Open Source Sip: Opensipstack compile error ...

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mouncifb

1 posts since
Jul 31, 2008

Jul 31, 2008 5:13 PM

Opensipstack compile error

I have downloaded opensipstack and opensbc from CVS I get this when I run: make bothnoshared

what should I do?

g++ -D_REENTRANT -D_REENTRANT -Wall -DPTRACING -I/root/opensbc/opensipstack/include -DPTRACING -I/root/opensbc/opensipstack/include -Os -c /root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx -o /root/opensbc/opensipstack/lib/obj_linux_x86_r/echocancel.o
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:106:24: error: speex_echo.h: No such file or directory
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:107:30: error: speex_preprocess.h: No such file or directory
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx: In destructor âvirtual OpalEchoCanceler::~OpalEchoCanceler()â:
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:158: error: âspeex_echo_state_destroyâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:163: error: âspeex_preprocess_state_destroyâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx: In member function âvoid OpalEchoCanceler::SetParameters(const OpalEchoCanceler::Params&)â:
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:187: error: âspeex_echo_state_destroyâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:192: error: âspeex_preprocess_state_destroyâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx: In member function âvirtual void OpalEchoCanceler::ReceivedPacket(RTP_DataFrame&, INT)â:
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:233: error: âspeex_echo_state_initâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:236: error: âspeex_preprocess_state_initâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:237: error: âSPEEX_PREPROCESS_SET_DENOISEâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:237: error: âspeex_preprocess_ctlâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:241: error: âspx_int16_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:241: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:241: error: expected `;' before âmallocâ
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:246: error: âspx_int32_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:249: error: âspx_int16_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:249: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:249: error: expected `;' before âmallocâ
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:251: error: âspx_int16_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:251: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:251: error: expected `;' before âmallocâ
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:257: error: âspx_int16_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:257: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:257: error: expected `)' before âref_bufâ
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:267: error: âspx_int16_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:267: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:267: error: âspeex_preprocessâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:268: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:277: error: âspx_int32_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:277: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:277: error: âspeex_echo_cancelâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:284: error: âspx_int16_tâ was not declared in this scope
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:284: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:284: error: expected primary-expression before â)â token
/root/opensbc/opensipstack/src/opal/src/codec/echocancel.cxx:284: error: âspeex_preprocessâ was not declared in this scope
make[3]: *** [/root/opensbc/opensipstack/lib/obj_linux_x86_r/echocancel.o] Error 1
make[3]: Leaving directory `/root/opensbc/opensipstack/src'
make[2]: *** [/root/opensbc/opensipstack/lib/libopensipstack_linux_x86_r_s.a] Error 2
make[2]: Leaving directory `/root/opensbc/opensbc'
make[1]: *** optnoshared Error 2
make[1]: Leaving directory `/root/opensbc/opensbc'
make: *** bothnoshared Error 2

joegen

136 posts since
Apr 28, 2007

1. Jul 31, 2008 8:57 PM

in response to: mouncifb

Re: Opensipstack compile error

Try ./configure --enable-localspeex and rebuild your library. If this does not work, please send the results of your configure output.

-joegen

mouncifb wrote:

I have downloaded opensipstack and opensbc from CVS I get this when I run: make bothnoshared
what should I do?

snip

Guest

2. Aug 6, 2008 10:07 AM

in response to: mouncifb

ATLSIP.dll dependancies

Hi,

I am using ATLSIP.dll (it works great, thanks) but am having troubles
registering it on one computer.

It gives error: application failed to initialize, try reinstalling the
application might fix the problem.

I was wondering about what dependencies the dll has ( .NET, Windows Service
Pack, ...).

Regards,

Robert Vos

Guest

3. Aug 6, 2008 11:07 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP.dll dependancies

Hi Robert,

I think, the the Library is missing some required DLL's.
You can find out which one is missing with a tool named
Depends.exe which is shipped with the VS.NET.

Kind regards

Christian

-----Ursprüngliche Nachricht-----
Von: opensipstack-devel-bounces@lists.sourceforge.net
opensipstack-devel-bounces@lists.sourceforge.net Im Auftrag von
Robert Vos
Gesendet: Mittwoch, 6. August 2008 16:07
An: opensipstack-devel@opensourcesip.org;
opensipstack-devel@lists.sourceforge.net
Betreff: OpenSIPStack ATLSIP.dll dependancies

Hi,

I am using ATLSIP.dll (it works great, thanks) but am having troubles
registering it on one computer.

