[OpenSIPS-Users] lookup

[Ghaith ALKAYYEM]

Hi list,

Does anybody know the possible parameters of "lookup" function, they're
not mentioned in the available documents or webinars.

There is: lookup("location") which is responsible for finding the
reference to the specified destination, but what (lookup("aliases") does
for example?

Thank you


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Re: [OpenSIPS-Users] static

[Ghaith ALKAYYEM]

Thank you,
I thought there's other restrictions imposed on module development. So,
it's just because using static is the correct choice in this case.


On Thu, 2009-10-01 at 16:36 +0200, Saúl Ibarra wrote:
> > What do you mean exactly with "static"?
> 
> He surely means why functions are like "static in myfunctionname...".
> Functions are static because the don't need to be exported so if you
> declare a funcion as static in C you can only use it within that .c
> file IIRC.
> 
> 


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[OpenSIPS-Users] static

[Ghaith ALKAYYEM]

Hi,
Could you tell me why the most functions and structures in the developed
modules are static?
Regards.



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Re: [OpenSIPS-Users] Opensips as SBC

[Ghaith ALKAYYEM]

Thank you for response,

Actually what I'm looking for is relaying and controlling the traffic
according to static/dynamic rules that take into consideration what's
available in the SIP header or by taking into account link conditions or
status if possible. So, it's a kind of RTP relaying through different
links/interfaces and according to different rules.

Regards.


On Wed, 2009-09-30 at 22:47 +0300, Bogdan-Andrei Iancu wrote:
> Hi Ghaith,
> 
> an SBC is a very generic term.it can do a lot of stuff (NAT , topi 
> hiding, net bridging, security, etc)...
> 
> So, what kind of functionalities you have in mind when you ask about a SBC?
> 
> Regards,
> Bogdan
> 
> Ghaith ALKAYYEM wrote:
> > Hello,
> >
> > Do you think that we can consider OpenSIPS as a real SBC? and if not
> > what do you think the missed functionalities are?
> > or one has to add some module in Opensips to act it as SBC
> >
> > are there some performance drawbacks and/or other issues while
> > converting Opensips to SBC?
> >
> > thanks for your reply in advance.
> > Regards
> >
> >
> >
> > ___
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
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[OpenSIPS-Users] Opensips as SBC

[Ghaith ALKAYYEM]

Hello,

Do you think that we can consider OpenSIPS as a real SBC? and if not
what do you think the missed functionalities are?
or one has to add some module in Opensips to act it as SBC

are there some performance drawbacks and/or other issues while
converting Opensips to SBC?

thanks for your reply in advance.
Regards



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Re: [OpenSIPS-Users] rtp proxy issue

[Ghaith ALKAYYEM]

Hi,
Maybe you have to try calling some functions before forcing the traffic
in order to change the parameter in SDP headers.
you can try something like that:

force_rport();
set_rtp_proxy_set("0");
fix_nated_contact();
force_rtp_proxy();

Because as far as I know that SER will change the "connection c=" field
in the SDP header.
It works fine with me.
Good luck


On Wed, 2009-09-30 at 01:57 -0700, Jan D. wrote:
> I'm having problems with force_rtp_proxy(). My final goal is to use rtp_proxy
> for user to user calls (not outbound).
> 
> I compiled opensips 1.5.3 on a Debian Unstable system. Als used apt-get to
> install rtpproxy. It runs under user opensips.
> 
> If I use force_rtp_proxy() the new IP address (77.20.20.1) (in the body) is
> placed behind the old IP address (89.10.10.1):
> Connection Address: 89.10.10.177.20.20.1
> 
> Here the nathelper config:
> 
> loadmodule "nathelper.so"
> modparam("nathelper", "natping_interval", 60)
> modparam("nathelper", "ping_nated_only", 1)
> modparam("nathelper", "sipping_from", "sip:natp...@sip3.sollie.nl")
> modparam("nathelper", "received_avp", "$avp(i:801)")
> modparam("nathelper", "sipping_bflag", 6)
> modparam("nathelper", "rtpproxy_sock",
> "unix:/var/run/rtpproxy/rtpproxy.sock")
> 
> This is how I use rtpproxy:
> 
> # rtp proxy for local to local
> if(has_body("application/sdp"))
> {
> xlog("L_INFO"," INFO: force_rtp_proxy\n");
> force_rtp_proxy();
> t_on_reply("2");
> }
> 
> t_relay("0x05");
> 
> I also tried to set options but without any success.
> 
> Can anyone help or give a hint?
> 
> Jan
> 


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Re: [OpenSIPS-Users] Diameter

[Ghaith ALKAYYEM]

this looks a very good starting point, this module is designed to work
with SER.
I'll try to check it.
Thank you.

Regards

On Tue, 2009-09-22 at 19:26 +0200, Stefan Sayer wrote:
> Hi,
> 
> o Ghaith ALKAYYEM [09/21/09 19:59]:
> > Hello lists,
> > 
> > I'm interested in AAA functions according to Diameter which is newer
> > than Radius.
> > There's a module in OpenSIPS which is called "AUTH_DIAMETER Module" and
> > it's mentioned that this module is obsolete. So I'd like your
> > recommendations about this matter, should I work from the scratch to
> > develop something that does this functionalities or is it possible to
> > integrate other open source software with OpenSIPS.
> also have a look at cdp module from openimscore 
> (http://www.openimscore.org/docs/ser_ims/CDP.html), its based on the old 
> disc implementation, but extended a lot from that point - chances are 
> you can use/reuse many things from there.
> 
> hth
> Stefan
> 
> 
> > 
> > Thank you very much.
> > 
> > 
> > 
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


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Re: [OpenSIPS-Users] Diameter

[Ghaith ALKAYYEM]

Actually, openblox project has two versions, one is proposed as a
commercial and the other as an open license version.
It's implemented in JAVA and they mentioned in their web site that
there's a C++ implementation but I couldn't reach it through their web
site.
you can see the product in the following link:
http://traffixsystems.com/site/content/t2.asp?Sid=49&Pid=241

Regards.

