establish a call using Sip 
> Trunk
>                                              
> ------------------------------>
>                
>
>                                             Call is established 
> successfully
>                                              
> <-------------------------------
>
>
>
> Regards,
>
> Dinesh
>
>
>
> ------------------------------------------------------------------------
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by:
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>   
> ------------------------------------------------------------------------
>
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> Version: 7.5.503 / Virus Database: 269.17.0/1180 - Release Date: 12/10/2007 2:51 PM
>   

Re: [OpenSBC] B2BUA Routes Based on source

[Joegen E. B. <joe...@gm...>]

You can't do it currently but I have added this in the feature request 
list for the final 1.1.5 release.

Cheers!

Chris Veazey wrote:
> Is it possible to create a B2BUA route based open the source of the SIP signaling?
>
>
> Customer A(IP 1.2.3.4)-->OSBC(IP 5.6.7.8)-->UAS(IP 6.7.8.9)
>
>
> B2BUA  Route:
>
>
> [sip:*@1.2.3.4*] sip:*@6.7.8.9:5060
>
>
>  
>
>
> Instead of:
>
>
>  [sip:*@5.6.7.8*] sip:*@6.7.8.9:5060
>
>
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by:
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> It's the best place to buy or sell services for
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>
>
>   

[OpenSBC] OpenSBC 1.1.6, Summary of Feature Requests and Bug Reports

[Joegen E. B. <joe...@gm...>]

Hi Everyone,

I am intending to finalize 1.1.5 release before the year ends so we have 
a clean slate for 1.1.6.  If you have feature requests or bug fixes 
which you think should be part of this release, please document them and 
post them to the list ASAP.  We are going to implement  CVS branches 
beginning 1.1.5 to allow bug fixes specific to a certain branch.   There 
will be some major architectural changes that will be implemented in 
version 1.1.6 which might not be anymore compatible with the 1.1.5 
branch.   I will enumerate the pending feature requests I summarized 
from previous post.  Let me know if I missed smething

Pending Feature Requests:

1.  Support for redirect in relay Routes - sam...@ve...
2.  Creation of folders (logs, registry, calea etc ) in places where 
user has permission - H.Kropf
3.  Rewriting of From URI - Warren Kreckler
4.  Routing based on  Packet  Source - chr...@gm...
5.  Routing based on recipient NIC - Warren Kreckler

If I  missed something feel free to reply inline.

Planned Features for 1.1.6

1.  I have already announced the plan for a separate configuration 
module on top of the current HTTP Admin.  
2.  We are also going to implement a separate  media proxy application 
to allow HA/Load Balancing of streams on top of the internal media proxy 
implementation.  This change would allow an SBC to use multiple 
instances of external media proxies giving way to a higher volume 
deployment.  
3.  We are also going to extract the Media Server Trunk and promote it 
as a fully featured VXML Media Server allowing PBX developers to create 
PBX applications using the OpenSIPStack Framework. 
4.  The plan to implement Diameter as the AAA implementation in OpenSBC.
5.  Finally, the long awaited resurrection of the TCP and TLS transports.

I have also Opened a  new development effort in SourceForge  called  
Plugin++ (  http://sourceforge.net/projects/pluginpp ).  This new 
project will be the entry point of a planned Plugin Framework/API for 
OpenSBC and other OpenSIPStack applications.

We are aiming for June 2008 as the ETC for the 1.1.6 release of 
OpenSBC.  We will come up with milestone releases for each functionality.

[OpenSBC] Call through Sip Trunk if Enum fails

[Dinesh D. <din...@gm...>]

Hi Joegen,

I am using Sip Trunk as well as Enum capabilities of OpenSBC. I have a
question.

I want that whenever an outbound call is established, it should first search
enum records corresponding to phone number dialed and if found make a call
directly.

But it may happen that the other party also has an OpenSBC installed and it
has put IP of my OpenSBC in List of Blocked IP's, so the other party will
send a 403 Forbidden message.

Now two situation can arise:
1. The call should be disconnected.
2. The call should now be made using Sip Trunk facilities of OpenSBC.

I want to know how can I handle the situation in which if a call is rejected
by other party through Enum then the call should be established through Sip
Trunk.