It gives error: application failed to initialize, try reinstalling the
application might fix the problem.

I was wondering about what dependencies the dll has ( .NET, Windows Service
Pack, ...).

Regards,

Robert Vos

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Guest

4. Aug 6, 2008 11:05 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP.dll dependancies

Robert,

Dependency walker should be able to tell you what DLLs are missing. See
http://dependencywalker.com/

Joegen

Robert Vos wrote:

Hi,

I am using ATLSIP.dll (it works great, thanks) but am having troubles
registering it on one computer.

It gives error: application failed to initialize, try reinstalling the
application might fix the problem.

I was wondering about what dependencies the dll has ( .NET, Windows Service
Pack, ...).

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
Grand prize is a trip for two to an Open Source event anywhere in the world
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opensipstack-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

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Guest

5. Oct 13, 2008 7:58 AM

in response to: Guest

[OpenSIPStack] ATLSIP multiple calls

Hi,

I was wondering if it possible to handle multiple calls with ATLSIP. I am
attempting to make an operator soft phone, where there can be multiple
incoming calls at the same time. I plan to use it with a SIP intercom
system, so it is especially important that the operator knows about all
incoming calls.

Is there anyway to do this with ATLSIP, or perhaps modify it to allow this?

I have thought of a possible work around, where one registers several sip
components, and then setup the asterisk server to phone them consecutively
until it finds one that is open. This might work, but it seems very
inefficient.

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
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Guest

6. Oct 13, 2008 12:13 PM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

Is there anyway to do this with ATLSIP, or perhaps modify it to allow this?

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applications with Moblin SDK & win great prizes
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opensipstack-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
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Guest

7. Oct 15, 2008 5:07 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

Hello Joegen,

Do we send any extra header in SIP Invite from Opensipclient to OpenSBC to
indicate wether it should encrypt RTP?...Or OpenSBC decrypts RTP by default
if SIP packets are encrypted?

The problem is i am not receiving any Voice at both ends when i enable
Encryption however i can see SIP packets being encrypted decrypted fine. I
am using OSSphone to initiate calls.

Regards,

Manoj

-----Original Message-----
From: Joegen E. Baclor joegen.baclor@gmail.com
Sent: Monday, October 13, 2008 9:43 PM
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
opensipstack-devel mailing list
opensipstack-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.8.0/1721 - Release Date: 10/12/2008

12:00 PM

--
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Checked by AVG Free Edition.
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Guest

8. Oct 15, 2008 5:27 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

OpenSBC should decrypt the RTP and SIP packets automatically. You might
want to run wireshark in OpenSBC and see if RTP is being decrypted
properly as well.

Manoj Joshi wrote:

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.8.0/1721 - Release Date: 10/12/2008

12:00 PM

--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 270.8.0/696 - Release Date: 10/9/2008
12:00 AM

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Guest

9. Oct 15, 2008 5:27 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

I ran some debug and found following messages at OpenSBC. "RTP_UDP Session
1, Received data packet too small: 36 bytes"
There are many messages like that.

Also...does it matter if i call "Encryption::Engine::Encrypt( in, out )" for
each RTP packet?..i guess headers will be encrypted too right?

Regards,

manoj

-----Original Message-----
From: Joegen E. Baclor joegen.baclor@gmail.com
Sent: Wednesday, October 15, 2008 2:46 PM
To: Manoj Joshi
Cc: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

OpenSBC should decrypt the RTP and SIP packets automatically. You might
want to run wireshark in OpenSBC and see if RTP is being decrypted
properly as well.

Manoj Joshi wrote:

Hello Joegen,

Do we send any extra header in SIP Invite from Opensipclient to OpenSBC to
indicate wether it should encrypt RTP?...Or OpenSBC decrypts RTP by

default

if SIP packets are encrypted?

The problem is i am not receiving any Voice at both ends when i enable
Encryption however i can see SIP packets being encrypted decrypted fine. I
am using OSSphone to initiate calls.

Regards,

Manoj

-----Original Message-----
From: Joegen E. Baclor joegen.baclor@gmail.com
Sent: Monday, October 13, 2008 9:43 PM
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

I was wondering if it possible to handle multiple calls with ATLSIP. I

attempting to make an operator soft phone, where there can be multiple
incoming calls at the same time. I plan to use it with a SIP intercom
system, so it is especially important that the operator knows about all
incoming calls.