On Tue, 2009-09-22 at 17:58 +0300, Bogdan-Andrei Iancu wrote:
> Hi Ghaith,
> 
> Ghaith ALKAYYEM wrote:
> > Thank you for response,
> >
> > I see in the details of that module (auth_diameter) this diagram:
> >
> >  ++ SIP INVITE   +=+  DIAMETER  +--+   +--+
> >  || no Auth hdr  #/#  AA-Request|  |   |  |
> >  ||-1--->#/#---2--->|  |---2-->|  |
> >  |UAC |  #UAS//#|DClnt |   |DSrv  |
> >  ||<-4---#(SER)#<--3|(DISC)|<--3---|(DISC)|
> >  || 401  #/#  DIAMETER  |  |   |  |
> >  ++ Unauthorized +=+  AA-Answer +--+   +--+
> >
> > We notice in this architecture that we have two diameter blocks, the
> > first one plays the role of diameter client(DClnt) and the second one
> > plays the role of diameter server(DSrv).
> >   
> as said, this was an old approach and both the design and software are 
> outdated.
> > But in Radius modules the OpenSIPS interacts with Radius server
> > directly, so maybe I have a misunderstood in this regard but I'd like to
> > know whether it's possible to make OpenSIPS interact with Diameter
> > server directly or this is not possible due to the nature of diameter
> > protocol.
> >   
> yes, opensips talks directly to RADIUS server because it is using the 
> libradiusclient-ng (which acts as radius client)
> > Opendiameter is written in C++ so I think it's not possible to integrate
> > it directly in OpenSIPS as a module, so we have to design something
> > similar to the above diagram, isn't it? What would be the type of
> > communication between OpenSIPS & Diameter Client, is it diameter based
> > also?
> >   
> no, the diameter client should be provided by a library and opensips 
> will link against that library (like we do for RADIUS now).
> 
> Also, looking at the opendiameter project, not sure how active it is - 
> there are no code changes since February 2008.
> > The implementation of Openblox looks promising as well, so do you think
> > it would be a good candidate for building the module?
> >   
> Do you have a link to the project? is this project GPL compatible? also, 
> does it provide a C API ?
> 
> Regards,
> Bogdan
> 
> > Regards.
> >
> >
> > On Tue, 2009-09-22 at 14:17 +0300, Bogdan-Andrei Iancu wrote:
> >   
> >> Hi Ghaith,
> >>
> >> Ghaith ALKAYYEM wrote:
> >> 
> >>> Hello lists,
> >>>
> >>> I'm interested in AAA functions according to Diameter which is newer
> >>> than Radius.
> >>>   
> >>>   
> >> yes, the new AAA interface will simplify a lot the addition of DIAMETER 
> >> in OpenSIPS. All modules using the AAA interface will be automatically 
> >> able to use the DIAMETER support.
> >> 
> >>> There's a module in OpenSIPS which is called "AUTH_DIAMETER Module" and
> >>> it's mentioned that this module is obsolete. 
> >>>   
> >> yes ,it is obsolete as it is using an old and obsolete DIAMETER 
> >> client-server implementation (DISC).
> >>
> >> 
> >>> So I'd like your
> >>> recommendations about this matter, should I work from the scratch to
> >>> develop something that does this functionalities or is it possible to
> >>> integrate other open source software with OpenSIPS.
> >>>   
> >>>   
> >> Our plan is to use some opensource libraries to build a DIAMETER 
> >> (aaa_diameter module)  implementation for the AAA API in OpenSIPS. We 
> >> tried to evaluate opendiameter project for this 
> >> (http://www.opendiameter.org/)
> >>
> >> Regards,
> >> Bogdan
> >> 
> >>> Thank you very much.
> >>>
> >>>
> >>>
> >>> ___
> >>> Users mailing list
> >>> Users@lists.opensips.org
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>
> >>>   
> >>>   
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >> 
> >
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
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Re: [OpenSIPS-Users] Diameter

[Ghaith ALKAYYEM]

Thank you for response,

I see in the details of that module (auth_diameter) this diagram:

 ++ SIP INVITE   +=+  DIAMETER  +--+   +--+
 || no Auth hdr  #/#  AA-Request|  |   |  |
 ||-1--->#/#---2--->|  |---2-->|  |
 |UAC |  #UAS//#|DClnt |   |DSrv  |
 ||<-4---#(SER)#<--3|(DISC)|<--3---|(DISC)|
 || 401  #/#  DIAMETER  |  |   |  |
 ++ Unauthorized +=+  AA-Answer +--+   +--+

We notice in this architecture that we have two diameter blocks, the
first one plays the role of diameter client(DClnt) and the second one
plays the role of diameter server(DSrv).
But in Radius modules the OpenSIPS interacts with Radius server
directly, so maybe I have a misunderstood in this regard but I'd like to
know whether it's possible to make OpenSIPS interact with Diameter
server directly or this is not possible due to the nature of diameter
protocol.

Opendiameter is written in C++ so I think it's not possible to integrate
it directly in OpenSIPS as a module, so we have to design something
similar to the above diagram, isn't it? What would be the type of
communication between OpenSIPS & Diameter Client, is it diameter based
also?

The implementation of Openblox looks promising as well, so do you think
it would be a good candidate for building the module?

Regards.


On Tue, 2009-09-22 at 14:17 +0300, Bogdan-Andrei Iancu wrote:
> Hi Ghaith,
> 
> Ghaith ALKAYYEM wrote:
> > Hello lists,
> >
> > I'm interested in AAA functions according to Diameter which is newer
> > than Radius.
> >   
> yes, the new AAA interface will simplify a lot the addition of DIAMETER 
> in OpenSIPS. All modules using the AAA interface will be automatically 
> able to use the DIAMETER support.
> > There's a module in OpenSIPS which is called "AUTH_DIAMETER Module" and
> > it's mentioned that this module is obsolete. 
> yes ,it is obsolete as it is using an old and obsolete DIAMETER 
> client-server implementation (DISC).
> 
> > So I'd like your
> > recommendations about this matter, should I work from the scratch to
> > develop something that does this functionalities or is it possible to
> > integrate other open source software with OpenSIPS.
> >   
> Our plan is to use some opensource libraries to build a DIAMETER 
> (aaa_diameter module)  implementation for the AAA API in OpenSIPS. We 
> tried to evaluate opendiameter project for this 
> (http://www.opendiameter.org/)
> 
> Regards,
> Bogdan
> > Thank you very much.
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
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[OpenSIPS-Users] Diameter

[Ghaith ALKAYYEM]

Hello lists,

I'm interested in AAA functions according to Diameter which is newer
than Radius.
There's a module in OpenSIPS which is called "AUTH_DIAMETER Module" and
it's mentioned that this module is obsolete. So I'd like your
recommendations about this matter, should I work from the scratch to
develop something that does this functionalities or is it possible to
integrate other open source software with OpenSIPS.

Thank you very much.



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[OpenSIPS-Users] opensips vs kamailio

[Ghaith ALKAYYEM]

Hello,
Could you please tell me the difference between these two products:
OpenSIPS & Kamailio
every time I check them I feel they are very similar.
Regards.



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Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed

[Ghaith ALKAYYEM]

Does the dispatcher work on the same machine also?
Could you provide me with more details about running these two
instances?

Regards.

