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Re: [OpenSIPStack] Last version OpenSIPStack Download

From: Andre S. <eds...@ya...> - 2007-05-22 00:31:19

Jeremy,
   
  Download TORTOISE (client cvs interface).
  - As soon as you installed it make a directory on where you would put the open sources.
  - right click on the directory amnd choose CVS CHECKOUT and a dialog box will appear.
   
  Insert the following on the entry boxes
   
  CVS CHECKOUT 
:pserver:ano...@op...:/cvsroot/opensipstack
   
  MODULE 
  atlsip or opensipstack or opensbc
   
  
Counet Jérémy <jer...@ze...> wrote:
                Hi, 
  I see that some updates of OpenSIPStack are distributed, but where? Im sorry but on the web, there is always the 1.1.5 version on download link.
  And I dont know how to make downloading from CVS
  Thanks for information,
   
  Jeremy Counet.

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[OpenSIPStack] Last version OpenSIPStack Download

From: <jer...@ze...> - 2007-05-21 13:49:32

Hi,=20

I see that some updates of OpenSIPStack are distributed, but where? I'm =
sorry but on the web, there is always the 1.1.5 version on download =
link.

And I don't know how to make downloading from CVS...

Thanks for information,

=20

Jeremy Counet.

Re: [OpenSIPStack] OpenSBC - Load balancing

From: Joegen E. B. <jb...@so...> - 2007-05-21 13:21:59

Costin Bularca wrote:
> Hello,
> I have some questions about how "load balancing" is working (e.g. I 
> have 3 instances of OpenSBC, the first is set in "ProxyOnlyMode" and 
> in its relay route I set the IP's of the second and the third 
> instance, which are running in "FullMode") :  
>  
> 1.  How the first instance(Proxy) distribute the requests (eg. 
> INVITE)?  Does it know which instance(FullMode) has less sessions?

It's done via round robin.  The load-balancer instance distributes 
evenly accross the three instances.


> 2. When an INVITE message arrives in Proxy, the message is sent to a 
> intance(FullMode) and there is created a session.  When BYE arrives 
> from the same client, does Proxy know to send the BYE message to that 
> intance(FullMode), which handles the session?

Yes.  When a dialog is created, the destination of requests sent within 
that dialog would be towards the contact address provided with the 200 
OK.  If the UA core that processed your invite is the B2BUA of the 
full-mode instance, then the contact would be the listener address of 
the full-mode instance.   If you used the proxy core of the full-mode 
instance, then it would have inserted a record route.  Thus subsequent 
request would always contain a route-set traversing that proxy instance.



>  
> Regards,
> Costin Bularca
>
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Re: [OpenSIPStack] OpenSIPStack

From: Joegen E. B. <jb...@so...> - 2007-05-21 12:56:35

Make sure you call Initialize() every time you make a change to the 
softphone properties

Counet Jérémy wrote:
>
> Hi,
>
> When I make change in SIP Account User, User ID for example, changes 
> are log in default.ini with no problems.
>
> But when I do a “DoLogin()”, It log with old configuration, loaded in 
> the starting up of the application. How can I do to make sure that 
> “DoLogin()” is logged with default.ini configuration?
>
> Thanks a lot,
>
> Jeremy Counet.
>
> ------------------------------------------------------------------------
>
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[OpenSIPStack] OpenSBC - Load balancing

From: Costin B. <cos...@ya...> - 2007-05-21 12:56:08

Hello, 
I have some questions about how "load balancing" is working (e.g. I have 3 instances of OpenSBC, the first is set in "ProxyOnlyMode" and in its relay route I set the IP's of the second and the third instance, which are running in "FullMode") :  

1.  How the first instance(Proxy) distribute the requests (eg. INVITE)?  Does it know which instance(FullMode) has less sessions?
2. When an INVITE message arrives in Proxy, the message is sent to a intance(FullMode) and there is created a session.  When BYE arrives from the same client, does Proxy know to send the BYE message to that intance(FullMode), which handles the session?

Regards,
Costin Bularca


       
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Re: [OpenSIPStack] OSSPhone impossible to re-login

From: Joegen E. B. <jb...@so...> - 2007-05-21 12:48:37

Andre/Jeremy

Come on guys, you give up so easily! If you are using OpenSBC, the 200 
OK sent to the Unregister does not contain a contact and an expires. 
This bug caused OSSPhone to not distinguish the response properly as a 
response to Unregister. So thanks to you, a new bug has been patched. I 
have also brought this thing up with Ilian. His next wave of updates 
should have made the Softphone immune to this situation.

Joegen

Andre Silo wrote:
> Jeremy,
> We just close the whole application when we logout. Re-login does not work
>
> */Counet Jérémy <jer...@ze...>/* wrote:
>
>     Hi,
>     When I logout from the server with OSSPhone, is it impossible to
>     re-login to the server after… Can you explain me why?
>     Regards,
>     Jeremy Counet.
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[OpenSIPStack] OpenSIPStack

From: <jer...@ze...> - 2007-05-21 12:30:08

Hi,

=20

When I make change in SIP Account User, User ID for example, changes are =
log in default.ini with no problems.

But when I do a "DoLogin()", It log with old configuration, loaded in =
the starting up of the application. How can I do to make sure that =
"DoLogin()" is logged with default.ini configuration?

=20

Thanks a lot,

=20

Jeremy Counet.

Re: [OpenSIPStack] OSSPhone impossible to re-login

From: Andre S. <eds...@ya...> - 2007-05-21 08:53:14

Jeremy,
   
  We just close the whole application when we logout. Re-login does not work

Counet Jérémy <jer...@ze...> wrote:
                Hi,
   
  When I logout from the server with OSSPhone, is it impossible to re-login to the server after Can you explain me why?
   Regards,
   
  Jeremy Counet.

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Re: [OpenSIPStack] Event_LogoutSuccessful

From: Andre S. <eds...@ya...> - 2007-05-21 08:51:49

Jeremy,
   
  What we did was force the logout every 10 seconds until the event pops out. 
  For the recording 1) We record using our asterisk, 2) inserted record/play library on our softphone UI.

Counet Jérémy <jer...@ze...> wrote:
                Hi,
  Ive a problem when I logout, there is no event LogoutSuccessful when I do DoLogout(). But I see perfectly that Im logout on the server.
  I would like to know if it possible to record audio is streaming during a conversation. Which layer in the stack allows me to do that?
   
  Thx
   
  Jeremy Counet

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[OpenSIPStack] OSSPhone impossible to re-login

From: <jer...@ze...> - 2007-05-21 08:15:37

Hi,

=20

When I logout from the server with OSSPhone, is it impossible to =
re-login to the server after... Can you explain me why?

 Regards,

=20

Jeremy Counet.

[OpenSIPStack] Event_LogoutSuccessful

From: <jer...@ze...> - 2007-05-21 07:47:25

Hi,

I've a problem when I logout, there is no event "LogoutSuccessful" when =
I do "DoLogout()". But I see perfectly that I'm logout on the server.

I would like to know if it possible to record audio is streaming during =
a conversation. Which layer in the stack allows me to do that?

=20

Thx

=20

Jeremy Counet

Re: [OpenSIPStack] Error compiling OpenSBC on Windows

From: Joegen E. B. <jb...@so...> - 2007-05-18 21:12:04

Make sure you do an update on OpenSBC as well.  This code was checked in 
yesterday.

