Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as INVITE?

[Tuomas Kaikkonen]

The dispatcher works now. I had forgot some iptables NAT rules from previous 
testing, and had to flush all my iptables NAT rules. The source IP is now 
correct.

Tuomas Kaikkonen
Software Developer | Twisted Pair Solutions
3131 Elliott Ave, Suite 200, Seattle, WA 98121
Tel: (206) 812-0732

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tuomas Kaikkonen
Sent: Tuesday, March 19, 2013 4:49 PM
To: OpenSIPS users mailling list; Bogdan-Andrei Iancu
Subject: Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as 
INVITE?

Trying dispatcher:


mysql> select * from dispatcher;
++---+---+---++---+-+
| id | setid | destination   | flags | weight | attrs | 
description |
++---+---+---++---+-+
|  1 | 1 | sip:WWW.WWW.WWW.WWW:5060;transport=udp  | 0 |  1 |   
| |
|  2 | 1 | sip:XXX.XXX.XXX.XXX:5060;transport=udp | 0 |  1 |   
| |
++---+---+---++---+-+

The WWW and XXX are IP addresses in my actual database, just scrubbing them for 
posting to this mailing group.

My opensips.cfg route for the dispatching opensips is this:

route{
if ( !mf_process_maxfwd_header("10") )
{
send_reply("483","To Many Hops");
exit;
};

if ( !ds_select_dst("1", "0") ) {
send_reply("500","Unable to route");
exit;
}

t_relay();
}


The database is shared by the three opensips servers. When I try to register my 
client with the SIP domain as the IP address of the dispatching OpenSIPs  
server as the SIP proxy on the client, the registration request times out. I do 
NOT see the destination OpenSIPs servers getting the REGISTER at all.

I see in tcpdump the client's register is received, then the dispatching 
OpenSIPs sends it to one of the two OpenSIPs servers configured in the 
dispatcher table, but when it is sending them the IP packets Source IP is 
marked as 192.168.1.1  -- which is NOT the internal or public IP address of 
this system. Somehow the OpenSIPs dispatcher will overwrite the source IP with 
"192.168.1.1" - is there a configuration that I should have to make it use its 
advertised_address?





Tuomas Kaikkonen
Software Developer | Twisted Pair Solutions
3131 Elliott Ave, Suite 200, Seattle, WA 98121
Tel: (206) 812-0732

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org]
 On Behalf Of Tuomas Kaikkonen
Sent: Tuesday, March 19, 2013 10:43 AM
To: Bogdan-Andrei Iancu; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as 
INVITE?

Bogdan,

What I am trying to do is this:  Have two or more OpenSIPs servers with each 
having their own rtpproxy server. The database is shared over ssh tunnel so it 
looks to each server as they are using a local database. The goal is to make 
the service scalable so that we can add more OpenSIPs hosts as the need comes. 
Also we want to have the system detect if a server has failed and not direct 
calls/registers to that server. All the servers are in a cloud system (which 
does not provide Multicast, only Unicast), and they will have both public IP 
and internal IP associated with them. The problem with the shared database is 
that the location table has the socket column storing the Internal IP of the 
OpenSIPs server who did the REGISTER of the SIP client with. I need to relay 
all INVITE/BYE/CANCEL requests via that server. I have no UDP load balancer to 
balance the SIP requests, and having the RTP Proxy proxy all audio packets is 
an extra burden to the design.  Our clients register with only one SIP server, 
there is no configuration to register with a primary and secondary SIP proxy.

What we worry about is putting one OpenSIPs server acting as a load balancer or 
dispatcher, is that we'd be then again dependent on one OpenSIPs server. That 
dispatching / load balancing OpenSIPs server should be also backed up by a fail 
over server. I've looked into some, mostly academic, papers on how they did 
their load balancing and fail over.

Tuomas Kaikkonen
Software Developer | Twisted Pair Solutions
3131 Elliott Ave, Suite 200, Seattle, WA 98121
Tel: (206) 812-0732

From: Bogdan-Andrei Iancu 
[mailto:bog...@opensips.org]
Sent: Tuesday, March 19, 2013 8:01 AM
To: OpenSIPS users mailling list
Cc: Tuomas Kaikkonen
Subject: Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as 
INVITE?

Hi Tuomas,

The LB modules is only for CALLs - it understands by +1 load only a call. You 
cannot use it for REGISTERsAnyhow REGISTERs and INVITEs a

Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as INVITE?

[Tuomas Kaikkonen]

Trying dispatcher:


mysql> select * from dispatcher;
++---+---+---++---+-+
| id | setid | destination   | flags | weight | attrs | 
description |
++---+---+---++---+-+
|  1 | 1 | sip:WWW.WWW.WWW.WWW:5060;transport=udp  | 0 |  1 |   
| |
|  2 | 1 | sip:XXX.XXX.XXX.XXX:5060;transport=udp | 0 |  1 |   
| |
++---+---+---++---+-+

The WWW and XXX are IP addresses in my actual database, just scrubbing them for 
posting to this mailing group.

My opensips.cfg route for the dispatching opensips is this:

route{
if ( !mf_process_maxfwd_header("10") )
{
send_reply("483","To Many Hops");
exit;
};

if ( !ds_select_dst("1", "0") ) {
send_reply("500","Unable to route");
exit;
}

t_relay();
}


The database is shared by the three opensips servers. When I try to register my 
client with the SIP domain as the IP address of the dispatching OpenSIPs  
server as the SIP proxy on the client, the registration request times out. I do 
NOT see the destination OpenSIPs servers getting the REGISTER at all.

I see in tcpdump the client's register is received, then the dispatching 
OpenSIPs sends it to one of the two OpenSIPs servers configured in the 
dispatcher table, but when it is sending them the IP packets Source IP is 
marked as 192.168.1.1  -- which is NOT the internal or public IP address of 
this system. Somehow the OpenSIPs dispatcher will overwrite the source IP with 
"192.168.1.1" - is there a configuration that I should have to make it use its 
advertised_address?





Tuomas Kaikkonen
Software Developer | Twisted Pair Solutions
3131 Elliott Ave, Suite 200, Seattle, WA 98121
Tel: (206) 812-0732

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tuomas Kaikkonen
Sent: Tuesday, March 19, 2013 10:43 AM
To: Bogdan-Andrei Iancu; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as 
INVITE?

Bogdan,

What I am trying to do is this:  Have two or more OpenSIPs servers with each 
having their own rtpproxy server. The database is shared over ssh tunnel so it 
looks to each server as they are using a local database. The goal is to make 
the service scalable so that we can add more OpenSIPs hosts as the need comes. 
Also we want to have the system detect if a server has failed and not direct 
calls/registers to that server. All the servers are in a cloud system (which 
does not provide Multicast, only Unicast), and they will have both public IP 
and internal IP associated with them. The problem with the shared database is 
that the location table has the socket column storing the Internal IP of the 
OpenSIPs server who did the REGISTER of the SIP client with. I need to relay 
all INVITE/BYE/CANCEL requests via that server. I have no UDP load balancer to 
balance the SIP requests, and having the RTP Proxy proxy all audio packets is 
an extra burden to the design.  Our clients register with only one SIP server, 
there is no configuration to register with a primary and secondary SIP proxy.

What we worry about is putting one OpenSIPs server acting as a load balancer or 
dispatcher, is that we'd be then again dependent on one OpenSIPs server. That 
dispatching / load balancing OpenSIPs server should be also backed up by a fail 
over server. I've looked into some, mostly academic, papers on how they did 
their load balancing and fail over.

Tuomas Kaikkonen
Software Developer | Twisted Pair Solutions
3131 Elliott Ave, Suite 200, Seattle, WA 98121
Tel: (206) 812-0732

From: Bogdan-Andrei Iancu 
[mailto:bog...@opensips.org]
Sent: Tuesday, March 19, 2013 8:01 AM
To: OpenSIPS users mailling list
Cc: Tuomas Kaikkonen
Subject: Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as 
INVITE?

Hi Tuomas,

The LB modules is only for CALLs - it understands by +1 load only a call. You 
cannot use it for REGISTERsAnyhow REGISTERs and INVITEs are as apples and 
onions :)...so you cannot put them in the same basket.

if you could provide more details on what you are trying to achieve, maybe I 
can advice you on the best balancing option (like maybe using dispatcher module)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 03/18/2013 07:56 PM, Tuomas Kaikkonen wrote:
Can the Load Balancer module be configured to balance REGISTERs as well as the 
INVITES so that the above mentioned setup would work? OR is the Load Balancer 
module just useful for balancing RTP Proxy / media server resources for INVITEs?

I'm new to the Load Balancer module of OpenSIPs. I a

Re: [OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Nick Khamis]

If you want to learn SIP, learn OpenSIPS :). Good luck on your project!

N.

On 3/19/13, Pink Cupcake  wrote:
> You're in the ballpark, Nick. I'm working on a design for a new system that
> includes a SIP component, and while I'd prefer if the end-to-end solution
> stuck with UDP, there are some technical reasons why TCP may be required,
> and it's not something I'd be able to work around.
>
> So far OpenSIPS has been relatively easy to work with and so I'm continuing
> my experiments with it, while a colleague is investigating Kamailio. I may
> stick with OpenSIPS irrespective of what my colleague goes ahead with,
> simply because I want a SIP server solution that will be the easiest to use
> in an automated testing environment -- my goal being to test a SIP user
> agent and not to care much about the server side.
>
> -PKCK
>
>
> On Tue, Mar 19, 2013 at 3:34 PM, Nick Khamis  wrote:
>
>> Hello PKCK,
>>
>> I am not sure but I can only think of a few reasons why you would like
>> to run SIP on the TCP protocol, and none that are even good reasons as
>> that. Are you sure you need to run OpenSIPS on TCP? UDP should suffice
>> 99% pf the time. I take it you're just experimenting.
>>
>> N.
>>
>> On 3/19/13, Pink Cupcake  wrote:
>> > It looks like I answered my question almost immediately after sending
>> > my
>> > last message (of course!) -- I was invoking opensips thusly:
>> >
>> > ./sbin/opensips -D -f
>> > /path/to/opensips/1.8.2/opensips-with-local-changes.cfg
>> >
>> > I just read that OpenSIPS will not be able to listen on more than one
>> > interface unless it forks, and since it always must listen on UDP, it
>> will
>> > skip listening on TCP if it must.
>> >
>> > It looks like the -D parameter prevents OpenSIPS from forking, which
>> means
>> > no TCP port connection; if I run that command without the -D, TCP
>> > connections seem to work.
>> >
>> > Of course I liked using -D because it made it easy to start/stop
>> > OpenSIPS
>> > (just with a ^C). I'm assuming that "daemon mode" means it's going to
>> leave
>> > a pid file around somewhere to make it easier to skill using a shell
>> > (or
>> > shell script), so I'm off to find out if that's the case.
>> >
>> > -PKCK
>> >
>> >
>> > On Tue, Mar 19, 2013 at 2:45 PM, Pink Cupcake
>> > wrote:
>> >
>> >> Thanks, Nick. I'm having a read through it now.
>> >>
>> >> I fixed my initial problem, though, and it was a pretty silly mistake:
>> >> I
>> >> had OpenSIPS and my SIP client (Jitsi) both using the same port. Oops!
>> >> I
>> >> switched Jitsi over to ports 55060/55061 and then things started
>> working.
>> >>
>> >> Working, at least with UDP. I need to get TCP working now.
>> >>
>> >> The default configuration file has TCP disabled. Searching for
>> >> information
>> >> on how to set up TCP is quite difficult. The PDF you linked to has a
>> >> few
>> >> tips, luckily. But not enough to get it working.
>> >>
>> >> Here's a portion of my config file:
>> >>
>> >> #listen=udp:127.0.0.1:5060   # CUSTOMIZE ME
>> >> listen=udp:172.16.23.79:5060   # CUSTOMIZE ME
>> >> listen=tcp:172.16.23.79:5060   # CUSTOMIZE ME
>> >> #disable_tcp=yes
>> >> disable_tcp=no
>> >> tcp_children=4
>> >>
>> >> When I try to connect with Jitsi, or with telnet, and sniff the
>> >> traffic
>> >> with Wireshark, I can see a SYN sent from Jitsi->OpenSIPS and a
>> >> RST/ACK
>> >> returned, but that's it. I don't think the TCP connection is getting
>> >> properly established, and Jitsi bails out saying it can't create a
>> >> connection.
>> >>
>> >> Jitsi does successfully connect to OpenSIPS if I set the proxy
>> >> transport
>> >> setting to UDP in Jitsi.
>> >>
>> >> What do I need to do to take OpenSIPS, from out of the box, and get it
>> to
>> >> accept REGISTERs/INVITEs on TCP and not UDP?
>> >>
>> >> -PKCK
>> >>
>> >>
>> >>
>> >> On Fri, Feb 8, 2013 at 4:49 PM, Nick Khamis  wrote:
>> >>
>> >>> Pink,
>> >>>
>> >>> A great resource that can get you up and going in at most a week can
>> >>> be found here:
>> >>>
>> >>>
>> >>>
>> >>>
>> http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w
>> >>>
>> >>> Happy Routing!
>> >>>
>> >>> Nick.
>> >>>
>> >>> On 2/8/13, Bogdan-Andrei Iancu  wrote:
>> >>> > Hi,
>> >>> >
>> >>> > On 02/08/2013 12:17 AM, Pink Cupcake wrote:
>> >>> >> Hi Bogdan-Andrei,
>> >>> >>
>> >>> >> Your response really doesn't help me or deal with the questions I
>> >>> >> laid
>> >>> >> out in my original post. Perhaps I was not being clear. Let me
>> >>> >> start
>> >>> >> over.
>> >>> >>
>> >>> > Well, my answer was related to the error logs you posted :).
>> >>> >
>> >>> >>
>> >>> >>
>> >>> >> I need to implement an automated testing scenario on an OS X build
>> >>> >> machine.
>> >>> >>
>> >>> >> The test requires two different SIP UAs -- which are both running
>> >>> >> locally on the sa

Re: [OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Pink Cupcake]

You're in the ballpark, Nick. I'm working on a design for a new system that
includes a SIP component, and while I'd prefer if the end-to-end solution
stuck with UDP, there are some technical reasons why TCP may be required,
and it's not something I'd be able to work around.