On Thu, 2009-09-10 at 23:39 +0800, Jiang Jinke wrote:
> Thanks for the detail instruction.
> I just use a symlink into the directory, it's working properly now.
> 
> Just like below:
> /usr/local/relay1/media-relay   -> /usr/bin/media-relay
> /usr/local/relay2/media-relay   -> /usr/bin/media-relay
> 
> Regards,
> Jinke Jiang
> 
> On Thu, Sep 10, 2009 at 10:02 PM, Dan Pascu  wrote:
> >
> > On 10 Sep 2009, at 15:47, Raúl Alexis Betancor Santana wrote:
> >
> >> On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote:
> >>> Hello,
> >>>
> >>> I think it's not possible to use two separate relays on the same
> >>> server,
> >>> I tried that a lot then I switched to RTPproxy.
> >>
> >> That's not true, you could run as many Realys as you want on the
> >> same server,
> >> only have to patch mediaproxy-relay to be able to call it with a
> >> diferent .cfg as the default one, have diferent listen ports and no
> >> more.
> >
> > You don't need to patch anything. Just unpack mediaproxy in as many
> > different directories as you need, run ./build_inplace and modify each
> > config.ini in those directories as needed. Then run mediaproxy from
> > those directories and each of them will use the local config.ini from
> > its own directory.
> >
> > Alternatively, if you want to use a system wide installation, you can
> > copy the binaries from /usr/bin to a number of different directories
> > and add a config.ini in each directory. Then run those binaries from
> > those directories instead of /usr/bin/ and each binary will use the
> > config.ini file in its own directory to overwrite settings from the
> > global /etc/mediaproxy/config.ini.
> >
> > Mediaproxy uses 2 configuration files. The global one resides in /etc/
> > mediaproxy/config.ini. On top of that if a config.ini is present in
> > the same directory as the binary (media-relay & media-dispatcher) that
> > one will be used to overwrite the settings from the global one having
> > priority over it.
> >
> > --
> > Dan
> >
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> 
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Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed

[Ghaith ALKAYYEM]

Hello,

I think it's not possible to use two separate relays on the same server,
I tried that a lot then I switched to RTPproxy.

Regards.


On Thu, 2009-09-10 at 18:22 +0800, Jiang Jinke wrote:
> Dear All,
> 
> I tried to make the relay to listen to the none default route
> interface, but failed.
> 
> Below is my configuration info and my env.
> 
> Using the latest mediaproxy-2.3.8 on CentOS 5.3.
> 
> I had media-dispatcher and media-relay running on my server which has
> multiple NIC interface,
> two public ips which are belong to two different provider.
> 
> for example:
> eth0: 192.168.88.12
> eth1: 203.xx.xx.xx
> eth2: 63.xx.xx.xx
> eth3: 10.11.12.21
> 
> route -n
> Kernel IP routing table
> Destination Gateway Genmask Flags Metric RefUse Iface
> 63.xx.xx.320.0.0.0 255.255.255.240 U 0  00 eth2
> 203.xx.xx.128  0.0.0.0 255.255.255.192 U 0  00 eth1
> 10.11.12.0  0.0.0.0 255.255.255.0   U 0  00 eth3
> 192.168.88.00.0.0.0 255.255.255.0   U 0  00 eth0
> 169.254.0.0 0.0.0.0 255.255.0.0 U 0  00 eth3
> 0.0.0.0 203.xx.xx.129  0.0.0.0 UG0  00 eth1
> 
> But I notice that the relay are always binding to the one that has
> default route
> as said in the document.
> I can use two machine of course, but is it possible to have two
> separate relay that will
> only sent out packet from the interface it's listening to.
> 
> my config file is:
> [Relay]
> dispatchers = 63.xx.xx.xx 203.xx.xx.xx
> port_range = 3:4
> log_level = DEBUG
> [Dispatcher]
> socket_path = dispatcher.sock
> listen_management = 127.0.0.1
> log_level = DEBUG
> relay_timeout = 5
> [TLS]
> certs_path = tls
> [Database]
> [Radius]
> [OpenSIPS]
> socket_path = '/var/run/opensips.sock'
> max_connections = 20
> 
> I tried to use the media_relay_avp before calling use_media_proxy();
> but it seems the relay still listening to the other interface which
> has the default route.
> 
> The script of opensips is below:
> $avp(s:media_relay) = "63.xx.xx.xx";
> use_media_proxy();
> 
> Regards,
> --
> Jinke Jiang
> 
> 
> 
> the debug log of mediaproxy:
> Sep 10 18:16:54 ssw1 media-dispatcher[21925]: debug: Issuing "update"
> command to relay at 63.xx.xx.xx
> Sep 10 18:16:54 ssw1 media-relay[21930]: debug: Received new SDP offer
> Sep 10 18:16:54 ssw1 media-relay[21930]:
> mediaproxy.mediacontrol.StreamListenerProtocol starting on 30028
> Sep 10 18:16:54 ssw1 media-relay[21930]:
> mediaproxy.mediacontrol.StreamListenerProtocol starting on 30029
> Sep 10 18:16:54 ssw1 media-relay[21930]:
> mediaproxy.mediacontrol.StreamListenerProtocol starting on 30030
> Sep 10 18:16:54 ssw1 media-relay[21930]:
> mediaproxy.mediacontrol.StreamListenerProtocol starting on 30031
> Sep 10 18:16:54 ssw1 media-relay[21930]: debug: Added new stream:
> (audio) 10.10.101.147:20562 (RTP: Unknown, RTCP: Unknown) <->
> 203.xx.xx.xx:30028 <->
> 
> 203.xx.xx.xx:30030 <-> Unknown (RTP: Unknown, RTCP: Unknown)
> Sep 10 18:16:54 ssw1 media-relay[21930]: debug: created new session
> 4022675...@10.10.101.147: 8...@test.com (3297793466) -->
> 0086135x...@test.com
> Sep 10 18:16:55 ssw1 media-dispatcher[21925]: debug: Issuing "update"
> command to relay at 63.xx.xx.xx
> Sep 10 18:16:55 ssw1 media-relay[21930]: debug: updating existing
> session 4022675...@10.10.101.147: 8...@test.com (3297793466) -->
> 0086135x...@test.com
> Sep 10 18:16:55 ssw1 media-relay[21930]: debug: Received updated SDP answer
> Sep 10 18:16:55 ssw1 media-relay[21930]: debug: Got initial answer
> from callee for stream: (audio) 10.10.101.147:20562 (RTP: Unknown,
> RTCP: Unknown) <->
> 
> 203.xx.xx.xx:30028 <-> 203.xx.xx.xx:30030 <-> 203.xx.xx.15x:18550
> (RTP: Unknown, RTCP: Unknown)
> Sep 10 18:16:56 ssw1 media-relay[21930]: debug: Got traffic
> information for stream: (audio) 10.10.101.147:20562 (RTP:
> 119.145.xx.xx:20562, RTCP: Unknown) <->
> 
> 203.xx.xx.xx:30028 <-> 203.xx.xx.xx:30030 <-> 203.xx.xx.15x:18550
> (RTP: Unknown, RTCP: Unknown)
> Sep 10 18:16:56 ssw1 media-relay[21930]: debug: Got traffic
> information for stream: (audio) 10.10.101.147:20562 (RTP:
> 119.145.xx.xx:20562, RTCP: Unknown) <->
> 
> 203.xx.xx.xx:30028 <-> 203.xx.xx.xx:30030 <-> 203.xx.xx.15x:18550
> (RTP: 203.xx.xx.15x:18550, RTCP: Unknown)
> Sep 10 18:16:58 ssw1 media-relay[21930]: debug: Got traffic
> information for stream: (audio) 10.10.101.147:20562 (RTP:
> 119.145.xx.xx:20562, RTCP: Unknown) <->
> 
> 203.xx.xx.xx:30028 <-> 203.xx.xx.xx:30030 <-> 203.xx.xx.15x:18550
> (RTP: 203.xx.xx.15x:18550, RTCP: 203.xx.xx.15x:18551)
> Sep 10 18:17:03 ssw1 media-dispatcher[21925]: debug: Issuing "update"
> command to relay at 63.xx.xx.xx
> Sep 10 18:17:03 ssw1 media-relay[21930]: debug: updating existing
> session 4022675...@10.10.101.147: 8...@test.com (3297793466) -->
> 0086135x...@test.com
> Sep 10 18:17:03 ssw1 media-relay[21930]: debug: Recei