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I have thought of a possible work around, where one registers several sip
components, and then setup the asterisk server to phone them

consecutively

until it finds one that is open. This might work, but it seems very
inefficient.

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.8.0/1721 - Release Date:

10/12/2008

12:00 PM

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challenge

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prizes
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--
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Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 270.8.0/696 - Release Date: 10/9/2008
12:00 AM

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Guest

10. Oct 15, 2008 5:37 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

Manoj Joshi wrote:

I ran some debug and found following messages at OpenSBC. "RTP_UDP Session
1, Received data packet too small: 36 bytes"
There are many messages like that.

This is an indication that the RTP packet is either truncated or is not
formatted correctly.

Also...does it matter if i call "Encryption::Engine::Encrypt( in, out )" for
each RTP packet?..i guess headers will be encrypted too right?

Headers will be encrypted too, yes

Regards,

manoj

-----Original Message-----
From: Joegen E. Baclor joegen.baclor@gmail.com
Sent: Wednesday, October 15, 2008 2:46 PM
To: Manoj Joshi
Cc: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

OpenSBC should decrypt the RTP and SIP packets automatically. You might
want to run wireshark in OpenSBC and see if RTP is being decrypted
properly as well.

Manoj Joshi wrote:

Hello Joegen,

Do we send any extra header in SIP Invite from Opensipclient to OpenSBC to
indicate wether it should encrypt RTP?...Or OpenSBC decrypts RTP by

default

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

I was wondering if it possible to handle multiple calls with ATLSIP. I

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I have thought of a possible work around, where one registers several sip
components, and then setup the asterisk server to phone them

consecutively

until it finds one that is open. This might work, but it seems very
inefficient.

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.8.0/1721 - Release Date:

10/12/2008

12:00 PM

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great
prizes
Grand prize is a trip for two to an Open Source event anywhere in the

world

--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 270.8.0/696 - Release Date: 10/9/2008
12:00 AM

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Guest

11. Oct 15, 2008 5:47 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

In your logic are you calling Encryption::Engine::Encrypt( in, out ) for
every RTP packet or what?...coz in SIP you are capturing whole message first
and then encrypting?

-----Original Message-----
From: joegen@opensipstack.org joegen.baclor@gmail.com
Sent: Wednesday, October 15, 2008 3:00 PM
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

Manoj Joshi wrote:

I ran some debug and found following messages at OpenSBC. "RTP_UDP Session
1, Received data packet too small: 36 bytes"
There are many messages like that.

This is an indication that the RTP packet is either truncated or is not
formatted correctly.

Also...does it matter if i call "Encryption::Engine::Encrypt( in, out )"

for

each RTP packet?..i guess headers will be encrypted too right?

Headers will be encrypted too, yes

OpenSBC should decrypt the RTP and SIP packets automatically. You might
want to run wireshark in OpenSBC and see if RTP is being decrypted
properly as well.

Manoj Joshi wrote:

Hello Joegen,

Do we send any extra header in SIP Invite from Opensipclient to OpenSBC

indicate wether it should encrypt RTP?...Or OpenSBC decrypts RTP by

default

if SIP packets are encrypted?

The problem is i am not receiving any Voice at both ends when i enable
Encryption however i can see SIP packets being encrypted decrypted fine.

am using OSSphone to initiate calls.

Regards,

Manoj

-----Original Message-----
From: Joegen E. Baclor joegen.baclor@gmail.com
Sent: Monday, October 13, 2008 9:43 PM
To: opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

I was wondering if it possible to handle multiple calls with ATLSIP. I

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I have thought of a possible work around, where one registers several

sip

components, and then setup the asterisk server to phone them

consecutively

until it finds one that is open. This might work, but it seems very
inefficient.

I would really appreciate any help on this.

Regards,

Robert Vos

------------------------------------------------------------------------

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

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Guest

12. Oct 16, 2008 3:27 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

Hello Joegan,

Could you please reply to my query?....Is this called for each packet?...or
for frame...does it matter if its is called for each frame or for packet.