Joegen

Antonio Higuera wrote:
> I get the following error compiling the last CVS code:
>
>  \OpenSBC.cxx(445) : error C2039: 'SetBackDoorEndPoint' : it is not 
> member of 'OpenSBC'
>
> The previous CVS code compilled fine.
>
> Regards,
> Antonio
>
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>
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[OpenSIPStack] Error compiling OpenSBC on Windows

From: Antonio H. <ahi...@gm...> - 2007-05-18 17:20:02

I get the following error compiling the last CVS code:

 \OpenSBC.cxx(445) : error C2039: 'SetBackDoorEndPoint' : it is not member
of 'OpenSBC'

The previous CVS code compilled fine.

Regards,
Antonio

Re: [OpenSIPStack] OSBC - load balancing

From: Joegen E. B. <jb...@so...> - 2007-05-18 09:56:11

Costin Bularca wrote:
> Hello,
> I saw in some descriptions for OpenSBC that this can be configured to 
> load balance sessions across other instances. Each instance may be run 
> on separate servers.
> For example: I have 2 instances of OpenSBC. Every of them runs on a PC.
> How can I configure this 2 instances to make load balancing?
>
> Regards,
> Costin Bularca
>

You can configure OpenSBC as a load balancer.   You are going to need to 
run the load balancer instance in one of your two boxes.  The load 
balancer should be configured to run in ProxyOnlyMode.  Then in Relay 
Routes, you can specify the address of the other two instances of opensbc

Example:

[*]  sip:opensbcipaddress_1:5080 | sip:opensbcipaddress_2:5060

Joegen

[OpenSIPStack] OSBC - load balancing

From: Costin B. <cos...@ya...> - 2007-05-18 09:44:03

Hello,
I saw in some descriptions for OpenSBC that this can be configured to load balance sessions across
other instances. Each instance may be run on separate servers.
For example: I have 2 instances of OpenSBC. Every of them runs on a PC.
How can I configure this 2 instances to make load balancing?

Regards,
Costin Bularca





 
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Re: [OpenSIPStack] FW: OpenSBC as Forking Proxy

From: Joegen E. B. <jb...@so...> - 2007-05-18 00:15:41

Gustavo,

Sorry, I forgot to get back to you.  If you have the latest CVS head 
code, check out BOOL SIPTransaction::DoDNSFailover() in 
SIPTransaction.cxx.   I have committed this a few days ago to 
demonstrate fail-over by forking using DNS/SRV records.   New 
transactions are created by calling FindTransactionAndAddEvent().   You 
can just change its behavior a bit and get the fail-over routes 
somewhere instead of DNS/SRV queries.

Joegen


Gustavo Curetti wrote:
>
> Joegen:
>
> I don't understand how to create a new client transaction when the 
> first invite fail in the FSM layer. Could you give some directions, 
> please?
>
> Thanks for your help.
>
> Gustavo.
>
>
>
>     ------------------------------------------------------------------------
>     From: cur...@ho...
>     To: jb...@so...
>     Subject: RE: [OpenSIPStack] OpenSBC as Forking Proxy
>     Date: Mon, 14 May 2007 16:18:57 +0200
>
>     Joegen:
>      
>     Thanks for your help. Do you suggest to do the serial forking in
>     FSM layer with a custom header?.Must each new try create a new
>     client transaction?
>      
>     Thanks
>     Gustavo
>
>     ------------------------------------------------------------------------
>
>     > Date: Thu, 10 May 2007 13:28:29 +0800
>     > From: jb...@so...
>     > To: cur...@gm...
>     > Subject: Re: [OpenSIPStack] OpenSBC as Forking Proxy
>     >
>     > Gustavo,
>     >
>     > This will be a bit tricky. It is not as simple as spawning an
>     outbound
>     > invite. There should be a clean mechanism to clone transactions and
>     > this is not present in the FSM currently. Forking should be done
>     in the
>     > FSM layer, not in the UACore layer. I will see what I can do to
>     help
>     > you. I will let you know when I have something you can use
>     cleanly to
>     > fork your calls. Perhaps over the weekend, but that isn't a promise.
>     >
>     > Joegen
>     >
>     > Gustavo Curetti wrote:
>     > > Joegen:
>     > >
>     > > What I want to do is a very simple sequential search. When one
>     > > destination don't answer or reject the call I want the OpenSBC
>     try
>     > > another.
>     > >
>     > > I made the following changes in the code for timer B
>     expiration just
>     > > for do some tests:
>     > >
>     > > void ProxySessionManager::OnTimerExpire(
>     > > SIPTimerExpire & timerEvent,
>     > > SIPSession * session
>     > > )
>     > > {
>     > > if( session != NULL )
>     > > {
>     > > LOG_IF_DEBUG( LogWarning(), "*** TIMER EXPIRATION *** for SIP
>     > > Session " << session->GetSessionId() );
>     > > if( timerEvent.GetTimer() == SIPTransactions::SIPTimerEvent::B)
>     > > {
>     > > SIPMessage msg = ((ProxySession *)session)->GetOriginalInvite();
>     > > session->EnqueueSessionEvent( new SIPSessionEvent( *session, 1,
>     > > msg ) );
>     > > }
>     > > session->OnTimerExpire( timerEvent );
>     > > }
>     > > }
>     > >
>     > > and
>     > >
>     > > void ProxySession::OnTimerExpire(
>     > > SIPTimerExpire & timerEvent
>     > > )
>     > > {
>     > > GCREF( "SIPSession::OnTimerExpire" );
>     > > if( timerEvent.GetTimer() == SIPTransactions::SIPTimerEvent::B ||
>     > > timerEvent.GetTimer() == SIPTransactions::SIPTimerEvent::F )
>     > > {
>     > > ///this is an ICT timeout
>     > > SIPMessage timeout;
>     > > GetCurrentUASRequest().CreateResponse( timeout,
>     > > SIPMessage::Code480_TemporarilyNotAvailable );
>     > > SendRequest( timeout );
>     > > }
>     > >
>     > > //Destroy();
>     > > }
>     > >
>     > > With these changes and a relay route:
>     > >
>     > > [sip:*@192.168.0.207:*] sip:192.168.0.1:5060,
>     sip:192.168.0.60:5060
>     > >
>     > > the OpenSBC made the second invite successfully. But what i
>     really
>     > > want is to use some custom headers with a list destination
>     addresses
>     > > instead of the relay routes and to do the same in case of a
>     reject. Do
>     > > you have any suggestions?
>     > >
>     > > Other question: Can i have two active ICT for a session?
>     Because in
>     > > the case of reject, I must start a new ICT for trying the next
>     > > destination but canceling throw the first ICT at the same time.
>     > >
>     > > Thanks for your help.
>     > >
>     > > Gustavo
>     > >
>     > >
>     ------------------------------------------------------------------------
>     > >
>     > > > Date: Sat, 5 May 2007 13:50:44 +0800
>     > > > From: jb...@so...
>     > > > To: cur...@gm...;
>     ope...@li...
>     > > > Subject: Re: [OpenSIPStack] OpenSBC as Forking Proxy
>     > > >
>     > > > Gustavo,
>     > > >
>     > > > Forking is not supported yet in OpenSBC.
>     > > >
>     > > > Joegen
>     > > >
>     > > > Gustavo Curetti wrote:
>     > > > >
>     > > > > Hi Joegen:
>     > > > >
>     > > > > I want to use the OpenSBC as a Forking Proxy. I want that the
>     > > > > OpenSBC try the differents Relays Routes one by one. Could
>     you give
>     > > > > some directions, please?
>     > > > >
>     > > > > Thanks for your help.
>     > > > >
>     > > > > Gustavo.
>     > > > >
>     > > > >
>     > >
>     ------------------------------------------------------------------------
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Re: [OpenSIPStack] Having problems with the class RTP_UDP

From: Joegen E. B. <jb...@so...> - 2007-05-17 12:35:14

Claudio Miceli wrote:
> Hello guys ,
>
>
> I am having a problem with PTimeInterval in RTP_UDP at rtp.h. I have 
> created  a class that  inhiteries this abstract class. But during 
> compilation the compiler can not recognize PTimeInterval. I am using 
> Microsft Visual Studio 2005. The most weird thing is that none of the 
> other classes give me trouble, just this one. Do you have any ideas on 
> how to solve this problem ?
>
>
> Claudio Miceli de Farias
>

Make sure you #inlude <ptlib.h> before you include rtp.h in your code.

