From resende.bruno at gmail.com  Thu Jul  2 14:29:29 2009
From: resende.bruno at gmail.com (Bruno Resende)
Date: Thu, 2 Jul 2009 14:29:29 +0200
Subject: [Osip] Header NULL in a 200_OK response
Message-ID: <3f2f32b30907020529g217a5204v37697671404cae9e@mail.gmail.com>

Hi,

I need to parser the Content-Type header inside of a 200_OK like response
for a INVITE.
The INVITE works fine and the 200_OK response with a Content-Type header
like I need, but the analyse of this header don't work. They show me a
"NULL" response for this header inside 200_OK.

This is code for the analyse when it listen a 200_OK:


case EXOSIP_CALL_ANSWERED:

        if((event_at->response)->status_code == 200)
            {

           osip_content_type_t *content_type;
           char* content;
           content_type = osip_message_get_content_type(event_at->response);
           osip_content_type_to_str ( content_type, &content );

          printf("\n\n%s\n\n",content);
           }
        break;

Did I make a mistake i this code?

From alain.ronce at nodcom.fr  Thu Jul  2 14:43:05 2009
From: alain.ronce at nodcom.fr (Alain Ronce)
Date: Thu, 2 Jul 2009 14:43:05 +0200
Subject: [Osip] Header NULL in a 200_OK response
References: <3f2f32b30907020529g217a5204v37697671404cae9e@mail.gmail.com>
Message-ID: <7F1A181F1A0444D38AA30547DAC7B630@RONCE3BIS>

Hello,
I think the best wait should be :

osip_body_t *body;
osip_message_get_body (pEvent->response, 0, &body);
if (body)
  {
     .... // use body->body for the text
     .... // and body->length
  }

Best Regards,
AR


----- Original Message ----- 
From: "Bruno Resende" <resende.bruno at gmail.com>
To: <osip at atosc.org>
Sent: Thursday, July 02, 2009 2:29 PM
Subject: [Osip] Header NULL in a 200_OK response


> Hi,
>
> I need to parser the Content-Type header inside of a 200_OK like response
> for a INVITE.
> The INVITE works fine and the 200_OK response with a Content-Type header
> like I need, but the analyse of this header don't work. They show me a
> "NULL" response for this header inside 200_OK.
>
> This is code for the analyse when it listen a 200_OK:
>
>
> case EXOSIP_CALL_ANSWERED:
>
>        if((event_at->response)->status_code == 200)
>            {
>
>           osip_content_type_t *content_type;
>           char* content;
>           content_type = 
> osip_message_get_content_type(event_at->response);
>           osip_content_type_to_str ( content_type, &content );
>
>          printf("\n\n%s\n\n",content);
>           }
>        break;
>
> Did I make a mistake i this code?
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip 


From alain.ronce at nodcom.fr  Thu Jul  2 15:02:06 2009
From: alain.ronce at nodcom.fr (Alain Ronce)
Date: Thu, 2 Jul 2009 15:02:06 +0200
Subject: [Osip] How to pass optimization flag to the compiler
Message-ID: <C569A3EB416E4EB6AB6C42BDDA556C5A@RONCE3BIS>

Hi,
Is there an easy way to pass -Os option to the compiler when using 
./configure line options. I am not confortable with automake, autoconf, 
etc... so I don't know what to do...
Best Regards
AR


From jack at atosc.org  Thu Jul  2 15:06:45 2009
From: jack at atosc.org (Aymeric Moizard)
Date: Thu, 2 Jul 2009 15:06:45 +0200 (CEST)
Subject: [Osip] Header NULL in a 200_OK response
In-Reply-To: <3f2f32b30907020529g217a5204v37697671404cae9e@mail.gmail.com>
References: <3f2f32b30907020529g217a5204v37697671404cae9e@mail.gmail.com>
Message-ID: <Pine.LNX.4.63.0907021505320.2161@sip2>



On Thu, 2 Jul 2009, Bruno Resende wrote:

> Hi,
>
> I need to parser the Content-Type header inside of a 200_OK like response
> for a INVITE.
> The INVITE works fine and the 200_OK response with a Content-Type header
> like I need, but the analyse of this header don't work. They show me a
> "NULL" response for this header inside 200_OK.
>
> This is code for the analyse when it listen a 200_OK:
>
>
> case EXOSIP_CALL_ANSWERED:
>
>        if((event_at->response)->status_code == 200)
>            {
>
>           osip_content_type_t *content_type;
>           char* content;
>           content_type = osip_message_get_content_type(event_at->response);
>           osip_content_type_to_str ( content_type, &content );
>
>          printf("\n\n%s\n\n",content);
>           }
>        break;
>
> Did I make a mistake i this code?

