Re: [SR-Users] 4.4 compiler warnings on debian jessie

[Ovidiu Sas]

If you have a patch, I can test it for you.

-ovidiu
On May 11, 2016 10:56,  wrote:

> > debian jessie c compiler complains about these in kamailio 4.4:
> >
> > CC (gcc) [sip-proxy]mem/tlsf_malloc.o
> > mem/tlsf_malloc.c: In function 'tlsf_malloc_init_pkg_manager':
> > mem/tlsf_malloc.c:1353:16: warning: assignment from incompatible pointer
> type
> >   ma.xmalloc= tlsf_malloc;
>
> Hi,
>
> is this happening on a x86 32 bits machine? This issue was pointed out
> already
> by Ovidu Sas: <
> http://lists.sip-router.org/pipermail/sr-dev/2016-January/thread.html#32757
> >,
> but I had no 32 bits machine to test this on, and also no idea why it does
> not
> work (I see no error message on my 64 bits Debian Jessie).
>
> All I see is that we are assigning a
> void* (*)(tlsf_t, size_t)
> to a
> void* (*)(void* , unsigned long)
> yet tlsf_t is defined as `typedef void* tlsf_t`, and size_t and unsigned
> long
> have the same width on this architecture...
>
> Do compiling with clang generates an error too? If yes, is the error
> message
> more specific about the issue?
>
> Any other idea?
>
> --
> Camille
>
>
> _
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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Ovidiu Sas]

If the SDP is correct, then you might have specific issues related to your
specific deployment case. Snippets from others config files won't help. You
really need to investigate and understand your particular issue that you
are facing and fix it accordingly.

Regards,
Ovidiu Sas
On Oct 8, 2015 09:04, "Dirk Teurlings - SIGNET B.V." <dteurli...@signet.nl>
wrote:

> On 30-09-15 13:29, Fred Posner wrote:
>
>>
>> Without a version of rtpproxy using the -A flag, you'll need to either
>> (1) update to a different version of rtpproxy or (2) skip rtpproxy and
>> have your asterisk handle all the rtp.
>>
>
> I tried rtpproxy v2, with the -A flag in bridge mode ( -A
> privateip/publicip ). This doesn't reflect anything in the SIP headers.
>
> The problem is a bit more complex I think, because all INVITEs to gateways
> contain the same internal IPs from Asterisk and Kamaialio in their From and
> To header. SDP information is correctly being displayed. But it seems that
> some UAs disregard what's in the SDP descriptors and just look at the SIP
> headers (To/From/Contact).
>
> Can anyone share their config snippets about how they've delt with the
> Asterisk behind NAT situation? It would really be appreciated!
>
> Cheers,
> Dirk
>
>
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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Ovidiu Sas]

If the SDP is correct, then you might have specific issues related to your
specific deployment case. Snippets from others config files won't help. You
really need to investigate and understand your particular issue that you
are facing and fix it accordingly.

Regards,
Ovidiu Sas
On Oct 8, 2015 09:04, "Dirk Teurlings - SIGNET B.V." <dteurli...@signet.nl>
wrote:

> On 30-09-15 13:29, Fred Posner wrote:
>
>>
>> Without a version of rtpproxy using the -A flag, you'll need to either
>> (1) update to a different version of rtpproxy or (2) skip rtpproxy and
>> have your asterisk handle all the rtp.
>>
>
> I tried rtpproxy v2, with the -A flag in bridge mode ( -A
> privateip/publicip ). This doesn't reflect anything in the SIP headers.
>
> The problem is a bit more complex I think, because all INVITEs to gateways
> contain the same internal IPs from Asterisk and Kamaialio in their From and
> To header. SDP information is correctly being displayed. But it seems that
> some UAs disregard what's in the SDP descriptors and just look at the SIP
> headers (To/From/Contact).
>
> Can anyone share their config snippets about how they've delt with the
> Asterisk behind NAT situation? It would really be appreciated!
>
> Cheers,
> Dirk
>
>
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Re: [SR-Users] Familiar topic about Kamailio with Asterisk behind NAT

[Ovidiu Sas]

If you have kamailio bound to both private and public interfaces, then
you need to run rtpproxy in bridge mode (vanilla rtpproxy).
Another option is to get rid of the rtpproxy altogether and let
asterisk handle the media, but you will need to make sure that:
 - kamailio is rewriting IPs in SDP provided by asterisk;
 - you perform port forwarding for the NATed RTP ports to asterisk.

Also, you need to be more specific about the client trying to connect
to the private address of asterisk. Are you referring to media? In
this case it seems that you didn't engage rtpproxy. Is it about
signalling? Then you might deal with a bogus SIP client.

Take a look at other clients that are working and find out why this
particular one doesn't work.

Regards,
Ovidiu Sas

On Wed, Sep 30, 2015 at 4:22 AM, Dirk Teurlings - SIGNET B.V.
<dteurli...@signet.nl> wrote:
> Hi,
>
> I found some existing topics on this but failed to get a solutions out of
> them.
>
> We're running into some issues with client devices connecting to our private
> addresses. The way it is setup now:
>
>  CLIENTS <-> (NAT) <-> INTERNET <-> KAMAILIO(4.2.5) with RTPPROXY(v1)
> <-> PRIVATE LAN <-> ASTERISK (v1.8)
>
> Our Kamaialio and Asterisk are in a private address range, but Kamailio also
> has a public interface. Most of the clients (about 95%) work well with this
> setup, but a couple don't. We have one case now where the CLIENT tries to
> connect to the private address of ASTERISK. And of course, that doesn't
> work.
>
> I'm kind of stuck as to where I need to fix this. I tried using the
> externaddr option in Asterisk to solve it on that end. But that didn't help
> anything. The NAT options in Kamailio are not really suited for this, as
> they tend to fix client NAT problems.
>
> Any pointer or help would be greatly appriciated.
>
> Cheers,
> Dirk
>
>
>
>
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Re: [SR-Users] Best practices for troubleshooting deadlocks?

[Ovidiu Sas]

There is 'kamctl trap' which does a backtrace on all kamailio
processes, similar with what your script does.
Use top to identify which processes are locked (100% CPU utilization)
and after that ... code inspection.

-ovidiu

On Mon, Sep 28, 2015 at 1:26 PM, Alex Balashov
 wrote:
> We just encountered another one of these famed deadlocks. Any suggestions
> for how to analyse them beyond what I've already trotted out here?
>
>
> On 09/14/2015 05:47 PM, Alex Balashov wrote:
>
>> Hello,
>>
>> Very occasionally, we encounter what appear to be deadlocks in all UDP
>> receiver threads. All Kamailio processes are running, but no SIP
>> messages are being processed.
>>
>> On one of our high-volume installation, this happens extremely
>> infrequently -- maybe once every month or two. On these occasions, the
>> operator restarts the proxy before we get a chance to go in and figure
>> out what's going on.
>>
>> So, I'm trying to provide the operator with a procedure to execute prior
>> to restarting the proxy on these occasions, so that we can see a
>> snapshot of where the receiver threads are stuck. As far as I can tell,
>> unless Kamailio itself segfaults, there's no specific PID that one can
>> attach GDB to in order to get an overhead snapshot of all the child
>> processes.
>>
>> Here's what I came up with:
>>
>> -
>> #!/bin/bash
>>
>> kamcmd -s /tmp/kamailio_ctl ps > thread_log.txt
>> echo >> thread_log.txt
>>
>> while read PID;
>> do
>>  gdb --pid=$PID<>thread_log.txt
>> set print elements 0
>> thread apply all bt full
>> generate-core-file
>> detach
>> EOF
>> done < <(kamcmd -s /tmp/kamailio_ctl ps | grep 'udp receiver' | awk
>> '{print $1}')
>> -
>>
>> As far as I can tell, this should give me the most ample visibility into
>> the state of the threads, with further core dumps to inspect if
>> follow-up is needed. Hopefully this will result in some fixes back to
>> the project.
>>
>> However, if there are any other suggestions for information to grab in
>> such a scenario, I'm all ears.
>>
>> Thanks in advance!
>>
>> -- Alex
>>
>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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Re: [SR-Users] Re-invites from carrier breaks the call

[Ovidiu Sas]

Well ... kamailio is a proxy (not a B2BUA), and in dialog requests should
not point at the proxy.
If you are paranoid about it, then you can alter signalling by mangling and
de-mangling the Contact header for requests and reply to achieve that.

Regards,
Ovidiu Sas

On Thu, Feb 19, 2015 at 12:59 PM, Andres and...@telesip.net wrote:

  On 2/18/15 9:44 PM, Will Ferrer wrote:

 Hi Alex

  Thanks so much for the reply.

  Is there anything that we could do perhaps that is a more creative
 solution, for instance not passing the re-invite all the way to the
 softphone and just responding from the kamailio box handling the call?

  We tried this as well actually, but we didn't get it to work. We just
 sent a 200 ok from the kamailio box, no sdp or anything on the packet since
 we sent it with just send_reply and the carrier just sent a bye.

  Hopefully there is something clever we could do to correct the problem,
 it is preventing us from using alot of our carriers since the re-invite
 breaks our clients softphones.

  Thanks again for the assistance.

 We have struggled with this issue ourselves.  The problem was that we did
 not want our SIP server to behave like an open relay.  We were seeing that
 the session-timer Re-Invites have a  Request-URI with the IP of the other
 endpoint instead of the Proxy.  If the SIP server is an open relay then no
 problem, but ours is not so the config file was very strict and dropped the
 Re-Invite (since the Request-URI had an external IP) thus dropping the
 call.  The config file could be enhanced by testing for has_totag() since
 the Re-Invite has the totag but an original Invite does not, but the hacker
 could put a bogus totag and make calls so its more secure to leave it this
 way.  We ended up disabling session-timers at some our clients PBXs.  Its
 always a balancing act between convenience/services and more security.  We
 chose more security.


  All the best.

  Will Ferrer

 On Wed, Feb 18, 2015 at 6:07 PM, Alex Balashov abalas...@evaristesys.com
 wrote:

  Kamailio cannot correct this. This is an endpoint issue. The whole
 point of Record-Route is to hairpin sequential requests (and indeed, their
 replies) through the proxy. The endpoints need to comply by affixing the
 correct Route header to the end-to-end ACK.

  --
 Sent from my BlackBerry. Please excuse errors and brevity.
*From: *Will Ferrer
 *Sent: *Wednesday, February 18, 2015 9:01 PM
 *To: *Kamailio (SER) - Users Mailing List
 *Reply To: *Kamailio (SER) - Users Mailing List
 *Subject: *[SR-Users] Re-invites from carrier breaks the call

  Hi All

  We have any issue with re invites coming from the carrier.

  When a reinvite occurs, our softphone client gets the invite, sends a
 100, and then sends 200 ok. However the 200 ok does not have the softphones
 ip in the record route. Since it's not in the record route the ack from the
 carrier never makes it's way all the back to the softphone.

  This causes the softphone to keep sending 200 oks since it never gets
 the ack.

  Eventually the softphone gets tired of sending 200 oks and sends a bye.

  Is there any way that Kamailio can help me correct for this, or do we
 need to have our clients use different softphones? If it has to be handled
 via softphones is there even a softphone that can account for this?

  Thanks for all your assistance in advance.

 All the best.

  Will Ferrer

  Switchsoft




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Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)

[Ovidiu Sas]

You could simply let the RTP traffic to flow directly between FS and
endpoints (no need for rtpproxy).
All you need to do is:
 - forward the appropriate RTP ports to FS;
 - fix the private IP in SDP by replacing it with the public IP for
the inbound rtp streams (to FS).

-ovidiu

On Mon, Feb 16, 2015 at 11:30 AM, Giovanni Maruzzelli gmar...@gmail.com wrote:
 dear Kamailians,

 I have Kamailio+rtpproxy in front of FreeSWITCH.

 Kamailio and FreeSWITCH are on the same private network.
 Public Internet IP address ports are redirected to Kamailio and
 rtpproxy (same situation as in Amazon EC2).
 Clients comes from Internet, and make calls to Internet, SIP signaling
 passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH
 originate an outbound B leg INVITE, and then bridge the legs).

 Using rtpproxy with -A advertise patch from Daniel, this topology
 works fine in a traditional telco way: rtp goes from caller to
 rtpproxy to callee, and viceversa.

 Now I want to maintain FreeSWITCH in the middle of rtp flow all the
 time, in a pure b2bua way, so it can control and analyze the media
 streams.

 So, I need rtpproxy to act paying attention to direction, as in
 caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa).

 Normally I would use Kamailio multihomed and rtpproxy in bridging
 mode. But I cannot have a NIC on the public address.

 How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like
 environment? (eg: no public address attached to machine, but ports
 redirection from public address).

 I read this trick from Hugh Waite:

 I have used rtpproxy (with the advertised address patch) in Amazon to
 bridge media between internet facing and private subnets in a VPC.
 I found that I couldn’t use different advertised addresses depending
 on which direction the signalling was going on a single private IP
 address. I worked around this by allocating a second private ip
 address to the instance and used that in the ‘bridge’.
 -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15

 Can you explain how to use this trick, or another way (without
 additional addresses is gladly accepted!) to reach the same result
 (rtp always passing through FreeSWITCH) ?

 Thank you all in advance,

 -giovanni

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Re: [SR-Users] xhttp_rpc - dispatcher html-rpc

[Ovidiu Sas]

Check if dispatcher.reload works via the rpc interface: kamcmd dispatcher.reload
Also, which version of kamailio are you using: kamailio -V

Regards,
Ovidiu Sas

On Mon, Jan 12, 2015 at 7:04 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Maybe you can run kamailio with debug=3 and see if you get anything useful
 in the syslog. Otherwise, I haven't used xhttp_rpc myself for dispatcher.

 Cheers,
 Daniel

 On 08/01/15 17:24, Yuriy Gorlichenko wrote:

 Sorry. Yes xhttp_rpc.

 I use this cfg for this module
 modparam(xhttp_rpc, xhttp_rpc_root, http_rpc)

 then I try to restart dispatcher from http
 http://10.0.1.12:8080/http_rpc/dispatcher/dispatcher.reload?arg=dispatcher.reload

 but nothing going on. Dispatcher not reloading

 2015-01-08 15:42 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

 Hello,

 do you refer to xmlrpc or xhttp_rpc? html-rpc is not something I could
 relate to something in Kamailio ...

 Cheers,
 Daniel

 On 08/01/15 04:11, Yuriy Gorlichenko wrote:

 Hello. How I must use this function for dynamic reload dispatcher without
 restarting me server?


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Re: [SR-Users] Kamailio fails to generate core dump

[Ovidiu Sas]

IIRC, you get that message if core dumping is not properly enabled at
system level. Check that you have those settings and kill kamailio with a
SIGSEGV to test.
It always worked for me. I can't afford to install a server without core
dumping enabled.

-ovidiu
On Jan 3, 2015 5:23 PM, Alex Balashov abalas...@evaristesys.com wrote:

 Thanks, Ovidiu. But since the message core was not generated comes from
 handle_sigs() in Kamailio, what I am really interested is in what reason
 Kamailio itself would have for not dumping core. I assume that the
 situation would look different if Kamailio tried to dump core, but was
 restrained from doing so by operating system factors such as ulimits.

 From main.c:handle_sigs(), it appears that the log message came from here:

 #ifdef WCOREDUMP
 ... LM_ALERT(core was %sgenerated\n, WCOREDUMP(chld_status) ?   : not
  );
 #endif

 WCOREDUMP() as I understand it allows one to examine the return code of a
 dead child to determine if it returned a core dump. Is further information
 available in such a case, like a kind of errno for such cases?