[OpenSIPStack] Having problems with the class RTP_UDP

From: Claudio M. <cmi...@gm...> - 2007-05-17 12:24:12

Hello guys ,


I am having a problem with PTimeInterval in RTP_UDP at rtp.h. I have
created  a class that  inhiteries this abstract class. But during
compilation the compiler can not recognize PTimeInterval. I am using
Microsft Visual Studio 2005. The most weird thing is that none of the other
classes give me trouble, just this one. Do you have any ideas on how to
solve this problem ?


Claudio Miceli de Farias

[OpenSIPStack] Control call length and rfc 3372

From: <jf...@id...> - 2007-05-17 09:24:52

Hi Guys,

I really wanted to control the call duration once it is connected but I
do'nt know how to use StartAutoDestructTimer and by the way do you have
any plans to implement rfc 3372?


Thanks,
Julian

Re: [OpenSIPStack] OpenSBC and SER, invites are not passed from the inside interface to UA

From: Joegen E. B. <jb...@so...> - 2007-05-17 04:07:03

For OpenSBC to process the INVITE coming from SER, it needs to be routed via the backdoor port.&nbsp;&nbsp; This is normally 65060.&nbsp;&nbsp; This is also the port presented as the contact address presented to SER during upper-reg.&nbsp; If SER statically routes to the defaut port 5060, OpenSBC would treat it as a loop and drop the request.<br /><br /><br />

[OpenSIPStack] OpenSBC configuration for public call

From: paolinho9 <pao...@li...> - 2007-05-16 14:17:56

Hello!
I'm trying to use OpenSBC as B2BBUA (or B2BUAUpperReg) with my wengo acco=
unt. 
I'm not be able to make a call PC to PC with one of my wengo contacts. Th=
e achitecture is:


my Generic SoftPhone ----- OpenSBC ---- internet ----- wengo contact

Could anyone suggest a correct configuration for OpenSBC?
Thanks to all!!

Paolo

  =0A=0A=0A------------------------------------------------------=0ALeggi=
 GRATIS le tue mail con il telefonino i-mode=99 di Wind=0Ahttp://i-mode.w=
ind.it/=0A

Re: [OpenSIPStack] Wrong registration on upper sip proxy.

From: Joegen E. B. <jb...@so...> - 2007-05-15 12:34:28

Hi Antonio,

1.  You need to explicitly specify a route in UpperRegistraton routes 
for OpenSBC to change the contact address.  eg: [sip:*myregistrar*] 
sip:myregistrar.   I know its a bit none-inuitive.   This is merely 
becasue OSBC allows you to rewrite the domain for registrarion.  in this 
case the domain is maintained.

2.  New code in CVS should now allow REGISTERs to be hijacked 
automatically as long as the mode is B2BUpperReg.   No need to specify a 
route anymore.

Joegen


Antonio Higuera wrote:
> Hi list,
>
> I have a questions related the following network configuration:
>
> UA (private IP)-------NAT(public IP)-----Internet-------(public 
> IP)NAT-----(private IP)OpenSBC-----(private IP)SER
>
> The NAT box before OpenSBC has open the 5060 sip and rtp port.
>  As you can see I have clients behind a NAT box triyng to get access 
> to the SIP proxy (SER), OpenSBC is in the middle acting as B2BUA to 
> rtp proxing if is needed.
> In this scenario the UA client can register normally on the SIP 
> registar (behind OpenSBC), but  the SIP  REGISTER message from OpenSBC 
> to SIP registar has in the contact header the original private IP 
> address of the client. The fact is the client can call but it cannot 
> be called because the private address.
>
> Is it normal this behaviour? Can I configure OpenSBC to replace the 
> contact header with the OpenSBC IP address? This way all the calls to 
> the UA pass through OpenSBC given it has control over the NAT.
>
> Thanks very much,
> Antonio.
> ------------------------------------------------------------------------
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>

[OpenSIPStack] Wrong registration on upper sip proxy.

From: Antonio H. <ahi...@gm...> - 2007-05-14 14:10:00

Hi list,

I have a questions related the following network configuration:

UA (private IP)-------NAT(public IP)-----Internet-------(public
IP)NAT-----(private IP)OpenSBC-----(private IP)SER

The NAT box before OpenSBC has open the 5060 sip and rtp port.
 As you can see I have clients behind a NAT box triyng to get access to the
SIP proxy (SER), OpenSBC is in the middle acting as B2BUA to rtp proxing if
is needed.
In this scenario the UA client can register normally on the SIP registar
(behind OpenSBC), but  the SIP  REGISTER message from OpenSBC to SIP
registar has in the contact header the original private IP address of the
client. The fact is the client can call but it cannot be called because the
private address.

Is it normal this behaviour? Can I configure OpenSBC to replace the contact
header with the OpenSBC IP address? This way all the calls to the UA pass
through OpenSBC given it has control over the NAT.

Thanks very much,
Antonio.

Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make call

From: bondan <bo...@ci...> - 2007-05-14 08:25:49

Dear Joegen,

Thank you for your information. I will try it.

Best regards,
Bondan Wisnuwardhana



            Send an Instant Message
-----Original Message-----
From: Joegen E. Baclor [mailto:joe...@gm...] 
Sent: Monday, May 14, 2007 9:51 AM
To: bondan
Cc: ope...@li...
Subject: Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make call

You need to download opensipstack using a CVS client.  
See http://sourceforge.net/cvs/?group_id=156710 for instructions.