At least, the code looks fine. (and you have to osip_free(content);
if it's not NULL at the end)

A debugguer will certainly help you!

tks,
Aymeric MOIZARD / ANTISIP
amsip - http://www.antisip.com
osip2 - http://www.osip.org
eXosip2 - http://savannah.nongnu.org/projects/exosip/


> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>

From jack at atosc.org  Thu Jul  2 15:23:11 2009
From: jack at atosc.org (Aymeric Moizard)
Date: Thu, 2 Jul 2009 15:23:11 +0200 (CEST)
Subject: [Osip] How to pass optimization flag to the compiler
In-Reply-To: <C569A3EB416E4EB6AB6C42BDDA556C5A@RONCE3BIS>
References: <C569A3EB416E4EB6AB6C42BDDA556C5A@RONCE3BIS>
Message-ID: <Pine.LNX.4.63.0907021522270.2161@sip2>



On Thu, 2 Jul 2009, Alain Ronce wrote:

> Hi,
> Is there an easy way to pass -Os option to the compiler when using
> ./configure line options. I am not confortable with automake, autoconf,
> etc... so I don't know what to do...

CFLAGS="-0s" ./configure

should work

Aymeric

> Best Regards
> AR
>
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>

From shilpa.walvekar at coreobjects.com  Tue Jul 14 11:16:40 2009
From: shilpa.walvekar at coreobjects.com (Shilpa Kedar Walvekar)
Date: Tue, 14 Jul 2009 14:46:40 +0530
Subject: [Osip] Help required on using oSip
Message-ID: <33A0EEB8D50198449794A1AB9DEB1E018662CFAC5B@copuex01.coreobjects.com>

Hi,

Just now I downloaded oSip and want to use the library to establish a voice call and receive an incoming call.

How to achieve this using oSip?

I tried to search a lot but couldn't find some sample application regarding this.

Thanks,
--Shilpa

From vikasraigupta at gmail.com  Tue Jul 14 15:48:53 2009
From: vikasraigupta at gmail.com (VG)
Date: Tue, 14 Jul 2009 19:18:53 +0530
Subject: [Osip] how to add record route in refer message
Message-ID: <c4fdd6d00907140648x53c3dbe1q6277e10a52cc839a@mail.gmail.com>

hello,

I like to add record route in refer generated by osip.

string like softswitch at 192.168.4.52

what api or method to use for that.

Regards,
Vikas

From jack at atosc.org  Tue Jul 14 18:31:06 2009
From: jack at atosc.org (Aymeric Moizard)
Date: Tue, 14 Jul 2009 18:31:06 +0200 (CEST)
Subject: [Osip] how to add record route in refer message
In-Reply-To: <c4fdd6d00907140648x53c3dbe1q6277e10a52cc839a@mail.gmail.com>
References: <c4fdd6d00907140648x53c3dbe1q6277e10a52cc839a@mail.gmail.com>
Message-ID: <Pine.LNX.4.63.0907141829350.28116@sip2>



On Tue, 14 Jul 2009, VG wrote:

> hello,
>
> I like to add record route in refer generated by osip.
>
> string like softswitch at 192.168.4.52

You can use the following API:
   osip_message_set_record_route(..)

Aymeric

> what api or method to use for that.
>
> Regards,
> Vikas
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>

From bon-ami at 163.com  Wed Jul 15 04:18:40 2009
From: bon-ami at 163.com (A117)
Date: Wed, 15 Jul 2009 10:18:40 +0800
Subject: [Osip] Help required on using oSip
In-Reply-To: <mailman.1.1247565602.10361.osip@atosc.org>
References: <mailman.1.1247565602.10361.osip@atosc.org>
Message-ID: <000001ca04f2$91bdeb20$b539c160$@com>

Shilpa,
	Linphone is a great project to begin with, which uses eXosip2 (,
which is already enough to learn how to use oSIP2, actually) to interact
with oSIP2.

	yours



From vikasraigupta at gmail.com  Wed Jul 15 15:28:20 2009
From: vikasraigupta at gmail.com (VG)
Date: Wed, 15 Jul 2009 18:58:20 +0530
Subject: [Osip] generating invite after receiving 302
Message-ID: <c4fdd6d00907150628p11e764d6q7d33a29135f117f8@mail.gmail.com>

hi,

I like to know where from this invite is generated on stack after receiving 302.