Regards,

Manoj

-----Original Message-----
From: Manoj Joshi manoj@ascenttelecom.com
Sent: Wednesday, October 15, 2008 3:10 PM
To: joegen@opensipstack.org; opensipstack-devel@lists.sourceforge.net
Subject: Re: OpenSIPStack ATLSIP multiple calls

In your logic are you calling Encryption::Engine::Encrypt( in, out ) for
every RTP packet or what?...coz in SIP you are capturing whole message first
and then encrypting?

Manoj Joshi wrote:

I ran some debug and found following messages at OpenSBC. "RTP_UDP Session
1, Received data packet too small: 36 bytes"
There are many messages like that.

This is an indication that the RTP packet is either truncated or is not
formatted correctly.

Also...does it matter if i call "Encryption::Engine::Encrypt( in, out )"

for

each RTP packet?..i guess headers will be encrypted too right?

Headers will be encrypted too, yes

OpenSBC should decrypt the RTP and SIP packets automatically. You might
want to run wireshark in OpenSBC and see if RTP is being decrypted
properly as well.

Manoj Joshi wrote:

Hello Joegen,

Do we send any extra header in SIP Invite from Opensipclient to OpenSBC

indicate wether it should encrypt RTP?...Or OpenSBC decrypts RTP by

default

if SIP packets are encrypted?

The problem is i am not receiving any Voice at both ends when i enable
Encryption however i can see SIP packets being encrypted decrypted fine.

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

I was wondering if it possible to handle multiple calls with ATLSIP. I

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I have thought of a possible work around, where one registers several

sip

components, and then setup the asterisk server to phone them

consecutively

until it finds one that is open. This might work, but it seems very
inefficient.

I would really appreciate any help on this.

Regards,

Robert Vos

------------------------------------------------------------------------

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No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.8.0/1721 - Release Date:

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Guest

13. Oct 16, 2008 3:37 AM

in response to: Guest

Re: [OpenSIPStack] ATLSIP multiple calls

Encryption is done on the entire RTP packet, not just the payload.
What are you trying to do exactly? Are you trying to make your own
custom encryption server based on XOR encryption of OpenSIPStack?

Manoj Joshi wrote:

In your logic are you calling Encryption::Engine::Encrypt( in, out ) for
every RTP packet or what?...coz in SIP you are capturing whole message first
and then encrypting?

Manoj Joshi wrote:

I ran some debug and found following messages at OpenSBC. "RTP_UDP Session
1, Received data packet too small: 36 bytes"
There are many messages like that.

This is an indication that the RTP packet is either truncated or is not
formatted correctly.

Also...does it matter if i call "Encryption::Engine::Encrypt( in, out )"

for

each RTP packet?..i guess headers will be encrypted too right?

Headers will be encrypted too, yes

OpenSBC should decrypt the RTP and SIP packets automatically. You might
want to run wireshark in OpenSBC and see if RTP is being decrypted
properly as well.

Manoj Joshi wrote:

Hello Joegen,

Do we send any extra header in SIP Invite from Opensipclient to OpenSBC

indicate wether it should encrypt RTP?...Or OpenSBC decrypts RTP by

default

if SIP packets are encrypted?

The problem is i am not receiving any Voice at both ends when i enable
Encryption however i can see SIP packets being encrypted decrypted fine.

Robert,

I regret to say it but ATLSIP only support single channel for now. It a
very nice to have functionality and I agree it has to be supported in
future versions.

Joegen

Robert Vos wrote:

Hi,

I was wondering if it possible to handle multiple calls with ATLSIP. I

Is there anyway to do this with ATLSIP, or perhaps modify it to allow

this?

I have thought of a possible work around, where one registers several

sip

components, and then setup the asterisk server to phone them

consecutively

until it finds one that is open. This might work, but it seems very
inefficient.

I would really appreciate any help on this.

Regards,

Robert Vos

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

No virus found in this incoming message.
Checked by AVG - http://www.avg.com
Version: 8.0.173 / Virus Database: 270.8.0/1721 - Release Date:

10/12/2008

12:00 PM

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great
prizes
Grand prize is a trip for two to an Open Source event anywhere in the

world

--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 270.8.0/696 - Release Date: 10/9/2008
12:00 AM

This SF.Net email is sponsored by the Moblin Your Move Developer's

challenge

Build the coolest Linux based applications with Moblin SDK & win great

prizes

Grand prize is a trip for two to an Open Source event anywhere in the

world

--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.446 / Virus Database: 270.8.0/696 - Release Date: 10/9/2008
12:00 AM

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Guest

14. Oct 16, 2008 4:27 AM