modules are opensipstack, atlsip and opensbc 

bondan wrote:
> Dear Joegen,
>
> I visit http://www.opensipstack.org/, and the current version of
> OpenSIPStack is 1.1.5 version. Also, the current version of ATLSIP is
1.0.2
> version. Would you inform to me where I have to download the newest
version
> of OpenSIPStack and ALTSIP?
>
>
> Many thanks,
> Bondan Wisnuwardhana
>
>
>             Send an Instant Message
>
> -----Original Message-----
> From: Joegen E. Baclor [mailto:jb...@so...] 
> Sent: Friday, May 11, 2007 11:38 AM
> To: jb...@so...; ope...@li...
> Cc: bondan
> Subject: Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make
call
>
> Bondan,
>
> One more thing.  Looking at your logs, it seems you are using an old 
> copy of OpenSIPStack?   Please download the latest CVS head.  Current 
> version is
>
> OpenSIPStack-1.1.6-133
>
>
> Joegen
>
> Joegen E. Baclor wrote:
>   
>> Hi Bondan,
>>
>> I am not able to replicate your problem.  I just recompiled OSSPhone 
>> from CVS and it's able to autheticate calls against OpenSBC terminating 
>> to an asterisk box.  See logs below.
>>
>> BTW, is this modified code you are working with?
>>
>> ----------------3:44:55.989----------------
>> *** LISTENER STARTED *** 127.0.0.1:5060
>> ----------------3:44:56.022----------------
>> *** LISTENER STARTED *** 192.168.0.161:5060 [*** DEFAULT LISTENER ***]
>> ----------------3:44:56.028----------------
>> *** LISTENER STARTED *** 192.168.6.1:5060
>> ----------------3:44:56.031----------------
>> *** LISTENER STARTED *** 192.168.15.1:5060
>> ----------------3:46:44.147----------------
>> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
>> (SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From:  <sip:186...@i2...>;tag=1359945326143
>> To:  <sip:63.116.254.88:12000>
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643ee5500004e3500000099;rport=6
> 2787;received=125.60.243.66
>   
>> CSeq: 64 OPTIONS
>> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
>> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
>> Content-Length: 0
>>
>> ----------------3:51:05.733----------------
>> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
>> (SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From:  <sip:186...@i2...>;tag=1386096812568
>> To:  <sip:63.116.254.88:12000>
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643ef5b000027b10000009a;rport=6
> 2787;received=125.60.243.66
>   
>> CSeq: 65 OPTIONS
>> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
>> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
>> Content-Length: 0
>>
>> ----------------3:54:05.787----------------
>> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
>> (SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From:  <sip:186...@i2...>;tag=1404096821517
>> To:  <sip:63.116.254.88:12000>
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643f00f000014b30000009b;rport=6
> 2787;received=125.60.243.66
>   
>> CSeq: 66 OPTIONS
>> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
>> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
>> Content-Length: 0
>>
>> ----------------3:56:49.262----------------
>> SEND: enc=0 507 Bytes to 192.168.6.1:5061:UDP (REGISTER 
>> sip:192.168.6.1:5061 SIP/2.0) Interface Address=
>> REGISTER sip:192.168.6.1:5061 SIP/2.0
>> From:  <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc
>> To: sip:192.168.6.1
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3825;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas
> -addr=192.168.6.1;rport
>   
>> CSeq: 1 REGISTER
>> Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc
>> Contact: <sip:192.168.6.1:5060;transport=udp>
>> User-Agent: OpenSIPStack-1.1.6-133
>> Expires: 3600
>> Max-Forwards: 10
>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
>> Content-Length: 0
>>
>> ----------------3:56:49.320----------------
>> RCV: enc=0 424 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
>> (SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From:  <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc
>> To: sip:192.168.6.1
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3185;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas
> -addr=192.168.6.1;rport=5060;received=192.168.6.1
>   
>> CSeq: 1 REGISTER
>> Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc
>> Contact: <sip:192.168.6.1:5060;transport=udp>
>> Server: OpenSIPStack-1.1.6-133
>> Expires: 3600
>> Content-Length: 0
>>
>> ----------------3:57:00.206----------------
>> SEND: enc=0 762 Bytes to 192.168.6.1:5061:UDP (INVITE 
>> sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1
>> INVITE sip:613@192.168.6.1 SIP/2.0
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To: sip:613@192.168.6.1
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
> -addr=192.168.6.1;rport
>   
>> CSeq: 4711 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Contact: "1111" <sip:1111@192.168.6.1:5060>
>> User-Agent: OpenSIPStack-1.1.6-133
>> Max-Forwards: 10
>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
>> Content-Type: application/sdp
>> Content-Length: 228
>>
>> v=0
>> o=- 1178857661 1178857661 IN IP4 192.168.6.1
>> s=OSS RTP Session
>> c=IN IP4 192.168.6.1
>> t=0 0
>> m=audio 5000 RTP/AVP 101 106 3
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtpmap:106 iLBC/8000
>> a=rtpmap:3 GSM/8000
>> ----------------3:57:00.222----------------
>> RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP 
>> (SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To: sip:613@192.168.6.1
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
> -addr=192.168.6.1
>   
>> CSeq: 4711 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Content-Length: 0
>>
>> ----------------3:57:00.248----------------
>> RCV: enc=0 557 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
>> (SIP/2.0 407 Proxy Authentication Required)
>> SIP/2.0 407 Proxy Authentication Required
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To:  <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
> -addr=192.168.6.1;rport=5060;received=192.168.6.1
>   
>> CSeq: 4711 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Proxy-Authenticate: Digest realm=192.168.6.1, 
>> nonce="92a98f3ef419bf98501149af46ee7604", 
>> opaque="e430240b53dfd47acdf7cbda42cdb64c", algorithm=MD5
>> Content-Length: 0
>>
>> ----------------3:57:00.252----------------
>> SEND: enc=0 535 Bytes to 192.168.6.1:5061:UDP (ACK sip:613@192.168.6.1 
>> SIP/2.0) Interface Address=192.168.6.1
>> ACK sip:613@192.168.6.1 SIP/2.0
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To:  <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
> -addr=192.168.6.1;rport
>   
>> CSeq: 4711 ACK
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Contact: "1111" <sip:1111@192.168.6.1:5060>
>> User-Agent: OpenSIPStack-1.1.6-133
>> Max-Forwards: 10
>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
>> Content-Length: 0
>>
>> ----------------3:57:00.258----------------
>> SEND: enc=0 957 Bytes to 192.168.6.1:5061:UDP (INVITE 
>> sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1
>> INVITE sip:613@192.168.6.1 SIP/2.0
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To: sip:613@192.168.6.1
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
> -addr=192.168.6.1;rport
>   
>> CSeq: 4712 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Contact: "1111" <sip:1111@192.168.6.1:5060>
>> User-Agent: OpenSIPStack-1.1.6-133
>> Max-Forwards: 10
>> Proxy-Authorization: Digest username="1111", realm="192.168.6.1", 
>> nonce="92a98f3ef419bf98501149af46ee7604", uri="sip:613@192.168.6.1", 
>> response="bf9a133f7e481849ae92fac139ad5525", algorithm=MD5
>> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
>> Content-Type: application/sdp
>> Content-Length: 228
>>
>> v=0
>> o=- 1178857661 1178857661 IN IP4 192.168.6.1