                                          Do an Enum lookup
    UA----->OpenSBC 1        ---------------------------->
Enum Server

                                         Enum lookup is successful
                                         <----------------------------
                                          return a Sip URI




               OpenSBC 1          Call directly to other party
                                           ----------------------------->
               OpenSBC 2


                                           Blocked IP of OpensBC 1
                                           <-----------------------------

                                            Sends a 403 Forbidden message


                                           Establish a call using Sip Trunk
                                             ------------------------------>


                                            Call is established successfully

<-------------------------------



Regards,

Dinesh

[OpenSBC] B2BUA Routes Based on source

[Chris V. <chr...@gm...>]

Is it possible to create a B2BUA route based open the source of the SIP signaling?


Customer A(IP 1.2.3.4)-->OSBC(IP 5.6.7.8)-->UAS(IP 6.7.8.9)


B2BUA  Route:


[sip:*@1.2.3.4*] sip:*@6.7.8.9:5060


 


Instead of:


 [sip:*@5.6.7.8*] sip:*@6.7.8.9:5060

Re: [OpenSBC] basic doubts

[Joegen E. B. <joe...@gm...>]

Cris wrote:
> Good Morning  everybody,
> I'm too raw in VoIP things, and iwould like to connect my Asterisk 
> with open SBC, how do I tell openSBC the ip address of SIP 
> Sever(Asterisk) I cant find the place ...Would you please help me?
> ThankYou
> Kind Regards
> Cris
>
>

Assuming you already know how to access the config pages, look for 
"B2BUA Routes" section.   In the route list enter the following entry

[sip:*] sip:192.168.0.1:5060  

Where 192.168.0.1:5060 is a place holder of the real ip address of your 
asterisk box.   [sip:*] simple means that you want to route all calls to 
asterisk.  If you want to be more specific, then you could specify the 
number in your filters

[sip:1212*]  sip:ip_of_asterisk_1

[sip:011*]  sip:ip_of_asterrisk_2

Joegen

[OpenSBC] basic doubts

[Cris <cr...@ti...>]

Good Morning  everybody,
I'm too raw in VoIP things, and iwould like to connect my Asterisk with 
open SBC, how do I tell openSBC the ip address of SIP Sever(Asterisk) I 
cant find the place ...Would you please help me?
ThankYou
Kind Regards
Cris

Re: [OpenSBC] Redirect functionality in OpenSBC

[Joegen E. B. <joe...@gm...>]

Try it in B2BUA Mode only.   I think redirect is only supported in the 
B2BUA and not the Proxy.


sandeep wrote:
> Hi,
>
>
>  I am running my OpenSBC in a Full Mode. Can OpenSBC work like a redirect server? What I tried is, I configured a rule in the B2BUA routes section as  [sip:613*] sip:61...@fw...;action="redirect" . But this is not working for me. Am I doing it correctly? or is there any other way?
>
>
>  
>
>
> Cheers
>
>
> -Sandeep 
>
>
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by: The Future of Linux Business White Paper
> from Novell.  From the desktop to the data center, Linux is going
> mainstream.  Let it simplify your IT future.
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>
>
>   

[OpenSBC] Redirect functionality in OpenSBC

[sandeep <sam...@ve...>]

Hi,


 I am running my OpenSBC in a Full Mode. Can OpenSBC work like a redirect server? What I tried is, I configured a rule in the B2BUA routes section as  [sip:613*] sip:61...@fw...;action="redirect" . But this is not working for me. Am I doing it correctly? or is there any other way?


 


Cheers


-Sandeep 

Re: [OpenSBC] (no subject)

[Cris <cr...@ti...>]