 -- Alex

 On 01/03/2015 05:00 PM, Ovidiu Sas wrote:

  Pretty annoying problem :(

 Here's how I enable core dumps on linux installs in sysctl (reboot
 required):
 fs.suid_dumpable = 1
 kernel.core_pattern = /tmp/core.%e.%u.%t
 kernel.core_uses_pid = 1

 On a live system (no need to restart):
 echo 1  /proc/sys/fs/suid_dumpable
 echo 1  /proc/sys/kernel/core_uses_pid
 echo /tmp/core.%e.%u.%t  /proc/sys/kernel/core_pattern

 Hope that now you will be able to enjoy the core dumps :-/

 Regards,
 Ovidiu Sas

 On Sat, Jan 3, 2015 at 4:38 PM, Alex Balashov abalas...@evaristesys.com
 wrote:

 Hi,

 I recently had a simultaneous crash on three production instances of
 Kamailio. Unfortunately, I have not been able to get to the bottom of the
 issue because Kamailio refused to produce a core dump:

 Dec 30 15:37:29 xx /usr/local/sbin/kamailio[13743]: ALERT: core
 [main.c:784]: handle_sigs(): child process 13763 exited by a signal 11
 Dec 30 15:37:29 xx /usr/local/sbin/kamailio[13743]: ALERT: core
 [main.c:787]: handle_sigs(): core was not generated

 I do not have disable_core_dump=yes (default is no), and checked ulimits
 for
 limits on core file size. Kamailio was running as root, and there was no
 limit.

 What are the other reasons why Kamailio may not generate a core dump in
 such
 a case?

 Thanks!

 -- Alex

 --
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 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States

 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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 Decatur, GA 30030
 United States

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 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] Thread safety of shared variables

[Ovidiu Sas]

It shouldn't ... Based on how many workers and cores you have, the
probability of having collisions should be pretty low.  You can
process more then the minimum of workers/cores available on your
server.

Regards,
Ovidiu Sas

On Wed, Nov 19, 2014 at 11:21 AM, Alex Balashov
abalas...@evaristesys.com wrote:
 Thanks, it was actually increment I was interested in.

 Can one get around it with an intermediate variable?

 $var(y) = $shv(x);
 $shv(x) = $var(y) + 1;

 I would just use a lock(), but I'm afraid that it will serialise message
 processing too much at high volume, due to the blocking.


 On 19 November 2014 01:23:14 GMT-05:00, Daniel-Constantin Mierla
 mico...@gmail.com wrote:

 For what kind of operation?

 Reading or setting the value are safe, but updating it with its own
 value used in an expression is not.

 Safe:

 xlog(value is $sht(x)\n);
 $sht(x) = 1;

 Race:

 $sht(x) = $sht(x) + 1;

 Cheers,
 Daniel

 On 19/11/14 00:27, Alex Balashov wrote:

  Does setting $shv()s in script require lock()ing, or is it inherently
  thread-safe?

  Thanks!



 --
 Sent from my Nexus 10, with all the figments of autocorrect that might
 imply.

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 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] Thread safety of shared variables

[Ovidiu Sas]

s/You can process/You can't process

Typo :)

-ovidiu

On Wed, Nov 19, 2014 at 11:35 AM, Ovidiu Sas o...@voipembedded.com wrote:
 It shouldn't ... Based on how many workers and cores you have, the
 probability of having collisions should be pretty low.  You can
 process more then the minimum of workers/cores available on your
 server.

 Regards,
 Ovidiu Sas

 On Wed, Nov 19, 2014 at 11:21 AM, Alex Balashov
 abalas...@evaristesys.com wrote:
 Thanks, it was actually increment I was interested in.

 Can one get around it with an intermediate variable?

 $var(y) = $shv(x);
 $shv(x) = $var(y) + 1;

 I would just use a lock(), but I'm afraid that it will serialise message
 processing too much at high volume, due to the blocking.


 On 19 November 2014 01:23:14 GMT-05:00, Daniel-Constantin Mierla
 mico...@gmail.com wrote:

 For what kind of operation?

 Reading or setting the value are safe, but updating it with its own
 value used in an expression is not.

 Safe:

 xlog(value is $sht(x)\n);
 $sht(x) = 1;

 Race:

 $sht(x) = $sht(x) + 1;

 Cheers,
 Daniel

 On 19/11/14 00:27, Alex Balashov wrote:

  Does setting $shv()s in script require lock()ing, or is it inherently
  thread-safe?

  Thanks!



 --
 Sent from my Nexus 10, with all the figments of autocorrect that might
 imply.

 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] cSeq increasing

[Ovidiu Sas]

Yes, you can. But this will break subsequent CSeq numbers in all requests
within the dialog. Also, you will need to restore proper CSeq for replies
to the INVITE for which the CSeq was altered.

-ovidiu
On Oct 30, 2014 9:59 AM, Yuriy Gorlichenko ovoshl...@gmail.com wrote:

 Does it possible increase cSeq manually (for example remove  and then
 append headers?) for UAC module when send INVITE messages with Auth, or
 kamailio have pseudovar for this header?

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Re: [SR-Users] RPM Based installs...

[Ovidiu Sas]

Maybe it has to do with the old dependency on libconfuse.
The dependency is no longer required since 4.1 and all release bigger
then 4.1 should provide the rpm.

Right now, the rpm is available for Fedora:
http://download.opensuse.org/repositories/home:/kamailio:/v4.2.x-rpms/Fedora_20/x86_64/kamailio-carrierroute-4.2.0-14.1.x86_64.rpm

It should be also built for CentOS and RHEL.
Please open a bug report.

-ovidiu

On Thu, Oct 23, 2014 at 12:52 PM, Derrick Bradbury derri...@halex.com wrote:
 Just a question for the RPM maintainers...

 Is there a reason for some of the modules (such as carrierroute) not being
 included in the RPMS?

 It's been bugging me for a while, when I do an installation, I have to go
 and compile all the sources  Not that it's an issue, just easier to do
 it from RPM if available.

 Thanks!
 Derrick


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Re: [SR-Users] dispatcher behind NAT - lost ACK

[Ovidiu Sas]

The issue here is that your carrier is detecting you as a NATed client
and this breaks your ACK routing.

I overloaded the meaning of interfaces.  Two sockets (same IP,
different ports) works fine in a multihomed setting.
One socket for internal traffic and one socket (with external
advertised IP) for external traffic.

Regards,
Ovidiu Sas

On Thu, Oct 23, 2014 at 11:40 PM, Nicholas Gill n...@etellicom.com wrote:
 Hi Ovidiu,

 On 18/10/14 00:37, Ovidiu Sas wrote:

 Which is bad, it should be the IP of the FS server.

 I investigated and I'm not sure this is the issue.

 Unfortunately when I named the various addresses it obscured the fact that
 the ip address kamailio.int is the IP address of the freeswitch server -
 i.e. FS and Kamailio were hosted on the same machine for that test.

 Regarding multiple interfaces the configuration does listen on two ports,
 one for internal and one for external. What benefit would making this on two
 distinct interfaces bring?

 Cheers,

 -nick


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Re: [SR-Users] dispatcher behind NAT - lost ACK

[Ovidiu Sas]

It seems that you SIP provider detects your NAT settings and it is
trying to fix it, which breaks your setup.
See the Contact in the 200OK: Contact:
sip:0390156...@kamailio.int:5070;transport=udp
Which is bad, it should be the IP of the FS server.
See the RURI in ACK: sip:0390156...@kamailio.ext:5061;transport=udp
The IP was switched from int to ext.

Probably a better setup would be to listen on multiple interfaces, one
for external traffic and one for internal traffic.
And then you will need to ask your SIP provider to stop messing with
the Contact header or you will need to circumvent that by doing some
header and routing manipulations.


Regards,
Ovidiu Sas

On Thu, Oct 16, 2014 at 9:09 PM, Nicholas Gill n...@etellicom.com wrote:
 Hello sr-users,

 We have Kamailio (behind NAT) configured acting as a proxy in front of some
 FreeSWITCH servers.

 There appears to be something amiss with my inbound (dispatcher)
 configuration which leads to misaddressing / misdelivery of the ACK after
 200 OK (outbound calls appear to be proxied correctly [1]).

 Calls from the sip provider incoming to Kamailio are distributed by the
 dispatcher module to the FS server (For testing in this simplified scenario
 FS and Kamailio are on the same machine):

 SIP provider (5060) sends an invite to Kamailio(5061), Kamailio uses the
 dispatcher module to select a backend server and forwards the INVITE to
 FS(5070) (see inbound-callflow.png; kamailio.ext and kamailio.int are the
 same machine, just public/private addresses).

 I notice that at no point is a Via/record-route header for the FS server
 inserted into the forwarded session. I'm not actually certain this is a
 requirement, but I can't think of another obvious way that Kamailio could
 proxy the same session to the same FS server.

 The 100 Trying seems unremarkable [2], I suspect the 200 OK [3] is
 problematic. It has been proxied from the FS server, however contains no
 reference to the FS server address either in the via headers nor
 record-route (kamailio.int:5070). The incoming ACK [4] then appears to be
 misdelivered / lost - Kamailio receives it on the private address and
 forwards it to the public address rather than the FS server.

 My configuration [5] is built based on the default configuration + examples
 from the dispatcher module. There are some provisions for FreeSWITCH
 internal/external profiles made so the configuration listens on 2 different
 ports. This particular scenario should only use the 5061 port as it involves
 calls to an external sip provider (briefly 5060 should be proxied to FS:5080
 and 5061 should be proxied to FS:5070 and vice-versa).


 If someone can see an issue with the configuration and/or point to an error
 in the call flow (i.e. should FS be inserting the Via header?) that would be
 greatly appreciated.


 Thanks,

 -nick

 [1] Outbound call flow (see also outbound-callflow.png)
 FS(port 5070) sends an invite to Kamailio(5061) (Kamailio is configured in
 FS as an outbound proxy), INVITE contains Via header for the FS server,
 Kamailio forwards to sip provider, and routes all messages back and forth
 correctly.

 [2] 100 Trying
 SIP/2.0 100 trying -- your call is important to us
 Via: SIP/2.0/UDP
 sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.0;rport=5060
 Via: SIP/2.0/UDP
 far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK
 From: Nicholas Gill sip:0384171...@far.external.ip;tag=j3KQmpvmg6mvr
 To: sip:0390156...@sip.provider.com
 Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9
 CSeq: 66413636 INVITE
 Server: kamailio (4.1.6 (x86_64/linux))
 Content-Length: 0

 [3] Proxied 200 OK (Kamailio - Sip Provider)
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP
 sip.provider.com:5060;rport=5060;branch=z9hG4bKfffb.7c81ee53.0
 Via: SIP/2.0/UDP
 far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK
 Record-Route: sip:kamailio.ext:5061;lr=on
 Record-Route: sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr;did=0ec.84ff82b1
 From: Nicholas Gill sip:0384171...@far.external.ip;tag=j3KQmpvmg6mvr
 To: sip:0390156...@sip.provider.com;tag=BFvQmggHrg74m
 Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9
 CSeq: 66413636 INVITE
 Contact: sip:0390156...@kamailio.int:5070;transport=udp
 User-Agent: IMX
 Accept: application/sdp
 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
 REFER, NOTIFY
 Supported: timer, path, replaces
 Allow-Events: talk, hold, conference, refer
 Content-Type: application/sdp
 Content-Disposition: session
 Content-Length: 270
 X-FS-Support: update_display,send_info

 [4] Incoming ACK (Sip Provider - Kamailio)
 ACK sip:0390156...@kamailio.ext:5061;transport=udp SIP/2.0
 Record-Route: sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr
 Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.2
 Via: SIP/2.0/UDP
 far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK509ZQBKB947aF
 Route: sip:kamailio.ext:5061;lr=on
 Max-Forwards: 69
 From: Nicholas Gill sip

Re: [SR-Users] Fwd: presence module issue with NOTIFY messsage.

[Ovidiu Sas]

Should we deprecate add_contact_alias()?
Add a warning message.

-ovidiu

On Tue, Oct 14, 2014 at 9:44 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Hello,

 short update on this topic - the master and 4.2 branches include a patch
 that should make presence work with set_contact_alias().

 It would be safer to use set_contact_alias(), especially if you have UA that
 checks R-URI for incoming requests.

 Cheers,
 Daniel


 On 13/10/14 04:42, Thanh Truong wrote:

 Hi All,

 I have added : fix_natted_contact() and it fix this issue.

 Thank all for help.


 Thanks,
 ThanhTruong.


 On Fri, Oct 10, 2014 at 8:39 PM, Ovidiu Sas o...@voipembedded.com wrote:

 I would prefer to have add_contact_alias deprecated and then dropped (less
 confusing this way).

 -ovidiu

 On Fri, Oct 10, 2014 at 9:31 AM, Daniel-Constantin Mierla
 mico...@gmail.com wrote:


 On 10/10/14 14:54, Ovidiu Sas wrote:

 I don't think this is documented anywhere and it's challenging for
 someone not familiar with kamailio to deal with it.
 Maybe we should open a bug report to make presence *_contact_alias
 friendly.

 add_contact_alias() is not going to be (very easy) friendly, by the way
 was coded -- probably targeting only the requests that are proxied.

 It was the reason I added set_contact_alias() for fixing that, but I was
 not sure how it will cope over all at the end. So the first follow up fix
 was done inside dialog module (because I could test it at that moment), I
 will push it now directly in tm to catch the other modules sending requests
 within dialog.

 Eventually add_contact_alias() will be removed or 'aliased' to
 set_contact_alias().

 Daniel


 -ovidiu

 On Fri, Oct 10, 2014 at 5:16 AM, Daniel-Constantin Mierla
 mico...@gmail.com wrote:

 Hello,

 if the presence server is first hop after a nat router, then use
 fix_natted_contact() for SUBSCRIBE requests instead of
 add/set_contact_alias().

 Cheers,
 Daniel

 On 05/10/14 06:52, Thanh Truong wrote:

 Hi all,

 I have install kamailio 4.2 latest version with presence module.

 But I cant get contact status and send message.

 []

 I see that SUBSCRIBE message send to wrong ip, it is sent to local IP
 and my sip phone do not receive it. But i do not know how to fix it.

 please suggest to get it.


 --
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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


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Re: [SR-Users] Fwd: presence module issue with NOTIFY messsage.

[Ovidiu Sas]

I don't think this is documented anywhere and it's challenging for someone
not familiar with kamailio to deal with it.
Maybe we should open a bug report to make presence *_contact_alias friendly.

-ovidiu

On Fri, Oct 10, 2014 at 5:16 AM, Daniel-Constantin Mierla mico...@gmail.com
 wrote:

  Hello,

 if the presence server is first hop after a nat router, then use
 fix_natted_contact() for SUBSCRIBE requests instead of
 add/set_contact_alias().

 Cheers,
 Daniel

 On 05/10/14 06:52, Thanh Truong wrote:

  Hi all,

  I have install kamailio 4.2 latest version with presence module.

  But I cant get contact status and send message.

 []

   I see that SUBSCRIBE message send to wrong ip, it is sent to local IP
 and my sip phone do not receive it. But i do not know how to fix it.

  please suggest to get it.


 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda


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Re: [SR-Users] Fwd: presence module issue with NOTIFY messsage.

[Ovidiu Sas]

I would prefer to have add_contact_alias deprecated and then dropped (less
confusing this way).

-ovidiu

On Fri, Oct 10, 2014 at 9:31 AM, Daniel-Constantin Mierla mico...@gmail.com
 wrote:


 On 10/10/14 14:54, Ovidiu Sas wrote:

 I don't think this is documented anywhere and it's challenging for someone
 not familiar with kamailio to deal with it.
 Maybe we should open a bug report to make presence *_contact_alias
 friendly.

 add_contact_alias() is not going to be (very easy) friendly, by the way
 was coded -- probably targeting only the requests that are proxied.

 It was the reason I added set_contact_alias() for fixing that, but I was
 not sure how it will cope over all at the end. So the first follow up fix
 was done inside dialog module (because I could test it at that moment), I
 will push it now directly in tm to catch the other modules sending requests
 within dialog.

 Eventually add_contact_alias() will be removed or 'aliased' to
 set_contact_alias().

 Daniel


  -ovidiu

 On Fri, Oct 10, 2014 at 5:16 AM, Daniel-Constantin Mierla 
 mico...@gmail.com wrote:

  Hello,

 if the presence server is first hop after a nat router, then use
 fix_natted_contact() for SUBSCRIBE requests instead of
 add/set_contact_alias().

 Cheers,
 Daniel

 On 05/10/14 06:52, Thanh Truong wrote:

  Hi all,

  I have install kamailio 4.2 latest version with presence module.

  But I cant get contact status and send message.

  []

   I see that SUBSCRIBE message send to wrong ip, it is sent to local IP
 and my sip phone do not receive it. But i do not know how to fix it.

  please suggest to get it.