>> s=OSS RTP Session
>> c=IN IP4 192.168.6.1
>> t=0 0
>> m=audio 5000 RTP/AVP 101 106 3
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtpmap:106 iLBC/8000
>> a=rtpmap:3 GSM/8000
>> ----------------3:57:00.276----------------
>> RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP 
>> (SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To: sip:613@192.168.6.1
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
> -addr=192.168.6.1
>   
>> CSeq: 4712 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Content-Length: 0
>>
>> ----------------3:57:01.535----------------
>> RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
>> (SIP/2.0 180 Ringing)
>> SIP/2.0 180 Ringing
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
> -addr=192.168.6.1;rport=5060;received=192.168.6.1
>   
>> CSeq: 4712 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Server: OpenSIPStack-1.1.6-133
>> Content-Length: 0
>>
>> ----------------3:57:01.652----------------
>> RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
>> (SIP/2.0 180 Ringing)
>> SIP/2.0 180 Ringing
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
> -addr=192.168.6.1;rport=5060;received=192.168.6.1
>   
>> CSeq: 4712 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Server: OpenSIPStack-1.1.6-133
>> Content-Length: 0
>>
>> ----------------3:57:05.840----------------
>> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
>> (SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From:  <sip:186...@i2...>;tag=1422096824326
>> To:  <sip:63.116.254.88:12000>
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643f0c3000002e90000009c;rport=6
> 2787;received=125.60.243.66
>   
>> CSeq: 67 OPTIONS
>> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
>> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
>> Content-Length: 0
>>
>> ----------------3:57:08.535----------------
>> RCV: enc=0 764 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
>> (SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
> -addr=192.168.6.1;rport=5060;received=192.168.6.1
>   
>> CSeq: 4712 INVITE
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Contact: <sip:192.168.6.1:5061>
>> Record-Route: <sip:192.168.6.1:5061;lr>
>> Server: OpenSIPStack-1.1.6-133
>> Content-Type: application/sdp
>> Content-Length: 241
>>
>> v=0
>> o=root 29469 29469 IN IP4 69.90.168.13
>> s=session
>> c=IN IP4 69.90.168.13
>> t=0 0
>> m=audio 13674 RTP/AVP 3 97 101
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:97 iLBC/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> ----------------3:57:08.782----------------
>> SEND: enc=0 439 Bytes to 192.168.6.1:5061:UDP (ACK sip:192.168.6.1:5061 
>> SIP/2.0) Interface Address=192.168.6.1
>> ACK sip:192.168.6.1:5061 SIP/2.0
>> From:  <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
>> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
>> Via: SIP/2.0/UDP 
>>
>>     
>
192.168.6.1:5060;branch=z9hG4bKdbb6f45ba7f818109c8e875f9c108cdc;uas-addr=192
> .168.6.1;rport
>   
>> CSeq: 4712 ACK
>> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
>> Contact: "1111" <sip:1111@192.168.6.1:5060>
>> Route: <sip:192.168.6.1:5061;lr>
>> Content-Length: 0
>>
>>
>>
>>
>> bondan wrote:
>>   
>>     
>>> Resend...
>>>
>>>
>>>
>>>             Send an Instant Message
>>> -----Original Message-----
>>> From: bondan [mailto:bo...@ci...] 
>>> Sent: Friday, May 04, 2007 10:30 AM
>>> To: 'jb...@so...';
>>>       
> 'ope...@li...'
>   
>>> Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan'
>>> Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call
>>>
>>>
>>> Dear Joegen,
>>>
>>> This attachment is the log file and capture screen of OSSPhone.
>>> Thank you for helping me...
>>>
>>> Best regards,
>>> Bondan Wisnuwardhana
>>>
>>>
>>>  
>>>             Send an Instant Message
>>> -----Original Message-----
>>> From: ope...@li...
>>> [mailto:ope...@li...] On Behalf Of
>>> Joegen E. Baclor
>>> Sent: Wednesday, May 02, 2007 8:45 PM
>>> To: ope...@li...
>>> Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang'
>>> Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call
>>>
>>> bondan wrote:
>>>   
>>>     
>>>       
>>>> Dear Joegen,
>>>>
>>>> I download OSSPhone source, and compile it into application. With the 
>>>> application, I could login into asterisk server and receive call too. 
>>>> But when I use the application to make a call, the error message 
>>>> "Status: 407 Proxy Authentication Required" is appear.
>>>>
>>>> I use codec setting :
>>>>
>>>> Low Bit Rate : iLBC-13k3
>>>>
>>>> High Bit Rate : G.711-ALaw-64k
>>>>
>>>> When I use OSSPhone (based on Windows installer) with the same 
>>>> setting, the OSSPhone can use to make a call.
>>>>
>>>> Would you give advice for me how to make call from OSSPhone source? 
>>>> Many thank for your helping,,,
>>>>
>>>> Best regards,
>>>>
>>>> Bondan Wisnuwardhana
>>>>
>>>>     
>>>>       
>>>>         
>>> Please send me the ATLSIP logs off list so I can see what went wrong.
>>>
>>>   
>>>     
>>>       
>>>>
------------------------------------------------------------------------
>>>>     
>>>>       
>>>>         
>>>
-------------------------------------------------------------------------
>>> This SF.net email is sponsored by DB2 Express
>>> Download DB2 Express C - the FREE version of DB2 express and take
>>> control of your XML. No limits. Just data. Click to get it now.
>>> http://sourceforge.net/powerbar/db2/
>>> _______________________________________________
>>> opensipstack-devel mailing list
>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>
>>> __________ NOD32 2235 (20070502) Information __________
>>>
>>> This message was checked by NOD32 antivirus system.
>>> http://www.eset.com
>>>
>>>   
>>> ------------------------------------------------------------------------
>>>
>>> ----------------54:03.175----------------
>>> *** LISTENER STARTED *** 127.0.0.1:5060
>>> ----------------54:03.250----------------
>>> *** LISTENER STARTED *** 192.168.1.70:5060 [*** DEFAULT LISTENER ***]
>>> ----------------54:04.340----------------
>>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER
sip:192.168.1.73
>>>       
> SIP/2.0) Interface Address=
>   
>>> REGISTER sip:192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>>> Expires: 3600
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> ----------------54:04.909----------------
>>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER
sip:192.168.1.73
>>>       
> SIP/2.0) Interface Address=
>   
>>> REGISTER sip:192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>>> Expires: 3600
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> ----------------54:05.899----------------
>>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER
sip:192.168.1.73
>>>       
> SIP/2.0) Interface Address=
>   
>>> REGISTER sip:192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>>> Expires: 3600
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> ----------------54:07.908----------------
>>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER
sip:192.168.1.73
>>>       
> SIP/2.0) Interface Address=
>   
>>> REGISTER sip:192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>>> Expires: 3600
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> ----------------54:09.453----------------
>>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
>>>       
> sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>   
>>> INVITE sip:102@192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>>> To: sip:102@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 4711 INVITE
>>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060>
>>> Max-Forwards: 70
>>> Content-Type: application/sdp
>>> Content-Length: 231
>>>
>>> v=0
>>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>>> s=OSS RTP Session