Joegen E. Baclor escribió:
> There were a lot of CVS activities yesterday. You might have downloaded 
> a none synchronized source. Try again. I just confirmed CVS now compiles 
> properly in GCC.
>
> Joegen
>
> CRISTINA DIEGO GUIJARRO wrote:
>   
>> Hi all, im trying to install osbc but after obtaining a succesfull
>> compiling with ./configure y go with make optnoshared and:
>> g++ -D_REENTRANT -D_REENTRANT -Wall  -DPTRACING
>> -I/home/cris/opensipstack/include -DPTRACING
>> -I/home/cris/opensipstack/include -Os    -c CallSession.cxx -o
>> /home/cris/opensipstack/lib/obj_linux_x86_r/CallSession.o
>> CallSession.cxx: In member function ‘virtual void
>> UACORE::CallSession::OnIncomingSIPMessage(SIP::SIPMessageArrival&)’:
>> CallSession.cxx:798: error: ‘NIST_RecvPrack’ no se declaró en este ámbito
>> CallSession.cxx: In member function ‘virtual void
>> UACORE::CallSession::OnSessionEvent(int, const SIPParser::SIPMessage&)’:
>> CallSession.cxx:963: error: ‘NIST_RecvPrack’ no se declaró en este ámbito
>> make[2]: *** [/home/cris/opensipstack/lib/obj_linux_x86_r/CallSession.o]
>> Error 1
>> make[2]: se sale del directorio `/home/cris/opensipstack/src'
>> make[1]: *** [optnoshared] Error 2
>> make[1]: se sale del directorio `/home/cris/opensipstack/src'
>> make: *** [optnoshared] Error 2
>> can anybody helpme please??
>> Lots of thanks
>>
>> -------------------------------------------------------------------------
>> SF.Net email is sponsored by: The Future of Linux Business White Paper
>> from Novell.  From the desktop to the data center, Linux is going
>> mainstream.  Let it simplify your IT future.
>> http://altfarm.mediaplex.com/ad/ck/8857-50307-18918-4
>> _______________________________________________
>> Opensipstack-osbcdevel mailing list
>> Ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>>
>>
>>   
>>     
>
>
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by: The Future of Linux Business White Paper
> from Novell.  From the desktop to the data center, Linux is going
> mainstream.  Let it simplify your IT future.
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> _______________________________________________
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> Ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>
> __________ Información de NOD32, revisión 2703 (20071205) __________
>
> Este mensaje ha sido analizado con NOD32 antivirus system
> http://www.nod32.com
>
>
>
>   
Hi again!!
You were right, i got a non sinchronized version.
Lots of  thanks!
Cris

Re: [OpenSBC] An enhanced HTTP Config Module for OpenSBC

[Joegen E. B. <joe...@gm...>]

To sum up.  One in the list is in favor of a python approach and another 
in favor of a built-in fast CGI approach.   Everyone seems unanimous in 
having SQL server support.  Just this morning, Ryan Colobong pointed me 
to http://www.cherrypy.org/ which is another python based solution to 
incorporate dynamic web pages to opensbc.   There seem to be a lot of 
resources we could use in a python based solution.   However, fast-cgi 
support in OpenSBC would carry on the tradition of simplicity in 
installation, it will take a lot longer just to develop the module that 
enabled fast-cgi support not to mention that HTTP is not really the turf 
OpenSIPStack.   At this point I am leaning towards a python based 
solution.   For those who  are willing to pitch in your ideas, now is 
the time to speak up.

Joegen



Joegen E. Baclor wrote:
> Hi Everyone,
>
> As I have hinted in the past, there is a plan to provide a new level of 
> administrative interface for OpenSBC.  As most of you might have 
> discovered by now, OpenSBC is very easy to install and requires 
> virtually no configuration for you to be able to run and use it.   This 
> is all because of a built-in HTTP server that allows for OpenSBC to be 
> configured remotely.   However, since version 1.1.4 and with the 
> introduction of more advanced features like SIP Trunking, the built-in 
> HTTP Config Pages is out-growing its simplicity.    We are now in a 
> point where we need to decide what technology to use to bring the 
> configuration modules to the next level.    We need to seriously 
> consider the following criteria in choosing the solution.
>
> 1.  It should be very easy to install and package
> 2.  Built-in access to back-end databases preferably Postgress
> 3.  Must be able to host Dynamic HTML Content
> 4.  Other criteria that might be useful are support for XML-RPC and SOAP
>
>
> Some of the basic choices that I pulled out straight from my thoughts are
>
> 1.  A separate web interface easily installable as standard Apache Web 
> application.  Choices are PHP, JSP Ruby with SQL back-end
> 2.  Stand-alone Python HTTP Server just like the trac project.  See 
> http://trac.edgewall.org/
> 3.  Enhance the HTTP Admin to support Fast-CGI extension to allow for 
> dynamic HTTP content to be hosted straight by OpenSBC
>
> I would want to hear from everyone if you are leaning towards a certain 
> approach or would want another approach altogether.
>
> Joegen
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by: The Future of Linux Business White Paper
> from Novell.  From the desktop to the data center, Linux is going
> mainstream.  Let it simplify your IT future.
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> _______________________________________________
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> Ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>
>
>   

Re: [OpenSBC] [OpenSIPStack] Configurable Host Access List now available

[Joegen E. B. <joe...@gm...>]

I have modified the host access list configuration to include a list of 
selected hosts to be banned access.   Two new items are added to the 
config page.