 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda


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 --
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Re: [SR-Users] Adding Dynamic Routing Module management to Siremis

[Ovidiu Sas]

As an alternative, you can try to use the kamailio embedded provisioning module:
http://kamailio.org/docs/modules/devel/modules/xhttp_pi

Regards,
Ovidiu Sas

On Fri, Oct 3, 2014 at 9:08 PM, cpcnetworking cpcnetwork...@gmail.com wrote:
 Anybody have some insight or pointers? Thx

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Re: [SR-Users] Looking for some help

[Ovidiu Sas]

Take a look here:
http://www.kamailio.org/w/business/

Regards,
Ovidiu Sas

On Thu, Sep 4, 2014 at 5:24 PM, Sharan Harkisoon sha...@sharktek.net wrote:
 I am in need of a Kamailio expert that has some availability for consulting
 services (remote is fine).  Feel free to contact me for details.

 Thanks,
 Sharan Harkisoon

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Re: [SR-Users] UAC remote registration - refreshing database

[Ovidiu Sas]

Porting the registrant module is not straight forward.
Best solution here - as Daniel pointed out - would be to enhance the
existing implementation.

-ovidiu

On Thu, Jul 3, 2014 at 6:32 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Hello,

 not having against adding alternatives to existing features/modules,
 apparently here is just about implementing the reload capability. Most of
 the features should be there, like loading from database (which is done at
 startup) and destroying exiting structures in memory (which is done at
 shutdown). Doing the second followed by the first operation upon a rpc
 command should get the feature (of course, there can be some extra
 bits/adjustments needed)

 Cheers,
 Daniel


 On 03/07/14 12:04, Dan Christian Bogos wrote:

 Hey Alex,

 Many thanks for so fast answer. Could be I have missed the past interest
 (hence re-posting).

 Anyway, I wonder what would be the chances that we get Ovidiu's interest
 so he can port registrant module to kamailio maybe? Unfortunately without
 reloads it is hard to push remote registration in enterprise solutions.

 Thanks again,
 DanB

 On 03.07.2014 12:00, sr-users-requ...@lists.sip-router.org wrote:

 On 07/02/2014 06:22 AM, Dan Christian Bogos wrote:

 Anybody aware if it is possible to refresh the list of remote
 registrations from the database without restarting the whole server?

 This gets asked a lot, and the answer is no. But it is probably a widely
 desired feature set by now.

 -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be
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Re: [SR-Users] Setting priority (q) for kamctl ul add

[Ovidiu Sas]

kamctl is a basic script to get users familiarized with kamailio tables.
You will need to use SQL command to configure specific fields.
Alternatively you can install siremis or enable the xhttp_pi module.

Regards,
Ovidiu Sas
On Jun 20, 2014 12:16 AM, David Wilson d...@zaq.com.au wrote:

 Hi All,

 I'm trying to add a permanent usrloc entry via kamctl ul add.

 This works, but the created entry has a q value of 1.0 which is higher
 than I need.

 Is there a way to either:

 1.  Specify a q value when using kamctl ul add, or
 2.  Edit the q value of an existing record by using a kamctl command.

 Cheers,
 Dave.


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Re: [SR-Users] PERMISSIONS module issue

[Ovidiu Sas]

Use the dialplan module for dealing with DIDs.

Regards,
Ovidiu Sas

On Tue, Apr 15, 2014 at 11:42 AM, PIERRE Laurent ltpie...@gmail.com wrote:
 Hi,

 Yes we're used to configuring allow/deny files because we need to manage a
 numbering plans ( authorize DID numbers, short numbers, international
 numbers.etc) with regular expressions.
 The allow_address refers to ip addresses only . So it's not useful

 Do you think we'll be able to have reload support on the next version ?

 Thanks

 Laurent PIERRE
 http://www.linkedin.com/in/lpierre

 ---

 On 14 April 2014 18:06, Daniel-Constantin Mierla mico...@gmail.com wrote:

 Hello,


 On 14/04/14 17:00, PIERRE Laurent wrote:

 Hi,

 Indeed, when we add a telephone number in the authorize file
 (default_allow_file)  using permission module then we have to restart
 Kamailio. It's too bad for life system :(

 if you use allow/deny files, then you have to use allow_routing() or
 allow_register functions. I guess there is no reload support for them.

 If you use address database table, then you have to use allow_address() or
 allow_source_address(). There is support to reload records from db table
 without restart, via mi or rpc command.

 Those using 4.0.x, have to upgrade to the latest fix release (4.0.6) -
 there was a fix on reloading command (git
 7aba649db775a00e28dc75a9145a3da50f797776).

 Cheers,
 Daniel


 Is it planned to improve in the next version ?

 Thanks

 --
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 http://www.linkedin.com/in/lpierre

 --

 On 8 November 2013 09:38, Daniel Tryba dan...@pocos.nl wrote:

 On Thursday 07 November 2013 19:53:47 Samuel Ware wrote:
  I having issue updating my allow list for the PERMISSIONS module.  I
  added
  an address to the ADDRESS table.  I have tried to do a service restart,
  kamctl address reload, and kamcmd permissions.addressReload.  The
  kamctl
  address show displays the new address; however kamcmd
  permissions.addressDump does not neither does kamcmd
  permissions.subnetDump.  The messages from this new address return a
  false
  to the !allow_source_address(1”) command in my routing logic.  I am
  wondering if this is a bug or I am doing something wrong.  I am on the
  most recent GIT version to the best of my knowledge.

 I noticed the same yesterday with 4.0.3, had to restart to get the adress
 added.

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Re: [SR-Users] Web interface for kamailio

[Ovidiu Sas]

If you want to gather/visualize statistics, you can use the xhttp_rpc
module:
http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html
It's a simple built in web interface for all rpc commands.

Regards,
Ovidiu Sas


On Tue, Mar 11, 2014 at 11:02 AM, malik sherif asheri...@hotmail.comwrote:

 Any recommendation web interface for gathering statistics other than
 SIREMIS? I installed SIREMIS but unable to resolve login problem and my
 request for help on ASIPTO didn't get any response also unable to find
 siremis mailing list. Your help is greatly appreciated.
 Thanks
 Abdulmailk

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Re: [SR-Users] Debian init script log_faiulre_msg

[Ovidiu Sas]

Pushed!
Thanks for the report.

Regards,
Ovidiu Sas

On Fri, Mar 7, 2014 at 10:42 AM, Corey Edwards ten...@zmonkey.org wrote:
 There's a typo in the Debian init script. Is this the correct place to
 report packaging bugs?

 --- /etc/init.d/kamailio2014-03-06 13:42:23.0 -0700
 +++ /tmp/kamailio2014-03-07 08:41:05.0 -0700
 @@ -52,7 +52,7 @@
  log_failure_msg Not starting $DESC: invalid configuration file!
  log_failure_msg
  log_failure_msg $out
 -log_faiulre_msg
 +log_failure_msg
  exit 1
  fi
  }


 Corey


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Re: [SR-Users] Incorrect Contact address

[Ovidiu Sas]

No, you are not dealing with a buggy client.
As Carlos pointed out, you are dealing with standard SIP loose routing
mechanism.
Time to read the RFC :)

Regards,
Ovidiu Sas

On Wed, Mar 5, 2014 at 2:24 PM, Marc Soda ms...@coredial.com wrote:
 Yeah, I think I'm dealing with a buggy client...

 Thanks all.


 On Wed, Mar 5, 2014 at 1:57 PM, Carlos Ruiz Díaz carlos.ruizd...@gmail.com
 wrote:

 Why is erroneous to have the contact header with the backend IP?

 With the record-route on the 200 Ok, the ACK should be directed to the
 backend IP, but containing a route header pointing to the Kamailio IP.
 Kamailio will loose_route() this request and send it to the backend server
 as expected.

 Regards,


 On Wed, Mar 5, 2014 at 3:53 PM, Marc Soda ms...@coredial.com wrote:

 Thanks Olle.  I am calling record_record() on the initial INVITE.  In
 fact, the OK has a Record-Route header:

 1.1.1.1 is the endpoint
 2.2.2.2 is the kamailio proxy
 3.3.3.3 is the backend server

 SIP/2.0 200 OK
 Via: SIP/2.0/UDP
 1.1.1.1:60077;rport=46110;branch=z9hG4bK-d8754z-eb9768c7e3a2d1e7-1---d8754z-
 Record-Route: sip:2.2.2.2;lr=on;ftag=db634167;nat=yes
 From: sip:sip7878_s...@edge.domain.com;transport=UDP;tag=db634167
 To: sip:215...@edge.coredial.com;transport=UDP;tag=as3f9cf263
 Call-ID: MzI5YTA3YmRkNzFiZjhhZTRkNTc2OGE1ZTc5ZjdjMmM.
 CSeq: 2 INVITE
 User-Agent: CoreDialPBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces
 Contact: sip:215@3.3.3.3
 Content-Type: application/sdp
 Content-Length: 266

 v=0
 o=root 13486 13487 IN IP4 3.3.3.3
 s=session
 c=IN IP4 3.3.3.3
 t=0 0
 m=audio 29990 RTP/AVP 0 8 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv

 I don't think 3.3.3.3 should show up anywhere, it should be rewritten to
 2.2.2.2.


 On Wed, Mar 5, 2014 at 1:34 PM, Olle E. Johansson o...@edvina.net wrote:


 On 05 Mar 2014, at 18:30, Marc Soda ms...@coredial.com wrote:

 I have Kamailio setup as a proxy in front of a backend server
 (Asterisk).  When I make a call through the proxy, the Contact header in 
 the
 200 OK that is returned to the client has the IP of the backend server in
 it.  Thus, the client is sending it's ACK directly to the backend server.

 Is there a special method to rewrite the Contact header to be Kamailio's
 IP?

 Check record_route() in the default configuration script. You need to
 add a route set by using record_route() in the initial transaction.

 Cheers,
 /O


 Where is a good place in the config to do this?  (my config is loosely
 based on this:
 http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb)

 Thanks!
 Marc

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 +595981146623

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 Office: (215) 297-4400 x203 | Fax: (215) 297-4401 | Email:
 ms...@coredial.com

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Re: [SR-Users] HTABLE update from DB

[Ovidiu Sas]

Declare the table without autoexpire. Check again the module README.

Regards,
Ovidiu Sas
On Feb 23, 2014 11:23 AM, Uri Shacked ushac...@gmail.com wrote:

 Hi,

 Following my issue with reloading data. I am thinking of a way to keep
 data updated in memory.
 I read the HTABLE module again and notice the db_expires option.
 It does not work for me...

 I try to set it so an item will expire in 60 sec, and when expired, it
 will be reloaded from DB and not deleted.

 Any ideas?

 Uri

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Re: [SR-Users] HTABLE update from DB

[Ovidiu Sas]

It is working for sure, because I'm using it (reloading from db to refresh
memory cache).

Regards,
Ovidiu Sas
On Feb 23, 2014 11:56 AM, Uri Shacked ushac...@gmail.com wrote:

 It does not work...I tried with autoexpire=0, I tried with no autoexpires 
 define, I Tried defining a number and see if the value is updated


 this is my table definition:

 modparam(htable, htable, A=size=8,dbtable=aa;initial=0,dbmobe=0;)


 I remind you i need the memory to be updated from the DB every interval, and 
 not the DB form the memory.


 thanks.



 Declare the table without autoexpire. Check again the module README.

 Regards,
 Ovidiu Sas
 On Feb 23, 2014 11:23 AM, Uri Shacked ushacked at gmail.com 
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users wrote:

 * Hi,
 ** Following my issue with reloading data. I am thinking of a way to keep
 ** data updated in memory.
 ** I read the HTABLE module again and notice the db_expires option.
 ** It does not work for me...
 ** I try to set it so an item will expire in 60 sec, and when expired, it
 ** will be reloaded from DB and not deleted.
 ** Any ideas?
 ** Uri*


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Re: [SR-Users] Replacing an ACME Packet Net-Net SBC

[Ovidiu Sas]

You don't need B2B functionality to provide SBC functionality.
SBC is a loose term these days. All you need is properly crafted signaling
to achieve you requirements.
Using a SIP proxy server as an SBC is pretty common these days.

Regards
Ovidiu Sas
On Feb 20, 2014 5:55 PM, Francesco Maria Magnini fmm1...@gmail.com
wrote:

 @Carsten
 I looked at http://www.iptel.org/sems and seems to be only broken links
 to downloads.
 Do you know if the project is still maintained?

 @Fred
 Are you using openser as a B2BUA?

 Il giorno 20/feb/2014, alle ore 19:42, Fred Posner f...@palner.com ha
 scritto:

  Alex's article is one of my favorites. That being said, we switched out
 an Acme SBC for openser (at the time) and was immediately thrilled.
 
  Fred Posner
  The Palner Group, Inc.
  503-914-0999 (direct)
  954-472-2896 (fax)
 
  On 02/20/2014 01:14 PM, Alex Balashov wrote:
  Francesco,
 
  Have a look at this blog post:
 
 
 http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/
 
 
  That said, I agree with Carsten's suggestion of SEMS.
 
  On 02/20/2014 11:04 AM, Francesco Maria Magnini wrote:
 
  Hi,
 
  I would like to have some suggestions about a full replacement of an
  ACME Packet Net-Net Session Border Controller.
  By now, ACME SBC performs all the SBC functionalities, mainly:
 
  - it is used as a SIP endpoint for SIP client registrations
  - it is used as a SIP endpoint for interconnection to multiple SIP
  carriers via SIP trunks
  - it is used for NAT traversal
 
  In this deployment, the SIP Server communicates only with the SBC and
  this one takes care of the communication between the SIP Server and
  the external SIP entities (UA clients, SIP Trunks).
  In this scenario, can I consider to replace the SBC with Kamailio?
 
 
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Re: [SR-Users] Performance impact with AVP and VAR's

[Ovidiu Sas]

or a dedicated module :)
On Feb 20, 2014 5:38 PM, Alex Balashov abalas...@evaristesys.com wrote:

 Can you give some example of your use cases for them?

 I cannot say for sure, but my intuition is that if you have three hundred
 variables in any program, you're doing something wrong. At that point
 you're in territory that clearly calls for some sort of non- scalar data
 structure, such as an associative array.


 Jijo realj...@gmail.com wrote:

 Hi All,

 I have around 300 AVP's and quite amount of VAR's are used in the config
 file?  Does that impact performance?, If so how can i improve it?

 Thanks
 Jijo


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 --
 Sent from my mobile, and thus lacking in the refinement one might expect
 from a fully fledged keyboard.

 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0671
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Re: [SR-Users] Performance impact with AVP and VAR's

[Ovidiu Sas]

AVPs are in shared men and protected by locks. VARs are not.
There shouldn't be a big impact on using lots of them. Are you experiencing
any issues?

Regards
Ovidiu Sas
On Feb 20, 2014 7:32 PM, Jijo realj...@gmail.com wrote:

 We have defined dedicated AVP variables for each feature. For example, SIP
 Trunks or Subscribers or Media Handling or Header Manipulation etc, So the
 no of variables (AVP)  has been increased in the initialization.
 At an instance the no of AVP's used/active might be quite low as each
 avp's are dedicated for the feature.

 Does avp read or write cause any Lock?

 The VAR's has been used locally through out the route for header
 manipulation and other functions.





 On Thu, Feb 20, 2014 at 5:37 PM, Alex Balashov 
 abalas...@evaristesys.comwrote:

 Can you give some example of your use cases for them?

 I cannot say for sure, but my intuition is that if you have three hundred
 variables in any program, you're doing something wrong. At that point
 you're in territory that clearly calls for some sort of non- scalar data
 structure, such as an associative array.


 Jijo realj...@gmail.com wrote:

  Hi All,

 I have around 300 AVP's and quite amount of VAR's are used in the config
 file?  Does that impact performance?, If so how can i improve it?

 Thanks
 Jijo


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 --
 Sent from my mobile, and thus lacking in the refinement one might expect
 from a fully fledged keyboard.

 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0671
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

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Re: [SR-Users] Dialog

[Ovidiu Sas]

There are many modules that will let you store data:
http://kamailio.org/docs/modules/devel/modules/memcached.html
http://kamailio.org/docs/modules/devel/modules/htable.html
...