So far OpenSIPS has been relatively easy to work with and so I'm continuing
my experiments with it, while a colleague is investigating Kamailio. I may
stick with OpenSIPS irrespective of what my colleague goes ahead with,
simply because I want a SIP server solution that will be the easiest to use
in an automated testing environment -- my goal being to test a SIP user
agent and not to care much about the server side.

-PKCK


On Tue, Mar 19, 2013 at 3:34 PM, Nick Khamis  wrote:

> Hello PKCK,
>
> I am not sure but I can only think of a few reasons why you would like
> to run SIP on the TCP protocol, and none that are even good reasons as
> that. Are you sure you need to run OpenSIPS on TCP? UDP should suffice
> 99% pf the time. I take it you're just experimenting.
>
> N.
>
> On 3/19/13, Pink Cupcake  wrote:
> > It looks like I answered my question almost immediately after sending my
> > last message (of course!) -- I was invoking opensips thusly:
> >
> > ./sbin/opensips -D -f
> > /path/to/opensips/1.8.2/opensips-with-local-changes.cfg
> >
> > I just read that OpenSIPS will not be able to listen on more than one
> > interface unless it forks, and since it always must listen on UDP, it
> will
> > skip listening on TCP if it must.
> >
> > It looks like the -D parameter prevents OpenSIPS from forking, which
> means
> > no TCP port connection; if I run that command without the -D, TCP
> > connections seem to work.
> >
> > Of course I liked using -D because it made it easy to start/stop OpenSIPS
> > (just with a ^C). I'm assuming that "daemon mode" means it's going to
> leave
> > a pid file around somewhere to make it easier to skill using a shell (or
> > shell script), so I'm off to find out if that's the case.
> >
> > -PKCK
> >
> >
> > On Tue, Mar 19, 2013 at 2:45 PM, Pink Cupcake
> > wrote:
> >
> >> Thanks, Nick. I'm having a read through it now.
> >>
> >> I fixed my initial problem, though, and it was a pretty silly mistake: I
> >> had OpenSIPS and my SIP client (Jitsi) both using the same port. Oops! I
> >> switched Jitsi over to ports 55060/55061 and then things started
> working.
> >>
> >> Working, at least with UDP. I need to get TCP working now.
> >>
> >> The default configuration file has TCP disabled. Searching for
> >> information
> >> on how to set up TCP is quite difficult. The PDF you linked to has a few
> >> tips, luckily. But not enough to get it working.
> >>
> >> Here's a portion of my config file:
> >>
> >> #listen=udp:127.0.0.1:5060   # CUSTOMIZE ME
> >> listen=udp:172.16.23.79:5060   # CUSTOMIZE ME
> >> listen=tcp:172.16.23.79:5060   # CUSTOMIZE ME
> >> #disable_tcp=yes
> >> disable_tcp=no
> >> tcp_children=4
> >>
> >> When I try to connect with Jitsi, or with telnet, and sniff the traffic
> >> with Wireshark, I can see a SYN sent from Jitsi->OpenSIPS and a RST/ACK
> >> returned, but that's it. I don't think the TCP connection is getting
> >> properly established, and Jitsi bails out saying it can't create a
> >> connection.
> >>
> >> Jitsi does successfully connect to OpenSIPS if I set the proxy transport
> >> setting to UDP in Jitsi.
> >>
> >> What do I need to do to take OpenSIPS, from out of the box, and get it
> to
> >> accept REGISTERs/INVITEs on TCP and not UDP?
> >>
> >> -PKCK
> >>
> >>
> >>
> >> On Fri, Feb 8, 2013 at 4:49 PM, Nick Khamis  wrote:
> >>
> >>> Pink,
> >>>
> >>> A great resource that can get you up and going in at most a week can
> >>> be found here:
> >>>
> >>>
> >>>
> >>>
> http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w
> >>>
> >>> Happy Routing!
> >>>
> >>> Nick.
> >>>
> >>> On 2/8/13, Bogdan-Andrei Iancu  wrote:
> >>> > Hi,
> >>> >
> >>> > On 02/08/2013 12:17 AM, Pink Cupcake wrote:
> >>> >> Hi Bogdan-Andrei,
> >>> >>
> >>> >> Your response really doesn't help me or deal with the questions I
> >>> >> laid
> >>> >> out in my original post. Perhaps I was not being clear. Let me start
> >>> >> over.
> >>> >>
> >>> > Well, my answer was related to the error logs you posted :).
> >>> >
> >>> >>
> >>> >>
> >>> >> I need to implement an automated testing scenario on an OS X build
> >>> >> machine.
> >>> >>
> >>> >> The test requires two different SIP UAs -- which are both running
> >>> >> locally on the same machine -- to successfully engage in a SIP
> >>> >> call. In order for two UAs to talk to one another, they need to be
> >>> >> registered with a SIP server.
> >>> >>
> >>> >> I am trying to determine if OpenSIPS can be used as the server
> >>> >> component in my testing scenario.
> >>> >>
> >>> >> Since this 

Re: [OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Nick Khamis]

Hello PKCK,

I am not sure but I can only think of a few reasons why you would like
to run SIP on the TCP protocol, and none that are even good reasons as
that. Are you sure you need to run OpenSIPS on TCP? UDP should suffice
99% pf the time. I take it you're just experimenting.

N.

On 3/19/13, Pink Cupcake  wrote:
> It looks like I answered my question almost immediately after sending my
> last message (of course!) -- I was invoking opensips thusly:
>
> ./sbin/opensips -D -f
> /path/to/opensips/1.8.2/opensips-with-local-changes.cfg
>
> I just read that OpenSIPS will not be able to listen on more than one
> interface unless it forks, and since it always must listen on UDP, it will
> skip listening on TCP if it must.
>
> It looks like the -D parameter prevents OpenSIPS from forking, which means
> no TCP port connection; if I run that command without the -D, TCP
> connections seem to work.
>
> Of course I liked using -D because it made it easy to start/stop OpenSIPS
> (just with a ^C). I'm assuming that "daemon mode" means it's going to leave
> a pid file around somewhere to make it easier to skill using a shell (or
> shell script), so I'm off to find out if that's the case.
>
> -PKCK
>
>
> On Tue, Mar 19, 2013 at 2:45 PM, Pink Cupcake
> wrote:
>
>> Thanks, Nick. I'm having a read through it now.
>>
>> I fixed my initial problem, though, and it was a pretty silly mistake: I
>> had OpenSIPS and my SIP client (Jitsi) both using the same port. Oops! I
>> switched Jitsi over to ports 55060/55061 and then things started working.
>>
>> Working, at least with UDP. I need to get TCP working now.
>>
>> The default configuration file has TCP disabled. Searching for
>> information
>> on how to set up TCP is quite difficult. The PDF you linked to has a few
>> tips, luckily. But not enough to get it working.
>>
>> Here's a portion of my config file:
>>
>> #listen=udp:127.0.0.1:5060   # CUSTOMIZE ME
>> listen=udp:172.16.23.79:5060   # CUSTOMIZE ME
>> listen=tcp:172.16.23.79:5060   # CUSTOMIZE ME
>> #disable_tcp=yes
>> disable_tcp=no
>> tcp_children=4
>>
>> When I try to connect with Jitsi, or with telnet, and sniff the traffic
>> with Wireshark, I can see a SYN sent from Jitsi->OpenSIPS and a RST/ACK
>> returned, but that's it. I don't think the TCP connection is getting
>> properly established, and Jitsi bails out saying it can't create a
>> connection.
>>
>> Jitsi does successfully connect to OpenSIPS if I set the proxy transport
>> setting to UDP in Jitsi.
>>
>> What do I need to do to take OpenSIPS, from out of the box, and get it to
>> accept REGISTERs/INVITEs on TCP and not UDP?
>>
>> -PKCK
>>
>>
>>
>> On Fri, Feb 8, 2013 at 4:49 PM, Nick Khamis  wrote:
>>
>>> Pink,
>>>
>>> A great resource that can get you up and going in at most a week can
>>> be found here:
>>>
>>>
>>>
>>> http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w
>>>
>>> Happy Routing!
>>>
>>> Nick.
>>>
>>> On 2/8/13, Bogdan-Andrei Iancu  wrote:
>>> > Hi,
>>> >
>>> > On 02/08/2013 12:17 AM, Pink Cupcake wrote:
>>> >> Hi Bogdan-Andrei,
>>> >>
>>> >> Your response really doesn't help me or deal with the questions I
>>> >> laid
>>> >> out in my original post. Perhaps I was not being clear. Let me start
>>> >> over.
>>> >>
>>> > Well, my answer was related to the error logs you posted :).
>>> >
>>> >>
>>> >>
>>> >> I need to implement an automated testing scenario on an OS X build
>>> >> machine.
>>> >>
>>> >> The test requires two different SIP UAs -- which are both running
>>> >> locally on the same machine -- to successfully engage in a SIP
>>> >> call. In order for two UAs to talk to one another, they need to be
>>> >> registered with a SIP server.
>>> >>
>>> >> I am trying to determine if OpenSIPS can be used as the server
>>> >> component in my testing scenario.
>>> >>
>>> >> Since this is an integration test and doesn't require any state being
>>> >> retained, the test steps look like this: (1) bring up an OpenSIPS
>>> >> server in userspace, (2) have the two SIP UA clients register with
>>> >> that server, as simply as possible, (3) have the two SIP UA clients
>>> >> engage in and then end a SIP call, (4) stop the SIP UA clients, (5)
>>> >> shut down the server.
>>> >>
>>> >> If anything is unclear from the above, please reply back.
>>> > Clear and no issues here,
>>> >
>>> >>
>>> >>
>>> >>
>>> >> Please can someone answer the following:
>>> >>
>>> >> If I can run OpenSIPS in userspace, I would also not like to have it
>>> >> "installed" on the build machine. I used the "prefix" parameter to
>>> >> `make install` into a separate directory and I am attempting to run
>>> >> OpenSIPS from that directory.
>>> >>  It looks like I can run OpenSIPS in userspace. Is that correct?
>>> > true
>>> >>
>>> >> From the documentation is looks like OpenSIPS does not use a databas

Re: [OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Pink Cupcake]

It looks like I answered my question almost immediately after sending my
last message (of course!) -- I was invoking opensips thusly:

./sbin/opensips -D -f
/path/to/opensips/1.8.2/opensips-with-local-changes.cfg

I just read that OpenSIPS will not be able to listen on more than one
interface unless it forks, and since it always must listen on UDP, it will
skip listening on TCP if it must.

It looks like the -D parameter prevents OpenSIPS from forking, which means
no TCP port connection; if I run that command without the -D, TCP
connections seem to work.

Of course I liked using -D because it made it easy to start/stop OpenSIPS
(just with a ^C). I'm assuming that "daemon mode" means it's going to leave
a pid file around somewhere to make it easier to skill using a shell (or
shell script), so I'm off to find out if that's the case.