Re: [OpenSIPS-Users] questions about log?

[Ghaith ALKAYYEM]

http://www.opensips.org/Resources/DocsTsStart



On Wed, 2009-09-09 at 08:04 +0200, Uwe Kastens wrote:
> Hi,
> 
> you can define the syslog facility in opensips.cfg. After that you can
> put the log to any location.
> 
> BR
> 
> Uwe
> 
> 
> 
> ASHWINI NAIDU schrieb:
> > By default the logging of opensips will be done in */var/log/syslog* in
> > debian systems and */var/log/messages* in redhat based systems
> > 
> > 2009/9/9 zhangchao1  > >
> > 
> > 
> > Hello everybody, dose anyone know where the log file is?
> > 
> > 
> > "中国制造",讲述中国60年往事
> > 
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
> > 
> > 
> > 
> > -- 
> > Thanking You,
> > Ashwini BR Naidu
> > 
> > 
> > 
> > 
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 


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Re: [OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

[Ghaith . ALKAYYEM]

I'm using Wireshark

Saúl Ibarra  a écrit :

> How are you making the packet captures? If tcpdump, did you use -s0?
>
>
> --
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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[OpenSIPS-Users] Checksum incorrect maybe caused by "UDP" checksum offload

[Ghaith . ALKAYYEM]

Hello,

Could anybody tell me what is the problem with such a configuration:


--- 
   
(UAC)192.168.10.1 |===> |192.168.10.10 [OpenSIPS]  
192.168.20.20|===> |(UAC)192.168.20.1 |
--- 
   


The mysterious point is that all packets orginating from the interface  
(192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
level. This happened with/without RTPproxy, so i'm wondering whether I  
should care about this problem or not because the sound is conveyed  
and everything
seems okay.

Regards.


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[OpenSIPS-Users] Development & SDP header

[Ghaith . ALKAYYEM]

Hello,

I'm trying to investigate the media type inside the SDP header, when I  
had a look at the structure of SDP inside sdp.h, I found that:
sdp_info contains a list of sessions and each session is constituted  
of more than one stream which contains the media header field.
So, how can I reach the real type of media?
Thank you very much.

Regards.


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Re: [OpenSIPS-Users] force_rtp_proxy([flags [, ip_address]]) failed to change IP adress in SDP

[Ghaith ALKAYYEM]

Hi,

I think when using RTP proxy you should expect the IP address of the
external proxy in the "Connection Information (c):" line.
I tested it and it was okay.

Regards.


On Thu, 2009-09-03 at 19:12 +0200, Yannick LE COENT wrote:
> Hi,
> 
> I am using RTP proxy in bridge mode.
> 
> When calling force_rtp_proxy("ie", "192.168.3.51"), I was expecting to have
> the IP address of the media line (m= ) set to 192.168.3.51.
> In fact the SDP contains one of the IP address of RTP proxy bridge.
> 
> What's wrong?
> 
> The aim of my configuration is to move openSIPS to a DMZ.
> 
> Thanks for any help,
> Yannick
> 
> 
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Re: [OpenSIPS-Users] IDE

[Ghaith ALKAYYEM]

What debugger for development do you recommend as well?

Regards.

On Thu, 2009-09-03 at 09:21 +0200, Saúl Ibarra wrote:
> What about this? http://www.vim.org/scripts/script.php?script_id=2242
> 
> It's not very updated, but after reading it a bit seems easy to add
> the new core parameters and functions so that it works with latest
> version of OpenSIPS.
> 
> Regards,
> 
> 


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[OpenSIPS-Users] IDE

[Ghaith ALKAYYEM]

Hello,

Could anybody suggest an IDE that will ease the development & debugging
of OpenSIPS?

Thank you


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Re: [OpenSIPS-Users] struct sip_msg

[Ghaith ALKAYYEM]

Thank you very much for these valuable information, actually I'm trying
to learn how to develop new module, and i expected that those fields
should be filled upon the receipt of any external message.
Regards.

On Wed, 2009-09-02 at 15:37 +0300, Anca Vamanu wrote:
> Hi Ghaith,
> 
> You must explicitly call parse_headers for the message to be parsed and 
> for the fields in sip_msg to be filled.
> Example:
> parse_headers(msg,HDR_EOH_F, 0); -  will parse all headers.
> 
> However beware that this function will not parse the headers value also. 
> You must call the parse function for that header and it will fill the 
> 'parsed' filed of the struct hdr_field with a structure specific for 
> that filed
> Example: If you want to parse the value of the Contact header filed, you 
> call
> parse_contact(msg->contact).
> and it will fill the parsed filed with a contact_body_t structure that 
> contains the parsed value of the Contact header.
> (contact_body_t* )msg->contact->parsed;
> 
> I suggest learning by example technique :), look in other modules that 
> use the parser and see how it is done there. One option is function 
> *extract_sdialog_info* function from modules/presence/subscribe.c file.
> 
> Regards,
> Anca
> 
> Ghaith ALKAYYEM wrote:
> > Hello list,
> >
> > I was trying to play with the SIP header, So when i tried to access the
> > fields (to,from) in the sip_msg structure through a module c function
> > they were NULL and everything was included in the field headers.
> > I'd like to know whether there's something wrong or it's natural for
> > these fields to be NULL.
> >
> > struct sip_msg {
> > unsigned int id;   /* message id, unique/process*/
> > struct msg_start first_line;   /* Message first line */
> > struct via_body* via1; /* The first via */
> > struct via_body* via2; /* The second via */
> > struct hdr_field* headers; /* All the parsed headers*/
> > struct hdr_field* last_header; /* Pointer to the last parsed header*/
> > hdr_flags_t parsed_flag;   /* Already parsed header field types */
> >
> > /* Via, To, CSeq, Call-Id, From, end of header*/
> > /* pointers to the first occurrences of these headers;
> >  * everything is also saved in 'headers' (see above)
> >  */
> >
> > /* shorcuts to known headers */
> > struct hdr_field* h_via1;
> > struct hdr_field* h_via2;
> > struct hdr_field* callid;
> > struct hdr_field* to;
> > struct hdr_field* cseq;
> > struct hdr_field* from;
> > ...
> > ...
> >
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
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[OpenSIPS-Users] struct sip_msg