1.  Ban Selected Host - Flag whether OpenSBC would permanently reject 
INVITEs from the listed Banned hosts from the internet.
2.  Banned Hosts - Entry for host can be a single IP Address or a range. 
Example: 192.168.0.1 or 10.0.0.1-10.0.0.255


Joegen E. Baclor wrote:
> Hi Everyone,
>
> I have added a new configuration section "Host Access List" for OpenSBC 
> that allows only certain IP address to send calls to OpenSBC.   It can 
> accept a list of single IP addresses or a network block.  There  are 
> only two parameters in this page
>
> 1.  Trust All Hosts - Flag whether OpenSBC would accept INVITEs from ANY 
> host from the internet. If this flag is set to TRUE OpenSBC will only 
> accept calls from the IP Addresses or Range listed.
> 2.  Hosts - Entry for host can be a single IP Address or a range. 
> Example: 192.168.0.1 or 10.0.0.1-10.0.0.255
>
> If the Access List is enabled and a call came in from an unknown host, 
> OpenSBC will answer with a forbidden with a Warning header.  A sample of 
> such response is below:
>
> SIP/2.0 403 Forbidden
> From: "unknown" <sip:20...@pb...>;tag=172561562940
> To:  <sip:10...@pb...>;tag=de055ec348fa18109fd8d5d74376c6ed
> Via: SIP/2.0/UDP 
> 127.0.0.1:1000;iid=5050;branch=z9hG4bKc0a800990000005847557d8900000c8200000117;rport=1000;received=192.168.120.1
> CSeq: 1 INVITE
> Call-ID: E7335922-1B42-4670-817A-EE1FEFC337EC@192.168.0.153
> Server: OpenSIPStack-1.1.7-18
> Warning: 399 192.168.120.1 Is Not Allowed In Our Access List
> Content-Length: 0
>
> Joegen
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by: The Future of Linux Business White Paper
> from Novell.  From the desktop to the data center, Linux is going
> mainstream.  Let it simplify your IT future.
> http://altfarm.mediaplex.com/ad/ck/8857-50307-18918-4
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSBC] (no subject)

[Joegen E. B. <joe...@gm...>]

There were a lot of CVS activities yesterday. You might have downloaded=20
a none synchronized source. Try again. I just confirmed CVS now compiles =

properly in GCC.

Joegen

CRISTINA DIEGO GUIJARRO wrote:
> Hi all, im trying to install osbc but after obtaining a succesfull
> compiling with ./configure y go with make optnoshared and:
> g++ -D_REENTRANT -D_REENTRANT -Wall  -DPTRACING
> -I/home/cris/opensipstack/include -DPTRACING
> -I/home/cris/opensipstack/include -Os    -c CallSession.cxx -o
> /home/cris/opensipstack/lib/obj_linux_x86_r/CallSession.o
> CallSession.cxx: In member function =91virtual void
> UACORE::CallSession::OnIncomingSIPMessage(SIP::SIPMessageArrival&)=92:
> CallSession.cxx:798: error: =91NIST_RecvPrack=92 no se declar=F3 en est=
e =E1mbito
> CallSession.cxx: In member function =91virtual void
> UACORE::CallSession::OnSessionEvent(int, const SIPParser::SIPMessage&)=92=
:
> CallSession.cxx:963: error: =91NIST_RecvPrack=92 no se declar=F3 en est=
e =E1mbito
> make[2]: *** [/home/cris/opensipstack/lib/obj_linux_x86_r/CallSession.o=
]
> Error 1
> make[2]: se sale del directorio `/home/cris/opensipstack/src'
> make[1]: *** [optnoshared] Error 2
> make[1]: se sale del directorio `/home/cris/opensipstack/src'
> make: *** [optnoshared] Error 2
> can anybody helpme please??
> Lots of thanks
>
> -----------------------------------------------------------------------=
--
> SF.Net email is sponsored by: The Future of Linux Business White Paper
> from Novell.  From the desktop to the data center, Linux is going
> mainstream.  Let it simplify your IT future.
> http://altfarm.mediaplex.com/ad/ck/8857-50307-18918-4
> _______________________________________________
> Opensipstack-osbcdevel mailing list
> Ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>
>
>  =20

[OpenSBC] (no subject)

[CRISTINA D. G. <cr...@ti...>]