-PKCK


On Tue, Mar 19, 2013 at 2:45 PM, Pink Cupcake wrote:

> Thanks, Nick. I'm having a read through it now.
>
> I fixed my initial problem, though, and it was a pretty silly mistake: I
> had OpenSIPS and my SIP client (Jitsi) both using the same port. Oops! I
> switched Jitsi over to ports 55060/55061 and then things started working.
>
> Working, at least with UDP. I need to get TCP working now.
>
> The default configuration file has TCP disabled. Searching for information
> on how to set up TCP is quite difficult. The PDF you linked to has a few
> tips, luckily. But not enough to get it working.
>
> Here's a portion of my config file:
>
> #listen=udp:127.0.0.1:5060   # CUSTOMIZE ME
> listen=udp:172.16.23.79:5060   # CUSTOMIZE ME
> listen=tcp:172.16.23.79:5060   # CUSTOMIZE ME
> #disable_tcp=yes
> disable_tcp=no
> tcp_children=4
>
> When I try to connect with Jitsi, or with telnet, and sniff the traffic
> with Wireshark, I can see a SYN sent from Jitsi->OpenSIPS and a RST/ACK
> returned, but that's it. I don't think the TCP connection is getting
> properly established, and Jitsi bails out saying it can't create a
> connection.
>
> Jitsi does successfully connect to OpenSIPS if I set the proxy transport
> setting to UDP in Jitsi.
>
> What do I need to do to take OpenSIPS, from out of the box, and get it to
> accept REGISTERs/INVITEs on TCP and not UDP?
>
> -PKCK
>
>
>
> On Fri, Feb 8, 2013 at 4:49 PM, Nick Khamis  wrote:
>
>> Pink,
>>
>> A great resource that can get you up and going in at most a week can
>> be found here:
>>
>>
>>
>> http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w
>>
>> Happy Routing!
>>
>> Nick.
>>
>> On 2/8/13, Bogdan-Andrei Iancu  wrote:
>> > Hi,
>> >
>> > On 02/08/2013 12:17 AM, Pink Cupcake wrote:
>> >> Hi Bogdan-Andrei,
>> >>
>> >> Your response really doesn't help me or deal with the questions I laid
>> >> out in my original post. Perhaps I was not being clear. Let me start
>> >> over.
>> >>
>> > Well, my answer was related to the error logs you posted :).
>> >
>> >>
>> >>
>> >> I need to implement an automated testing scenario on an OS X build
>> >> machine.
>> >>
>> >> The test requires two different SIP UAs -- which are both running
>> >> locally on the same machine -- to successfully engage in a SIP
>> >> call. In order for two UAs to talk to one another, they need to be
>> >> registered with a SIP server.
>> >>
>> >> I am trying to determine if OpenSIPS can be used as the server
>> >> component in my testing scenario.
>> >>
>> >> Since this is an integration test and doesn't require any state being
>> >> retained, the test steps look like this: (1) bring up an OpenSIPS
>> >> server in userspace, (2) have the two SIP UA clients register with
>> >> that server, as simply as possible, (3) have the two SIP UA clients
>> >> engage in and then end a SIP call, (4) stop the SIP UA clients, (5)
>> >> shut down the server.
>> >>
>> >> If anything is unclear from the above, please reply back.
>> > Clear and no issues here,
>> >
>> >>
>> >>
>> >>
>> >> Please can someone answer the following:
>> >>
>> >> If I can run OpenSIPS in userspace, I would also not like to have it
>> >> "installed" on the build machine. I used the "prefix" parameter to
>> >> `make install` into a separate directory and I am attempting to run
>> >> OpenSIPS from that directory.
>> >>  It looks like I can run OpenSIPS in userspace. Is that correct?
>> > true
>> >>
>> >> From the documentation is looks like OpenSIPS does not use a database
>> >> by default, and keeps everything in memory. Is that correct?
>> > true
>> >>
>> >> Can any SIP UA client "REGISTER" with OpenSIPS when it is launched in
>> >> the "default" mode? If so, is there any special way the clients should
>> >> send the request?
>> > if using the opensips default script, no authentication will be required
>> > - but you need to use in the REGISTER RURI the IP of the server (so that
>> > opensips will consider them be handled locally)

Re: [OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Pink Cupcake]

Thanks, Nick. I'm having a read through it now.

I fixed my initial problem, though, and it was a pretty silly mistake: I
had OpenSIPS and my SIP client (Jitsi) both using the same port. Oops! I
switched Jitsi over to ports 55060/55061 and then things started working.

Working, at least with UDP. I need to get TCP working now.

The default configuration file has TCP disabled. Searching for information
on how to set up TCP is quite difficult. The PDF you linked to has a few
tips, luckily. But not enough to get it working.

Here's a portion of my config file:

#listen=udp:127.0.0.1:5060   # CUSTOMIZE ME
listen=udp:172.16.23.79:5060   # CUSTOMIZE ME
listen=tcp:172.16.23.79:5060   # CUSTOMIZE ME
#disable_tcp=yes
disable_tcp=no
tcp_children=4

When I try to connect with Jitsi, or with telnet, and sniff the traffic
with Wireshark, I can see a SYN sent from Jitsi->OpenSIPS and a RST/ACK
returned, but that's it. I don't think the TCP connection is getting
properly established, and Jitsi bails out saying it can't create a
connection.

Jitsi does successfully connect to OpenSIPS if I set the proxy transport
setting to UDP in Jitsi.

What do I need to do to take OpenSIPS, from out of the box, and get it to
accept REGISTERs/INVITEs on TCP and not UDP?

-PKCK



On Fri, Feb 8, 2013 at 4:49 PM, Nick Khamis  wrote:

> Pink,
>
> A great resource that can get you up and going in at most a week can
> be found here:
>
>
>
> http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w
>
> Happy Routing!
>
> Nick.
>
> On 2/8/13, Bogdan-Andrei Iancu  wrote:
> > Hi,
> >
> > On 02/08/2013 12:17 AM, Pink Cupcake wrote:
> >> Hi Bogdan-Andrei,
> >>
> >> Your response really doesn't help me or deal with the questions I laid
> >> out in my original post. Perhaps I was not being clear. Let me start
> >> over.
> >>
> > Well, my answer was related to the error logs you posted :).
> >
> >>
> >>
> >> I need to implement an automated testing scenario on an OS X build
> >> machine.
> >>
> >> The test requires two different SIP UAs -- which are both running
> >> locally on the same machine -- to successfully engage in a SIP
> >> call. In order for two UAs to talk to one another, they need to be
> >> registered with a SIP server.
> >>
> >> I am trying to determine if OpenSIPS can be used as the server
> >> component in my testing scenario.
> >>
> >> Since this is an integration test and doesn't require any state being
> >> retained, the test steps look like this: (1) bring up an OpenSIPS
> >> server in userspace, (2) have the two SIP UA clients register with
> >> that server, as simply as possible, (3) have the two SIP UA clients
> >> engage in and then end a SIP call, (4) stop the SIP UA clients, (5)
> >> shut down the server.
> >>
> >> If anything is unclear from the above, please reply back.
> > Clear and no issues here,
> >
> >>
> >>
> >>
> >> Please can someone answer the following:
> >>
> >> If I can run OpenSIPS in userspace, I would also not like to have it
> >> "installed" on the build machine. I used the "prefix" parameter to
> >> `make install` into a separate directory and I am attempting to run
> >> OpenSIPS from that directory.
> >>  It looks like I can run OpenSIPS in userspace. Is that correct?
> > true
> >>
> >> From the documentation is looks like OpenSIPS does not use a database
> >> by default, and keeps everything in memory. Is that correct?
> > true
> >>
> >> Can any SIP UA client "REGISTER" with OpenSIPS when it is launched in
> >> the "default" mode? If so, is there any special way the clients should
> >> send the request?
> > if using the opensips default script, no authentication will be required
> > - but you need to use in the REGISTER RURI the IP of the server (so that
> > opensips will consider them be handled locally).
> >>
> >> If it is necessary for OpenSIPS to be run with a database in order to
> >> allow clients to register? If so, will the db_text module suffice? If
> >> so, how do I perform this configuration (given my testing scenario)?
> > no, no need for DB - by default, in cfg, the usrloc module comes with no
> > DB support.
> >
> > Regards,
> > Bogdan
> >>
> >>
> >> PKCK
> >>
> >>
> >>
> >> On Thu, Feb 7, 2013 at 5:13 AM, Bogdan-Andrei Iancu
> >> mailto:bog...@opensips.org>> wrote:
> >>
> >> Hi,
> >>
> >> Without a trace I cannot tell for sure, but I suspect your clients
> >> send several REGISTER requests without increasing the CSEQ no
> >> (which is mandatory) - this is the meaning of the error you get.
> >>
> >> So, to be sure, make a network capture with the sip traffic
> >> (ngrep) and see what are the replies from opensips.
> >>
> >> Regards,
> >>
> >> Bogdan-Andrei Iancu
> >> OpenSIPS Founder and Developer
> >> http://www.opensips-solutions.com
> >>
> >>
> >> On 02/07/2013 12:0

[OpenSIPS-Users] Opensips taking too much time to load

[Darin Vivekananad]

My opensips is taking 10 min to load drouting. Can some suggest any ideas
as we are using 6 gb RAM and ver 1.5. All calls are routing properly only
issue is wit loading droutingbles  tables

Regards
Darin
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] opensips taking too much time to load drouting

[Darin Vivekananad]

I tried to reload opensips and its taking 10 min to reload the drouting
tables. we tried updating the versions and still no luck

 tm:inuse_transactions = 52
 one comment from a savvy
what is this 52 value means
We use 4Gb
/usr/local/opensip/sbin/opensip
sip/etc/opensips/opensips.cfg
Same with 6 GB
 version opensips-1.5.3-notls/
   Drouting
Revision $Revision: 5983 $ $Date: 2009-07-16 13:54:3
Revision $Revision: 5983 $ $Date: 2009-07-16 13:54:34 +0300
   (Thu, 16 Jul 2009) $


Please give some suggessions to fix this
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] opensips + freeradius + acc

[andrzej . ciupek]

Hello

I don't know where was problem, but I have downloaded  
cdrtool_9.0.1squeeze.tar.gz


and after:

mysql -u root -p radius < radius_accounting.proc

it works fine, so I had to have some kind of bug in previous  
radius_accounting.proc.


Greetings
Andrzej C

Cytowanie Tijmen de Mes :


Hi,

Something is not right. I used the exact same query as you did and here
it is successful:
mysql> CALL update_radacct_record( 'radius', '2013-03-19  14:14:07',
'0', '', '',
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 'df421e69',
'as287910d5' );
Query OK, 0 rows affected (0.00 sec)

That zero rows affected is because I don't have that session. Nothing
about wrong arguments or any other error.

Are you sure you have the right procedure and table layout? FYI we are
running  mysql 5.1.49-3

Best regards,

--
Tijmen de Mes
AG Projects


On 03/19/2013 04:05 PM, andrzej.ciu...@asterisk.edu.pl wrote:

Hello

Yes database is radius, table is radacct. But when I change it to   
database name I get error:


ERROR 1210 (HY000): Incorrect arguments to EXECUTE

But @v_db_name = var_db_name, is not used in procedure, after that:

table name is from:

@var_tbl_begin = (SELECT set_radacct_table_name()),

with concat of month.

Greetings
Andrzej

Cytowanie Tijmen de Mes :


Hi,

Are you sure database called 'radacct' ? Usually it is something like
'radius'. If it is radius the call needs to be:
CALL update_radacct_record('radius', '2013-03-19 14:14:07','0', '',
'','MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.',
'df421e69','as287910d5');

Best regards,

--
Tijmen de Mes
AG Projects



On 03/19/2013 02:45 PM, andrzej.ciu...@asterisk.edu.pl wrote:

Hello

I have started to use freeradius for acc. Using scripts from cdr-tool.
I have problem with update procedure after BYE.
Insert after answered call goes to the table "radacct201303", but  
  after BYE, when prodecure update_radacct_record is called,  
there  is  no result.

When I create UPDATE based on log from sqltrace log It works.
It looks like:

CALL update_radacct_record( 'radacct', '2013-03-19  14:14:07',   
'0', '', '',   
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 'df421e69',
  'as287910d5' );


doesn't update record but handmade UPDATE:

UPDATE radacct201303  SET AcctStopTime = '2013-03-19 14:14:07',
AcctSessionTime = CONCAT( TIME_TO_SEC(TIMEDIFF('2013-03-19
14:14:07', AcctStartTime)), '.', MICROSECOND(TIMEDIFF('2013-03-19  
  14:14:07', AcctStartTime)) ),AcctStopDelay =  
'',  ConnectInfo_stop = '', RTPStatistics = '', Normalized = '0'  
WHERE   AcctSessionId =  
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.'  AND  ((SipToTag =  
'as287910d5' AND SipFromTag = 'df421e69') OR   (SipFromTag =  
'as287910d5' AND SipToTag = 'df421e69')) AND   ConnectInfo_stop  
IS NULL AND (MediaInfo IS NULL OR MediaInfo !=   'timeout');


does.

Has someone had same problem ?

Greetings
Andrzej





___
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




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as INVITE?