The scenario is like:

A -- B
B sends 302 and
automatically invite is generated for the specific contact.

As i am using osip in switch then it creates problem.
I like to stop this please specify me the flow of this message so that
i can comment it and make it according to me.

Regards,
Vikas

From jack at atosc.org  Wed Jul 15 16:17:15 2009
From: jack at atosc.org (Aymeric Moizard)
Date: Wed, 15 Jul 2009 16:17:15 +0200 (CEST)
Subject: [Osip] generating invite after receiving 302
In-Reply-To: <c4fdd6d00907150628p11e764d6q7d33a29135f117f8@mail.gmail.com>
References: <c4fdd6d00907150628p11e764d6q7d33a29135f117f8@mail.gmail.com>
Message-ID: <Pine.LNX.4.63.0907151615570.28116@sip2>



On Wed, 15 Jul 2009, VG wrote:

> hi,
>
> I like to know where from this invite is generated on stack after receiving 302.
>
> The scenario is like:
>
> A -- B
> B sends 302 and
> automatically invite is generated for the specific contact.
>
> As i am using osip in switch then it creates problem.
> I like to stop this please specify me the flow of this message so that
> i can comment it and make it according to me.

I don't understand your problem: if you have automatic INVITE sent
after a 302, I guess you are using eXosip2? I this case, automatic
INVITE is most probably sent by "eXosip_automatic_action" which
your application is probably calling.

tks,
Aymeric MOIZARD / ANTISIP
amsip - http://www.antisip.com
osip2 - http://www.osip.org
eXosip2 - http://savannah.nongnu.org/projects/exosip/


> Regards,
> Vikas
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>

From gemi.tsai at gmail.com  Thu Jul 16 10:22:19 2009
From: gemi.tsai at gmail.com (=?Big5?B?R2VtaSBUc2FpIL2ysOqu9Q==?=)
Date: Thu, 16 Jul 2009 16:22:19 +0800
Subject: [Osip]  the eXosip_masquerade_contact issue
Message-ID: <825f198c0907160122x6b9a4823j9aa979b9c1f490c7@mail.gmail.com>

Hi all and Aymeric,

I tried to use the eXosip_masquerade_contact function, and input the
public address and port. (got by stun server)

But I found that eXosip only modify the Contact header of SIP Messages
to public address and port.
Why the Via header doesn't be modified?

The packet is attached.
Thanks very much!

Gemi

From jack at atosc.org  Thu Jul 16 12:23:17 2009
From: jack at atosc.org (Aymeric Moizard)
Date: Thu, 16 Jul 2009 12:23:17 +0200 (CEST)
Subject: [Osip] the eXosip_masquerade_contact issue
In-Reply-To: <825f198c0907160122x6b9a4823j9aa979b9c1f490c7@mail.gmail.com>
References: <825f198c0907160122x6b9a4823j9aa979b9c1f490c7@mail.gmail.com>
Message-ID: <Pine.LNX.4.63.0907161217510.28116@sip2>



On Thu, 16 Jul 2009, Gemi Tsai ?????? wrote:

> Hi all and Aymeric,
>
> I tried to use the eXosip_masquerade_contact function, and input the
> public address and port. (got by stun server)
>
> But I found that eXosip only modify the Contact header of SIP Messages
> to public address and port.
> Why the Via header doesn't be modified?

There is no reason to modify the Via IP and port. They are not
used for SIP routing. ("received" parameter is used by rfc3261 and "rport" 
extension is mandatory for deployment behind NAT)

Aymeric

> The packet is attached.
> Thanks very much!
>
> Gemi
>

From dave.harley at lincor.com  Thu Jul 16 13:48:31 2009
From: dave.harley at lincor.com (Dave Harley)
Date: Thu, 16 Jul 2009 12:48:31 +0100
Subject: [Osip] IS there support for re-invite with no sdp within
 established dialog with ref to rfc 3261 para 14.1. ?
Message-ID: <1247744911.2632.544.camel@dave-ws.cork.lincor.com>

Hi There, 

I have an interop issue with a mitel sip enabled pabx and a  Aastra
A5000 iPBX.  Both send a re-invite with no SDP as part of an on hold
( and also transfer) request within an established dialog.   