Or you can use shared memory variables:
http://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#shv_name_-_shared_memory_variables

Store anything that you need upon receiving a 18x provisional reply
and delete the data when a final reply to an initial INVITE is
received.
In between, you can access/retrieve data while handling a new INVITE
that you want to modify/alter.

Regards,
Ovidiu Sas


On Sun, Feb 9, 2014 at 3:31 AM, John Murray
john.mur...@skyracktelecom.com wrote:
 Hi Ovidiu,

 BLF requires support from the handset / UA. I have analogue handsets.

 How would the cache work you described?

 Regards

 John

 On 9 Feb 2014 02:27, Ovidiu Sas o...@voipembedded.com wrote:

 This is the typical BLF case. Why don't you use BLF/presence to do it?
 If you really want to do it from the script you can save all the required
 info in a cache and retrieve it from there, instead of messing with the
 dialog module.

 Regards,
 Ovidiu Sas

 On Feb 7, 2014 7:37 AM, John Murray john.mur...@skyracktelecom.com
 wrote:

 Daniel,



 On an incoming call I need to get the call-id from a ringing call in a
 specific dialog profile. I then add that called to a replaces header and
 send to a B2BUA which connects the current call to the ringing one.



 Thanks



 John



 From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
 Sent: 07 February 2014 12:32
 To: John Murray; 'Kamailio (SER) - Users Mailing List'
 Subject: Re: [SR-Users] Dialog



 Hello,

 in config you have access to SIP message that is currently processed and
 there you can simply use $ci. Or is there a special event route where you
 need the call-id?

 Cheers,
 Daniel

 On 07/02/14 13:29, John Murray wrote:

 Hi Daniel,



 Yes I need it in the kamailio config.



 Thanks



 John



 From: sr-users-boun...@lists.sip-router.org
 [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
 Daniel-Constantin Mierla
 Sent: 07 February 2014 12:15
 To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] Dialog



 Hello,

 do you need it in kamailio config or from an external application? From
 external application the mi/rpc command has to be used.

 Cheers,
 Daniel

 On 06/02/14 21:45, John Murray wrote:

 Hi,



 I need to get call-id and from-tag from a call in ringing state (2).

 If I use dlg_manage() and put the call into a profile using
 set_dlg_profile(usr,1).

 Then do a avp_db_query() I don't see the call until it is connected.

 I am using db_mode 1.



 Yet if I do a sercmd dlg.dlg_list I see the call in ringing state and
 when it is connected.



 How do I get the call-id when it is in ringing state?



 Regards



 John





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Re: [SR-Users] Dialog

[Ovidiu Sas]

This is the typical BLF case. Why don't you use BLF/presence to do it?
If you really want to do it from the script you can save all the required
info in a cache and retrieve it from there, instead of messing with the
dialog module.

Regards,
Ovidiu Sas
On Feb 7, 2014 7:37 AM, John Murray john.mur...@skyracktelecom.com
wrote:

 Daniel,



 On an incoming call I need to get the call-id from a ringing call in a
 specific dialog profile. I then add that called to a replaces header and
 send to a B2BUA which connects the current call to the ringing one.



 Thanks



 John



 *From:* Daniel-Constantin Mierla [mailto:mico...@gmail.com]
 *Sent:* 07 February 2014 12:32
 *To:* John Murray; 'Kamailio (SER) - Users Mailing List'
 *Subject:* Re: [SR-Users] Dialog



 Hello,

 in config you have access to SIP message that is currently processed and
 there you can simply use $ci. Or is there a special event route where you
 need the call-id?

 Cheers,
 Daniel

 On 07/02/14 13:29, John Murray wrote:

 Hi Daniel,



 Yes I need it in the kamailio config.



 Thanks



 John



 *From:* sr-users-boun...@lists.sip-router.org [
 mailto:sr-users-boun...@lists.sip-router.orgsr-users-boun...@lists.sip-router.org]
 *On Behalf Of *Daniel-Constantin Mierla
 *Sent:* 07 February 2014 12:15
 *To:* Kamailio (SER) - Users Mailing List
 *Subject:* Re: [SR-Users] Dialog



 Hello,

 do you need it in kamailio config or from an external application? From
 external application the mi/rpc command has to be used.

 Cheers,
 Daniel

 On 06/02/14 21:45, John Murray wrote:

 Hi,



 I need to get call-id and from-tag from a call in ringing state (2).

 If I use dlg_manage() and put the call into a profile using
 set_dlg_profile(usr,1).

 Then do a avp_db_query() I don't see the call until it is connected.

 I am using db_mode 1.



 Yet if I do a sercmd dlg.dlg_list I see the call in ringing state and when
 it is connected.



 How do I get the call-id when it is in ringing state?



 Regards



 John





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 --

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Re: [SR-Users] Asterisk re-INVITE race condition, error 500.

[Ovidiu Sas]

Maybe adding a Retry-After header to the 500 might help.

Regards,
Ovidiu Sas

On Mon, Jun 3, 2013 at 8:20 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
 This is a known problem without simple solution. It can also happen if the
 ACK gets lost somewhere.

 The proper solution is to fix Freeswitch. The reINVITE is an implicit ACK
 (as there wouldn't be a reINVITE if there wouldn't have been an ACK). Thus,
 FS should accept the reINVITE and implicitly behaves like the ACK was
 received. Later, when the ACK arrives, it should be ignored by FS.

 Actually Asterisk should also handle the 500 correctly and try the reINVITE
 again (or works although the reINVITE failed).

 regards
 Klaus


 On 03.06.2013 21:23, David K wrote:

 Hello all,

 So I have three machines, we don't care about audio for this problem, so
 everything I mention here is SIP related.

 Freeswitch -- Kamailio 3.3.2 -- Asterisk

 1. Asterisk sends an INVITE to Freeswitch through the Kamailio proxy.
 2. Kamailio replies 100 Trying and forwards to Freeswitch
 3. Freeswitch replies 100 Trying
 4. Freeswitch replies 180 Ringing to Kamailio
 5. Kamailio routes the answer to Asterisk
 6. Freeswitch replies 200 OK to Kamailio
 7. Kamailio replies 200 OK to Asterisk
 8. Asterisk replies ACK to Kamailio
 9. Asterisk sends a re-INVITE to Freeswitch through Kamailio
 10. Kamailio routes the re-INVITE to freeswitch
 11. Kamailio routes the ACK to freeswitch.
 12. Freeswitch replies 500 Server error because it got a re-INVITE
 before the ACK.

 So, my problem is that Kamailio seems to process my re-INVITE more
 quickly than the ACK. So Freeswitch replies an error because it got the
 re-INVITE before the ACK.

 So my solution is to add a usleep(20); for re-INVITEs on Kamailio, but
 I think this is a lousy solution.

 Has anyone here had to deal with problems where Kamailio routes a
 re-INVITE faster than an ACK causing endpoints to return error
 messages?  Has anyone had to deal with a similar issue?

 Thanks,

 David



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Re: [SR-Users] Kamailio behind NAT

[Ovidiu Sas]

Yes, it will.

Regards,
Ovidiu Sas

On Tue, Jan 21, 2014 at 9:03 AM, John Smith jsmith...@mail.com wrote:
 Using advertised IP address in manage_rttproxy would work with unpatched 
 rtpproxy?

 Thank you

 - Original Message -
 From: Klaus Darilion
 Sent: 01/21/14 05:18 AM
 To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] Kamailio behind NAT

 On 21.01.2014 12:27, Fred Posner wrote:
  With a patched version of rtpproxy you can advertise your private ip.
 
  http://www.fredposner.com/voip/1457/kamailio-behind-nat/

 Aha, nice. Haven't known of this one.

 I always specified the adverstised IP address when calling
 manage_rtpproxy(). That should work too.

 regards
 Klaus

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Re: [SR-Users] ingress timer

[Ovidiu Sas]

Take a look at the following module parameters:
http://kamailio.org/docs/modules/4.1.x/modules/tm.html#fr_timer_avp
http://kamailio.org/docs/modules/4.1.x/modules/tm.html#fr_inv_timer_avp

Regards,
Ovidiu Sas
On Jan 2, 2014 12:03 AM, Kelvin Chua kel...@gmail.com wrote:

 i would like to have an fr_inv_timer functionality on inbound INVITEs.

 the only way i imagine this to work is to use timer module, set a
 predefined timer value, and when reaching that value after the INVITE,
 execute a ROUTE that
 cancels the callee and sends a 500 or timeout message back to the caller.

 problem is, the timeout value for module is static. i need the timeout to
 be dynamic
 for different callers.

 are there any other methods on achieving this?


 Kelvin Chua

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Re: [SR-Users] [sr-dev] Kamailio v4.1.0 Released - new major version is out

[Ovidiu Sas]

For those who like to run kamailio on small routers or other embedded
systems, the optware feeds are updated:
http://www.nslu2-linux.org/wiki/Optware/HomePage

Regards,
Ovidiu Sas

On Wed, Dec 4, 2013 at 10:10 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Kamailio v4.1.0 is out – a new major release, bringing out as usually a very
 large set of new features and improvements.

 You can read detailed release notes at:

 * http://www.kamailio.org/w/kamailio-v4-1-0-release-notes/

 Many thanks to all developers and community members that made possible this
 release.

 Enjoy Kamailio v4.1.0 and have a great time during winter holidays!

 Daniel

 --
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Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Ovidiu Sas]

Try to attach gdb to the kamailio processes and run a full backtrace.

Regards,
Ovidiu Sas

On Wed, Nov 27, 2013 at 5:16 AM, Sotas Development sotas...@gmail.com wrote:
 Hi,

 In the mean time we have gathered more information on this problem:

 As given below, kamailio stops grabbing UDP SIP messages (SIP registrations)
 after running a while on an embedded ARM and PPC platform (which runs linux
 2.6.33 kernel). Some times the hangup occures within hours and some times
 after couple of days running.

 NETSTAT OUTPUT:
 root# netstat -pl | grep kam
 udp   1047968  0 (null):sip  (null):*
 8416/kamailio
 raw0  0 (null):255  (null):*255
 8416/kamailio
 unix  2  [ ACC ] STREAM LISTENING 755205 8429/kamailio
 /tmp/kamailio_ctl

 Kamailio is started with the following options = -m 4 -n 3 -f cfg -D

 Other relevant info:
 - When Kamailio hangs, I also noticed that the flag inuse_transactions has
 always the value of '1'. Readout with kamctl monitor.
 - A simple cat to /proc/kamailio_pid/wchan gives us the function:
 futex_wait_queue_me.
 - All possible polling methods are used with -W parameter (sigio_rt, poll,
 select etc) during these tests. Non of these options did solve this problem.

 I hope the additional info will clarify more. Thanks in advance.

 Best regards,
 Orhan Yilmaz



 On Wed, Nov 13, 2013 at 6:12 PM, Ovidiu Sas o...@voipembedded.com wrote:

 In a previous e-mail, you posted a warning that you had while compiling:
 no native memory barrier implementations, falling back to slow lock
 based workarround
 which means that you are already running without atomic locks.

 Regards,
 Ovidiu Sas

 On Wed, Nov 13, 2013 at 10:40 AM, Sotas Development sotas...@gmail.com
 wrote:
  Hi,
 
  Here's an update of this topic. We've tried again with the latest stable
  version 4.0.4. Unfortunately the problem still exists.
 
  In mails above it is mentioned to use kamailio without atomic locks. How
  do
  we this (e.g. which makefile options)?
 
  Kind regards,
 
  Bert
  (on behalf of Michiel Veldkamp)
 
 
 
  On Mon, Jan 28, 2013 at 4:44 PM, Ovidiu Sas o...@voipembedded.com
  wrote:
 
  4.0 (current trunk) is in code freeze.  I would suggest to test the
  trunk version (next 4.0).
  Even openser 1.3 requires patches to be properly cross compiled.
 
  Regards,
  Ovidiu Sas
 
  --
  VoIP Embedded, Inc.
  http://www.voipembedded.com
 
  -- Forwarded message --
  From: Sotas Development sotas...@gmail.com
  Date: Mon, Jan 28, 2013 at 10:08 AM
  Subject: Re: [SR-Users] Kamailio stability/timing problem w.r.t.
  registrations?
  To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
  Users Mailing List sr-users@lists.sip-router.org
 
 
  Hi Ovidiu,
 
  Thanks for the warning! We did not yet have much success running the
  current master branch, though this may well be a resource problem on
  the target platform.
 
  For the moment, we decided to switch back to openser 1.3.5 and wait
  for the official 4.0 release.
 
  Regards,
  Michiel Veldkamp
 
 
  On Thu, Jan 17, 2013 at 7:01 PM, Ovidiu Sas o...@voipembedded.com
  wrote:
  
   If you are running the stable version, there's need for heavy
   Makefile
   patching in order to properly cross compile (not to include and link
   to host libs).
   The trunk has everything fixed and it's cross-compiling properly for
   most of the modules.
   Make sure that your binaries are properly cross compiled.
  
   Depending on your ARM CPU, atomic locks may or may not work.
   I tested openser without atomic locks (using regular locks) and it
   worked fine.
  
   Regards,
   Ovidiu Sas
  
  
   --
   VoIP Embedded, Inc.
   http://www.voipembedded.com
 
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Re: [SR-Users] Books

[Ovidiu Sas]

Your first read should be: http://tools.ietf.org/html/rfc3261
After that, dealing with kamailio will be much easier :)

Regards,
Ovidiu Sas

On Wed, Nov 27, 2013 at 11:33 AM, Joli Martinez mrjoli...@gmail.com wrote:
 Hello,

 Is there any books you would recommend reading so I can learn more on 
 Kamailio?

 thanks,


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Re: [SR-Users] Log files

[Ovidiu Sas]

Most likely it's a bogus script.
Sometimes just sending a dummy reply, will stop the script sending SIP requests.
Check the User-Agent header and from username to see if you can
identify the script and google around for it.

Regards,
Ovidiu Sas

On Tue, Nov 26, 2013 at 4:17 PM, Joli Martinez mrjoli...@gmail.com wrote:
 I am running Kamailio in CentOS.  I ran tcpdump and noticed that we are 
 getting attacked from IP 188.138.32.72.  I have already blocked it on 
 IPtables, but he keeps on attacking the server.  If I look at 
 /var/log/secure there are no SIP messages.  My question is where is the log 
 file for Kamailio and how can I prevent this type of attacks in the future.

 Thanks,
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Re: [SR-Users] Log files

[Ovidiu Sas]

Google around for friendly-scanner to learn more about it.
In the mean time, allow the packets to be handled by kamailio and send
a 200ok back - maybe this will stop the attack.
After the attack is stopped, simply drop all friendly-scanner SIP requests :)

Regards,
Ovidiu Sas

On Tue, Nov 26, 2013 at 4:32 PM, Joli Martinez mrjoli...@gmail.com wrote:
 it is comming from friendly-scanner The other issue I have is that 
 /var/log/secure is not getting the sip requests so the only way I realize 
 it is happeing is from tcpdump.  If the secure file is not picking it up then 
 iptables wont know about it.  How can I tell iptables to listen for sip 
 requests?  I have already added the IP to the blocked IP's but he still keeps 
 on comming.

 Thanks,

 On Nov 26, 2013, at 4:28 PM, Ovidiu Sas o...@voipembedded.com wrote:

 Most likely it's a bogus script.
 Sometimes just sending a dummy reply, will stop the script sending SIP 
 requests.
 Check the User-Agent header and from username to see if you can
 identify the script and google around for it.

 Regards,
 Ovidiu Sas

 On Tue, Nov 26, 2013 at 4:17 PM, Joli Martinez mrjoli...@gmail.com wrote:
 I am running Kamailio in CentOS.  I ran tcpdump and noticed that we are 
 getting attacked from IP 188.138.32.72.  I have already blocked it on 
 IPtables, but he keeps on attacking the server.  If I look at 
 /var/log/secure there are no SIP messages.  My question is where is the 
 log file for Kamailio and how can I prevent this type of attacks in the 
 future.

 Thanks,
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Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Ovidiu Sas]

In a previous e-mail, you posted a warning that you had while compiling:
no native memory barrier implementations, falling back to slow lock
based workarround
which means that you are already running without atomic locks.

Regards,
Ovidiu Sas

On Wed, Nov 13, 2013 at 10:40 AM, Sotas Development sotas...@gmail.com wrote:
 Hi,

 Here's an update of this topic. We've tried again with the latest stable
 version 4.0.4. Unfortunately the problem still exists.