[Tuomas Kaikkonen]

Bogdan,

What I am trying to do is this:  Have two or more OpenSIPs servers with each 
having their own rtpproxy server. The database is shared over ssh tunnel so it 
looks to each server as they are using a local database. The goal is to make 
the service scalable so that we can add more OpenSIPs hosts as the need comes. 
Also we want to have the system detect if a server has failed and not direct 
calls/registers to that server. All the servers are in a cloud system (which 
does not provide Multicast, only Unicast), and they will have both public IP 
and internal IP associated with them. The problem with the shared database is 
that the location table has the socket column storing the Internal IP of the 
OpenSIPs server who did the REGISTER of the SIP client with. I need to relay 
all INVITE/BYE/CANCEL requests via that server. I have no UDP load balancer to 
balance the SIP requests, and having the RTP Proxy proxy all audio packets is 
an extra burden to the design.  Our clients register with only one SIP server, 
there is no configuration to register with a primary and secondary SIP proxy.

What we worry about is putting one OpenSIPs server acting as a load balancer or 
dispatcher, is that we'd be then again dependent on one OpenSIPs server. That 
dispatching / load balancing OpenSIPs server should be also backed up by a fail 
over server. I've looked into some, mostly academic, papers on how they did 
their load balancing and fail over.

Tuomas Kaikkonen
Software Developer | Twisted Pair Solutions
3131 Elliott Ave, Suite 200, Seattle, WA 98121
Tel: (206) 812-0732

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Tuesday, March 19, 2013 8:01 AM
To: OpenSIPS users mailling list
Cc: Tuomas Kaikkonen
Subject: Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as 
INVITE?

Hi Tuomas,

The LB modules is only for CALLs - it understands by +1 load only a call. You 
cannot use it for REGISTERsAnyhow REGISTERs and INVITEs are as apples and 
onions :)...so you cannot put them in the same basket.

if you could provide more details on what you are trying to achieve, maybe I 
can advice you on the best balancing option (like maybe using dispatcher module)

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 03/18/2013 07:56 PM, Tuomas Kaikkonen wrote:
Can the Load Balancer module be configured to balance REGISTERs as well as the 
INVITES so that the above mentioned setup would work? OR is the Load Balancer 
module just useful for balancing RTP Proxy / media server resources for INVITEs?

I'm new to the Load Balancer module of OpenSIPs. I am running OpenSIPs stable 
branch 1.7 - just by looking examples from the documentation it looks like 
INVITE load balancing is supported.

Tuomas Kaikkonen






___

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

Re: [OpenSIPS-Users] ACK Loop when changing contact on_reply: Please Help!!!

[Nick Khamis]

My approach was covered here:

http://openser.org/pipermail/users/2008-September/000473.html

Will use record_route_preset(public vs. private) in our logic, and
change back the value of advertised address back to the private IP for
local traffic to route logically

Thanks Again Bogdan!

Nick.

On 3/19/13, Nick Khamis  wrote:
> Hello Bogdan,
>
> Thank you so much for your response. We did have an RR problem that
> did not allow for an "ACK" to our "200 OK". Our solution was to change
> "advertised_address" to use the public IP instead of the local net,
> which seemed to get the RR problem solved. The server related global
> parameters we are using are as follow:
>
> alias=
> auto_aliases=no
> disable_dns_failover=yes
> sip_warning=no
>
> port=5060
> listen=udp:192.168.2.5:5060
> advertised_address=
>
> This got the external ACK responses to our 200, but only one way audio
> (probably RTP proxy related, and started a new message for that
> issue).
>
> The question is, Should I change "advertised_address" back to private
> IP, and use "record_route_preset" instead? In the meanwhile, I will
> try it.
>
> Nick.
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Nick Khamis]

Hello Razvan,

I should have mentioned that we only experienced this issue with this
particular DID provider. With others everything works perfectly. We
suspect the issue is because the RTP stream is coming from a different
source that of the SIP messages. So I think it's a matter of lining up
rtpproxy_offer/answer parameters (i.e., co).

Unfortunately, their service to our zone today is down. Will post
detailed logs as soon as we can initiate some calls.

Nick.

On 3/19/13, Răzvan Crainea  wrote:
> Hi, Nick!
>
> You said that you can see logs for RTPProxy. Can you set the debug level
> to DBUG and paste (preferably on pastebin) the logs of the session?
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 03/19/2013 03:52 PM, Nick Khamis wrote:
>> I wanted to mention that the same setup works perfectly with VoIP.ms
>> but not Voxbone. I think the problem is that the SIP messages and RTP
>> stream for voxbone are coming from different sources. With other
>> origination providers SIP and RTP streams came from the same source,
>> so we never experienced a problem.
>>
>> We are currently looking into rtpproxy_orffer/answer parameters (i..e,
>> "reico"...) to see if we can line things up nicely.
>>
>> Nichola.
>>
>> On 3/19/13, Nick Khamis  wrote:
>>> RTPProxy does work behind NAT. It's mediaporxy that must be on a public
>>> ip.
>>>
>>> Thanks for your help.
>>>
>>> Nick.
>>>
>>> On 3/19/13, Muhammad Shahzad  wrote:
 If you are unfamiliar with rtp proxy and how it works, then it would be
 better for you to use engage_rtp_proxy rather then offer / answer model.
 Also RTP Proxy requires public IP address, its likely not to work on
 private subnets (unless you have all SIP entities on same LAN, in which
 case theoretically it should work but i have never tested it myself).

 Thank you.


 On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis  wrote:

> I am not sure if this is the correct place to post OpenSIPS+RTPProxy
> questions however, I tried to subscribing to the RTP proxy mailing
> list and never heard from them since. If it is ok to post RTP proxy
> related questions here I am trying to test OpenSIPS with RTP proxy
> with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
> RTPPoxy) just for testing.
>
> The code I am using is:
>
> route {
>   force_rport();
> }
> route[1] {
>  if (is_method("INVITE")) {
>  t_on_branch("1");
>  t_on_reply("1");
>  t_on_failure("1");
>
>  if (has_body("application/sdp"))  rtpproxy_offer();
>  }
>  else if (is_method("BYE|CANCEL")) {
>  unforce_rtp_proxy();
>  }
>
>  if (!t_relay()) {
>  sl_reply_error();
>  };
>  exit;
> }
> onreply_route[1] {
>   if (has_body("application/sdp")) rtpproxy_answer();
> }
>
>
> There is no way audio using RTP proxy, but audio is fine between the
> UA without including the RTP proxy related script. Looking at the log
> I found that RTP is prefilling the callers address twice, but not the
> callees address.
>
>
> INFO:main: rtpproxy started, pid 7287
> INFO:handle_command: new session
> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
> d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
> INFO:handle_command: new session on a port 35010 created, tag
> d23f0168-538e-e211-8550-001b7700a65b;1
> INFO:handle_command: pre-filling caller's address with
> 192.168.2.101:5062
> INFO:handle_command: new session
> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
> d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
> INFO:handle_command: new session on a port 22982 created, tag
> d23f0168-538e-e211-8550-001b7700a65b;2
> INFO:handle_command: pre-filling caller's address with
> 192.168.2.101:5064
> INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 35010/0 is cleaned up
> INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 22982/0 is cleaned up
>
> Is it possible to test RTP relaying with everything on the same
> network?
>
> Thanks in Advance,
>
> Nick.
>
> __

Re: [OpenSIPS-Users] ACK Loop when changing contact on_reply: Please Help!!!

[Nick Khamis]

Hello Bogdan,

Thank you so much for your response. We did have an RR problem that
did not allow for an "ACK" to our "200 OK". Our solution was to change
"advertised_address" to use the public IP instead of the local net,
which seemed to get the RR problem solved. The server related global
parameters we are using are as follow:

alias=
auto_aliases=no
disable_dns_failover=yes
sip_warning=no

port=5060
listen=udp:192.168.2.5:5060
advertised_address=

This got the external ACK responses to our 200, but only one way audio
(probably RTP proxy related, and started a new message for that
issue).

The question is, Should I change "advertised_address" back to private
IP, and use "record_route_preset" instead? In the meanwhile, I will
try it.

Nick.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] opensips + freeradius + acc

[Tijmen de Mes]

Hi,

Something is not right. I used the exact same query as you did and here 
it is successful:
mysql> CALL update_radacct_record( 'radius', '2013-03-19  14:14:07', 
'0', '', '',  
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 
'df421e69', 'as287910d5' );

Query OK, 0 rows affected (0.00 sec)

That zero rows affected is because I don't have that session. Nothing 
about wrong arguments or any other error.


Are you sure you have the right procedure and table layout? FYI we are 
running  mysql 5.1.49-3


Best regards,

--
Tijmen de Mes
AG Projects


On 03/19/2013 04:05 PM, andrzej.ciu...@asterisk.edu.pl wrote:

Hello

Yes database is radius, table is radacct. But when I change it to 
database name I get error:


ERROR 1210 (HY000): Incorrect arguments to EXECUTE

But @v_db_name = var_db_name, is not used in procedure, after that:

table name is from:

@var_tbl_begin = (SELECT set_radacct_table_name()),

with concat of month.

Greetings
Andrzej

Cytowanie Tijmen de Mes :


Hi,

Are you sure database called 'radacct' ? Usually it is something like
'radius'. If it is radius the call needs to be:
CALL update_radacct_record('radius', '2013-03-19 14:14:07','0', '',
'','MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.',
'df421e69','as287910d5');

Best regards,

--
Tijmen de Mes
AG Projects



On 03/19/2013 02:45 PM, andrzej.ciu...@asterisk.edu.pl wrote:

Hello

I have started to use freeradius for acc. Using scripts from cdr-tool.
I have problem with update procedure after BYE.
Insert after answered call goes to the table "radacct201303", but  
after BYE, when prodecure update_radacct_record is called, there is 
 no result.

When I create UPDATE based on log from sqltrace log It works.
It looks like:

CALL update_radacct_record( 'radacct', '2013-03-19  14:14:07', '0', 
'', '', 
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 'df421e69', 
'as287910d5' );


doesn't update record but handmade UPDATE:

UPDATE radacct201303  SET AcctStopTime = '2013-03-19 14:14:07',  
AcctSessionTime = CONCAT( TIME_TO_SEC(TIMEDIFF('2013-03-19  
14:14:07', AcctStartTime)), '.', MICROSECOND(TIMEDIFF('2013-03-19  
14:14:07', AcctStartTime)) ),AcctStopDelay = '', 
ConnectInfo_stop = '', RTPStatistics = '', Normalized = '0' WHERE  
AcctSessionId = 'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.' AND  
((SipToTag = 'as287910d5' AND SipFromTag = 'df421e69') OR  
(SipFromTag = 'as287910d5' AND SipToTag = 'df421e69')) AND  
ConnectInfo_stop IS NULL AND (MediaInfo IS NULL OR MediaInfo !=  
'timeout');


does.

Has someone had same problem ?

Greetings
Andrzej





___
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 


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RTP Proxy Problem - No Way Audio (RTP Traces Within)

[Răzvan Crainea]

Hi, Nick!

From your traces, I can see that the RTPProxy session is properly 
established (you have both an offer and an answer). But on the media 
level, all I can see is that Asterisk (the callee) is sending RTP to 
caller, but the caller doesn't send anything. Also, this is what 
RTPProxy indicates (RTP stats: 86 in from callee, 0 in from caller). 
Most likely you should checkwhere is the NAT Box trying to sendRTP.


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/15/2013 02:00 AM, Nick Khamis wrote:

Hello Everyone,

I am having problem getting RTP packets flowing smoothly. The setup is

NAT Box (192.168.2.1) <-> OpenSIPS/RTPProxy (192.168.2.5) <-> Asterisk
(192.168.2.10)

I know that media is reaching the boxes since I see:

OpenSIPS (192.168.2.5)

0.00 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
   0.99  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272
   0.017956 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
   0.018028  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272
   0.037760 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
   0.037814  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272


Asterisk CLI (192.168.2.10)

Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)

RTPProxy Messages:

INFO:handle_command: new session
KN74JOEJTRFDVOR3PEH7I5XGBA@81.201.85.45, tag 86219;1 requested, type
strong
INFO:handle_command: new session on a port 20198 created, tag 86219;1
INFO:handle_command: pre-filling caller's address with 81.201.85.45:13272
INFO:handle_command: lookup on ports 20198/39810, session timer restarted
INFO:handle_command: pre-filling callee's address with 192.168.2.10:24454
INFO:handle_delete: forcefully deleting session 1 on ports 20198/39810
INFO:remove_session: RTP stats: 86 in from callee, 0 in from caller,
86 relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 20198/39810 is cleaned up


It says 86 in from callee but we do not even have incoming audio. I'm
pretty sure it's "rtpproxy_offer/answer" issue so bellow is my
configuration:

route[1] {
 xlog("Start Call Route For: [ fu=$fu/ tu=$tu /ru=$ru/
ci=$ci]\n");

 if (has_body("application/sdp")) {
 xlog("Has SDP: $fu\n");
 rtpproxy_offer();
 }
}

onreply_route[1] {
 xlog("Reply Route 1: [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]\n");
 if (has_body("application/sdp")) {
 xlog("Answering  RTP Proxy: $fu\n");
 rtpproxy_answer();
   }
}

Your help is greatly appreciated,

Nick.