Having had a quick look at RFC 3261 14.1 and 13.2.1 ( referenced in
14.1 ), it seems that I should be sending my SDP back in the 200OK to
the received invite.  Its not and after testing with an xlite client on
a windows box, look like its stack is. 

I'm using a really old version of libOsip so I don't expect any help
with getting it working.  Really what I'm hoping to find out is, does
the current top of the tree support re-invites with no SDP bodies within
established dialogs.  

I'd go test myself first but getting access to the required pabx
remotely requires sys admins to break firewalling policies and generally
do things they don't want to do so I'm trying to avoid this.   Had a
look at the code already but can't see this support in there so I
thought I'd ask to be sure.

Thanks for your time, 

Dave 

From jack at atosc.org  Thu Jul 16 14:42:34 2009
From: jack at atosc.org (Aymeric Moizard)
Date: Thu, 16 Jul 2009 14:42:34 +0200 (CEST)
Subject: [Osip] IS there support for re-invite with no sdp within
 established dialog with ref to rfc 3261 para 14.1. ?
In-Reply-To: <1247744911.2632.544.camel@dave-ws.cork.lincor.com>
References: <1247744911.2632.544.camel@dave-ws.cork.lincor.com>
Message-ID: <Pine.LNX.4.63.0907161441180.28116@sip2>



On Thu, 16 Jul 2009, Dave Harley wrote:

> Hi There,
>
> I have an interop issue with a mitel sip enabled pabx and a  Aastra
> A5000 iPBX.  Both send a re-invite with no SDP as part of an on hold
> ( and also transfer) request within an established dialog.
>
> Having had a quick look at RFC 3261 14.1 and 13.2.1 ( referenced in
> 14.1 ), it seems that I should be sending my SDP back in the 200OK to
> the received invite.  Its not and after testing with an xlite client on
> a windows box, look like its stack is.
>
> I'm using a really old version of libOsip so I don't expect any help
> with getting it working.  Really what I'm hoping to find out is, does
> the current top of the tree support re-invites with no SDP bodies within
> established dialogs.
>
> I'd go test myself first but getting access to the required pabx
> remotely requires sys admins to break firewalling policies and generally
> do things they don't want to do so I'm trying to avoid this.   Had a
> look at the code already but can't see this support in there so I
> thought I'd ask to be sure.

osip (and even eXosip2) does not handle SDP so this issue is to be
handled in your application anyway! whatever version your are using.

Regards,
Aymeric MOIZARD / ANTISIP
amsip - http://www.antisip.com
osip2 - http://www.osip.org
eXosip2 - http://savannah.nongnu.org/projects/exosip/


> Thanks for your time,
>
> Dave
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>

From gemi.tsai at gmail.com  Fri Jul 17 05:11:22 2009
From: gemi.tsai at gmail.com (=?Big5?B?R2VtaSBUc2FpIL2ysOqu9Q==?=)
Date: Fri, 17 Jul 2009 11:11:22 +0800
Subject: [Osip] the eXosip_masquerade_contact issue
In-Reply-To: <Pine.LNX.4.63.0907161217510.28116@sip2>
References: <825f198c0907160122x6b9a4823j9aa979b9c1f490c7@mail.gmail.com>
	<Pine.LNX.4.63.0907161217510.28116@sip2>
Message-ID: <825f198c0907162011n2346a046x2ef4c781a41de5b7@mail.gmail.com>

Hi Aymeric and all,

I tested with OpenIMSCore, and found this Via issue. (maybe it caused
by OpenIMSCore)

Please see attached files, when I didn't modify the Via header field
and originated a call, the UA received the 403 error response.

When I originate a call, I found the log of OpenIMSCore, the
originating contact that it got is private_addr:public_port
(192.168.180.63:3410).
So the OpenIMSCore can't look for this account, and return 403.

After you replied, I think that it is maybe an OpenIMSCore issue, do you agree?

Thanks for your response.