>>> c=IN IP4 192.168.1.70
>>> t=0 0
>>> m=audio 5000 RTP/AVP 101 106 8
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=rtpmap:106 iLBC/8000
>>> a=rtpmap:8 PCMA/8000
>>>
>>>
>>> ----------------54:10.007----------------
>>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
>>>       
> sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>   
>>> INVITE sip:102@192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>>> To: sip:102@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 4711 INVITE
>>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060>
>>> Max-Forwards: 70
>>> Content-Type: application/sdp
>>> Content-Length: 231
>>>
>>> v=0
>>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>>> s=OSS RTP Session
>>> c=IN IP4 192.168.1.70
>>> t=0 0
>>> m=audio 5000 RTP/AVP 101 106 8
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=rtpmap:106 iLBC/8000
>>> a=rtpmap:8 PCMA/8000
>>>
>>>
>>> ----------------54:11.004----------------
>>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
>>>       
> sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>   
>>> INVITE sip:102@192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>>> To: sip:102@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 4711 INVITE
>>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060>
>>> Max-Forwards: 70
>>> Content-Type: application/sdp
>>> Content-Length: 231
>>>
>>> v=0
>>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>>> s=OSS RTP Session
>>> c=IN IP4 192.168.1.70
>>> t=0 0
>>> m=audio 5000 RTP/AVP 101 106 8
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=rtpmap:106 iLBC/8000
>>> a=rtpmap:8 PCMA/8000
>>>
>>>
>>> ----------------54:11.924----------------
>>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER
sip:192.168.1.73
>>>       
> SIP/2.0) Interface Address=
>   
>>> REGISTER sip:192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>>> Expires: 3600
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> ----------------54:13.007----------------
>>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
>>>       
> sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>   
>>> INVITE sip:102@192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>>> To: sip:102@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 4711 INVITE
>>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060>
>>> Max-Forwards: 70
>>> Content-Type: application/sdp
>>> Content-Length: 231
>>>
>>> v=0
>>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>>> s=OSS RTP Session
>>> c=IN IP4 192.168.1.70
>>> t=0 0
>>> m=audio 5000 RTP/AVP 101 106 8
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=rtpmap:106 iLBC/8000
>>> a=rtpmap:8 PCMA/8000
>>>
>>>
>>> ----------------54:15.573----------------
>>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 100 Trying)
>   
>>> SIP/2.0 100 Trying
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.597----------------
>>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 401 Unauthorized)
>   
>>> SIP/2.0 401 Unauthorized
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73;tag=as087cafd3
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
>>>       
> nonce="5c7fc4df"
>   
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.620----------------
>>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 100 Trying)
>   
>>> SIP/2.0 100 Trying
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.661----------------
>>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 401 Unauthorized)
>   
>>> SIP/2.0 401 Unauthorized
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73;tag=as087cafd3
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
>>>       
> nonce="5c7fc4df"
>   
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.734----------------
>>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 100 Trying)
>   
>>> SIP/2.0 100 Trying
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.802----------------
>>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 401 Unauthorized)
>   
>>> SIP/2.0 401 Unauthorized
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73;tag=as087cafd3
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
>>>       
> nonce="5c7fc4df"
>   
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.827----------------
>>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 100 Trying)
>   
>>> SIP/2.0 100 Trying
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.896----------------
>>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 401 Unauthorized)
>   
>>> SIP/2.0 401 Unauthorized
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73;tag=as087cafd3
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
>>>       
> nonce="5c7fc4df"
>   
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.961----------------
>>> RCV: XOR=0 586 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 407 Proxy Authentication Required)
>   
>>> SIP/2.0 407 Proxy Authentication Required
>>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>>> To: sip:102@192.168.1.73;tag=as63b5a09f
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 4711 INVITE
>>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>>> Contact: <sip:102@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
>>>       
> nonce="49982f66"
>   
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:15.991----------------
>>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 100 Trying)
>   
>>> SIP/2.0 100 Trying
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:16.053----------------
>>> SEND: XOR=0 409 Bytes to 192.168.1.73:5060:UDP (ACK sip:102@192.168.1.73
>>>       
> SIP/2.0) Interface Address=192.168.1.70
>   
>>> ACK sip:102@192.168.1.73 SIP/2.0
>>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>>> To: sip:102@192.168.1.73;tag=as63b5a09f
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;rport
>   
>>> CSeq: 4711 ACK
>>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>>> Contact: "101" <sip:101@192.168.1.70:5060>
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> ----------------54:16.164----------------
>>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 401 Unauthorized)
>   
>>> SIP/2.0 401 Unauthorized
>>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>>> To: sip:101@192.168.1.73;tag=as087cafd3
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 1 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
>>>       
> nonce="5c7fc4df"
>   
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:16.321----------------
>>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 100 Trying)
>   
>>> SIP/2.0 100 Trying
>>> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204
>>> To: sip:101@192.168.1.73
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 2 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.73>
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>> ----------------54:16.353----------------
>>> RCV: XOR=0 564 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
>>>       
> (SIP/2.0 200 OK)
>   
>>> SIP/2.0 200 OK
>>> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204
>>> To: sip:101@192.168.1.73;tag=as087cafd3
>>> Via: SIP/2.0/UDP
>>>       
>
192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a
> ddr=192.168.1.73;received=192.168.1.70
>   
>>> CSeq: 2 REGISTER
>>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>>> Contact: <sip:101@192.168.1.70:5060;transport=udp>;expires=3600
>>> Date: Fri, 04 May 2007 02:42:07 GMT
>>> User-Agent: Asterisk PBX
>>> Expires: 3600
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>>> Content-Length: 0
>>>
>>>   
>>>
>>> ------------------------------------------------------------------------
>>>
>>>     
>>>       
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Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make call