___
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

Re: [OpenSIPS-Users] Record-routing & failover (drouting)

[Max Mühlbronner]

Hi,

ok, thanks.

I will look into this, but we got record_route() in the request route 
(initial request) still i can reproduce the error (i can see the invite 
going out without record-route, on gw failover) and if i add another 
record_route it is fine. Maybe i can get some more details.



Best Regards


On 03/19/2013 04:24 PM, Bogdan-Andrei Iancu wrote:

Hi Max,

If you do the RR on request route, the header will be present in all 
branches of that INVITE. If you do it later in branch route or failure 
route, it will be visible only for that branch.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03/15/2013 09:11 AM, Max Mühlbronner wrote:

Hi,

thanks very much for your reply. That's also what i thought, first it 
was just a siptrace where opensips sends to the second gw but the 
invite got no Record-route header.


But later on, I was able to reproduce the problem and when adding 
another record_route() the second invite is indeed fixed (contains a 
record-route header)? Maybe there is some other reason for this 
behavior i am not seeing yet?



Best Regards

Max M.

On 03/14/2013 05:35 PM, Muhammad Shahzad wrote:
No, I think the serial forked invite contains same RR and in general 
all changes you did to original invite just before creating the 
transaction (by calling t_newtrans or t_relay or any t_* function 
that creates transaction).


Thank you.


On Thu, Mar 14, 2013 at 5:15 PM, Max Mühlbronner > wrote:


Hi,

I am not sure about record-routing in combination with failover
of drouting. Maybe someone knows for sure :)

If i got a configuration where i am record_routing on inital
invite, but later there is a failover (use_next_gw() returns
true) and the call is sent to the next gateway. But the serial
forked call (second INVITE) is missing the Record-route header?


Does this mean i just have to explicitly call record routing
again on failover? But to me it seems like this can't be right,
or is this correct/expected behaviour?

||

if (use_next_gw()) {
...
record_route();

}



Best Regards

-- 
Max Mühlbronner


42com Telecommunication GmbH
Straße der Pariser Kommune 12-16
10243 Berlin

E-Mail:m...@42com.com  
Web:www.42com.com  

Firmenangaben/Company information:
Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
Umsatzsteuer-ID/VAT-ID: DE223812306
Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese 
sind möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. 
Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten haben, so 
informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail.

This message is intended only for the use of the individual or entity to 
which it is addressed. If you have received this message by mistake, please 
notify us immediately.


___
Users mailing list
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com 
Email: shaherya...@googlemail.com 


___
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



___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Safe place and way to change the Contact header without killing loose route

[Bogdan-Andrei Iancu]

Hi Nick,

Guess this is related to another email of yours (a later one) - you 
should not change the contact as your opensips is not end point, but you 
should change the RR header to reflect the public IP of your server.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/14/2013 07:33 PM, Nick Khamis wrote:

Hello Everyone,

We have our OpenSIPS box behind nat, and would like a safe place to
change the Contact header to a public IP without messing up our
loose/recorded routes. We attempted to do it in on_reply but got the
following result:

Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.
Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.
Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.
Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.
Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.
Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.
Via: SIP/2.0/UDP
73.14.112.21;rport=5060;received=73.14.112.21;branch=z9hG4bKe1c.5cc516a1.2.

Kind Regards,

Nick

___
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

Re: [OpenSIPS-Users] Routing to Proxy Issue

[Bogdan-Andrei Iancu]

Yes, as said, the first server should use "send()" function instead of 
"t_relay()" -> it will be skipped by the reply.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/14/2013 07:41 PM, Nathaniel L Keeling III wrote:
Thanks for the response. What I am trying to do is to have the first 
proxy route the request to the second proxy server that will actually 
process the request. In this case I have another opensips server that 
I want to just handle register requests. This proxy server would then 
send the replies back to the client and not the first proxy server. Is 
this the idea way of performing this functionality? I was trying to 
separate some functions to separate sip servers ( a sip server for 
registrations, another sip server for pstn gateway, etc) but have a 
central point of entry.


Thanks

Nathaniel


On 3/14/13 11:55 AM, Bogdan-Andrei Iancu wrote:

Hi Nathaniel,

According to SIP RFC, replies are following the same path as requests 
(in revert order, of course) - this back routing is done based on VIA 
hdr (added in requests, consumed from replies).


if you want to avoid opensips to add the VIA hdr (so, to be skipped 
by replies), use the send() function for relaying the requests - be 
careful this is a stateless function, so no failure_route !


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/14/2013 09:30 AM, Nathaniel L Keeling III wrote:

Hello,

I would like to know if this routing is possible and if so how. I 
have a scenario where I have a opensips proxy that I want to use 
just to send requests to other opensips proxy witch would then 
process the request. Here is the scenario:  client  --> p1 --> p2 
--> client --> p2. I have been able to route to proxy 2 but the 
replies are still being routed back via proxy 1 then to the client. 
I was hoping to send the replies directly to the client and not back 
thru the first proxy. Is this possible? I am currently using the 
dispatcher module to send the requests to the second proxy. I also 
noticed that the SIP packets have a VIA header of the first proxies 
IP address when it reaches the second proxy. I thought the 
record_route was causing this, so I moved it after I send the 
request to the second proxy. Here is a small snipet of my code:


if (is_method("REGISTER")) {
   if (!ds_select_dst("2", "3")) {
sl_send_reply("503", "Rawr!!");
   };
   t_relay();
}


Thanks

Nathaniel


___
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



___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Record-routing & failover (drouting)

[Bogdan-Andrei Iancu]

Hi Max,

If you do the RR on request route, the header will be present in all 
branches of that INVITE. If you do it later in branch route or failure 
route, it will be visible only for that branch.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/15/2013 09:11 AM, Max Mühlbronner wrote:

Hi,

thanks very much for your reply. That's also what i thought, first it 
was just a siptrace where opensips sends to the second gw but the 
invite got no Record-route header.


But later on, I was able to reproduce the problem and when adding 
another record_route() the second invite is indeed fixed (contains a 
record-route header)? Maybe there is some other reason for this 
behavior i am not seeing yet?



Best Regards

Max M.

On 03/14/2013 05:35 PM, Muhammad Shahzad wrote:
No, I think the serial forked invite contains same RR and in general 
all changes you did to original invite just before creating the 
transaction (by calling t_newtrans or t_relay or any t_* function 
that creates transaction).


Thank you.


On Thu, Mar 14, 2013 at 5:15 PM, Max Mühlbronner > wrote:


Hi,

I am not sure about record-routing in combination with failover
of drouting. Maybe someone knows for sure :)

If i got a configuration where i am record_routing on inital
invite, but later there is a failover (use_next_gw() returns
true) and the call is sent to the next gateway. But the serial
forked call (second INVITE) is missing the Record-route header?


Does this mean i just have to explicitly call record routing
again on failover? But to me it seems like this can't be right,
or is this correct/expected behaviour?

||

if (use_next_gw()) {
...
record_route();

}



Best Regards

-- 
Max Mühlbronner


42com Telecommunication GmbH
Straße der Pariser Kommune 12-16
10243 Berlin

E-Mail:m...@42com.com  
Web:www.42com.com  

Firmenangaben/Company information:
Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
Umsatzsteuer-ID/VAT-ID: DE223812306
Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese 
sind möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. 
Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten haben, so 
informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail.

This message is intended only for the use of the individual or entity to 
which it is addressed. If you have received this message by mistake, please 
notify us immediately.


___
Users mailing list
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com 
Email: shaherya...@googlemail.com 


___
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
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Set a specific signalling IP to trunk

[Bogdan-Andrei Iancu]

  
  
Hi Khaled,
  
  I suggest keeping the IP of the outbound interface (the $avp(termip))
  in the gateways attrs - those values are automatically loaded for
  script usage when the GW is selected, so you can do the setting
  for $fs .
  
  Look at the "attrs" column in the dr_gateways table and at the
  gw_attrs param in dr module.

  Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03/15/2013 06:28 PM, M.Khaled W Chehab wrote:

  
  
  
  
  
Dears,
 
While testing and reading I find out how I
  can change the outgoing signaling IP using 
if (!has_totag()
&&is_method("INVITE")) {
$var(a) =
"udp:"+$avp(termip)+":5060";

$fs = $var(a) ;
….
But that will change the  signaling realm
  for  all outgoing calls for all  trunks  ,how can I change the
  outgoing signaling realm for a specified trunk which located
  in my dr_gateways 
 
Regards
 
 
 
 
Khaled Chehab
Senior NGN Engineer

Operations Office - Lebanon
Office    : +961 1 515155 ext 300
Mobile  : +961 3 045212
E-mail    : kche...@icucall.com
MSN ID :khalidche...@hotmail.com 
Skype    : k_chehab 
Web Site: http://www.icucall.com
 http://www.allohi.com
 
  
  

___
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

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Răzvan Crainea]

Hi, Nick!

You said that you can see logs for RTPProxy. Can you set the debug level 
to DBUG and paste (preferably on pastebin) the logs of the session?


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/19/2013 03:52 PM, Nick Khamis wrote:

I wanted to mention that the same setup works perfectly with VoIP.ms
but not Voxbone. I think the problem is that the SIP messages and RTP
stream for voxbone are coming from different sources. With other
origination providers SIP and RTP streams came from the same source,
so we never experienced a problem.

We are currently looking into rtpproxy_orffer/answer parameters (i..e,
"reico"...) to see if we can line things up nicely.

Nichola.

On 3/19/13, Nick Khamis  wrote:

RTPProxy does work behind NAT. It's mediaporxy that must be on a public ip.

Thanks for your help.

Nick.

On 3/19/13, Muhammad Shahzad  wrote:

If you are unfamiliar with rtp proxy and how it works, then it would be
better for you to use engage_rtp_proxy rather then offer / answer model.
Also RTP Proxy requires public IP address, its likely not to work on
private subnets (unless you have all SIP entities on same LAN, in which
case theoretically it should work but i have never tested it myself).

Thank you.


On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis  wrote:


I am not sure if this is the correct place to post OpenSIPS+RTPProxy
questions however, I tried to subscribing to the RTP proxy mailing
list and never heard from them since. If it is ok to post RTP proxy
related questions here I am trying to test OpenSIPS with RTP proxy
with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
RTPPoxy) just for testing.

The code I am using is:

route {
  force_rport();
}
route[1] {
 if (is_method("INVITE")) {
 t_on_branch("1");
 t_on_reply("1");
 t_on_failure("1");

 if (has_body("application/sdp"))  rtpproxy_offer();
 }
 else if (is_method("BYE|CANCEL")) {
 unforce_rtp_proxy();
 }

 if (!t_relay()) {
 sl_reply_error();
 };
 exit;
}
onreply_route[1] {
  if (has_body("application/sdp")) rtpproxy_answer();
}


There is no way audio using RTP proxy, but audio is fine between the
UA without including the RTP proxy related script. Looking at the log
I found that RTP is prefilling the callers address twice, but not the
callees address.


INFO:main: rtpproxy started, pid 7287
INFO:handle_command: new session
ae450168-538e-e211-8550-001b7700a65b@oakville, tag
d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
INFO:handle_command: new session on a port 35010 created, tag
d23f0168-538e-e211-8550-001b7700a65b;1
INFO:handle_command: pre-filling caller's address with
192.168.2.101:5062
INFO:handle_command: new session
ae450168-538e-e211-8550-001b7700a65b@oakville, tag
d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
INFO:handle_command: new session on a port 22982 created, tag
d23f0168-538e-e211-8550-001b7700a65b;2
INFO:handle_command: pre-filling caller's address with
192.168.2.101:5064
INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 35010/0 is cleaned up
INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 22982/0 is cleaned up

Is it possible to test RTP relaying with everything on the same network?

Thanks in Advance,

Nick.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com


___
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

Re: [OpenSIPS-Users] ACK Loop when changing contact on_reply: Please Help!!!

[Bogdan-Andrei Iancu]

Hi Nick,

As I suspect that your opensips is not an end-point in the call (but 
simply a proxy), I guess the right approach is to reflect the network 
changing in the RR headers, and not in Contact (contact reflects the end 
points in dialog).


I suggest using the record_route_preset() and pushing all the time the 
public IP of opensips in the RR header.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/19/2013 04:43 PM, Nick Khamis wrote:

Hello Muhammad, thanks again for your response. On our test
environment, our opensips+rtpproxy server is behind NAT, and the
reason we are modifying the contact header is to point to
1001@  instead of 1001@. UA
1001 is also behind the same NAT.