 In mails above it is mentioned to use kamailio without atomic locks. How do
 we this (e.g. which makefile options)?

 Kind regards,

 Bert
 (on behalf of Michiel Veldkamp)



 On Mon, Jan 28, 2013 at 4:44 PM, Ovidiu Sas o...@voipembedded.com wrote:

 4.0 (current trunk) is in code freeze.  I would suggest to test the
 trunk version (next 4.0).
 Even openser 1.3 requires patches to be properly cross compiled.

 Regards,
 Ovidiu Sas

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

 -- Forwarded message --
 From: Sotas Development sotas...@gmail.com
 Date: Mon, Jan 28, 2013 at 10:08 AM
 Subject: Re: [SR-Users] Kamailio stability/timing problem w.r.t.
 registrations?
 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
 Users Mailing List sr-users@lists.sip-router.org


 Hi Ovidiu,

 Thanks for the warning! We did not yet have much success running the
 current master branch, though this may well be a resource problem on
 the target platform.

 For the moment, we decided to switch back to openser 1.3.5 and wait
 for the official 4.0 release.

 Regards,
 Michiel Veldkamp


 On Thu, Jan 17, 2013 at 7:01 PM, Ovidiu Sas o...@voipembedded.com wrote:
 
  If you are running the stable version, there's need for heavy Makefile
  patching in order to properly cross compile (not to include and link
  to host libs).
  The trunk has everything fixed and it's cross-compiling properly for
  most of the modules.
  Make sure that your binaries are properly cross compiled.
 
  Depending on your ARM CPU, atomic locks may or may not work.
  I tested openser without atomic locks (using regular locks) and it
  worked fine.
 
  Regards,
  Ovidiu Sas
 
 
  --
  VoIP Embedded, Inc.
  http://www.voipembedded.com

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Re: [SR-Users] xHttp_PI Module

[Ovidiu Sas]

Please post your questions to the mailing list instead of sending
private e-mails.

The subscriber provisioning via xhttp is supported only for plaintext
passwords. Set
modparam(auth_db, calculate_ha1, 1)
and it will work fine.

Regards,
Ovidiu Sas

On Sat, Nov 2, 2013 at 12:22 PM, Abdul Hakeem alhak...@gmail.com wrote:
 Hello,

 I just want to enquire if the subscriber full provisioning is ready.
 Best regards,
 Abdul Hakeem





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Re: [SR-Users] [sr-dev] Kamailio Tech Admin Group

[Ovidiu Sas]

I can give a hand on this.
I already maintain the kamailio optware feeds for embedded systems.

Regards,
Ovidiu Sas

On Mon, Oct 28, 2013 at 4:49 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Hello,

 being discussed during last Devel IRC Meeting, we are planing to build a
 Kamailio Project Technical Administration Group:

 https://www.kamailio.org/wiki/devel/irc-meetings/2013blog#technical_administration_group

 Its goal is to get a bunch of people that volunteer to do administration
 tasks for the project, such as:
 - helping with releases (e.g., patch backports, packaging, uploading files
 for download, etc)
 - doing sysadmin tasks for our servers (e.g., performing upgrades to wiki,
 web site, etc)
 - preparing technical decisions and doing them (e.g., what applications to
 use to make operations easier, cloning git repository to github, ...)

 From the devel meeting, so far we have Victor Seva, Fred Posner, Peter
 Dunkley and Olle Johansson. Existing people doing admin tasks will probably
 stay in (if they don't opt out): me, Elena-Ramona Modroiu, Henning
 Westerholt (owner of devel.kamailio.org hardware), Jan Janak (owner of
 sip-router.org hardware), Jesus Rodriguez and Oriol Capsada (owners of
 kamailio.org hardware).

 Requirements for candidates and other details:
 - volunteer to do the work, it is not a paid job
 - an existing record of activity within the project is a plus (e.g.,
 developer, active mailing list member)
 - reply to the lists detailing where and how you can help
 - possibility to spend 1-4 hours a week for project administration (more is
 welcome, sometime is not necessary at all)

 Rewards:
 - you will be listed as part of project administration on the website
 - get to interact more with the project and the nice guys around it ;-)
 - more spam - admin list address will be public and the list open so
 everyone can send in case of critical situations (content/archive will be
 kept private)

 Note that we will try to build a group of an adequate size, thus not
 everyone willing to participate may get in (at least on the first phase).
 One criteria is to have skills that complement existing team knowledge.

 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla - http://www.asipto.com
 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
 Kamailio Advanced Trainings - Berlin, Nov 25-28
   - more details about Kamailio trainings at http://www.asipto.com -


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Re: [SR-Users] Decoding HTTP URLs in event_route[xhttp:request]

[Ovidiu Sas]

Great!

Now don't forget to update the wiki:
 - 
http://www.kamailio.org/wiki/cookbooks/devel/transformations#parameters_list_transformations
and create the new entry for url transformations:
 - 
http://www.kamailio.org/wiki/cookbooks/devel/transformations#url_transformations

Regards,
Ovidiu Sas

On Wed, Sep 25, 2013 at 11:15 AM, Peter Dunkley
peter.dunk...@crocodilertc.net wrote:
 Hello,

 I have added a transformation to the xhttp module that breaks a URL into a
 path and a querystring
 - {url.path}
 - {url.querystring}

 I have also added an optional delimiter parameter to the {param.}
 transformations.

 Regards,

 Peter


 On 22 September 2013 14:55, Ovidiu Sas o...@voipembedded.com wrote:

 You can use {s.select,index,separator} to extract the path and the
 parameters into two different variables.
 Or here you could create a new url transformation to break it in two:
  - {url.path}
  - {url.searchpath}

 After that, the existing code for param transformation may be reused
 (by making the separator configurable (using '' instead of ';') and
 we could have a new transformation:
  - {urlsearchpath.value,name}
 Or maybe we can enhance the existing param transformation to pass as
 an optional argument - the param delimiter:
  - {param.value,name,[param_delimiter]}.
  - {param.valueat,index,[param_delimiter]}
  - {param.name,index,[param_delimiter]}
  - {param.count,[param_delimiter]}


 Regards,
 Ovidiu Sas

 On Sun, Sep 22, 2013 at 4:50 AM, Peter Dunkley
 peter.dunk...@crocodilertc.net wrote:
  Hello,
 
  Does anyone have any ideas about this?
 
  If not it's something I want to try and do before the freeze (any
  suggestions as to how would be appreciated) as it will be a nice
  finishing
  touch to the WebSocket/outbound/stun/auth_ephemeral stuff I've worked on
  over the last couple of releases.
 
  Thanks,
 
  Peter
 
 
  On 19 September 2013 21:36, Peter Dunkley
  peter.dunk...@crocodilertc.net
  wrote:
 
  Hello,
 
  I was wondering if there was an easy way to decode HTTP URLs in
  event_route[xhttp:request]?
 
  For example, it would be good to be able to breakdown a URL like:
/sip?apiKey=abcdefgusername=1234567890:al...@example.com
  into path/on/server (/sip in this case) and a set of parameters.
  For
  the parameters something like the {param.value,name} transformation for
  SIP
  header parameters would be ideal (which works perfectly for picking
  values
  out of HTTP Cookie: headers).
 
  I noticed that there is already an {s.urldecode.param} transformation
  in
  the PV module but I couldn't find any documentation for it in the wiki
  and
  looking at the code it doesn't appear to do this anyway.
 
  Regards,
 
  Peter
 
 
  --
  Peter Dunkley
  Technical Director
  Crocodile RCS Ltd
 
 
 
 
  --
  Peter Dunkley
  Technical Director
  Crocodile RCS Ltd
 
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Re: [SR-Users] Decoding HTTP URLs in event_route[xhttp:request]

[Ovidiu Sas]

You can use {s.select,index,separator} to extract the path and the
parameters into two different variables.
Or here you could create a new url transformation to break it in two:
 - {url.path}
 - {url.searchpath}

After that, the existing code for param transformation may be reused
(by making the separator configurable (using '' instead of ';') and
we could have a new transformation:
 - {urlsearchpath.value,name}
Or maybe we can enhance the existing param transformation to pass as
an optional argument - the param delimiter:
 - {param.value,name,[param_delimiter]}.
 - {param.valueat,index,[param_delimiter]}
 - {param.name,index,[param_delimiter]}
 - {param.count,[param_delimiter]}


Regards,
Ovidiu Sas

On Sun, Sep 22, 2013 at 4:50 AM, Peter Dunkley
peter.dunk...@crocodilertc.net wrote:
 Hello,

 Does anyone have any ideas about this?

 If not it's something I want to try and do before the freeze (any
 suggestions as to how would be appreciated) as it will be a nice finishing
 touch to the WebSocket/outbound/stun/auth_ephemeral stuff I've worked on
 over the last couple of releases.

 Thanks,

 Peter


 On 19 September 2013 21:36, Peter Dunkley peter.dunk...@crocodilertc.net
 wrote:

 Hello,

 I was wondering if there was an easy way to decode HTTP URLs in
 event_route[xhttp:request]?

 For example, it would be good to be able to breakdown a URL like:
   /sip?apiKey=abcdefgusername=1234567890:al...@example.com
 into path/on/server (/sip in this case) and a set of parameters.  For
 the parameters something like the {param.value,name} transformation for SIP
 header parameters would be ideal (which works perfectly for picking values
 out of HTTP Cookie: headers).

 I noticed that there is already an {s.urldecode.param} transformation in
 the PV module but I couldn't find any documentation for it in the wiki and
 looking at the code it doesn't appear to do this anyway.

 Regards,

 Peter


 --
 Peter Dunkley
 Technical Director
 Crocodile RCS Ltd




 --
 Peter Dunkley
 Technical Director
 Crocodile RCS Ltd

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-- 
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http://www.voipembedded.com

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Re: [SR-Users] Cross-Compilation Problems for mips (gcc 3.4.2)

[Ovidiu Sas]

Please open a bug report about it.
And please check that your cross compilation is sane (no includes from
your local build system).

Regards,
Ovidiu Sas

On Thu, Sep 12, 2013 at 10:10 AM, Tirant Lo Blanc
tirantloblan...@yahoo.es wrote:


 I managed to fix it by adding:

 #include linux/types.h
 to sipcapture.c and socket_info.c

 Thanks to all anyway



 

 Hi,

 I've been exploring the possibility to port Kamailio 3.3 (SER) to some MIPS 
 boards. I didn't have any problem with the first one, with a GCC 4.3.4 
 toolchain. But on my second board (gcc 3.4.2)  I am having problems when 
 compiling. Are there any requirements for gcc/binutils/kernel versions?

 This is the log I am getting:

 CC (mipsel-linux-uclibc-gcc) [ser]sip_msg_clone.o
 In file included from atomic_ops.h:181,
  from sip_msg_clone.c:43:
 atomic/atomic_unknown.h:59:2: warning: #warning no native memory barrier 
 implementations, falling back to slow lock based workarround
 CC (mipsel-linux-uclibc-gcc) [ser]socket_info.o
 In file included from socket_info.c:836:
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:22:
  error: parse error before __u32
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:28:
  error: parse error before __u32
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:30:
  error: parse error before nlmsg_flags
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:31:
  error: parse error before nlmsg_seq
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:32:
  error: parse error before nlmsg_pid
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:83:
  error: field `msg' has incomplete type
 In file included from socket_info.c:837:
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:253:
  error: parse error before __u32
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:255:
  error: parse error before rta_expires
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:256:
  error: parse error before rta_error
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:257:
  error: parse error before rta_used
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:260:
  error: parse error before rta_id
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:261:
  error: parse error before rta_ts
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:262:
  error: parse error before rta_tsage
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:333:
  error: parse error before __s32
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:359:
  error: parse error before __u16
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:361:
  error: parse error before ndm_type
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:404:
  error: parse error before __u32
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:406:
  error: parse error before ndm_updated
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:407:
  error: parse error before ndm_refcnt
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:519:
  error: parse error before __u32
 /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:521:
  error: parse error before tcm_info
 socket_info.c: In function `addattr_l':
 socket_info.c:874: error: dereferencing pointer to incomplete type
 socket_info.c:878: error: dereferencing pointer to incomplete type
 socket_info.c:882: error: dereferencing pointer to incomplete type
 socket_info.c:882: error: dereferencing pointer to incomplete type
 socket_info.c: In function `nl_bound_sock

[SR-Users] rtpproxy behind NAT

[Ovidiu Sas]

I've seen a lot of discussions about running rtpproxy behind NAT.
The fact is that standard vanilla rtpproxy can run behind NAT without
any issues (no patches required).
A few things must be addressed:
 - the proper ports must be forwarded from the public IP to the private IP;
 - when calling rtpproxy_offer/answer, the second parameter must be
properly populated with the external public IP for streams from public
network.

I hope this brings some light over this highly debated topic.


Regards,
Ovidiu Sas

--
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http://www.voipembedded.com

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Re: [SR-Users] Siremis 4.0 with Kamailio 4.0

[Ovidiu Sas]

If you simply want a web interface for MI/RPC commands, take a look at
the xhttp_rpc module:
http://kamailio.org/docs/modules/devel/modules/xhttp_rpc

Regards,
Ovidiu Sas

On Fri, Jun 28, 2013 at 5:09 PM, Geoffrey Mina geoffreym...@gmail.com wrote:
 I am having some trouble getting Siremis 4.0 to work with Kamailio 4.0.  The
 PHP application is functioning fine and all the DB access is working as
 intended.  Where I am having an issue is with the Command Services
 section.  I am unclear weather I want the MI or XMLRPC.  The basic
 functions I want to use would be:

 sip_trace on/off
 debug 
 lcr_reload
 address_reload

 I know they are going to be different coming from 1.5 to 4.0, but I am
 hoping there is still a way to get Siremis to invoke these commands.  When I
 click on the XMLRPC section, I get this error:

 [2013-06-28 21:09:16 (GMT)] An exception occurred while executing this
 script:
 Error message: #2, require_once(XML/RPC.php) [function.require-once]: failed
 to open stream: No such file or directory
 Script name and line number of error:
 /var/www/siremis-4.0.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2

 function: errorHandler ( 2, require_once(XML/RPC.php) [a
 href='function.require-once'funct...,
 /var/www/siremis-4.0.0/siremis/modules/ser/service/asipto/libs/c..., 2,
 Array(12) ) @ /var/www/siremis-4.0.0/openbiz/bin/sysheader.inc 117
 function: userErrorHandler ( 2, require_once(XML/RPC.php) [a
 href='function.require-once'funct...,
 /var/www/siremis-4.0.0/siremis/modules/ser/service/asipto/libs/c..., 2,
 Array(12) ) @
 function: require_once ( ) @
 /var/www/siremis-4.0.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php
 2
 function: include_once (
 /var/www/siremis-4.0.0/siremis/modules/ser/service/asipto/libs/c... ) @
 /var/www/siremis-4.0.0/siremis/modules/ser/cms/form/XrcmdsForm.php 2
 function: include_once (
 /var/www/siremis-4.0.0/siremis/modules/ser/cms/form/XrcmdsForm.p... ) @
 /var/www/siremis-4.0.0/openbiz/bin/ObjectFactory.php 162
 function: constructObject ( ser.cms.form.XrcmdsForm ) @
 /var/www/siremis-4.0.0/openbiz/bin/ObjectFactory.php 56
 function: getObject ( ser.cms.form.XrcmdsForm ) @
 /var/www/siremis-4.0.0/openbiz/bin/easy/EasyView.php 348
 function: initAllForms ( ) @
 /var/www/siremis-4.0.0/openbiz/bin/easy/EasyView.php 232
 function: render ( ) @ /var/www/siremis-4.0.0/openbiz/bin/BizController.php
 221
 function: renderView ( ser.view.XrcmdsView, , , Null,  ) @
 /var/www/siremis-4.0.0/openbiz/bin/BizController.php 107
 function: dispatchRequest ( ) @
 /var/www/siremis-4.0.0/openbiz/bin/BizController.php 32
 function: include_once (
 /var/www/siremis-4.0.0/openbiz/bin/BizController.php ) @
 /var/www/siremis-4.0.0/siremis/bin/controller.php 6
 function: include ( /var/www/siremis-4.0.0/siremis/bin/controller.php ) @
 /var/www/siremis-4.0.0/siremis/bin/_forward.php 102
 function: include ( /var/www/siremis-4.0.0/siremis/bin/_forward.php ) @
 /var/www/siremis-4.0.0/siremis/index.php 3


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Re: [SR-Users] Xhttp_pi DB Query Clause Operators

[Ovidiu Sas]

Hello Simpson,

The xhttp_pi module is built on top of the existing kamailio generic
db api which doesn't provide a contains or like SQL query.
The supported clause operators are listed in the autogenerated
pi_framework.xml, but I will need to add them also to the module
readme:
/* clause_cols operator */
- ''  'lt;'
- ''  'gt;'
- '='  '='
- '=' 'lt;='
- '=' 'gt;='
- '!=' '!='

Regards,
Ovidiu Sas

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

On Thu, Jun 27, 2013 at 7:01 AM, Simpson Chua simpsonc...@yahoo.com wrote:
 Hi,

 I'm using the xhttp_pi module to query the kamailio user registration table
 location. Per the defined clause_cols operators in the module, it doesn't
 seem possible to achieve a query equivalent to a SQL contains or like.
 An exact match is required to query the record; e.g.,
 http://xxx/location/QueryContacts?cmd=on0=sip:8675309@a.b.c.d:5060;transport=udp.
 Is my understanding correct that there is no way to match on an regular
 expression? My end goal is to be able to query for a user's AOR and
 expiration status based on a search string (e.g. 8675309) via a webservice.
 Can anyone share a better method to do this? Perhaps asynchronously? All
 feedback is appreciated.