Hi all=2C im trying to install osbc but after obtaining a succesfull
compiling with =2E/configure y go with make optnoshared and=3A
g++ -D=5FREENTRANT -D=5FREENTRANT -Wall  -DPTRACING
-I/home/cris/opensipstack/include -DPTRACING
-I/home/cris/opensipstack/include -Os    -c CallSession=2Ecxx -o
/home/cris/opensipstack/lib/obj=5Flinux=5Fx86=5Fr/CallSession=2Eo
CallSession=2Ecxx=3A In member function =91virtual void
UACORE=3A=3ACallSession=3A=3AOnIncomingSIPMessage(SIP=3A=3ASIPMessageArri=
val=26)=92=3A
CallSession=2Ecxx=3A798=3A error=3A =91NIST=5FRecvPrack=92 no se declar=F3=
 en este =E1mbito
CallSession=2Ecxx=3A In member function =91virtual void
UACORE=3A=3ACallSession=3A=3AOnSessionEvent(int=2C const SIPParser=3A=3AS=
IPMessage=26)=92=3A
CallSession=2Ecxx=3A963=3A error=3A =91NIST=5FRecvPrack=92 no se declar=F3=
 en este =E1mbito
make=5B2=5D=3A *** =5B/home/cris/opensipstack/lib/obj=5Flinux=5Fx86=5Fr/C=
allSession=2Eo=5D
Error 1
make=5B2=5D=3A se sale del directorio =60/home/cris/opensipstack/src=27
make=5B1=5D=3A *** =5Boptnoshared=5D Error 2
make=5B1=5D=3A se sale del directorio =60/home/cris/opensipstack/src=27
make=3A *** =5Boptnoshared=5D Error 2
can anybody helpme please=3F=3F
Lots of thanks

[OpenSBC] Configurable Host Access List now available

[Joegen E. B. <joe...@gm...>]

Hi Everyone,

I have added a new configuration section "Host Access List" for OpenSBC 
that allows only certain IP address to send calls to OpenSBC.   It can 
accept a list of single IP addresses or a network block.  There  are 
only two parameters in this page

1.  Trust All Hosts - Flag whether OpenSBC would accept INVITEs from ANY 
host from the internet. If this flag is set to TRUE OpenSBC will only 
accept calls from the IP Addresses or Range listed.
2.  Hosts - Entry for host can be a single IP Address or a range. 
Example: 192.168.0.1 or 10.0.0.1-10.0.0.255

If the Access List is enabled and a call came in from an unknown host, 
OpenSBC will answer with a forbidden with a Warning header.  A sample of 
such response is below:

SIP/2.0 403 Forbidden
From: "unknown" <sip:20...@pb...>;tag=172561562940
To:  <sip:10...@pb...>;tag=de055ec348fa18109fd8d5d74376c6ed
Via: SIP/2.0/UDP 
127.0.0.1:1000;iid=5050;branch=z9hG4bKc0a800990000005847557d8900000c8200000117;rport=1000;received=192.168.120.1
CSeq: 1 INVITE
Call-ID: E7335922-1B42-4670-817A-EE1FEFC337EC@192.168.0.153
Server: OpenSIPStack-1.1.7-18
Warning: 399 192.168.120.1 Is Not Allowed In Our Access List
Content-Length: 0

Joegen

Re: [OpenSBC] Unable to validate XML file against XML Schema

[Joegen E. B. <joe...@gm...>]

PXML is a wrapper around expat.  Expat is a NONE validating parser.  
It's the trade off for simplicity and speed. 


Dinesh Dialani wrote:
> Hi Joegen,
>
> I am using PXML.h file to parse an XML file but I also want to 
> validate it against XML Schema.
>
> Is there any function to validate XML file against XML schema in OpenSBC?
>
> Thanks
>
> Dinesh
>  
>
>
> ------------------------------------------------------------------------
>
> -------------------------------------------------------------------------
> SF.Net email is sponsored by: The Future of Linux Business White Paper
> from Novell.  From the desktop to the data center, Linux is going
> mainstream.  Let it simplify your IT future.
> http://altfarm.mediaplex.com/ad/ck/8857-50307-18918-4
> ------------------------------------------------------------------------
>
> _______________________________________________
> Opensipstack-osbcdevel mailing list
> Ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel
>   
> ------------------------------------------------------------------------
>
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.503 / Virus Database: 269.16.13/1169 - Release Date: 12/3/2007 10:56 PM
>   

[OpenSBC] Unable to validate XML file against XML Schema

[Dinesh D. <din...@gm...>]

Hi Joegen,

I am using PXML.h file to parse an XML file but I also want to validate it
against XML Schema.

Is there any function to validate XML file against XML schema in OpenSBC?

Thanks

Dinesh