My first question, do I need to modify the contact header? Or should I
be paying closer attention to the SDP payload. Making sure c=, and o=
are pointing to the right locations?

Your help is greatly appreciated.

Nick.

On 3/19/13, Muhammad Shahzad  wrote:

Yup, that's expected to happen. ACK is sent to Contact header of 200 OK.
So, if you mess up with it, then unexpected results will happen, as in your
case you are perhaps setting Contact address of 200 OK pointing to opensips
itself, instead of destination party, so ACK will obviously loop as
expected.

Thank you.


On Mon, Mar 18, 2013 at 5:55 PM, Nick Khamis  wrote:


Hello Everyone,

We are changing the "Contact" header in the on_reply to a public ip
address using:

onreply_route[1] {
 xlog("incoming reply\n");
 if (has_body("application/sdp")) {
 remove_hf("Contact");
 append_hf("Contact:
\r\n");
 append_hf("P-hint: Onreply-route -
fixcontact \r\n");

 }
}

When doing so, ACK is going into a loop:

U 2013/03/18 13:42:11.021017 75.15.201.2:5060 ->  192.168.2.5:5060
ACK sip:75.15.201.2;lr;did=b03.4af9f8f3 SIP/2.0.
Call-ID: VQUK2UGSQBCPHEW27UN5NBJIQM@81.201.86.45.
CSeq: 102 ACK.
From: "15178334003";tag=91641.
To:;tag=2643FD58-346926A7.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
Via: SIP/2.0/UDP 7


Your help is greatly appreciated,

Nick.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com


___
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

Re: [OpenSIPS-Users] two UACs behind the same nat

[Bogdan-Andrei Iancu]

Hi Nick,

So, your opensips should do the NAT traversal for the signaling 
(fix_nated_contact() stuff), but it should not do anything in regards to 
media (rtpproxy) if caller and callee are behind the same NAT - how to 
test that ? after lookup(location), when you know the destination IP of 
the callee, you can simply check if SRC IP of the call is the same with 
the destination IP - if so, it means caller and callee are behind the 
same NAT.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/18/2013 04:20 PM, Nick wrote:

Hello

I checked it. It don't to go to the route[nat_fixups].
But My UACs both behind the same NAT.

How to check why not go to the route[nat_fixups]??

Thanks
Nick


2013/3/18 Nick Khamis mailto:sym...@gmail.com>>

It will change it if fired. Make sure that you get to
"route[nat_fixups]" for the call. And for testing, try removing the
flags" (isflagset(5) || isbflagset(6))" etc... xlog("Why me god:
$ru\n") always seems to have mercy on our souls. ;)


Nick.

On 3/18/13, Nick mailto:nick.ch...@kland.com.tw>> wrote:
> Hello
>
> I modify my config.
>
> route[nat_fixups] {
> xlog("L_NOTICE","4 NAT FIXED - $fu call $ru ");
> # RTP Proxy handling ---#
> if (is_method("BYE|CANCEL")) {
> unforce_rtp_proxy();
> } else if (is_method("INVITE")){
> # Activates the RTP Proxy for the CALLEE ---#
> rtpproxy_offer("c","61.67.128.46");
> };
> # catch and fix replies
> t_on_reply("2");
> }
>
> onreply_route[2] {
> #xlog("incoming reply\n");
>
> # Handling of the SDP for the 200 or 183 reply #
> # If behind nat (flags 5 or 6) start RTP Proxy #
> #Activates the RTP Proxy for the CALLER#
> if ( is_method("INVITE") &&
> (isflagset(5) || isbflagset(6)) &&
> has_body("application/sdp") ){
> rtpproxy_answer("c","61.67.128.46");
> }
>
> # If the CALLEE is behind NAT, fix the CONTACT HF #
> if (isbflagset(6)) {
> fix_nated_contact();
> search_append('Contact:.*sip:[^>[:cntrl:]]*',
';nat=yes');
> }
> exit;
> }
>
> Then, I used ngrep get packet.
>
> Connection Information still not Opensips SERVER of IP.
> It's not change to my setting.
>
> Do you have any suggestion??
>
> Thanks
> Nick
>
>
>
> 2013/3/15 Nick Khamis mailto:sym...@gmail.com>>
>
>> Hello Nick,
>>
>> As Bogdan pointed out please look at the important parts of
your SDP
>> payload:
>>
>> sip:55688@61.67.128.46:5060 
>> Contact: http://sip:55688@211.75.166.164:5060>>
>> c=IN IP4 211.75.166.164
>>
>> sip:@61.67.128.46 
>> Contact: http://sip:@192.168.4.197:5060>>
>> c=IN IP4 192.168.4.197
>>
>> Kind Regards,
>>
>> Nick.
>>
>> On 3/15/13, Bogdan-Andrei Iancu mailto:bog...@opensips.org>> wrote:
>> > Hi Nick,
>> >
>> > But in the SDP-s received by the UACs, in the "c" line, you
see the IP
>> > of the other UAC and not the IP of OpenSIPS, right ?
>> >
>> > Regards,
>> >
>> > Bogdan-Andrei Iancu
>> > OpenSIPS Founder and Developer
>> > http://www.opensips-solutions.com
>> >
>> >
>> > On 03/15/2013 03:28 AM, Nick Chang wrote:
>> >>
>> >> Hello Bodgan
>> >>
>> >> 2 UACs behind the same nat, I check my routing.
>> >>
>> >> UAC don't route to nat_fixups.
>> >>
>> >> But UACs checked My OpenSIPS private IP (10.10.12.111), Not
Public IP.
>> >>
>> >> I don't know why? Do you have any suggestion??
>> >>
>> >> My Config
>> >>
>> >> route[generic_relay] {
>> >>
>> >> xlog("L_NOTICE","3 invite $si");
>> >>
>> >> # for INVITEs enable some additional helper routes
>> >>
>> >> # Helper route, if nat=yes in the R-URI  set
flag 6 #
>> >>
>> >> #This is used to Process REINVITES  
   #

>> >>
>> >> if (subst_uri('/((sip:.*)||(sip:.*:.*));nat=yes/\1/')){
>> >>
>> >> setbflag(6);
>> >>
>> >> };
>> >>
>> >> # If caller(flag 5) or callee(flag 6) are behind
NAT  ---#
>> >>
>> >> # Call the route(6) to force the use of the RTP
Proxy ---#
>> >>
>> >> if (isflagset(5) || isbflagset(6)) {
>> >>
>> >> route(nat_fixups);
>> >>
>> >> };
>> >>
>> >> if (!t_relay()) {
>> >>
>> >> 

Re: [OpenSIPS-Users] opensips + freeradius + acc

[andrzej . ciupek]

Hello

Yes database is radius, table is radacct. But when I change it to  
database name I get error:


ERROR 1210 (HY000): Incorrect arguments to EXECUTE

But @v_db_name = var_db_name, is not used in procedure, after that:

table name is from:

@var_tbl_begin = (SELECT set_radacct_table_name()),

with concat of month.

Greetings
Andrzej

Cytowanie Tijmen de Mes :


Hi,

Are you sure database called 'radacct' ? Usually it is something like
'radius'. If it is radius the call needs to be:
CALL update_radacct_record('radius', '2013-03-19 14:14:07','0', '',
'','MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.',
'df421e69','as287910d5');

Best regards,

--
Tijmen de Mes
AG Projects



On 03/19/2013 02:45 PM, andrzej.ciu...@asterisk.edu.pl wrote:

Hello

I have started to use freeradius for acc. Using scripts from cdr-tool.
I have problem with update procedure after BYE.
Insert after answered call goes to the table "radacct201303", but   
after BYE, when prodecure update_radacct_record is called, there is  
 no result.

When I create UPDATE based on log from sqltrace log It works.
It looks like:

CALL update_radacct_record( 'radacct', '2013-03-19   
14:14:07', '0', '', '',   
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 'df421e69',  
'as287910d5' );


doesn't update record but handmade UPDATE:

UPDATE radacct201303  SET AcctStopTime = '2013-03-19 14:14:07',   
AcctSessionTime = CONCAT( TIME_TO_SEC(TIMEDIFF('2013-03-19   
14:14:07', AcctStartTime)), '.', MICROSECOND(TIMEDIFF('2013-03-19   
14:14:07', AcctStartTime)) ),AcctStopDelay = '',   
ConnectInfo_stop = '', RTPStatistics = '', Normalized = '0' WHERE   
AcctSessionId = 'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.' AND   
((SipToTag = 'as287910d5' AND SipFromTag = 'df421e69') OR   
(SipFromTag = 'as287910d5' AND SipToTag = 'df421e69')) AND   
ConnectInfo_stop IS NULL AND (MediaInfo IS NULL OR MediaInfo !=   
'timeout');


does.

Has someone had same problem ?

Greetings
Andrzej





___
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




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPs Database Maintenance

[Bogdan-Andrei Iancu]

  
  
Hi David,
  
  The performance degradation of the cdr procedure (from CP) may be
  because:
      - acc table got larger as polluted with old acc STOP/START
  records which never matched into a call; I suggest to move into an
  archive table whatever acc record is older than 2 days (you do not
  have calls longer than 2 days :) )
      - cdr table may become large if you are continuously
  accumulating data inthere. Be sure you are rotating or consuming
  the data from the CDR table.
  
  Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03/18/2013 04:36 PM, David Crow wrote:

  
  
  
  
  
I’ve got an opensips system that we’ve been
  running for about 3 years now.  Recently I’ve noticed that the
  load average on the system has been creeping up slowly over
  time.
 
I’ve narrowed this down to the cdr
  generation script in the opensips-cp, which seems to be taking
  longer and longer to finish. 
  
 
I was looking at the mysql database and it
  looks like both the acc and cdrs tables just continue to
  grow.  I know that I can remove stuff from the CDR table, but
  I’m not really sure about the ACC table.  Does anyone have
  scripts that regularly purge old stuff from these databases? 
  Also what do I do now with them as large as they are?
 
Thanks,
 
David Crow
| Senior Systems
  Architect
1301 Gervais Street, Suite 1800 | Columbia, SC
  29201
(d) 803.978.2727
|
(f)
  803.733.5888
david.c...@vc3.com|
  www.VC3.com
Follow us:
  
 
  
  
 
 
  
  

___
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

Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as INVITE?

[Bogdan-Andrei Iancu]

Hi Tuomas,

The LB modules is only for CALLs - it understands by +1 load only a 
call. You cannot use it for REGISTERsAnyhow REGISTERs and INVITEs 
are as apples and onions :)...so you cannot put them in the same basket.


if you could provide more details on what you are trying to achieve, 
maybe I can advice you on the best balancing option (like maybe using 
dispatcher module)


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/18/2013 07:56 PM, Tuomas Kaikkonen wrote:


Can the Load Balancer module be configured to balance REGISTERs as 
well as the INVITES so that the above mentioned setup would work? OR 
is the Load Balancer module just useful for balancing RTP Proxy / 
media server resources for INVITEs?


I'm new to the Load Balancer module of OpenSIPs. I am running OpenSIPs 
stable branch 1.7 -- just by looking examples from the documentation 
it looks like INVITE load balancing is supported.


Tuomas Kaikkonen


___
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

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Nick Khamis]

I wanted to mention that the same setup works perfectly with VoIP.ms
but not Voxbone. I think the problem is that the SIP messages and RTP
stream for voxbone are coming from different sources. With other
origination providers SIP and RTP streams came from the same source,
so we never experienced a problem.

We are currently looking into rtpproxy_orffer/answer parameters (i..e,
"reico"...) to see if we can line things up nicely.

Nichola.