Gemi


2009/7/16 Aymeric Moizard <jack at atosc.org>:
>
>
> On Thu, 16 Jul 2009, Gemi Tsai ?????? wrote:
>
>> Hi all and Aymeric,
>>
>> I tried to use the eXosip_masquerade_contact function, and input the
>> public address and port. (got by stun server)
>>
>> But I found that eXosip only modify the Contact header of SIP Messages
>> to public address and port.
>> Why the Via header doesn't be modified?
>
> There is no reason to modify the Via IP and port. They are not
> used for SIP routing. ("received" parameter is used by rfc3261 and "rport"
> extension is mandatory for deployment behind NAT)
>
> Aymeric
>
>> The packet is attached.
>> Thanks very much!
>>
>> Gemi
>
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>
>

From gemi.tsai at gmail.com  Fri Jul 17 07:57:30 2009
From: gemi.tsai at gmail.com (=?Big5?B?R2VtaSBUc2FpIL2ysOqu9Q==?=)
Date: Fri, 17 Jul 2009 13:57:30 +0800
Subject: [Osip] the eXosip_masquerade_contact issue
In-Reply-To: <825f198c0907162011n2346a046x2ef4c781a41de5b7@mail.gmail.com>
References: <825f198c0907160122x6b9a4823j9aa979b9c1f490c7@mail.gmail.com>
	<Pine.LNX.4.63.0907161217510.28116@sip2>
	<825f198c0907162011n2346a046x2ef4c781a41de5b7@mail.gmail.com>
Message-ID: <825f198c0907162257v2611bcd5q5da11b376f1c24cc@mail.gmail.com>

Hi Aymeric and all,

I'm sorry.
I found that is not the OpenIMScore issue, too.
Because the P-CSCF acts the outbound proxy, so we didn't need to use
the eXosip_masquerade_contact function in the IMS.

It means that we don't do anything for SIP Signaling, and UAs can
traverse the NAT.
But, the SDP is must handled by ourselves.

Thanks.
Gemi


2009/7/17 Gemi Tsai ??? <gemi.tsai at gmail.com>:
> Hi Aymeric and all,
>
> I tested with OpenIMSCore, and found this Via issue. (maybe it caused
> by OpenIMSCore)
>
> Please see attached files, when I didn't modify the Via header field
> and originated a call, the UA received the 403 error response.
>
> When I originate a call, I found the log of OpenIMSCore, the
> originating contact that it got is private_addr:public_port
> (192.168.180.63:3410).
> So the OpenIMSCore can't look for this account, and return 403.
>
> After you replied, I think that it is maybe an OpenIMSCore issue, do you agree?
>
> Thanks for your response.
>
> Gemi
>
>
> 2009/7/16 Aymeric Moizard <jack at atosc.org>:
>>
>>
>> On Thu, 16 Jul 2009, Gemi Tsai ?????? wrote:
>>
>>> Hi all and Aymeric,
>>>
>>> I tried to use the eXosip_masquerade_contact function, and input the
>>> public address and port. (got by stun server)
>>>
>>> But I found that eXosip only modify the Contact header of SIP Messages
>>> to public address and port.
>>> Why the Via header doesn't be modified?
>>
>> There is no reason to modify the Via IP and port. They are not
>> used for SIP routing. ("received" parameter is used by rfc3261 and "rport"
>> extension is mandatory for deployment behind NAT)
>>
>> Aymeric
>>
>>> The packet is attached.
>>> Thanks very much!
>>>
>>> Gemi
>>
>> _______________________________________________
>> Osip mailing list
>> Osip at atosc.org
>> http://www.atosc.org/mailinglist/listinfo/osip
>>
>>
>

From vikasraigupta at gmail.com  Sat Jul 18 07:54:17 2009
From: vikasraigupta at gmail.com (VG)
Date: Sat, 18 Jul 2009 11:24:17 +0530
Subject: [Osip] generating invite after receiving 302
In-Reply-To: <Pine.LNX.4.63.0907151615570.28116@sip2>
References: <c4fdd6d00907150628p11e764d6q7d33a29135f117f8@mail.gmail.com>
	<Pine.LNX.4.63.0907151615570.28116@sip2>
Message-ID: <c4fdd6d00907172254k477798f3h8d4e6f603d0093b1@mail.gmail.com>

Thanks for your reply yes I am using exoip and using the function
eXosip_automatic_action

I like to use eXosip_automatic_action but  when my application recieve
302 then eXosip_automatic_action automatically generates INVITE for
that .