[Ghaith ALKAYYEM]

Hello list,

I was trying to play with the SIP header, So when i tried to access the
fields (to,from) in the sip_msg structure through a module c function
they were NULL and everything was included in the field headers.
I'd like to know whether there's something wrong or it's natural for
these fields to be NULL.

struct sip_msg {
unsigned int id;   /* message id, unique/process*/
struct msg_start first_line;   /* Message first line */
struct via_body* via1; /* The first via */
struct via_body* via2; /* The second via */
struct hdr_field* headers; /* All the parsed headers*/
struct hdr_field* last_header; /* Pointer to the last parsed header*/
hdr_flags_t parsed_flag;   /* Already parsed header field types */

/* Via, To, CSeq, Call-Id, From, end of header*/
/* pointers to the first occurrences of these headers;
 * everything is also saved in 'headers' (see above)
 */

/* shorcuts to known headers */
struct hdr_field* h_via1;
struct hdr_field* h_via2;
struct hdr_field* callid;
struct hdr_field* to;
struct hdr_field* cseq;
struct hdr_field* from;
...
...


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Re: [OpenSIPS-Users] SIP Trunking

[Ghaith ALKAYYEM]

Hi,

Is it possible also to make bridging dependent on a variable value by
passing a variable as a parameter to force_send_socket() as following:

$var(a) = "x.x.x.x:xx";
force_send_socket("$var(a)");

because the above configuration gave me an error but when I used the
variable in xlog function it was okay:
xlog("$var(a)");

I might do some code modification in this regard.

Regards.

On Mon, 2009-08-24 at 18:03 +0300, Bogdan-Andrei Iancu wrote:
> Hi Matthew,
> 
> There 2 things when comes bridging:
> 
> 1) signalling part - selecting the proper outbound interface (private or 
> public)
> a) this can be automatically done by opensips (based on the 
> destination IP) if you enable the mhomed parameter in core ; this is 
> simple by not so efficient
> 
> b) you can do it manually, by selecting from script the correct 
> interface - see the force_send_socket() function
> 
> 2) media part
>  a) rtpproxy - when enabling RTPproxy (at request and reply time) 
> you can explicitly select which interface to use (see the e and i flags 
> - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362)
> 
> 
> Best regards,
> Bogdan
> 
> Matthew S. Crocker wrote:
> > Hello,
> >
> >  I'm brand new to OpenSIPS, just going through the make process now.  
> >
> >  I need to configure OpenSIPS to act like a SBC for some SIP trunks coming 
> > off a VoIP switch.  Where should I look for Documentation/Examples of a 
> > working config?
> >
> > Here is my scenario:
> >
> > OpenSIPS has two interfaces,  private & public.  
> > VoIP Gateway is on private LAN with no gateway configured (it can only talk 
> > to local machines, no routing)
> >
> > End user has an Asterisk server on a private lan behind their firewall (NAT)
> >
> > I need to configure OpenSIPS to listen for SIP messages on :5060 from the 
> > end user firewall.  It then need to rewrite the SIP message and send it to 
> > the Gateway.  The Gateway would see the messages coming from the internal 
> > IP of the OpenSIPS server.  Once all of the SIP messages get processed I 
> > then need the OpenSIPS server to proxy the RTP streams (plan on using 
> > mediaproxy) between the Asterisk server and VoIP Gateway.
> >
> > Any helpful hints on where to look?
> >
> > -Matt
> >
> >
> >   
> 
> 
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Re: [OpenSIPS-Users] SIP Trunking

[Ghaith ALKAYYEM]

Hello,

I tried to use mediaproxy, it includes two softwares (dispatcher &
relay), I tried a lot to run more than one relay on the same server in
order to bind them to different interfaces. But unfortunately this
didn't work and I think it's not possible.
I recommend using RTPProxy which is designed to work in bridging mode
between two networks and you can run multiple instance of RTPProxy on
the same server.

Regards.