On 3/19/13, Nick Khamis  wrote:
> RTPProxy does work behind NAT. It's mediaporxy that must be on a public ip.
>
> Thanks for your help.
>
> Nick.
>
> On 3/19/13, Muhammad Shahzad  wrote:
>> If you are unfamiliar with rtp proxy and how it works, then it would be
>> better for you to use engage_rtp_proxy rather then offer / answer model.
>> Also RTP Proxy requires public IP address, its likely not to work on
>> private subnets (unless you have all SIP entities on same LAN, in which
>> case theoretically it should work but i have never tested it myself).
>>
>> Thank you.
>>
>>
>> On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis  wrote:
>>
>>> I am not sure if this is the correct place to post OpenSIPS+RTPProxy
>>> questions however, I tried to subscribing to the RTP proxy mailing
>>> list and never heard from them since. If it is ok to post RTP proxy
>>> related questions here I am trying to test OpenSIPS with RTP proxy
>>> with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
>>> RTPPoxy) just for testing.
>>>
>>> The code I am using is:
>>>
>>> route {
>>>  force_rport();
>>> }
>>> route[1] {
>>> if (is_method("INVITE")) {
>>> t_on_branch("1");
>>> t_on_reply("1");
>>> t_on_failure("1");
>>>
>>> if (has_body("application/sdp"))  rtpproxy_offer();
>>> }
>>> else if (is_method("BYE|CANCEL")) {
>>> unforce_rtp_proxy();
>>> }
>>>
>>> if (!t_relay()) {
>>> sl_reply_error();
>>> };
>>> exit;
>>> }
>>> onreply_route[1] {
>>>  if (has_body("application/sdp")) rtpproxy_answer();
>>> }
>>>
>>>
>>> There is no way audio using RTP proxy, but audio is fine between the
>>> UA without including the RTP proxy related script. Looking at the log
>>> I found that RTP is prefilling the callers address twice, but not the
>>> callees address.
>>>
>>>
>>> INFO:main: rtpproxy started, pid 7287
>>> INFO:handle_command: new session
>>> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
>>> d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
>>> INFO:handle_command: new session on a port 35010 created, tag
>>> d23f0168-538e-e211-8550-001b7700a65b;1
>>> INFO:handle_command: pre-filling caller's address with
>>> 192.168.2.101:5062
>>> INFO:handle_command: new session
>>> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
>>> d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
>>> INFO:handle_command: new session on a port 22982 created, tag
>>> d23f0168-538e-e211-8550-001b7700a65b;2
>>> INFO:handle_command: pre-filling caller's address with
>>> 192.168.2.101:5064
>>> INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
>>> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
>>> relayed, 0 dropped
>>> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
>>> relayed, 0 dropped
>>> INFO:remove_session: session on ports 35010/0 is cleaned up
>>> INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
>>> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
>>> relayed, 0 dropped
>>> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
>>> relayed, 0 dropped
>>> INFO:remove_session: session on ports 22982/0 is cleaned up
>>>
>>> Is it possible to test RTP relaying with everything on the same network?
>>>
>>> Thanks in Advance,
>>>
>>> Nick.
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>>
>> --
>> Mit freundlichen Grüßen
>> Muhammad Shahzad
>> ---
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.com
>>
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK Loop when changing contact on_reply: Please Help!!!

[Nick Khamis]

Hello Muhammad, thanks again for your response. On our test
environment, our opensips+rtpproxy server is behind NAT, and the
reason we are modifying the contact header is to point to
1001@ instead of 1001@. UA
1001 is also behind the same NAT.

My first question, do I need to modify the contact header? Or should I
be paying closer attention to the SDP payload. Making sure c=, and o=
are pointing to the right locations?

Your help is greatly appreciated.

Nick.

On 3/19/13, Muhammad Shahzad  wrote:
> Yup, that's expected to happen. ACK is sent to Contact header of 200 OK.
> So, if you mess up with it, then unexpected results will happen, as in your
> case you are perhaps setting Contact address of 200 OK pointing to opensips
> itself, instead of destination party, so ACK will obviously loop as
> expected.
>
> Thank you.
>
>
> On Mon, Mar 18, 2013 at 5:55 PM, Nick Khamis  wrote:
>
>> Hello Everyone,
>>
>> We are changing the "Contact" header in the on_reply to a public ip
>> address using:
>>
>> onreply_route[1] {
>> xlog("incoming reply\n");
>> if (has_body("application/sdp")) {
>> remove_hf("Contact");
>> append_hf("Contact:
>> \r\n");
>> append_hf("P-hint: Onreply-route -
>> fixcontact \r\n");
>>
>> }
>> }
>>
>> When doing so, ACK is going into a loop:
>>
>> U 2013/03/18 13:42:11.021017 75.15.201.2:5060 -> 192.168.2.5:5060
>> ACK sip:75.15.201.2;lr;did=b03.4af9f8f3 SIP/2.0.
>> Call-ID: VQUK2UGSQBCPHEW27UN5NBJIQM@81.201.86.45.
>> CSeq: 102 ACK.
>> From: "15178334003" ;tag=91641.
>> To: ;tag=2643FD58-346926A7.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
>> Via: SIP/2.0/UDP 7
>>
>>
>> Your help is greatly appreciated,
>>
>> Nick.
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [NEW] mi_xmlrpc_ng module

[Bogdan-Andrei Iancu]

Hi Ovidiu,

I'm really glad to hear this - the old mi_xmlrpc module already became a 
pain in maintaining - unfortunately the libxmlrpc-c3 library have 
several bugs and limitations and in time it proved hard to use in the 
opensips context.


Getting rid of it will make me personally sleep better ;) Thanks for the 
work !


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 03/19/2013 03:06 PM, Ovidiu Sas wrote:

Hello all,


A new module is available in dev svn: mi_xmlrpc_ng:
http://www.opensips.org/html/docs/modules/devel/mi_xmlrpc_ng.html
The intent is to replace the existing mi_xmlrpc module with this new
one and remove the dependency on the libxmlrpc-c3 library.

A sample config file is provided in the svn tree (in the examples directory):
http://opensips.svn.sourceforge.net/svnroot/opensips/trunk/examples/httpd.cfg
Testing and feedback is really appreciated.


Regards,
Ovidiu Sas



___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] opensips + freeradius + acc

[Tijmen de Mes]

Hi,

Are you sure database called 'radacct' ? Usually it is something like 
'radius'. If it is radius the call needs to be:
CALL update_radacct_record('radius', '2013-03-19 14:14:07','0', '', 
'','MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 
'df421e69','as287910d5');


Best regards,

--
Tijmen de Mes
AG Projects



On 03/19/2013 02:45 PM, andrzej.ciu...@asterisk.edu.pl wrote:

Hello

I have started to use freeradius for acc. Using scripts from cdr-tool.
I have problem with update procedure after BYE.
Insert after answered call goes to the table "radacct201303", but 
after BYE, when prodecure update_radacct_record is called, there is no 
result.

When I create UPDATE based on log from sqltrace log It works.
It looks like:

CALL update_radacct_record( 'radacct', '2013-03-19 
14:14:07', '0', '', '', 
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.', 
'df421e69', 'as287910d5' );


doesn't update record but handmade UPDATE:

UPDATE radacct201303  SET AcctStopTime = '2013-03-19 14:14:07', 
AcctSessionTime = CONCAT( TIME_TO_SEC(TIMEDIFF('2013-03-19 14:14:07', 
AcctStartTime)), '.', MICROSECOND(TIMEDIFF('2013-03-19 14:14:07', 
AcctStartTime)) ),AcctStopDelay = '', ConnectInfo_stop 
= '', RTPStatistics = '', Normalized = '0' WHERE AcctSessionId = 
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.' AND ((SipToTag = 
'as287910d5' AND SipFromTag = 'df421e69') OR (SipFromTag = 
'as287910d5' AND SipToTag = 'df421e69')) AND ConnectInfo_stop IS NULL 
AND (MediaInfo IS NULL OR MediaInfo != 'timeout');


does.

Has someone had same problem ?

Greetings
Andrzej





___
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

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Nick Khamis]

RTPProxy does work behind NAT. It's mediaporxy that must be on a public ip.

Thanks for your help.

Nick.

On 3/19/13, Muhammad Shahzad  wrote:
> If you are unfamiliar with rtp proxy and how it works, then it would be
> better for you to use engage_rtp_proxy rather then offer / answer model.
> Also RTP Proxy requires public IP address, its likely not to work on
> private subnets (unless you have all SIP entities on same LAN, in which
> case theoretically it should work but i have never tested it myself).
>
> Thank you.
>
>
> On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis  wrote:
>
>> I am not sure if this is the correct place to post OpenSIPS+RTPProxy
>> questions however, I tried to subscribing to the RTP proxy mailing
>> list and never heard from them since. If it is ok to post RTP proxy
>> related questions here I am trying to test OpenSIPS with RTP proxy
>> with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
>> RTPPoxy) just for testing.
>>
>> The code I am using is:
>>
>> route {
>>  force_rport();
>> }
>> route[1] {
>> if (is_method("INVITE")) {
>> t_on_branch("1");
>> t_on_reply("1");
>> t_on_failure("1");
>>
>> if (has_body("application/sdp"))  rtpproxy_offer();
>> }
>> else if (is_method("BYE|CANCEL")) {
>> unforce_rtp_proxy();
>> }
>>
>> if (!t_relay()) {
>> sl_reply_error();
>> };
>> exit;
>> }
>> onreply_route[1] {
>>  if (has_body("application/sdp")) rtpproxy_answer();
>> }
>>
>>
>> There is no way audio using RTP proxy, but audio is fine between the
>> UA without including the RTP proxy related script. Looking at the log
>> I found that RTP is prefilling the callers address twice, but not the
>> callees address.
>>
>>
>> INFO:main: rtpproxy started, pid 7287
>> INFO:handle_command: new session
>> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
>> d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
>> INFO:handle_command: new session on a port 35010 created, tag
>> d23f0168-538e-e211-8550-001b7700a65b;1
>> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5062
>> INFO:handle_command: new session
>> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
>> d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
>> INFO:handle_command: new session on a port 22982 created, tag
>> d23f0168-538e-e211-8550-001b7700a65b;2
>> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5064
>> INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
>> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
>> relayed, 0 dropped
>> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
>> relayed, 0 dropped
>> INFO:remove_session: session on ports 35010/0 is cleaned up
>> INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
>> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
>> relayed, 0 dropped
>> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
>> relayed, 0 dropped
>> INFO:remove_session: session on ports 22982/0 is cleaned up
>>
>> Is it possible to test RTP relaying with everything on the same network?
>>
>> Thanks in Advance,
>>
>> Nick.
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] opensips + freeradius + acc

[andrzej . ciupek]

Hello

I have started to use freeradius for acc. Using scripts from cdr-tool.
I have problem with update procedure after BYE.
Insert after answered call goes to the table "radacct201303", but  
after BYE, when prodecure update_radacct_record is called, there is no  
result.

When I create UPDATE based on log from sqltrace log It works.
It looks like:

CALL update_radacct_record( 'radacct',  
'2013-03-19 14:14:07', '0', '', 
 '',   
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.',  
'df421e69', 'as287910d5' );


doesn't update record but handmade UPDATE:

UPDATE radacct201303  SET AcctStopTime = '2013-03-19 14:14:07',  
AcctSessionTime = CONCAT( TIME_TO_SEC(TIMEDIFF('2013-03-19 14:14:07',  
AcctStartTime)), '.', MICROSECOND(TIMEDIFF('2013-03-19 14:14:07',  
AcctStartTime)) ),AcctStopDelay = '', ConnectInfo_stop  
= '', RTPStatistics = '', Normalized = '0' WHERE AcctSessionId =  
'MmJiNjRiYTNkNjBmNWM3YmZhNGRhOWRkZWJiMDA4ZmU.' AND ((SipToTag =  
'as287910d5' AND SipFromTag = 'df421e69') OR (SipFromTag =  
'as287910d5' AND SipToTag = 'df421e69')) AND ConnectInfo_stop IS NULL  
AND (MediaInfo IS NULL OR MediaInfo != 'timeout');


does.

Has someone had same problem ?

Greetings
Andrzej





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] [NEW] mi_xmlrpc_ng module

[Ovidiu Sas]

Hello all,


A new module is available in dev svn: mi_xmlrpc_ng:
http://www.opensips.org/html/docs/modules/devel/mi_xmlrpc_ng.html
The intent is to replace the existing mi_xmlrpc module with this new
one and remove the dependency on the libxmlrpc-c3 library.

A sample config file is provided in the svn tree (in the examples directory):
http://opensips.svn.sourceforge.net/svnroot/opensips/trunk/examples/httpd.cfg
Testing and feedback is really appreciated.


Regards,
Ovidiu Sas

-- 
VoIP Embedded, Inc.
http://www.voipembedded.com

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [VoiceOps] Older Cisco SIP firmware download

[Max Mühlbronner]

Sorry, wrong list :)


On 03/19/2013 12:24 PM, Max Mühlbronner wrote:


Google:

inurl:cmterm-7941_7961-sip.8-5-4.zip


But its a russian site, not sure if this is legit?


Best Regards


On 03/18/2013 11:37 PM, Adam Baird wrote:

Hi all.

I have been tasked with performing a SIP interop with the Cisco 7941 
model IP phone.  I've failed to get it working with the latest SIP 
9.x which is still available from Cisco w/o a contract.


From what I've found in my research, an older version 8.5.4 might be 
the way to go but I can't download it from Cisco without a contract. 
  I'm not sure why they chose to protect the older versions but not 
the newer.


Anyway, would any of you happen to know an alternate place for me to 
download the file?  I'm not sure I'll be able to purchase a contract 
to allow me to download the file since this model went EOS so long ago.


I'm looking for: cmterm-7941_7961-sip.8-5-4.zip

Feel free to direct message me if you prefer.