From: bondan <bo...@ci...> - 2007-05-14 07:47:55

Dear Joegen,

I visit http://www.opensipstack.org/, and the current version of
OpenSIPStack is 1.1.5 version. Also, the current version of ATLSIP is 1.0.2
version. Would you inform to me where I have to download the newest version
of OpenSIPStack and ALTSIP?


Many thanks,
Bondan Wisnuwardhana


            Send an Instant Message

-----Original Message-----
From: Joegen E. Baclor [mailto:jb...@so...] 
Sent: Friday, May 11, 2007 11:38 AM
To: jb...@so...; ope...@li...
Cc: bondan
Subject: Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make call

Bondan,

One more thing.  Looking at your logs, it seems you are using an old 
copy of OpenSIPStack?   Please download the latest CVS head.  Current 
version is

OpenSIPStack-1.1.6-133


Joegen

Joegen E. Baclor wrote:
> Hi Bondan,
>
> I am not able to replicate your problem.  I just recompiled OSSPhone 
> from CVS and it's able to autheticate calls against OpenSBC terminating 
> to an asterisk box.  See logs below.
>
> BTW, is this modified code you are working with?
>
> ----------------3:44:55.989----------------
> *** LISTENER STARTED *** 127.0.0.1:5060
> ----------------3:44:56.022----------------
> *** LISTENER STARTED *** 192.168.0.161:5060 [*** DEFAULT LISTENER ***]
> ----------------3:44:56.028----------------
> *** LISTENER STARTED *** 192.168.6.1:5060
> ----------------3:44:56.031----------------
> *** LISTENER STARTED *** 192.168.15.1:5060
> ----------------3:46:44.147----------------
> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
> (SIP/2.0 200 OK)
> SIP/2.0 200 OK
> From:  <sip:186...@i2...>;tag=1359945326143
> To:  <sip:63.116.254.88:12000>
> Via: SIP/2.0/UDP 
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643ee5500004e3500000099;rport=6
2787;received=125.60.243.66
> CSeq: 64 OPTIONS
> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
> Content-Length: 0
>
> ----------------3:51:05.733----------------
> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
> (SIP/2.0 200 OK)
> SIP/2.0 200 OK
> From:  <sip:186...@i2...>;tag=1386096812568
> To:  <sip:63.116.254.88:12000>
> Via: SIP/2.0/UDP 
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643ef5b000027b10000009a;rport=6
2787;received=125.60.243.66
> CSeq: 65 OPTIONS
> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
> Content-Length: 0
>
> ----------------3:54:05.787----------------
> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
> (SIP/2.0 200 OK)
> SIP/2.0 200 OK
> From:  <sip:186...@i2...>;tag=1404096821517
> To:  <sip:63.116.254.88:12000>
> Via: SIP/2.0/UDP 
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643f00f000014b30000009b;rport=6
2787;received=125.60.243.66
> CSeq: 66 OPTIONS
> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
> Content-Length: 0
>
> ----------------3:56:49.262----------------
> SEND: enc=0 507 Bytes to 192.168.6.1:5061:UDP (REGISTER 
> sip:192.168.6.1:5061 SIP/2.0) Interface Address=
> REGISTER sip:192.168.6.1:5061 SIP/2.0
> From:  <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc
> To: sip:192.168.6.1
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3825;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas
-addr=192.168.6.1;rport
> CSeq: 1 REGISTER
> Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc
> Contact: <sip:192.168.6.1:5060;transport=udp>
> User-Agent: OpenSIPStack-1.1.6-133
> Expires: 3600
> Max-Forwards: 10
> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
> Content-Length: 0
>
> ----------------3:56:49.320----------------
> RCV: enc=0 424 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
> (SIP/2.0 200 OK)
> SIP/2.0 200 OK
> From:  <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc
> To: sip:192.168.6.1
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3185;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas
-addr=192.168.6.1;rport=5060;received=192.168.6.1
> CSeq: 1 REGISTER
> Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc
> Contact: <sip:192.168.6.1:5060;transport=udp>
> Server: OpenSIPStack-1.1.6-133
> Expires: 3600
> Content-Length: 0
>
> ----------------3:57:00.206----------------
> SEND: enc=0 762 Bytes to 192.168.6.1:5061:UDP (INVITE 
> sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1
> INVITE sip:613@192.168.6.1 SIP/2.0
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To: sip:613@192.168.6.1
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
-addr=192.168.6.1;rport
> CSeq: 4711 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Contact: "1111" <sip:1111@192.168.6.1:5060>
> User-Agent: OpenSIPStack-1.1.6-133
> Max-Forwards: 10
> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
> Content-Type: application/sdp
> Content-Length: 228
>
> v=0
> o=- 1178857661 1178857661 IN IP4 192.168.6.1
> s=OSS RTP Session
> c=IN IP4 192.168.6.1
> t=0 0
> m=audio 5000 RTP/AVP 101 106 3
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:106 iLBC/8000
> a=rtpmap:3 GSM/8000
> ----------------3:57:00.222----------------
> RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP 
> (SIP/2.0 100 Trying)
> SIP/2.0 100 Trying
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To: sip:613@192.168.6.1
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
-addr=192.168.6.1
> CSeq: 4711 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Content-Length: 0
>
> ----------------3:57:00.248----------------
> RCV: enc=0 557 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
> (SIP/2.0 407 Proxy Authentication Required)
> SIP/2.0 407 Proxy Authentication Required
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To:  <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
-addr=192.168.6.1;rport=5060;received=192.168.6.1
> CSeq: 4711 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Proxy-Authenticate: Digest realm=192.168.6.1, 
> nonce="92a98f3ef419bf98501149af46ee7604", 
> opaque="e430240b53dfd47acdf7cbda42cdb64c", algorithm=MD5
> Content-Length: 0
>
> ----------------3:57:00.252----------------
> SEND: enc=0 535 Bytes to 192.168.6.1:5061:UDP (ACK sip:613@192.168.6.1 
> SIP/2.0) Interface Address=192.168.6.1
> ACK sip:613@192.168.6.1 SIP/2.0
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To:  <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas
-addr=192.168.6.1;rport
> CSeq: 4711 ACK
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Contact: "1111" <sip:1111@192.168.6.1:5060>
> User-Agent: OpenSIPStack-1.1.6-133
> Max-Forwards: 10
> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
> Content-Length: 0
>
> ----------------3:57:00.258----------------
> SEND: enc=0 957 Bytes to 192.168.6.1:5061:UDP (INVITE 
> sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1
> INVITE sip:613@192.168.6.1 SIP/2.0
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To: sip:613@192.168.6.1
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
-addr=192.168.6.1;rport
> CSeq: 4712 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Contact: "1111" <sip:1111@192.168.6.1:5060>
> User-Agent: OpenSIPStack-1.1.6-133
> Max-Forwards: 10
> Proxy-Authorization: Digest username="1111", realm="192.168.6.1", 
> nonce="92a98f3ef419bf98501149af46ee7604", uri="sip:613@192.168.6.1", 
> response="bf9a133f7e481849ae92fac139ad5525", algorithm=MD5
> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS
> Content-Type: application/sdp
> Content-Length: 228
>
> v=0
> o=- 1178857661 1178857661 IN IP4 192.168.6.1
> s=OSS RTP Session
> c=IN IP4 192.168.6.1
> t=0 0
> m=audio 5000 RTP/AVP 101 106 3
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:106 iLBC/8000
> a=rtpmap:3 GSM/8000
> ----------------3:57:00.276----------------
> RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP 
> (SIP/2.0 100 Trying)
> SIP/2.0 100 Trying
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To: sip:613@192.168.6.1
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
-addr=192.168.6.1
> CSeq: 4712 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Content-Length: 0
>
> ----------------3:57:01.535----------------
> RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
> (SIP/2.0 180 Ringing)
> SIP/2.0 180 Ringing
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
-addr=192.168.6.1;rport=5060;received=192.168.6.1
> CSeq: 4712 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Server: OpenSIPStack-1.1.6-133
> Content-Length: 0
>
> ----------------3:57:01.652----------------
> RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
> (SIP/2.0 180 Ringing)
> SIP/2.0 180 Ringing
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
-addr=192.168.6.1;rport=5060;received=192.168.6.1
> CSeq: 4712 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Server: OpenSIPStack-1.1.6-133
> Content-Length: 0
>
> ----------------3:57:05.840----------------
> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP 
> (SIP/2.0 200 OK)
> SIP/2.0 200 OK
> From:  <sip:186...@i2...>;tag=1422096824326
> To:  <sip:63.116.254.88:12000>
> Via: SIP/2.0/UDP 
>
192.168.0.161;branch=z9hG4bKc0a800a1000000104643f0c3000002e90000009c;rport=6
2787;received=125.60.243.66
> CSeq: 67 OPTIONS
> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161
> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY
> Content-Length: 0
>
> ----------------3:57:08.535----------------
> RCV: enc=0 764 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP 
> (SIP/2.0 200 OK)
> SIP/2.0 200 OK
> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas
-addr=192.168.6.1;rport=5060;received=192.168.6.1
> CSeq: 4712 INVITE
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Contact: <sip:192.168.6.1:5061>
> Record-Route: <sip:192.168.6.1:5061;lr>
> Server: OpenSIPStack-1.1.6-133
> Content-Type: application/sdp
> Content-Length: 241
>
> v=0
> o=root 29469 29469 IN IP4 69.90.168.13
> s=session
> c=IN IP4 69.90.168.13
> t=0 0
> m=audio 13674 RTP/AVP 3 97 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> ----------------3:57:08.782----------------
> SEND: enc=0 439 Bytes to 192.168.6.1:5061:UDP (ACK sip:192.168.6.1:5061 
> SIP/2.0) Interface Address=192.168.6.1
> ACK sip:192.168.6.1:5061 SIP/2.0
> From:  <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc
> To:  <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df
> Via: SIP/2.0/UDP 
>
192.168.6.1:5060;branch=z9hG4bKdbb6f45ba7f818109c8e875f9c108cdc;uas-addr=192
.168.6.1;rport
> CSeq: 4712 ACK
> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc
> Contact: "1111" <sip:1111@192.168.6.1:5060>
> Route: <sip:192.168.6.1:5061;lr>
> Content-Length: 0
>
>
>
>
> bondan wrote:
>   
>> Resend...
>>
>>
>>
>>             Send an Instant Message
>> -----Original Message-----
>> From: bondan [mailto:bo...@ci...] 