I handled this situation by using:
if(event->type != EXOSIP_CALL_REDIRECTED)

	eXosip_automatic_action");
			
}



But still getting INVITE how could I stop this only for 302. No
decision taken by Application.
Also specify any open source code that is based on exosip for code
refrence regarding subscribe/notify .



Regards,
VG
On Wed, Jul 15, 2009 at 7:47 PM, Aymeric Moizard<jack at atosc.org> wrote:
>
>
> On Wed, 15 Jul 2009, VG wrote:
>
>> hi,
>>
>> I like to know where from this invite is generated on stack after
>> receiving 302.
>>
>> The scenario is like:
>>
>> A -- B
>> B sends 302 and
>> automatically invite is generated for the specific contact.
>>
>> As i am using osip in switch then it creates problem.
>> I like to stop this please specify me the flow of this message so that
>> i can comment it and make it according to me.
>
> I don't understand your problem: if you have automatic INVITE sent
> after a 302, I guess you are using eXosip2? I this case, automatic
> INVITE is most probably sent by "eXosip_automatic_action" which
> your application is probably calling.
>
> tks,
> Aymeric MOIZARD / ANTISIP
> amsip - http://www.antisip.com
> osip2 - http://www.osip.org
> eXosip2 - http://savannah.nongnu.org/projects/exosip/
>
>
>> Regards,
>> Vikas
>> _______________________________________________
>> Osip mailing list
>> Osip at atosc.org
>> http://www.atosc.org/mailinglist/listinfo/osip
>>
>

From akhil4es at gmail.com  Tue Jul 21 12:52:06 2009
From: akhil4es at gmail.com (mohammmed akhil)
Date: Tue, 21 Jul 2009 16:22:06 +0530
Subject: [Osip] hi
Message-ID: <11bec9d20907210352h2c0d1275ube862381ce6704c9@mail.gmail.com>

hi all,
I have installed linphone on my ubuntu system.From where can i get my sip
address ? Do i need  to subscribe some where to have atleast the voice call
being activated.

thanks and regards
mohammed akhil

From vikasraigupta at gmail.com  Thu Jul 23 14:39:47 2009
From: vikasraigupta at gmail.com (VG)
Date: Thu, 23 Jul 2009 18:09:47 +0530
Subject: [Osip] exosip on TCP and UDP Both
Message-ID: <c4fdd6d00907230539r54e00518taa52000f9974888e@mail.gmail.com>

Hi,

I am building an exosip based switching application. I am in a fix.
I like to communicate with Microsoft exchange for voice mail.
This Application only works on TCP.
As I am using exosip so I configured this Protocol as UDP.
Now All the communication that is done on UDP is going well but when
it comes for Microsoft Exchange Communication then got ICMP packet
from MSExchange.
Any body have implemented this kind of situation using exosip.

Is it possible to configure exosip on both TCP and UDP and selective
method to send request using TCP or UDP for further Communication.

What I want exactly is Exosip sit on Both TCP and UDP.
If i want to communicate with Phone then communication should be on UDP.
If the communication is with MS Exchange then it should be on TCP.

Any help is highly appreciated.

Regards,
VG

From benjaminmodra at europe.com  Fri Jul 24 03:33:14 2009
From: benjaminmodra at europe.com (Benjamin Modra)
Date: Thu, 23 Jul 2009 20:33:14 -0500
Subject: [Osip] can't send a register packet to ser
Message-ID: <20090724013314.CF8DC606865@ws1-4.us4.outblaze.com>

hi all ...

I am using libeXosip and libosip for my application... i can't send
aregister packet to ser .


How can I overcome this problems?

?