Thanks
-Adam


___
VoiceOps mailing list
voice...@voiceops.org
https://puck.nether.net/mailman/listinfo/voiceops




___
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

Re: [OpenSIPS-Users] [VoiceOps] Older Cisco SIP firmware download

[Max Mühlbronner]

Google:

inurl:cmterm-7941_7961-sip.8-5-4.zip


But its a russian site, not sure if this is legit?


Best Regards


On 03/18/2013 11:37 PM, Adam Baird wrote:

Hi all.

I have been tasked with performing a SIP interop with the Cisco 7941 
model IP phone.  I've failed to get it working with the latest SIP 9.x 
which is still available from Cisco w/o a contract.


From what I've found in my research, an older version 8.5.4 might be 
the way to go but I can't download it from Cisco without a contract.   
I'm not sure why they chose to protect the older versions but not the 
newer.


Anyway, would any of you happen to know an alternate place for me to 
download the file?  I'm not sure I'll be able to purchase a contract 
to allow me to download the file since this model went EOS so long ago.


I'm looking for: cmterm-7941_7961-sip.8-5-4.zip

Feel free to direct message me if you prefer.

Thanks
-Adam


___
VoiceOps mailing list
voice...@voiceops.org
https://puck.nether.net/mailman/listinfo/voiceops


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Muhammad Shahzad]

If you are unfamiliar with rtp proxy and how it works, then it would be
better for you to use engage_rtp_proxy rather then offer / answer model.
Also RTP Proxy requires public IP address, its likely not to work on
private subnets (unless you have all SIP entities on same LAN, in which
case theoretically it should work but i have never tested it myself).

Thank you.


On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis  wrote:

> I am not sure if this is the correct place to post OpenSIPS+RTPProxy
> questions however, I tried to subscribing to the RTP proxy mailing
> list and never heard from them since. If it is ok to post RTP proxy
> related questions here I am trying to test OpenSIPS with RTP proxy
> with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
> RTPPoxy) just for testing.
>
> The code I am using is:
>
> route {
>  force_rport();
> }
> route[1] {
> if (is_method("INVITE")) {
> t_on_branch("1");
> t_on_reply("1");
> t_on_failure("1");
>
> if (has_body("application/sdp"))  rtpproxy_offer();
> }
> else if (is_method("BYE|CANCEL")) {
> unforce_rtp_proxy();
> }
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
> onreply_route[1] {
>  if (has_body("application/sdp")) rtpproxy_answer();
> }
>
>
> There is no way audio using RTP proxy, but audio is fine between the
> UA without including the RTP proxy related script. Looking at the log
> I found that RTP is prefilling the callers address twice, but not the
> callees address.
>
>
> INFO:main: rtpproxy started, pid 7287
> INFO:handle_command: new session
> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
> d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
> INFO:handle_command: new session on a port 35010 created, tag
> d23f0168-538e-e211-8550-001b7700a65b;1
> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5062
> INFO:handle_command: new session
> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
> d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
> INFO:handle_command: new session on a port 22982 created, tag
> d23f0168-538e-e211-8550-001b7700a65b;2
> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5064
> INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 35010/0 is cleaned up
> INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 22982/0 is cleaned up
>
> Is it possible to test RTP relaying with everything on the same network?
>
> Thanks in Advance,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as INVITE?

[Muhammad Shahzad]

Load balancer depends on SIP address only for routing, so just about any
SIP server and just about any SIP method, you can use this module for load
balancing. However, you need to plan carefully how on-net calls and
presence will work in such an architecture!

Thank you.


On Mon, Mar 18, 2013 at 5:56 PM, Tuomas Kaikkonen <
tuomas.kaikko...@twistpair.com> wrote:

> Can the Load Balancer module be configured to balance REGISTERs as well as
> the INVITES so that the above mentioned setup would work? OR is the Load
> Balancer module just useful for balancing RTP Proxy / media server
> resources for INVITEs?
>
> ** **
>
> I’m new to the Load Balancer module of OpenSIPs. I am running OpenSIPs
> stable branch 1.7 – just by looking examples from the documentation it
> looks like INVITE load balancing is supported.
>
> ** **
>
> Tuomas Kaikkonen
>
> ** **
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK Loop when changing contact on_reply: Please Help!!!

[Muhammad Shahzad]

Yup, that's expected to happen. ACK is sent to Contact header of 200 OK.
So, if you mess up with it, then unexpected results will happen, as in your
case you are perhaps setting Contact address of 200 OK pointing to opensips
itself, instead of destination party, so ACK will obviously loop as
expected.

Thank you.


On Mon, Mar 18, 2013 at 5:55 PM, Nick Khamis  wrote:

> Hello Everyone,
>
> We are changing the "Contact" header in the on_reply to a public ip
> address using:
>
> onreply_route[1] {
> xlog("incoming reply\n");
> if (has_body("application/sdp")) {
> remove_hf("Contact");
> append_hf("Contact:
> \r\n");
> append_hf("P-hint: Onreply-route -
> fixcontact \r\n");
>
> }
> }
>
> When doing so, ACK is going into a loop:
>
> U 2013/03/18 13:42:11.021017 75.15.201.2:5060 -> 192.168.2.5:5060
> ACK sip:75.15.201.2;lr;did=b03.4af9f8f3 SIP/2.0.
> Call-ID: VQUK2UGSQBCPHEW27UN5NBJIQM@81.201.86.45.
> CSeq: 102 ACK.
> From: "15178334003" ;tag=91641.
> To: ;tag=2643FD58-346926A7.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 7
>
>
> Your help is greatly appreciated,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips RTP issue

[Muhammad Shahzad]

Well, if call establishes successfully (even without media) then opensips
is perfectly fine. The problem is on Asterisk side. Check NAT settings in
sip.conf for OpenSIPs trunk.

Thank you.


On Mon, Mar 18, 2013 at 9:25 PM, Jagadish Thoutam wrote:

>
> Can any one  help Me on this
>
> i configured Opensips i can able to receive the call on Asterisk server
> whn i dial my did
>
>
> ###
> External DID Provider --->  Opensips(NAT with Public
> IP)(192.168.1.62)->Asterisk (192.168.1.222)
>
> ###
>
> My Opensips is on NAT with Public Ip
>
> But i cant able to hear anything on call  (No audio)
>
>
> Here i my opensips.cfg file
>
>
> XX.XX.XX.XX  = My Public Ip
>
> ##
> ## Core Parameters
> ##
> # chroot=
> # group="opensips"
> # user="opensips"
> # dbversion_table=
> disable_core_dump=no
> max_while_loops=100
> maxbuffer=262144
> memdump=3
> memlog=2
> # open_files_limit=2048
> server_signature=no
> server_header="Server: OpenSIPS"
> user_agent_header="User-Agent: OpenSIPS"
>
> ##
> ## Core Fork Parameters
> ##
> fork=yes
> children=8
> tcp_children=8
>
> ##
> ## Core Logging Parameters
> ##
> debug=3
> sip_warning=0
> log_stderror=no
> log_facility=LOG_LOCAL0
> log_name="opensips"
>
> ##
> ## Aliases
> ##
> auto_aliases=yes
> #alias=localhost
> #alias=localhost.localdomain
>
> ##
> ## Connectivity
> ##
> #listen=udp:eth0:5060
> #listen=udp:eth0:5060
> #listen=tcp:eth0:5060
> #listen=udp:eth0:7000
> #listen=tcp:eth0:7000
> #listen=udp:eth0:7000
> #listen=tcp:eth0:7000
> # listen=udp:eth1:5060
> #tos=IPTOS_LOWDELAY
> # advertised_address=174.129.131.38
> # advertised_port=5060
> mcast_loopback=no
> mcast_ttl=1
> mhomed=0
> # tcp_accept_aliases
> tcp_connect_timeout=3
> tcp_connection_lifetime=120
> tcp_max_connections=2048
> # tcp_poll_method=select
>
> ##
> ## DNS
> ##
> dns=no
> dns_retr_time=1
> dns_retr_no=3
> # dns_servers_no=2
> dns_try_ipv6=no
> disable_dns_blacklist=yes
> disable_dns_failover=no
> dns_use_search_list=no
> rev_dns=no
>
> ##
> ## SIP
> ##
> check_via=0
> #! disable_503_translation=no
> disable_stateless_fwd=no
> disable_tcp=no
> # disable_tls=no
> #! reply_to_via=1
>
> ##
> ## TLS
> ##
> # disable_tls=no
> # listen=tls:your_IP:5061
> # tls_verify_server=1
> # tls_verify_client=1
> # tls_require_client_certificate=0
> # tls_method=TLSv1
> # tls_certificate="/usr/local/etc/opensips/tls/user/user-cert.pem"
> # tls_private_key="/usr/local/etc/opensips/tls/user/user-privkey.pem"
> # tls_ca_list="/usr/local/etc/opensips/tls/user/user-calist.pem"
>
> ##
> ## Destination Blacklist
> ##
> # dst_blacklist=gw:{( tcp , 192.168.2.100 , 5060 , "" ),( any ,
> 192.168.2.101 , 0 , "" )}
> # dst_blacklist=net_filter2:{ !( any , 192.168.30.0/255.255.255.0 , 0 ,
> "" )}
>
> ##
> ## Attribute Value Pairs
> ##
> # avp_aliases="uuid=I:660;email=s:email_addr;fwd=i:753"
>
> ##
> ## Module Loading
> ##
> mpath="/usr/local/lib64/opensips/modules/"
>
> loadmodule "db_mysql.so"
> loadmodule "localcache.so"
> loadmodule "signaling.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "dialog.so"
> loadmodule "maxfwd.so"
> loadmodule "rr.so"
> loadmodule "path.so"
> loadmodule "uri.so"
> loadmodule "textops.so"
> loadmodule "usrloc.so"
> loadmodule "nathelper.so"
> loadmodule "nat_traversal.so"
> loadmodule "uac_redirect.so"
> loadmodule "dispat

Re: [OpenSIPS-Users] ACK Timer

[Răzvan Crainea]

Hi, Khaled!

Not really without an if close. You can use something like:

if (is_method("ACK"))
$avp(timeout2) = 3540.

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/18/2013 11:15 PM, M.Khaled W Chehab wrote:


Hi, Razvan .

You mean to add $avp(timeout2) = 3540; before  match dialog with 
no if close


if (has_totag() && (uri == myself) && 
is_method("INVITE|ACK|BYE|UPDATE")) {


*$avp(timeout2) = 3540;*

if(match_dialog()) {

xlog("L_INFO", "< in-dialog topology hiding request - $DLG_dir 
> \n");


if (check_route_param("nat=yes")) {

setflag(5);

setbflag(6);

}

if ( is_method("INVITE")) {

$avp(timeout2) = 3;

} else if (is_method("ACK")) {

$avp(timeout2) = 3540;

}

route(1);

Regards

*From:*users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Razvan Crainea

*Sent:* Monday, March 18, 2013 4:53 PM
*To:* users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] ACK Timer

Hi, Khaled!

The match_dialog() function also matched the dialog and updates the 
timer, therefore you should also set the pseudo variable for ACK 
before the match_dialog() call.


Best regards,


Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/18/2013 03:38 PM, M.Khaled W Chehab wrote:

Dear Bogdan,

I set the timer as below and all my calls have now 3 sec duration
,please advice

if (has_totag() && (uri == myself) &&
is_method("INVITE|ACK|BYE|UPDATE")) {

if(match_dialog()) {

xlog("L_INFO", "< in-dialog topology hiding request - $DLG_dir
> \n");

if (check_route_param("nat=yes")) {

setflag(5);

setbflag(6);

}

if ( is_method("INVITE")) {

$avp(timeout2) = 3;

} else if (is_method("ACK")) {

$avp(timeout2) = 3540;

}

route(1);

exit;

  }

if (has_totag()) {

  if (is_method("ACK")) {

$avp(timeout2) = 3540;

  }

  # sequential request withing a dialog should

  # take the path determined by record-routing

  if (loose_route()) {

etc……..

} else {

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;

   } else {

  if ( is_method("INVITE")) {

$avp(timeout2) = 3;

  }

   }

Regards

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Thursday, March 14, 2013 6:59 PM
*To:* OpenSIPS users mailling list
*Cc:* M.Khaled W Chehab; 'Brito Nicolas'
*Subject:* Re: [OpenSIPS-Users] ACK Timer

Hi Khaled,

your mistake here is to set the timeout for INVITE under the
has_totag() branch - initial INVITEs do not have TO tags.

Try:


if (has_totag()) {
if (is_method("ACK")) {
$avp(timeout2) = 3540;
}
  . (loose_route)

} else {

  if ( is_method("INVITE")) {
$avp(timeout2) = 3;
}

...

}

Regards,
Bogdan

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

  





___

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


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users