>> Sent: Friday, May 04, 2007 10:30 AM
>> To: 'jb...@so...';
'ope...@li...'
>> Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan'
>> Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call
>>
>>
>> Dear Joegen,
>>
>> This attachment is the log file and capture screen of OSSPhone.
>> Thank you for helping me...
>>
>> Best regards,
>> Bondan Wisnuwardhana
>>
>>
>>  
>>             Send an Instant Message
>> -----Original Message-----
>> From: ope...@li...
>> [mailto:ope...@li...] On Behalf Of
>> Joegen E. Baclor
>> Sent: Wednesday, May 02, 2007 8:45 PM
>> To: ope...@li...
>> Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang'
>> Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call
>>
>> bondan wrote:
>>   
>>     
>>> Dear Joegen,
>>>
>>> I download OSSPhone source, and compile it into application. With the 
>>> application, I could login into asterisk server and receive call too. 
>>> But when I use the application to make a call, the error message 
>>> "Status: 407 Proxy Authentication Required" is appear.
>>>
>>> I use codec setting :
>>>
>>> Low Bit Rate : iLBC-13k3
>>>
>>> High Bit Rate : G.711-ALaw-64k
>>>
>>> When I use OSSPhone (based on Windows installer) with the same 
>>> setting, the OSSPhone can use to make a call.
>>>
>>> Would you give advice for me how to make call from OSSPhone source? 
>>> Many thank for your helping,,,
>>>
>>> Best regards,
>>>
>>> Bondan Wisnuwardhana
>>>
>>>     
>>>       
>> Please send me the ATLSIP logs off list so I can see what went wrong.
>>
>>   
>>     
>>> ------------------------------------------------------------------------
>>>     
>>>       
>> -------------------------------------------------------------------------
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>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>> __________ NOD32 2235 (20070502) Information __________
>>
>> This message was checked by NOD32 antivirus system.
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>>
>>   
>> ------------------------------------------------------------------------
>>
>> ----------------54:03.175----------------
>> *** LISTENER STARTED *** 127.0.0.1:5060
>> ----------------54:03.250----------------
>> *** LISTENER STARTED *** 192.168.1.70:5060 [*** DEFAULT LISTENER ***]
>> ----------------54:04.340----------------
>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73
SIP/2.0) Interface Address=
>> REGISTER sip:192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>> Expires: 3600
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> ----------------54:04.909----------------
>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73
SIP/2.0) Interface Address=
>> REGISTER sip:192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>> Expires: 3600
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> ----------------54:05.899----------------
>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73
SIP/2.0) Interface Address=
>> REGISTER sip:192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>> Expires: 3600
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> ----------------54:07.908----------------
>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73
SIP/2.0) Interface Address=
>> REGISTER sip:192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>> Expires: 3600
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> ----------------54:09.453----------------
>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>> INVITE sip:102@192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>> To: sip:102@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 4711 INVITE
>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060>
>> Max-Forwards: 70
>> Content-Type: application/sdp
>> Content-Length: 231
>>
>> v=0
>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>> s=OSS RTP Session
>> c=IN IP4 192.168.1.70
>> t=0 0
>> m=audio 5000 RTP/AVP 101 106 8
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtpmap:106 iLBC/8000
>> a=rtpmap:8 PCMA/8000
>>
>>
>> ----------------54:10.007----------------
>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>> INVITE sip:102@192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>> To: sip:102@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 4711 INVITE
>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060>
>> Max-Forwards: 70
>> Content-Type: application/sdp
>> Content-Length: 231
>>
>> v=0
>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>> s=OSS RTP Session
>> c=IN IP4 192.168.1.70
>> t=0 0
>> m=audio 5000 RTP/AVP 101 106 8
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtpmap:106 iLBC/8000
>> a=rtpmap:8 PCMA/8000
>>
>>
>> ----------------54:11.004----------------
>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>> INVITE sip:102@192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>> To: sip:102@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 4711 INVITE
>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060>
>> Max-Forwards: 70
>> Content-Type: application/sdp
>> Content-Length: 231
>>
>> v=0
>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>> s=OSS RTP Session
>> c=IN IP4 192.168.1.70
>> t=0 0
>> m=audio 5000 RTP/AVP 101 106 8
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtpmap:106 iLBC/8000
>> a=rtpmap:8 PCMA/8000
>>
>>
>> ----------------54:11.924----------------
>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73
SIP/2.0) Interface Address=
>> REGISTER sip:192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp>
>> Expires: 3600
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> ----------------54:13.007----------------
>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE
sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70
>> INVITE sip:102@192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>> To: sip:102@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 4711 INVITE
>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060>
>> Max-Forwards: 70
>> Content-Type: application/sdp
>> Content-Length: 231
>>
>> v=0
>> o=- 1178246522 1178246522 IN IP4 192.168.1.70
>> s=OSS RTP Session
>> c=IN IP4 192.168.1.70
>> t=0 0
>> m=audio 5000 RTP/AVP 101 106 8
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtpmap:106 iLBC/8000
>> a=rtpmap:8 PCMA/8000
>>
>>
>> ----------------54:15.573----------------
>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.597----------------
>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 401 Unauthorized)
>> SIP/2.0 401 Unauthorized
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73;tag=as087cafd3
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5c7fc4df"
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.620----------------
>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.661----------------
>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 401 Unauthorized)
>> SIP/2.0 401 Unauthorized
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73;tag=as087cafd3
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5c7fc4df"
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.734----------------
>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.802----------------
>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 401 Unauthorized)
>> SIP/2.0 401 Unauthorized
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73;tag=as087cafd3
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5c7fc4df"
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.827----------------
>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.896----------------
>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 401 Unauthorized)
>> SIP/2.0 401 Unauthorized
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73;tag=as087cafd3
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5c7fc4df"
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.961----------------
>> RCV: XOR=0 586 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 407 Proxy Authentication Required)
>> SIP/2.0 407 Proxy Authentication Required
>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>> To: sip:102@192.168.1.73;tag=as63b5a09f
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 4711 INVITE
>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>> Contact: <sip:102@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="49982f66"
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:15.991----------------
>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:16.053----------------
>> SEND: XOR=0 409 Bytes to 192.168.1.73:5060:UDP (ACK sip:102@192.168.1.73
SIP/2.0) Interface Address=192.168.1.70
>> ACK sip:102@192.168.1.73 SIP/2.0
>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204
>> To: sip:102@192.168.1.73;tag=as63b5a09f
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a
ddr=192.168.1.73;rport
>> CSeq: 4711 ACK
>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204
>> Contact: "101" <sip:101@192.168.1.70:5060>
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> ----------------54:16.164----------------
>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 401 Unauthorized)
>> SIP/2.0 401 Unauthorized
>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204
>> To: sip:101@192.168.1.73;tag=as087cafd3
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 1 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5c7fc4df"
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:16.321----------------
>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 100 Trying)
>> SIP/2.0 100 Trying
>> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204
>> To: sip:101@192.168.1.73
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 2 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.73>
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>> ----------------54:16.353----------------
>> RCV: XOR=0 564 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP
(SIP/2.0 200 OK)
>> SIP/2.0 200 OK
>> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204
>> To: sip:101@192.168.1.73;tag=as087cafd3
>> Via: SIP/2.0/UDP
192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a
ddr=192.168.1.73;received=192.168.1.70
>> CSeq: 2 REGISTER
>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204
>> Contact: <sip:101@192.168.1.70:5060;transport=udp>;expires=3600
>> Date: Fri, 04 May 2007 02:42:07 GMT
>> User-Agent: Asterisk PBX
>> Expires: 3600
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Content-Length: 0
>>
>>   
>>
>> ------------------------------------------------------------------------
>>
>>     
>
>
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>   


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