On Thu, 2009-08-20 at 15:48 -0400, Matthew S. Crocker wrote:
> Can mediaproxy glue two RTP streams together (CallerA to CallerB)?
> Can mediaproxy glue two RTP streams together from different interfaces/IPs 
> (eth0 & eth1) ?
> 
> If so then it should be able to glue two calls together between public IP 
> (eth0) and private IP (eth1).
> If the two RTP streams have to be on the same interface for mediaproxy to 
> work then I would expect to run into issues.
> 
> EndUser <-> (eth0) MediaProxy (eth1) <-> SIP Gateway
> 
> 
> - "Jeff Pyle"  wrote:
> 
> > From: "Jeff Pyle" 
> > To: "OpenSIPS users mailling list" 
> > Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern
> > Subject: Re: [OpenSIPS-Users] SIP Trunking
> >
> > Matthew,
> > 
> > While I'm no Mediaproxy expert, I have seen many conversations on this
> > list
> > where Mediaproxy is described as a part of a far-end NAT solution.  It
> > was
> > not designed to have a private IP attached to it.  For that, you most
> > likely
> > will want to look at the rtpproxy application.
> > 
> > It sounds like you are constructing a local ALG to connect private
> > and
> > public networks.  You don't necessarily need a full-blown Acme for
> > that.
> > I've had great luck with Edgewater Networks' "Edgemarc" devices, for
> > example.  That's just one.  There are many.
> > 
> > 
> > - Jeff
> > 
> > 
> > 
> > On 8/20/09 2:49 PM, "Matthew S. Crocker" 
> > wrote:
> > 
> > > 
> > > I understand that OpenSIPS is not a full blown SBC (I can't afford
> > an
> > > ACMEPacket).  Will it perform the functions to proxy the SIP & RTP
> > streams
> > > (via mediaproxy) between my end users and my internal gateway?
> > > 
> > > At some point I plan on increasing the use of openSIPS to handle
> > registration,
> > > presence, routing, etc.
> > > 
> > > -Matt
> > > 
> > > - "Alex Balashov"  wrote:
> > > 
> > >> From: "Alex Balashov" 
> > >> To: "OpenSIPS users mailling list" 
> > >> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada
> > Eastern
> > >> Subject: Re: [OpenSIPS-Users] SIP Trunking
> > >> 
> > >> Matthew,
> > >> 
> > >> Look for the mediaproxy module.
> > >> 
> > >> That said, do be aware that a proxy is, by definition, not like an
> > >> SBC. 
> > >>   SBCs have many other capabilities a proxy does not;  a proxy is
> > a
> > >> relatively "thin" interoperation layer.
> > >> 
> > >> Perhaps the recently introduced b2bua module is brought to bear on
> > >> that 
> > >> somewhat, but classically, OpenSIPS is a proxy.
> > >> 
> > >> -- Alex
> > >> 
> > >> Matthew S. Crocker wrote:
> > >> 
> > >>> Hello,
> > >>> 
> > >>>  I'm brand new to OpenSIPS, just going through the make process
> > now.
> > >>  
> > >>> 
> > >>>  I need to configure OpenSIPS to act like a SBC for some SIP
> > trunks
> > >> coming off a VoIP switch.  Where should I look for
> > >> Documentation/Examples of a working config?
> > >>> 
> > >>> Here is my scenario:
> > >>> 
> > >>> OpenSIPS has two interfaces,  private & public.
> > >>> VoIP Gateway is on private LAN with no gateway configured (it can
> > >> only talk to local machines, no routing)
> > >>> 
> > >>> End user has an Asterisk server on a private lan behind their
> > >> firewall (NAT)
> > >>> 
> > >>> I need to configure OpenSIPS to listen for SIP messages on :5060
> > >> from the end user firewall.  It then need to rewrite the SIP
> > message
> > >> and send it to the Gateway.  The Gateway would see the messages
> > coming
> > >> from the internal IP of the OpenSIPS server.  Once all of the SIP
> > >> messages get processed I then need the OpenSIPS server to proxy
> > the
> > >> RTP streams (plan on using mediaproxy) between the Asterisk server
> > and
> > >> VoIP Gateway.
> > >>> 
> > >>> Any helpful hints on where to look?
> > >>> 
> > >>> -Matt
> > >>> 
> > >>> 
> > >> 
> > >> 
> > >> -- 
> > >> Alex Balashov - Principal
> > >> Evariste Systems
> > >> Web : http://www.evaristesys.com/
> > >> Tel : (+1) (678) 954-0670
> > >> Direct  : (+1) (678) 954-0671
> > >> 
> > >> ___
> > >> Users mailing list
> > >> Users@lists.opensips.org
> > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
> > 
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: [OpenSIPS-Users] SIP Trunking

[Ghaith ALKAYYEM]

It's possible also to use RTPproxy, because it's designed to work in
bridging mode where it can forward the traffic from the internal network
towards external one.

Regards

On Thu, 2009-08-20 at 12:55 -0600, Darren Sessions wrote:
> Media Proxy will work with public internet addresses but, and this is  
> just my understanding, was not built for use with private IPs let  
> alone bridging from one subnet to another.
> 
> If your keeping the gateways on a private network is for security  
> purposes, you may consider giving them public ips on the same subnet  
> as your opensips and mediaproxy setup, but not specifying a default  
> gateway.
> 
> Essentially, this would allow the media proxy to do its job relaying  
> the audio, while still preventing 99% of any unwanted traffic to your  
> gateways. Couple that will firehol or some other cool iptables app (or  
> manually configure it if you like) and you'd be sitting pretty secure  
> I would think.
> 
> Really depends on what you've designed (and why).
> 
> - Darren
> 
> 
> On Aug 20, 2009, at 12:49 PM, Matthew S. Crocker wrote:
> 
> >
> > I understand that OpenSIPS is not a full blown SBC (I can't afford  
> > an ACMEPacket).  Will it perform the functions to proxy the SIP &  
> > RTP streams (via mediaproxy) between my end users and my internal  
> > gateway?
> >
> > At some point I plan on increasing the use of openSIPS to handle  
> > registration, presence, routing, etc.
> >
> > -Matt
> >
> > - "Alex Balashov"  wrote:
> >
> >> From: "Alex Balashov" 
> >> To: "OpenSIPS users mailling list" 
> >> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada  
> >> Eastern
> >> Subject: Re: [OpenSIPS-Users] SIP Trunking
> >>
> >> Matthew,
> >>
> >> Look for the mediaproxy module.
> >>
> >> That said, do be aware that a proxy is, by definition, not like an
> >> SBC.
> >>  SBCs have many other capabilities a proxy does not;  a proxy is a
> >> relatively "thin" interoperation layer.
> >>
> >> Perhaps the recently introduced b2bua module is brought to bear on
> >> that
> >> somewhat, but classically, OpenSIPS is a proxy.
> >>
> >> -- Alex
> >>
> >> Matthew S. Crocker wrote:
> >>
> >>> Hello,
> >>>
> >>> I'm brand new to OpenSIPS, just going through the make process now.
> >>
> >>>
> >>> I need to configure OpenSIPS to act like a SBC for some SIP trunks
> >> coming off a VoIP switch.  Where should I look for
> >> Documentation/Examples of a working config?
> >>>
> >>> Here is my scenario:
> >>>
> >>> OpenSIPS has two interfaces,  private & public.
> >>> VoIP Gateway is on private LAN with no gateway configured (it can
> >> only talk to local machines, no routing)
> >>>
> >>> End user has an Asterisk server on a private lan behind their
> >> firewall (NAT)
> >>>
> >>> I need to configure OpenSIPS to listen for SIP messages on :5060
> >> from the end user firewall.  It then need to rewrite the SIP message
> >> and send it to the Gateway.  The Gateway would see the messages  
> >> coming
> >> from the internal IP of the OpenSIPS server.  Once all of the SIP
> >> messages get processed I then need the OpenSIPS server to proxy the
> >> RTP streams (plan on using mediaproxy) between the Asterisk server  
> >> and
> >> VoIP Gateway.
> >>>
> >>> Any helpful hints on where to look?
> >>>
> >>> -Matt
> >>>
> >>>
> >>
> >>
> >> -- 
> >> Alex Balashov - Principal
> >> Evariste Systems
> >> Web : http://www.evaristesys.com/
> >> Tel : (+1) (678) 954-0670
> >> Direct  : (+1) (678) 954-0671
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > -- 
> > Matthew S. Crocker
> > President
> > Crocker Communications, Inc.
> > PO BOX 710
> > Greenfield, MA 01302-0710
> > http://www.crocker.com
> > P: 413-746-2760
> >
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
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Re: [OpenSIPS-Users] AVP/ Programing Documentation

[Ghaith ALKAYYEM]

I totally agree with you. I'm really sorry for that.