 Sample contact field:
 sip:8675309@a.b.c.d:5060;transport=udp

 Sample Framework:
 cmdcmd_nameQueryContacts/cmd_name
 db_table_idlocation/db_table_id
 cmd_typeDB1_QUERY/cmd_type
 clause_cols
 colfieldcontact/fieldoperator=/operator/col
 /clause_cols
 query_cols
 colfieldusername/field/col
 colfieldcontact/field/col
 colfieldexpires/field/col
 /query_cols
 /cmd

 Thanks,
 Simpson

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Re: [SR-Users] [sr-dev] releasing v4.0.2

[Ovidiu Sas]

Here's a list of compiler warnings:


CC (gcc) [kamailio] cfg/cfg_ctx.o
cfg/cfg_ctx.c: In function âcfg_set_nowâ:
cfg/cfg_ctx.c:490:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:494:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:564:4: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:564:4: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:565:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:584:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c: In function âcfg_commitâ:
cfg/cfg_ctx.c:1125:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1133:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1185:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1185:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1190:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1225:4: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1226:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1226:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1228:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c: In function âcfg_add_group_instâ:
cfg/cfg_ctx.c:1582:2: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1583:2: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1585:2: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1594:3: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c: In function âcfg_del_group_instâ:
cfg/cfg_ctx.c:1678:2: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1679:2: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1681:2: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1692:5: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1710:6: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1712:6: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]
cfg/cfg_ctx.c:1718:3: warning: dereferencing type-punned pointer will
break strict-aliasing rules [-Wstrict-aliasing]

CC (gcc) [M pdb.so] pdb.o
pdb.c: In function âpdb_queryâ:
pdb.c:273:7: warning: dereferencing type-punned pointer will break
strict-aliasing rules [-Wstrict-aliasing]

CC (gcc) [M xmlrpc.so]  xmlrpc.o
xmlrpc.c: In function âselect_methodâ:
xmlrpc.c:2408:9: warning: âdoc.lenâ may be used uninitialized in this
function [-Wmaybe-uninitialized]
xmlrpc.c: In function âdispatch_rpcâ:
xmlrpc.c:2010:26: warning: âdoc.lenâ may be used uninitialized in this
function [-Wmaybe-uninitialized]
xmlrpc.c:1998:6: note: âdoc.lenâ was declared here

CC (gcc) [M memcached.so]   mcd_var.o
mcd_var.c: In function âpv_get_mcd_value_helperâ:
mcd_var.c:102:2: warning: field precision specifier â.*â expects
argument of type âintâ, but argument 8 has type âsize_tâ [-Wformat]
mcd_var.c:103:1: warning: field precision specifier â.*â expects
argument of type âintâ, but argument 6 has type âsize_tâ [-Wformat]
mcd_var.c:103:1: warning: field precision specifier â.*â expects
argument of type âintâ, but argument 7 has type âsize_tâ [-Wformat]
mcd_var.c:103:1: warning: field precision specifier â.*â expects
argument of type âintâ, but argument 5 has type âsize_tâ [-Wformat]
mcd_var.c:103:1: warning: field precision specifier â.*â expects
argument of type âintâ, but argument 5 has type âsize_tâ [-Wformat]
mcd_var.c: In function âpv_mcd_atomic_helperâ:
mcd_var.c:238:2: warning: suggest parentheses around operand of â!â or
change ââ to ââ or â!â to â~â [-Wparentheses]
mcd_var.c:258:2: warning: field precision specifier â.*â expects
argument of type âintâ, but argument 8 has type âsize_tâ [-Wformat]
mcd_var.c:258:1: warning: field precision specifier 

Re: [SR-Users] [sr-dev] releasing v4.0.2

[Ovidiu Sas]

All this warnings pop up on the latest debian stable.
Pretty sure that ubuntu will pop them up too.
Centos is a little behind, so no warnings there.

On Fri, Jun 7, 2013 at 12:10 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Probably they are specific for some os/gcc version. I am not getting them on
 my devel computer, perhaps other devs can jump and fix some if they get
 them.

 Cheers,
 Daniel


 On 6/7/13 6:06 PM, Ovidiu Sas wrote:

 Here's a list of compiler warnings:


 CC (gcc) [kamailio] cfg/cfg_ctx.o
 cfg/cfg_ctx.c: In function ācfg_set_nowā:

 cfg/cfg_ctx.c:490:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:494:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:564:4: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:564:4: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:565:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:584:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c: In function ācfg_commitā:

 cfg/cfg_ctx.c:1125:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1133:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1185:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1185:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1190:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1225:4: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1226:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1226:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1228:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c: In function ācfg_add_group_instā:

 cfg/cfg_ctx.c:1582:2: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1583:2: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1585:2: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1594:3: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c: In function ācfg_del_group_instā:

 cfg/cfg_ctx.c:1678:2: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1679:2: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1681:2: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1692:5: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1710:6: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1712:6: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]
 cfg/cfg_ctx.c:1718:3: warning: dereferencing type-punned pointer will
 break strict-aliasing rules [-Wstrict-aliasing]

 CC (gcc) [M pdb.so] pdb.o
 pdb.c: In function āpdb_queryā:

 pdb.c:273:7: warning: dereferencing type-punned pointer will break
 strict-aliasing rules [-Wstrict-aliasing]

 CC (gcc) [M xmlrpc.so]  xmlrpc.o
 xmlrpc.c: In function āselect_methodā:
 xmlrpc.c:2408:9: warning: ādoc.lenā may be used uninitialized in this
 function [-Wmaybe-uninitialized]
 xmlrpc.c: In function ādispatch_rpcā:
 xmlrpc.c:2010:26: warning: ādoc.lenā may be used uninitialized in this
 function [-Wmaybe-uninitialized]
 xmlrpc.c:1998:6: note: ādoc.lenā was declared here


 CC (gcc) [M memcached.so]   mcd_var.o
 mcd_var.c: In function āpv_get_mcd_value_helperā:
 mcd_var.c:102:2: warning: field precision specifier ā.*ā expects
 argument of type āintā, but argument 8 has type āsize_tā [-Wformat]
 mcd_var.c:103:1: warning: field precision specifier ā.*ā expects
 argument of type āintā, but argument 6 has type āsize_tā [-Wformat]
 mcd_var.c:103:1: warning: field precision specifier ā.*ā expects
 argument of type āintā, but argument 7 has type āsize_tā [-Wformat]
 mcd_var.c:103:1: warning: field precision specifier ā.*ā expects
 argument

Re: [SR-Users] kamcmd vs kamctl

[Ovidiu Sas]

Hello Dan,

You can also use the built in web provisioning interface:
http://kamailio.org/docs/modules/4.0.x/modules/xhttp_pi.html
You can build your own provisioning layout (you can preset fields) and
also pre-validate data before pushing it into the database (like URI
or socket type validation).

Also, for running RPC commands via a web interface see:
http://kamailio.org/docs/modules/4.0.x/modules/xhttp_rpc.html

Regards,
Ovidiu Sas

On Tue, Jun 4, 2013 at 4:07 AM, DanB danb.li...@gmail.com wrote:
 Hey Daniel,

 Thanks for the answer. It makes more sense now.

 Have a good one!
 DanB



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Re: [SR-Users] home pbx server experience

[Ovidiu Sas]

The biggest issue with using a SIP proxy as a PBX is performing
authentication on outgoing calls to carriers.
I use asterisk front-end-ed by the proxy.  Like this, I can provision
authentication credentials on asterisk and route the call from
asterisk to carrier through the proxy.
I don't like the idea of running the proxy on the router (if I change
the router or the firmware on the router I need to do more work) and
therefor I run the proxy and the asterisk on two small arm boxes and I
route calls between them.  I register the subscribers on the proxy and
I route through asterisk only when I need to.

Regards,
Ovidiu Sas

On Tue, May 14, 2013 at 10:27 AM, u ueberwachungsst...@googlemail.com wrote:
 I would like to share my experience with kamailio and other home pbx servers.

 Kamailio on my kirkwood home router for my 6 SIP users is perhaps
 overkill: I don't really need mysql and scalability. But at last I
 finally managed to make calling between registered users work stable.
 My voip clients only work in all NAT scenarios if I work around some
 bugs: to use csipsimple on android I had to change rtpproxy_manage()
 to rtpproxy_manage(c) in kamailio's default config, so that problems
 with conflicting c: entries in the SDP go away.

 I propose kamailio could ship with a special example
 kamailio-compatible.cfg that doesn't try to be RFC compliant, but
 compatible to the most common voip clients. Right now the only thing I
 would change for this is the option for rtpproxy_manage, but I'm sure
 others will know more common quirks that could safely be enabled to
 increase compatibility. I think this compatibility idea is what yate
 sticks to for their defaults. In freeswitch you also have to do it all
 manually, and it's much more work to figure things out in their
 enormous config files.

 The other SIP proxies I had tried before kamailio officially fit all
 my requirements, including support for multihomed dynamic IPs, but
 contrary to their claims it didn't work.
 Yate was easy to set up, but the default dialplan is more confusing
 than powerful and after having made everything work I realised yate
 was clogging my CPU and RAM and after some time always randomly
 stopped working. This is with only 2 users connected! It also wasn't
 possible to fix NAT sdp while leaving the codecs section in the SDP
 alone at the same time. I tried to debug the code, but the C++ was so
 complex that I had to give up.
 Freeswitch was much more difficult to setup, a multihomed setup with
 dynamic IP was super buggy and it also didn't help that the
 unintuitive configuration is all in complex unreadable XML
 configuration files.

 Kamailio and rtpproxy don't officially support dynamic IP address, but
 I can just restart both each time my DSL provider forces me to a new
 IP address. This happens automatically in the night and is no big
 hassle really. The most simple, least-featureful solution works best
 it seems.

 Now the last problem I have with kamailio: I don't know how to connect
 my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont).
 I would like a simple way to do this, preferably without other
 features that always seem to complicate the matters. Is there
 something more lightweight and simple than asterisk, freeswitch and
 yate, that people use successfully for this task together with
 kamailio and rtpproxy?

 u

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Re: [SR-Users] Porting kamailio4 to OpenWrt

[Ovidiu Sas]

You are cross compiling and you need to use CROSS_COMPILE.
Take a look at the Makefile used to build for optware on all kind of
embedded platforms:
http://svn.nslu2-linux.org/svnroot/optware/trunk/make/kamailio.mk

Regards,
Ovidiu Sas

On Tue, Mar 12, 2013 at 11:02 AM, Jiri Slachta slac...@cesnet.cz wrote:
 Hello everyone,

 I am new to this list and I am a newbie when it comes to Kamailio details, so 
 sorry for any of my misunderstandings. I am the author of kamailio3 package 
 in OpenWrt and currently I am trying to port recent major release of kamailio 
 to OpenWrt. Let's jump directly to my question.

 The current state of my Kamailio4 package is that all modules which does not 
 depend on any external libraries, are succesfully built. Any module that 
 depends on external library is not built at all.

 But if I pass LD=$(TARGET_CC) to the linker via:

 make -C/path/to/module LD=$(TARGET_CC)

 then the module is succesfully compiled. I build kamailio4 via macro 
 Build/Compile in following Makefile - 
 http://liptel.vsb.cz/svn/besip/Trunk/packages-trunk/net/kamailio4/Makefile

 My question is - has anything changed in build procedures or variables 
 significantly? It seems that LD variable is not passed into Makefiles of 
 specific modules. Am I doing anything wrong?

 I am open to any suggestions.

 Thank you!

 ~ Jiri Slachta

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

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Re: [SR-Users] [sr-dev] Update existing module or create new?

[Ovidiu Sas]

http://sip-router.org/contribute/
http://sip-router.org/tracker/

Regards,
Ovidiu Sas

On Thu, Feb 21, 2013 at 5:37 PM, Charles Chance
charles.cha...@sipcentric.com wrote:
 Hi,


 On 06/02/2013 15:16, Henning Westerholt wrote:

 Am Dienstag, 5. Februar 2013, 13:55:41 schrieb Charles Chance:

 as the original author of the module I'd think that changing or replacing
 the existing module would be the way to go. So far I'd not recieved that
 much of bug reports against the existing module. And as Alex Balashov also
 mentioned recently, there are some other issues with the current library.

 If existing users need to stay with the old module, its available in the
 git and the existing releases, for the new release we should go with a
 module which supports the newer library.

 It would be nice if you could stay with the existing PV API, which I
 modelled somehow after the htable module. If you need to change something,
 just announce it on the devel list and ask for feedback.

 We have indeed used the module in the past with no issues - so thank you for
 writing and sharing :)

 Very happy to stay with existing PVs if possible. The only thing I'd like to
 see different is to set value and expiry at the same time, instead of
 having to set value, then alter expiration. This has to be better than
 setting a value with some default expiry, getting that same value back
 again, then re-setting the value once more with a different expiry?

 Could this be implemented at PV level? Something like $mct(key:expiry) =
 value? And if expiry is omitted, we use default set in params.

 Hi Charles,

 thanks, good to know that you use it. :-) With regards to the expiry value,
 yes I think this could be implemented like this. Just one remark, the syntax
 that other PVs uses is =, like in
 http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#sht_htable_key

 Then it would be $mct(key=expiry) = value

 Best regards,

 Henning Westerholt



 We now have an updated memcached module, working with libmemcached and also
 with the added ability to (optionally) specify expiry in the format
 $mct(key=expiry).

 How do we get these changes pushed back into the master?


 Regards,

 Charles

 --
 www.sipcentric.com

 Follow us on twitter @sipcentric

 Sipcentric Ltd. Company registered in England  Wales no. 7365592.
 Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South,
 Birmingham B7 4EJ.

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Re: [SR-Users] modifying global data-structures using Rtimer module

[Ovidiu Sas]

Why don't you just simply use db_text?
http://kamailio.org/docs/modules/devel/modules/db_text.html

Regards,
Ovidiu Sas

On Mon, Feb 4, 2013 at 11:34 PM, Kiran Bhosale kiranbhosa...@gmail.com wrote:
 Hi

 we  have  developed the custom module which stores the registered users  in
 a file.now we are trying to remove the expired  contacts using the rtimer
 module.while saving  the  registered users  to the file  we also store the
 expires  values in static  array.but when  we  try  to decrement the  these
 values  in a function called  with the help of rtimer  module. the values
 used  by this  periodic  function are not modified ones but the initial
 which are zero. is it that  we cant pass  the modified  values of  the
 global variables to the timed  functions. to get around  this  problem, we
 also registered the timer in our module but got  same results !!!

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Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Ovidiu Sas]

4.0 (current trunk) is in code freeze.  I would suggest to test the
trunk version (next 4.0).
Even openser 1.3 requires patches to be properly cross compiled.