Regards,

Benjamin

-- 
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From uniojnqoifazy at gmail.com  Fri Jul 24 04:50:25 2009
From: uniojnqoifazy at gmail.com (jay)
Date: Fri, 24 Jul 2009 10:50:25 +0800
Subject: [Osip] sip_reg compile error
Message-ID: <006b01ca0c09$80e12c40$82a384c0$@com>

Hello 

I have download libosip2-3.3.0 , but when I to compile sip_reg.c using "gcc
sip_reg.c -o sip_reg -leXosip -L/usr/local/lib -I../../include/
-DHAVE_STRUCT_TIMEVAL -losipparser2"

,I got some error msg :

sip_reg.c:55:28: error: eXosip2/eXosip.h: No such file or directory

sip_reg.c: In function 'main':

sip_reg.c:336: error: 'osip_message_t' undeclared (first use in this
function)

sip_reg.c:336: error: (Each undeclared identifier is reported only once

sip_reg.c:336: error: for each function it appears in.)

sip_reg.c:336: error: 'reg' undeclared (first use in this function)

sip_reg.c:366: error: 'eXosip_event_t' undeclared (first use in this
function)

sip_reg.c:366: error: 'event' undeclared (first use in this function)

sip_reg.c:385: error: 'EXOSIP_REGISTRATION_NEW' undeclared (first use in
this function)

sip_reg.c:388: error: 'EXOSIP_REGISTRATION_SUCCESS' undeclared (first use in
this function)

sip_reg.c:391: error: 'EXOSIP_REGISTRATION_FAILURE' undeclared (first use in
this function)

sip_reg.c:394: error: 'EXOSIP_REGISTRATION_TERMINATED' undeclared (first use
in this function)

 

 

 

how can I overcome the problem ?

 

cheers,

jay


From uniojnqoifazy at gmail.com  Fri Jul 24 06:11:07 2009
From: uniojnqoifazy at gmail.com (jay)
Date: Fri, 24 Jul 2009 12:11:07 +0800
Subject: [Osip] FW: /usr/bin/ld: cannot find -leXosip
Message-ID: <007501ca0c14$c70ba9b0$5522fd10$@com>

Hello 
When I using ?gcc sip_reg.c -o sip_reg -leXosip -losip2 -losipparser2 -lpthread -L/usr/local/lib? to compile sip_reg.c , I get the error msg as follows :
/usr/bin/ld: cannot find -leXosip
collect2: ld returned 1 exit status?

how can I overcome of errors?

Cheers,
Jay


From akhil4es at gmail.com  Fri Jul 24 08:49:23 2009
From: akhil4es at gmail.com (mohammmed akhil)
Date: Fri, 24 Jul 2009 12:19:23 +0530
Subject: [Osip] FW: /usr/bin/ld: cannot find -leXosip
In-Reply-To: <007501ca0c14$c70ba9b0$5522fd10$@com>
References: <007501ca0c14$c70ba9b0$5522fd10$@com>
Message-ID: <11bec9d20907232349l300e3d82k6d0f192659338f96@mail.gmail.com>

hi,
just add the path of other component's for eg

-I /usr/local/include/osipparser2               //(in my case path for
osipparser library)
similarly add the path for your libexosip library and compile that should
not flash any error.


you are getting  that error because you have not specified the path for your
exosip library and by default it's searching "/usr/bin/ld" where it's not
finding .so for exosip,hence you are getting error.

regards
Mohammed Akhil

On Fri, Jul 24, 2009 at 9:41 AM, jay <uniojnqoifazy at gmail.com> wrote:

> Hello
> When I using ?gcc sip_reg.c -o sip_reg -leXosip -losip2 -losipparser2
> -lpthread -L/usr/local/lib? to compile sip_reg.c , I get the error msg as
> follows :
> /usr/bin/ld: cannot find -leXosip
> collect2: ld returned 1 exit status?
>
> how can I overcome of errors?
>
> Cheers,
> Jay
>
> _______________________________________________
> Osip mailing list
> Osip at atosc.org
> http://www.atosc.org/mailinglist/listinfo/osip
>

From akhil4es at gmail.com  Fri Jul 24 10:50:39 2009
From: akhil4es at gmail.com (mohammmed akhil)
Date: Fri, 24 Jul 2009 14:20:39 +0530
Subject: [Osip] Need some pointers
Message-ID: <11bec9d20907240150t1bddacb2g50316e49cea8a96f@mail.gmail.com>

Hi aymeric and all,

I am trying to make a sip server,and i am totally new to sip, can you give
me some pointers regarding it.
I have downloaded and installed osip library,compiled the applications in it
and i follow the rfc 3261.
Some guide lines on using API to communicate between two end user will be
highly appriciated.


thanks and regards
mohammed akhil