On Wed, 2009-08-19 at 12:11 -0500, Khan wrote:
> Ghaith,
> 
> That would be violation of copyright law won't it or else we wont buy
> this book for $40+
> Authour has putin hardwork in putting it together, dont you think he
> should benefit a little ?
> 
> After all OpenSIPS is free but not everything else lol...
> 
> On Wed, Aug 19, 2009 at 11:41 AM, Ghaith
> ALKAYYEM wrote:
> > Hi,,
> > I think this book is available through rapidshare in the following link:
> >

> >
> >
> >
> > On Wed, 2009-08-19 at 11:26 -0500, osiris123d wrote:
> >> Also a good idea is to read Flavio E. Goncalves book "
> >> http://www.packtpub.com/building-telephony-systems-with-openser/book
> >> Building Telephony systems with OpenSER ".  Though the book does deal with
> >> an old version of OpenSER it still has good documentation and examples.
> >>
> >> There is also a section on AVPOPS
> >>
> >>
> >>
> >> roger wilbert wrote:
> >> >
> >> > Can anyone point me to documentation other than the module docs that can
> >> > explain how to AVP?  Not that the module docs don’t provide good
> >> > information. But it assumes knowledge that is missing from someone who is
> >> > not readily familiar with Opensips.
> >> >
> >> > ___
> >> > Users mailing list
> >> > Users@lists.opensips.org
> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >> >
> >> >
> >>
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> 
> 
> 


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Re: [OpenSIPS-Users] AVP/ Programing Documentation

[Ghaith ALKAYYEM]

Hi,,
I think this book is available through rapidshare in the following link:

http://rs211.rapidshare.com/files/143531736/Building_Telephony_Systems_with_Openser.pdf



On Wed, 2009-08-19 at 11:26 -0500, osiris123d wrote:
> Also a good idea is to read Flavio E. Goncalves book "
> http://www.packtpub.com/building-telephony-systems-with-openser/book
> Building Telephony systems with OpenSER ".  Though the book does deal with
> an old version of OpenSER it still has good documentation and examples.
> 
> There is also a section on AVPOPS
> 
>  
> 
> roger wilbert wrote:
> > 
> > Can anyone point me to documentation other than the module docs that can
> > explain how to AVP?  Not that the module docs don’t provide good
> > information. But it assumes knowledge that is missing from someone who is
> > not readily familiar with Opensips.
> > 
> > ___
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
> > 
> 


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[OpenSIPS-Users] rtpproxy from external to internal network

[Ghaith ALKAYYEM]

Hello,

I'm trying to test rtpproxy with nathelper module to forward RTP traffic
between two networks, my architecture is as follow:




-----

UAC (172.23.1.22) |||
| UAC (172.1.1.21) |
  |===>| 172.23.1.20 (OpenSIPS) 172.1.1.166 |
===>|  |
Internal Network  |||
| External Network |
-----






I run rtpproxy:
rtpproxy -fF -l 172.1.1.166/172.23.1.20
Then I tried to establish a call from internal net (172.23.1.22) to
external net(172.1.1.21) and the connection established succussfully
with perfect RTP relay through (172.1.1.166) interface.

But when I tried to call from external network towards internal, the RTP
traffic relay didn't work?.

Following is the part related to nathelper and routing logic in my
configuration file:



loadmodule "nathelper.so"

modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
modparam("nathelper", "received_avp", "$avp(i:42)")
modparam("registrar", "received_avp", "$avp(i:42)")

modparam("usrloc", "nat_bflag", 6)

modparam("registrar", "method_filtering", 1)

mhomed=1

route {

  # -
  # Sanity Check Section
  # -
  if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
#break;
return();
  };

  if (msg:len > max_len) {
sl_send_reply("513", "Message Overflow");
#break;
return();
  };

  # -
  # Record Route Section
  # -
  if (method!="REGISTER") {
record_route();
  };

  if (method=="BYE" || method=="CANCEL") {
unforce_rtp_proxy();
  }

  # -
  # Loose Route Section
  # -

  if (loose_route()) {

if ((method=="INVITE" || method=="REFER") && !has_totag()) {
  sl_send_reply("403", "Forbidden");
  #break;
return();
};

if (method=="INVITE") {

  #if (!proxy_authorize("","subscriber")) {
  #  proxy_challenge("","0");
  #  break;
  #} else if (!check_from()) {
  #  sl_send_reply("403", "Use From=ID");
  #  break;
  #};

  #consume_credentials();

  if (nat_uac_test("19")) {
setflag(6);
force_rport();
fix_nated_contact();
  };
  force_rtp_proxy("l");
};

route(1);
#break;
return();
  };


  # -
  # Call Type Processing Section
  # -
  if (uri!=myself) {
route(4);
route(1);
#break;
return();
  };

  if (method=="ACK") {
route(1);
#break;
return();
  } if (method=="CANCEL") {
route(1);
#break;
return();
  } else if (method=="INVITE") {
route(3);
#break;
return();
  } else  if (method=="REGISTER") {
route(2);
#break;
return();
  };

  lookup("aliases");
  if (uri!=myself) {
route(4);
route(1);
#break;
return();

  };

  if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
#break;
return();
  };

  route(1);
}


route[1] {

  # -
  # Default Message Handler
  # -

  t_on_reply("1");

  if (!t_relay()) {
if (method=="INVITE" && isflagset(6)) {
  unforce_rtp_proxy();
};
sl_reply_error();
  };
}

route[2] {

  # -
  # REGISTER Message Handler
  # 

  if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) {
setflag(6);
fix_nated_register();
force_rport();
  };

  sl_send_reply("100", "Trying");

  #if (!www_authorize("","subscriber")) {
  #  www_challenge("","0");
  #  break;
  #};

  #if (!check_to()) {
  #  sl_send_reply("401", "Unauthorized");
  #  break;
  #};

  #   consume_credentials();

  if (!save("location")) {
sl_reply_error();
  };
}

route[3] {

  # -
  # INVITE Message Handler
  # -

  #if (!proxy_authorize("","subscriber")) {
  #  proxy_challenge("","0");
  #  break;
  #} else if (!check_from()) {
  #  sl_send_reply("403", "Use From=ID");
  #  break;
  #};

  #   consume_credentials();

  

[OpenSIPS-Users] mediaproxy relay

[Ghaith ALKAYYEM]

Hi,
Could you tell me what is happening to mediaproxy relay when I'm trying
to run it?
The relay is not starting and this message appears:

Set resource limit for maximum open file descriptors to 11000
debug: Adding new dispatcher at 
O j: operation is not possible without initialized secure memory
Aborted



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[OpenSIPS-Users] media proxy

[Ghaith ALKAYYEM]

Hi,

Is it possible to define multiple relays on the same server that runs
Opensips + Mediaproxy.