Regards,
Ovidiu Sas

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

-- Forwarded message --
From: Sotas Development sotas...@gmail.com
Date: Mon, Jan 28, 2013 at 10:08 AM
Subject: Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List sr-users@lists.sip-router.org


Hi Ovidiu,

Thanks for the warning! We did not yet have much success running the
current master branch, though this may well be a resource problem on
the target platform.

For the moment, we decided to switch back to openser 1.3.5 and wait
for the official 4.0 release.

Regards,
Michiel Veldkamp


On Thu, Jan 17, 2013 at 7:01 PM, Ovidiu Sas o...@voipembedded.com wrote:

 If you are running the stable version, there's need for heavy Makefile
 patching in order to properly cross compile (not to include and link
 to host libs).
 The trunk has everything fixed and it's cross-compiling properly for
 most of the modules.
 Make sure that your binaries are properly cross compiled.

 Depending on your ARM CPU, atomic locks may or may not work.
 I tested openser without atomic locks (using regular locks) and it worked 
 fine.

 Regards,
 Ovidiu Sas


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

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Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Ovidiu Sas]

Hello Michiel ,

You should check with top to see the status of all kamailio processes.
If you see processes that are using 100% CPU, you may have a deadlock.
Connect with gdb to the process to investigate what's going on.

Also, you are running on arm.  How did you compiled kamailio: native or cross.
When you run kamailio on an embedded system, you need to check if you
have enough memory.  If you start running out of memory, the OS may
start killing processes randomly (check your OS logs).

You definitely need to look at this issue from a broader prospective
as it's not a regular x86 deployment with plenty of CPU and memory.
Also tuning the config (by removing everything that you don't need
might help with memory utilization).


Regards,
Ovidiu Sas

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

On Thu, Jan 17, 2013 at 8:41 AM, Sotas Development sotas...@gmail.com wrote:
 Daniel,


 can you add an xlog() at the start of the main route block and log a
 message for any
 request received? [...] You can put another xlog before the save()
 function to see if
 registration requests are getting there. [...] You can switch to kamailio
 flavour modules
 and see if reproduces.

 Thanks for the help! We switched to kamailio flavour, added some xlog
 messages and managed to reproduce. See below for some logging.

 To our surprise, the last message handled was an INVITE. We'll add more
 logging to see whether it is handled successfully or gets stuck somewhere.

 Those numbers at the start of the log message, are these child IDs? At first
 they alternate, but apparently child 1 and 2 stop running early on (each
 with an INVITE as last message).

 Regards,

 Michiel Veldkamp

  0(304) WARNING: core [socket_info.c:1392]: WARNING: fix_hostname: could
 not rev. resolve 192.168.10.1
  0(304) INFO: core [tcp_main.c:4832]: init_tcp: using epoll_lt as the io
 watch method (auto detected)
  0(306) INFO: usrloc [hslot.c:53]: locks array size 512
  0(306) INFO: core [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is
 initially 108544
  0(306) INFO: core [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is
 finally 217088
  7(315) INFO: ctl [io_listener.c:225]: io_listen_loop:  using epoll_lt io
 watch method (config)
  1(309) ERROR: script: request_route start -- method=REGISTER
  2(310) ERROR: script: request_route start -- method=REGISTER
  3(311) ERROR: script: request_route start -- method=REGISTER
  3(311) ERROR: script: route[REGISTRAR] start
  2(310) ERROR: script: route[REGISTRAR] start
  3(311) ERROR: script: route[REGISTRAR] saving location...
  2(310) ERROR: script: route[REGISTRAR] saving location...
  1(309) ERROR: script: route[REGISTRAR] start
  1(309) ERROR: script: route[REGISTRAR] saving location...
  1(309) ERROR: script: route[REGISTRAR] saving location... done
  3(311) ERROR: script: route[REGISTRAR] saving location... done
  2(310) ERROR: script: route[REGISTRAR] saving location... done
  3(311) ERROR: script: request_route start -- method=REGISTER
  3(311) ERROR: script: route[REGISTRAR] start
  3(311) ERROR: script: route[REGISTRAR] saving location...
  3(311) ERROR: script: route[REGISTRAR] saving location... done
  2(310) ERROR: script: request_route start -- method=REGISTER
  1(309) ERROR: script: request_route start -- method=REGISTER
  1(309) ERROR: script: route[REGISTRAR] start
  1(309) ERROR: script: route[REGISTRAR] saving location...
  2(310) ERROR: script: route[REGISTRAR] start
  2(310) ERROR: script: route[REGISTRAR] saving location...
  1(309) ERROR: script: route[REGISTRAR] saving location... done
  2(310) ERROR: script: route[REGISTRAR] saving location... done
  3(311) ERROR: script: request_route start -- method=INVITE
  3(311) ERROR: script: route[REGISTRAR] start
  1(309) ERROR: script: request_route start -- method=INVITE
  1(309) ERROR: script: route[REGISTRAR] start
  2(310) ERROR: script: request_route start -- method=INVITE
  2(310) ERROR: script: route[REGISTRAR] start
  1(309) ERROR: script: request_route start -- method=INVITE
  1(309) ERROR: script: route[REGISTRAR] start
  2(310) ERROR: script: request_route start -- method=INVITE
  2(310) ERROR: script: route[REGISTRAR] start
  2(310) ERROR: script: request_route start -- method=ACK
  1(309) NOTICE: acc [acc.c:275]: ACC: transaction answered:
 timestamp=1350034404;method=INVITE;from_tag=966002447;to_tag=3a8473f2;call_id=331189810@192.168.10.2;code=200;reason=OK;src_user=Radio1_device;src_domain=testnet;src_ip=192.168.10.2;dst_ouser=Radio1;dst_user=3;dst_domain=192.168.10.2
  3(311) NOTICE: acc [acc.c:275]: ACC: transaction answered:
 timestamp=1350034404;method=INVITE;from_tag=1718943109;to_tag=22a97aa7;call_id=956233863@192.168.10.2;code=200;reason=OK;src_user=User3;src_domain=testnet;src_ip=192.168.10.2;dst_ouser=IC1;dst_user=1;dst_domain=192.168.10.2
  3(311) ERROR: script: request_route start -- method=ACK
  1(309) NOTICE: acc [acc.c:275]: ACC: transaction answered:
 timestamp=1350034404;method=INVITE;from_tag=1709043140

Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Ovidiu Sas]

If you are running the stable version, there's need for heavy Makefile
patching in order to properly cross compile (not to include and link
to host libs).
The trunk has everything fixed and it's cross-compiling properly for
most of the modules.
Make sure that your binaries are properly cross compiled.

Depending on your ARM CPU, atomic locks may or may not work.
I tested openser without atomic locks (using regular locks) and it worked fine.

Regards,
Ovidiu Sas


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

On Thu, Jan 17, 2013 at 12:43 PM, Sotas Development sotas...@gmail.com wrote:
 Hi Ovidiu,

 Good points. We're not running out of memory, and all processes keep
 running.
 Top shows 0% or 1% CPU load for the Kamailio processes.

 We cross-compile with the CodeSourcery toolchain. The default build options
 drag in the file atomic_unknown.h, that produces the following compile
 warning:

 no native memory barrier implementations, falling back to slow lock based
 workarround

 Currently we're testing a version compiled with -DNOSMP (no
 atomic_unknown.h)
 that will run over the weekend.

 Regards,
 Michiel Veldkamp


 On Thu, Jan 17, 2013 at 3:20 PM, Ovidiu Sas o...@voipembedded.com wrote:

 Hello Michiel ,

 You should check with top to see the status of all kamailio processes.
 If you see processes that are using 100% CPU, you may have a deadlock.
 Connect with gdb to the process to investigate what's going on.

 Also, you are running on arm.  How did you compiled kamailio: native or
 cross.
 When you run kamailio on an embedded system, you need to check if you
 have enough memory.  If you start running out of memory, the OS may
 start killing processes randomly (check your OS logs).

 You definitely need to look at this issue from a broader prospective
 as it's not a regular x86 deployment with plenty of CPU and memory.
 Also tuning the config (by removing everything that you don't need
 might help with memory utilization).


 Regards,
 Ovidiu Sas

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Re: [SR-Users] Simple command to insert attributes to domain_attrs

[Ovidiu Sas]

Not all tables are supported by kamctl.
If you are looking for a simple way to provision your tables, you can
try the new provisioning module xhttp_pi:
http://kamailio.org/docs/modules/devel/modules/xhttp_pi

To enable the module in your config simply add the lines provided in
the example to your config:
http://kamailio.org/docs/modules/devel/modules/xhttp_pi#id2531339

When you compile and install the module, a sample framework will be
constructed with all the existing tables under:
/usr/local/share/kamailio/xhttp_pi/pi_framework.xml
Then you can edit the file by removing all the tables that are not
used and add new db commands/actions as you need.  A full example with
all the possible commands and features can be found here:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/pi_framework.xml
Or in the source tree: ./examples/pi_framework.xml

In the future, a new script will be available to generate the the
framework for tables defined in kamctlrc (same behavior as kamdbctl).


Regards,
Ovidiu Sas

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

On Wed, Jan 9, 2013 at 9:54 AM, Philippe Sultan
philippe.sul...@gmail.com wrote:
 Hi,

 I'm migrating from sip-router/Kamailio 3.1, SER flavour to the latest dev
 version, with MySQL support.

 I need to add attributes to domains, and therefore use the domain_attrs
 table. A handy command I used for that was ser_attr :
 ser_attr add domain=anydomain.voip attr=value

 Is there something similar I can use under the Kamailio flavour? kamctl does
 not seem to help here, unless I missed an option.

 Regards,

 Philippe

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Re: [SR-Users] Kamailio behind NAT - best practice

[Ovidiu Sas]

Hello Klauss,

I use record_route_preset for this kind of scenarios:
http://kamailio.org/docs/modules/3.3.x/modules_k/rr.html#id2550086
That was the main reason that I enhanced record_route_preset with the
second parameter (see the Note on string2).

I haven't tried your idea with two sockets.  Let us know if it's working.
If you need to use the same port on the internal and external
interface, you could add a new IP to the host and listen on two
sockets on the same port and force the socket when sending a request
out.
listen=udp:10.10.0.2
listen=udp:10.10.0.3 advertise pu.bl.ic.ip


Regards,
Ovidiu Sas


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

On Thu, Jan 3, 2013 at 5:11 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
 Hi!

 Up to now I could avoid Kamailio setups with Kamailio behind NAt. But this
 time I have to deal with it. That's why I want to ask what others did as
 best practice.

 The scenario is:


 Asterisk 1\   Kamailio+RTPPROXY
\ |10.10.0.2
 Asterisk n--\|
  |- FW --SIP-trunk--- ITSP
 Freeswitch 1/  10.10.0.1   public-IP
/
 Freeswitch n--/
10.10.0.x

 Kamailio and rtpproxy have a private IP. Internal communication uses private
 IPs, external communication uses a public IP which is NATed 1:1 to
 Kamailio's IP address. No registrations, just forwarding of messages.

 Using the global advertised_address setting with the public IP does not
 work, as there is also internal communication. Using
 set_advertised_address() is also cumbersome.

 So it seems, the easiest solution would be to use 2 sockets on Kamailio,
 e.g. port 5050 and port 5060. Then I could use the listen with dedicated
 advertised addresses:
 listen=udp:10.10.0.2:5050
 listen=udp:10.10.0.2:5060 advertise pu.bl.ic.ip:5060

 If I understand it correctly, this should solve all issues with
 Record-Routing and Via-headers.

 For RTP-Proxy it seems necessary to detect the direction of each message and
 set the IP address in rtpproxy_manage(,ip.add.re.ss) manually.

 Thus, it seems straight forward - or do I miss something? Any comments and
 practical experience?

 Thanks
 Klaus

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Re: [SR-Users] How to set variable externally

[Ovidiu Sas]

If you just want to control the debug level externally, take a look at
the debug parameter:
http://www.kamailio.org/wiki/cookbooks/3.3.x/core#debug
It can be controlled via sercmd (kamcmd in future versions).

If you want to play with global flags, take a look at cfgutils:
http://kamailio.org/docs/modules/3.3.x/modules_k/cfgutils.html
http://kamailio.org/docs/modules/3.3.x/modules_k/cfgutils.html#id2533439
http://kamailio.org/docs/modules/3.3.x/modules_k/cfgutils.html#id2494518
http://kamailio.org/docs/modules/3.3.x/modules_k/cfgutils.html#id2494559


Regards,
Ovidiu Sas

On Fri, Dec 21, 2012 at 7:17 AM, Mino Haluz mino.ha...@gmail.com wrote:
 Hi,

 I would like to set my custom different debug levels (with flag?) externally
 with kamctl command. So I neednt restart kamailio if I want to
 enable/disable debug.

 Which module should I use in that case?

 Thanks,
 Mino

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Re: [SR-Users] [sr-dev] kamailio-ser integration - status update

[Ovidiu Sas]

On Thu, Dec 20, 2012 at 11:04 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 On 12/20/12 4:13 PM, Ovidiu Sas wrote:

 On Thu, Dec 20, 2012 at 9:39 AM, Daniel-Constantin Mierla
 mico...@gmail.com wrote:

 - nathelper - some extra functionality, not sure if can be kept
 completely

 Maybe Andreas can look into this, as there is a lot of work going on
 with
 nathelper and the new rtpproxy anyways.

 I think Ovidiu Sas looked at it when he split the rtpproxy out in a
 dedicated module.

 IIRC, ping_contact whas the extra functionality in nathelper:

 http://sip-router.org/docbook/sip-router/branch/master/modules_s/nathelper/nathelper.html#ping_contact
 I don't know how widely used is this functionality.

 Maybe we should have a separate thread per module (in user mailing
 list to gather more imput) and see if it's worth merging the code or
 use only the k version.

 I re-cc-ed the thread to users in case someone has comments to it.



 Also, on a separate note, I saw the we have a few db2_[module].
 I think it would make sense for these modules to rename them into
 [module]_db[1|2].
 For example: ldap - we should have both versions under modules:
   - ldap_db1
   - ldap_db2
 Just a suggestion ...

 The type of the two ldap modules are different, modules_k/ldap is a
 connector to ldap server from configuration file, offering possibility to do
 ldap search queries from config.

 The former modules_s/ldap (now db2_ldap) is a DB API v2 implementation
 driver module, so it can be used as a replacement for db_mysql (for example)
 when using some modules (such as db2_ops). I prefixed with db2_ to indicate
 that is not implementing DB API v1.

I see now.  I failed to check the README file.  Now it's all clear.
Maybe we should have the module as db_ldap with a TODO item in the
README (about implementing db API v1).
I think it will be more clear.  And then we cam move ldap module from
modules_k to modules.

-ovidiu

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Re: [SR-Users] [sr-dev] kamailio-ser integration - db modules

[Ovidiu Sas]

On Thu, Dec 20, 2012 at 11:20 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 What's the difference between the db1 and db2 interface?

 It is not the case here, modules_k/ldap does not have any relation to
 database interface.


Any database module db_[DatabaseType] implements an API.
SER was using one API and opensips/kamailio a different API (and
therefore today we have version 1 and 2).
All db modules in the stable release are supporting both APIs.
The new db ldap module has support only for the SER API version.
The API is documented in each lib/srdb[1|]/*.h header files.

The ldap module from ser is completely different than ldap module from
kamailio (despite the fact that both are using the same name).
It's the same with the dialog module.  Check the README file for each.

Hope this brings a little bit of light :)


-ovidiu

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Re: [SR-Users] [sr-dev] New developer: Konstantin Mosesov

[Ovidiu Sas]

Hello Konstantin,

Welcome to the team!
Happy kamailio pythoning ez!

-ovidiu

On Mon, Dec 10, 2012 at 12:28 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
  Hello,

 I want to announce that a new developer got GIT write access to repository:
 Konstantin Mosesov - he has contributed patches to app_python and joins the
 team to help maintaining and developing this module.

 His git commit id is: ez

 My warm welcome and looking forward to future work within the project!

 Cheers,
 Daniel

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[SR-Users] kamailio vs.ser

[Ovidiu Sas]

Hello all,

By inspecting the source code, the only difference that I could see
between kamailio and ser flavours is that kamailio has support for the
tm:local-request.
Are there any constrains in having the tm:local-request present for ser flavour?
Does it make sense to continue to build two flavours?
My suggestion would be to focus one one single flavour and go forward
with it for the next release.
Comments?