I tried to define two interfaces in the configuration file, like:

relay_ip = 1.1.1.1 2.2.2.2
or
relay_ip = 1.1.1.1
relay_ip = 2.2.2.2

and I tried also to force the traffic to go through specific interface
by calling: $avp(s:media_relay) = "1.1.1.1";
and the traffic goes always through one interface.

Regards.



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Re: [OpenSIPS-Users] PROBLEM: Opensips stops replying to SIP packets

[Ghaith ALKAYYEM]

Yeah,
I agree with you this also happened with me in less than one day of
continuous working.

On Sat, 2009-08-08 at 19:59 -0700, James Lamanna wrote:
> Hi,
> I'm running the svn 1.5 branch of opensips.
> I've noticed after some amount of time, usually a day or so, opensips
> just completely stops
> responding to incoming SIP requests, REGISTER, NOTIFY, etc...
> The only way to recover from this is to restart opensips.
> 
> In fact it seems to stop doing anything, including writing debug output.
> 
> Thank you.
> 
> -- JAmes
> 
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> 


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[OpenSIPS-Users] load balancing in opensips

[Ghaith ALKAYYEM]

Hi,

I'm trying to connect OpenSIP to 3 WAN links, and I want to distribute
the traffic between those links according to the dialing number. So, I
tried some modules like lcr and load_balance but the test didn't succeed
because those modules are forwarding the traffic to another SIP gateway
or media gateway.

I tried to change the outbound interface via force_send_socket and
setting mhomed=1 and OpenSIPS still connect through the main interface.

The IPs of the system are:
172.23.1.20
172.23.1.19
172.23.1.18
and the originating IP is always 172.23.1.20

please find a part of my configuration file:




# main request routing logic

mhomed=1

route{

if($ct=~"2000.*")
{
   #load_balance("2","conf");
   force_send_socket(172.23.1.19:5060);
};
if($ct=~"3000.*")
{
   #load_balance("1","conf");
   force_send_socket(172.23.1.18:5060);
};

   if (!mf_process_maxfwd_header("10")) {
   sl_send_reply("483","Too Many Hops");
   exit;
   }

   if (has_totag()) {
   # sequential request withing a dialog should
   # take the path determined by record-routing
   if (loose_route()) {
   if (is_method("BYE")) {
   setflag(1); # do accounting ...
   setflag(3); # ... even if the transaction
fails
   } else if (is_method("INVITE")) {
   # even if in most of the cases is
useless, do RR for
   # re-INVITEs alos, as some buggy clients
do change route set
   # during the dialog.
   record_route();
   }
   # route it out to whatever destination was set by
loose_route()
   # in $du (destination URI).
   route(1);
   } else {
   /* uncomment the following lines if you want to
enable presence */
   ##if (is_method("SUBSCRIBE") && $rd ==
"your.server.ip.address") {
   ##  # in-dialog subscribe requests
   ##  route(2);
   ##  exit;
   ##}
   if ( is_method("ACK") ) {
   if ( t_check_trans() ) {
   # non loose-route, but stateful
ACK; must be an ACK after
   # a 487 or e.g. 404 from upstream
server
   t_relay();
   exit;
   } else {
   # ACK without matching
transaction ->
   # ignore and discard
   exit;
   }
   }
   sl_send_reply("404","Not here");
   }
   exit;
   }

   #initial requests

   # CANCEL processing
   if (is_method("CANCEL"))
   {
   if (t_check_trans())
   t_relay();
   exit;
   }

   t_check_trans();

   # authenticate if from local subscriber (uncomment to enable
auth)
   # authenticate all initial non-REGISTER request that pretend to
be

   # preloaded route checking
   if (loose_route()) {
   xlog("L_ERR",
   "Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]");
   if (!is_method("ACK"))
   sl_send_reply("403","Preload Route denied");
   exit;
   }
   # record routing
   if (!is_method("REGISTER|MESSAGE"))
   record_route();

   # account only INVITEs
   if (is_method("INVITE")) {
   setflag(1); # do accounting
   }
   if (!uri==myself)
   ## replace with following line if multi-domain support is used
   ##if (!is_uri_host_local())
   {
   append_hf("P-hint: outbound\r\n");
   # if you have some interdomain connections via TLS
   ##if($rd=="tls_domain1.net") {
   ##  t_relay("tls:domain1.net");
   ##  exit;
   ##} else if($rd=="tls_domain2.net") {
   ##  t_relay("tls:domain2.net");
   ##  exit;
   ##}
   route(1);
   }

   # requests for my domain

   ## uncomment this if you want to enable presence server
   ##   and comment the next 'if' block
   ##   NOTE: uncomment also the definition of route[2] from  below
   ##if( is_method("PUBLISH|SUBSCRIBE"))
   ##  route(2);

   if (is_method("PUBLISH"))
   {
   sl_send_reply("503", "Service Unavailable");
   exit;
   }

   if (is_method("REGISTER"))
   {
   # authenticate the REGISTER requests (un

[OpenSIPS-Users] Just as a proxy server

[Ghaith ALKAYYEM]

Hello,

Does anybody know how we can cancel the authentication functionality in
OpenSIPS and make it run as a proxy server & router.

Thank you



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Re: [OpenSIPS-Users] Trouble with perl module

[Ghaith ALKAYYEM]

Hi;
The same thing happened to me, But when i tried to call the script
through the configuration file, it was okay and executed successfully.


On Mon, 2009-07-13 at 13:55 +0400, M C wrote:
> Hello,
> 
> I ve installed perl module. Also, i copied directory with perl lib
> OpenSIPS.pm to /usr/lib/perl. But when i am trying to execute a simple
> code:
> use OpenSIPS::Message;
> print "Test\n";
> 
> or some of examples, line functions.pl, i have error:
> 
> Can't locate object method "bootstrap" via package "OpenSIPS"
> at /usr/lib/perl/5.10/OpenSIPS/
> Message.pm line 32.
> 
> How can i fix it?
> 
> -- 
> Best regards, Maksim.
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[OpenSIPS-Users] perl module

[Ghaith ALKAYYEM]

Hi guys,

I'm trying to use the perl modules available with OpenSIPS, but after
installing it I encountered a problem when i tried to compile a script:

Can't locate object method "bootstrap" via package "OpenSIPS"
at /usr/lib/perl5/OpenSIPS/Message.pm line 32.
Compilation failed in require at /usr/lib/perl5/OpenSIPS.pm line 34.
BEGIN failed--compilation aborted at /usr/lib/perl5/OpenSIPS.pm line 34.
Compilation failed in require at lib/opensips/perl/functions.pl line 4.
BEGIN failed--compilation aborted at lib/opensips/perl/functions.pl line
4.

does anybody have any idea about this issue?
Thanks in advance.



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