Regards,
Ovidiu Sas

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

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Re: [SR-Users] Kamailio v3.3.2 Released

[Ovidiu Sas]

Hello all,

For those who like running kamailio on routers and/or other small
embedded systems, the latest kamailio stable is available for
download.
For more info, please check: http://www.nslu2-linux.org/wiki/Optware/HomePage
For a list of supported platforms, please check:
http://www.nslu2-linux.org/wiki/Optware/Platforms

Regards,
Ovidiu Sas

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

On Tue, Oct 16, 2012 at 11:00 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Hello,

 Kamailio SIP Server v3.3.2 stable release is out.

 This is a maintenance release of the latest stable branch, 3.3, that
 includes fixes since release of v3.3.1. There is no change to database
 schema or configuration language structure that you have to do on
 installations of v3.3.0 or v3.3.1. Deployments running previous v3.x.x
 versions are strongly recommended to be upgraded to v3.3.2.

 For more details about version 3.3.2 (including links and hints to download
 the tarball or from GIT repository), visit:

   * http://www.kamailio.org/w/2012/10/kamailio-v3-3-2-released/

 RPM, Debian/Ubuntu packages will be available soon as well.

 Cheers,
 Daniel

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Re: [SR-Users] Presence on Cisco/Linksys handsets

[Ovidiu Sas]

You need to load pua_dialoginfo and configure the server accordingly:
http://kamailio.org/docs/modules/devel/modules_k/pua_dialoginfo.html

Take a look at the overview section to have an idea about
notifications are triggered:
http://kamailio.org/docs/modules/devel/modules_k/presence_dialoginfo.html#id2544906


-ovidiu

On Thu, Nov 29, 2012 at 2:19 PM, Mark Boyce m...@darkorigins.com wrote:
 Hi Ovidiu

 I see the following;

 - Phone registers

 - Phone subscribes / Kamailio sends 200 OK

 - PBX immediately responds with a Notify / Phone sends 200 OK

 If I now make a call on  the phone which is being watched by the
 subscription Kamailio doesn't generate any notify.

 Is there something I need to drop in the INVITE logic to trigger the notify?

 At the moment I'm loading presence.so and presence_dialoginfo.so modules
 with a config which is pretty much the same as the shipped sample config.

 Cheers

 Mark
 --
 Mark Boyce



 On 29 Nov 2012, at 19:09, Ovidiu Sas wrote:

 Do you see a subscription from the phone?
 A SUBSCRIBE message being sent when the phone registers?

 -ovidiu

 On Thu, Nov 29, 2012 at 1:52 PM, Mark Boyce m...@darkorigins.com wrote:

 Hi All


 I'm slowly working out what goes where with this.  So far I have;


 The Linksys / Cisco SCA / Shared Call Appearance requires Andrews SCA module

 not the standard presence ones.  This is the feature on the phone which was

 generating the call-info events.


 With SCA turned off on the phones and Server Type set to RFC3265_4235 the

 phones are generating normal subscribe requests.


 Kamailio is logging the invites in the active_watchers table fine.



 The problem seems to be that nothing in the system actually generates the

 notify back to the phones when a call is made.


 From the docs it looks like presence_dialoginfo should be doing this but

 isn't.



 What have I missed?


 Cheers


 Mark

 --

 Mark Boyce




 On 27 Nov 2012, at 19:44, Andrew Mortensen wrote:


 I recently added an SCA module to the project. If you're willing to try the

 master branch, I'd appreciate hearing how things work for you.


 Here are instructions for building from the git repository:


 http://www.kamailio.org/wiki/install/devel/git


 Add sca to the include_modules value in the make command, e.g.:


 make FLAVOUR=kamailio include_modules=db_mysql sca


 Take a look at the documentation here (it's not up on kamailio.org yet):


 https://github.com/fitterhappier/sca


 Scroll down past the list of files to find the formatted README text.


 Best,

 andrew


 On Nov 27, 2012, at 12:32 PM, Mark Boyce m...@darkorigins.com wrote:


 Hi All,



 I've been trying to get some Cisco/Linksys SPA phones working with Kamailio

 (current stable). All seems to be ok apart from getting Presence/BLF/SCA

 working.



 The phones are set to


 Server_Type : RFC3265_4235



 Line Key 2 Extension : Disabled


 Line Key 2 Share Call Appearance : Private


 Line Key 2 Extended Function :

 fnc=blf+sd+cp;sub=1...@ourserver.com;ext=1...@ourserver.com



 Where 1001 is the other phone.



 The phone appears to be sending event type 'call-info' rather than dialog

 etc.



 Phone is being sent '489 Bad Event' and the error in the logs is : presence

 [subscribe.c:1007]: Unsupported event header field value call-info



 Whatever I set the Server Type to on the phones it still tries to send

 call-info events on subscribes





 Has anyone got Presence working with Cisco SPA handset?




 The module needed seems to be available on OpenSIPs


 http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html




 Thanks!



 Mark

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Re: [SR-Users] Shared Call Appearances module

[Ovidiu Sas]

On Tue, Nov 20, 2012 at 5:17 PM, Andrew Mortensen
admor...@isc.upenn.edu wrote:
 On Nov 20, 2012, at 3:43 PM, Ovidiu Sas o...@voipembedded.com wrote:
 Hello Andrew,

 First of all, thank you for sharing your work.
 I was following this thread and I have a couple of questions.  Why do
 you need to bind to the usrloc module? The subscription itself should
 be sufficient because if a phone will unregister, it will also
 unsubscribe,

 Yes, that's good point. However, if the phone goes off-line instead of 
 unregistering, the contact's registration will expire, and no unsubscribe 
 will take place. As I mentioned in my reply to Daniel, I'm most concerned 
 about catching this so I can release any appearances seized by the 
 expired/deleted contact, and notify other members of the group. I'm certainly 
 open to alternatives.

I deployed sca and I didn't need to rely on usrloc for clearing up the
stale appearances.  The call was monitored and based on that the stale
appearance was removed.
Let's assume that a phone just registered for 1 hour and a call is
made.  During the call, the power is lost.  If you wait for the
contact to expire, you will end up with a 1h stale appearance.  If you
monitor the call/dialog, the stale appearance can be removed much
faster.

Another issues that I'm seeing here is if the sca server is behind a
registrar, then this setup will not work (registrations are held on a
different server).  It would be nice to have a parameter to disable
usrloc binding.  I don't know if usrloc binding is mandatory or not.


 which leads me to the following question: why not add a
 new dedicated module for call-info presence and reuse the existing
 infrastructure from kamailio for handling subscriptions/presence.  In
 this case, your module should just push PUBLISH events to the presence
 server and the server will automatically send out notifications for
 subscribers that subscribed to call-info events.

 We've worked extensively with the existing presence  pua modules, albeit 
 primarily dialog;sla using OpenSIPS behind the proxy. (Thanks, Anca!)

 SCA's unusual entanglement with call processing made me hesitate about 
 building on top of the existing presence module. For a first pass, I also 
 felt working with an entirely distinct module gave me more control and 
 transparency during development of a very loosely documented event package, 
 especially as I became more familiar with the available API.

 I don't have any objection to revisiting design decisions, of course. I'm 
 sure the module will continue to evolve, and it would be nice to eliminate 
 redundancy if possible.

As I pointed in my previous e-mail, the sca logic can be kept in one
module and the presence built on top of the existing presence
infrastructure.  The sca module will just need to send PUBLISH
messages to the presence server.  I built a call-info extension for
presence in opensips (I was working with Anca at that point in time)
and my goal was (and still is) to port the changes here (but the
project was delayed due to other projects).


 Based on your README file, you inspect SIP requests/replies and based
 on the presence of the call-info header, you create call-info events.
 This is great for sharing the appearances between phones, but how do
 you perform the retrieval of a call that was put on hold?  Are you
 using a dedicated B2BUA behind kamailio?

 No B2BUAs are involved.

 The INVITE retrieving a call held by another member of an SCA group has a 
 particular set of characteristics: RURI, To and From URIs are all the SCA 
 AoR; new unconfirmed dialog (no to-tag); and a Call-Info header referring to 
 the index of the held call. The module detects this type of INVITE, looks up 
 the dialog associated with the information in the Call-Info header, and 
 injects a Replaces header with the dialog of the held call before relaying it 
 to the remote party. The remote party must support RFC3891. I've only worked 
 with Polycoms and Cisco gateways to this point, and both do support that RFC.

That is very clever. I really like your idea messing with Replaces
header.  I know exactly how the special INVITE retrieving a held call
looks like.  Only the server behind the proxy must support RFC3891.
For the phones it should be transparent.
I'm looking forward to test your implementation.


 I'm very interested in hearing from users with Cisco (and Snom and Aastra?) 
 SCA setups. I have no doubt they'll find bugs resulting from assumptions I've 
 made in the code because I've only tested with Polycom, not least the 
 dependency on RFC3891 support.

I tested mainly with Cisco and Linksys phones (and a few Polycom
phones) and everything works great.  I didn't had any particular
issues with stale appearances.
Sometimes, I had issues with CISCO phones connected over WiFi and not
receiving notifications and the stale appearance was present only on
that set.  A new call (made from any other set) cleared up the stale
appearance (the new

[SR-Users] NEW: new module: xhttp_pi

[Ovidiu Sas]

Hello all,


A new module is available in trunk.
It provides a web provisioning interface:
http://kamailio.org/docs/modules/devel/modules/xhttp_pi.html

A sample config example is provided in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=examples/xhttp.cfg;hb=refs/heads/master

An additional xml file is required for web config framework (what to
configure, where and how).
A sample xml file is provided in the source tree for dispatcher and
dialplan tables:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=examples/pi_framework.xml;hb=refs/heads/master

When kamailio is up and running, a web provisioning interface for
managing db records will be available.
No supplementary web server needs to be installed in order to manage
the db records.
Also, it provides a tight control over what can be modified and how.

Testing and feedback is appreciated.


Regards,
Ovidiu Sas

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

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Re: [SR-Users] has_sdp()??

[Ovidiu Sas]

Making the rtpproxy user friendly would be the right approach here.
Then based on return code, the script admin should be able to handle
all scenarios.

-ovidiu

On Wed, Oct 17, 2012 at 12:09 PM, Juha Heinanen j...@tutpro.com wrote:
 Ovidiu Sas writes:

 That could be relatively easy to implement because the sdp parser is
 able to handle SDP in a multipart/mixed body.

 ovidiu,

 do you mean a new textops function that tells if sdp application/sdp
 bodypart exists in the body or making rtpproxy functions user friendly?

 -- juha

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Re: [SR-Users] has_sdp()??

[Ovidiu Sas]

On the other hand, is_audio_on_hold():
http://kamailio.org/docs/modules/devel/modules_k/textops.html#id2523065
could be re-worked/re-named to provide media status.

-ovidiu

On Wed, Oct 17, 2012 at 12:16 PM, Ovidiu Sas o...@voipembedded.com wrote:
 Making the rtpproxy user friendly would be the right approach here.
 Then based on return code, the script admin should be able to handle
 all scenarios.

 -ovidiu

 On Wed, Oct 17, 2012 at 12:09 PM, Juha Heinanen j...@tutpro.com wrote:
 Ovidiu Sas writes:

 That could be relatively easy to implement because the sdp parser is
 able to handle SDP in a multipart/mixed body.

 ovidiu,

 do you mean a new textops function that tells if sdp application/sdp
 bodypart exists in the body or making rtpproxy functions user friendly?

 -- juha

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Re: [SR-Users] $fs syntax

[Ovidiu Sas]

IIRC, port is also optional.

-ovidiu

On Wed, Oct 17, 2012 at 11:27 AM, Juha Heinanen j...@tutpro.com wrote:
 Daniel-Constantin Mierla writes:

 the value returned by $fs is always with proto, afaik. When you assign
 something to it is just to identify the listening socket, i.e., the
 given string value is parsed and used to search in the list of local
 listen sockets and if something matching is found, then $fs is linked to
 that socket structure.

 I think the description is good in regard to its value, maybe would be
 good to add notes about assignment value.

 ok, i added a sentence to the description text as an attempt to clarify
 the issue:

 It is R/W variable (you can assign values to it directly in
 configuration file). Transport proto can be omitted when assigning
 value, in which case it is taken from destination URI of the message.

 -- juha

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Re: [SR-Users] has_sdp()??

[Ovidiu Sas]

Then parse_sdp needs to be updated to return explicit error codes :(

-ovidiu

On Wed, Oct 17, 2012 at 12:35 PM, Juha Heinanen j...@tutpro.com wrote:
 Ovidiu Sas writes:

 On the other hand, is_audio_on_hold():
 http://kamailio.org/docs/modules/devel/modules_k/textops.html#id2523065
 could be re-worked/re-named to provide media status.

 i looked at the code and it has this:

 if (0 == parse_sdp(msg)) {
...
 }
 return -1;

 the problem is that parse_sdp returns -1 if there is no sdp, but also
 when there is some error.

 -- juha

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Re: [SR-Users] $fs syntax

[Ovidiu Sas]

In this case, you will need to specify the port.
If no port is specified, then the default 5060 is assumed (I think).
If you listen on two interfaces with non standard ports, then not
setting the port in fs should not match any interface.

-ovidiu

On Wed, Oct 17, 2012 at 12:41 PM, Juha Heinanen j...@tutpro.com wrote:
 Ovidiu Sas writes:

 IIRC, port is also optional.

 lets say that you have two listening ports on the same ip, one that is
 used for external traffic and the other for internal.  if you leave port
 out when you force the socket, which socket it is using?

 -- juha

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Re: [SR-Users] has_sdp()??

[Ovidiu Sas]

I guess not many rtpproxy deployments are handling SIP traffic with
multipart/mixed body content and therefor has_body(application/sdp)
works just fine.
In the case that multipart/mixed body is present, blindly invoking
rtpproxy_offer for INVITE will work ok if SDP is present (and that's
the most used pattern: SDP in INVITE/200ok as opposed to 200ok/AC.

So ... yes, not all cases are covered, but it seems that most common
scenarios are covered.

-ovidiu

On Wed, Oct 17, 2012 at 12:57 PM, Juha Heinanen j...@tutpro.com wrote:
 Ovidiu Sas writes:

 Making the rtpproxy user friendly would be the right approach here.
 Then based on return code, the script admin should be able to handle
 all scenarios.

 i thought that using rtpproxy is the mainstream thing.  now it appears
 that it cannot be used with currently existing script functions (if we
 leave raw regex matching out) without getting Unable to parse sdp
 errors to syslog.

 -- juha

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Re: [SR-Users] no socket found for match second RR

[Ovidiu Sas]

Sorry Juha,

I forgot to reply to you that I added that warning in the code for the
same reason.
What do you mean by a parameter that disables looking for the socket?

-ovidiu

On Fri, Sep 21, 2012 at 9:28 AM, Juha Heinanen j...@tutpro.com wrote:
 Daniel-Constantin Mierla writes:

 the second is better in this case, because will avoid a loop through
 local sockets. Also, your case is very rare, the warning is good to spot
 if someone changed the uri in route headers.

 daniel,

 sorry about the noise.  there is already rr mod param

 enable_socket_mismatch_warning

 which can be used to globally turn off the warning.  it is at the moment
 good enough for me, but a parameter could be added later that also
 disables looking for the socket in cases when it is known not to be found.

 -- juha

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Re: [SR-Users] no socket found for match second RR

[Ovidiu Sas]

It would need to be per loose_route() call, because a server can have
mixed traffic (real sockets only and real and advertised).
For real sockets, you would want to run the checks.
For advertised, you would not want to run the checks.

Then in the config, you will need to track calls through advertised
addresses and call appropriately loose_route().
Because of this, I implemented the warning disable parameter.
It is not optimal either, because for traffic through real sockets the
warning should be printed.


Regards.
Ovidiu Sas

On Fri, Sep 21, 2012 at 10:18 AM, Juha Heinanen j...@tutpro.com wrote:
 Ovidiu Sas writes:

 I forgot to reply to you that I added that warning in the code for the
 same reason.
 What do you mean by a parameter that disables looking for the socket?

 hi ovidiu,

 the piece of code currently looks like this:

 if (enable_double_rr  is_2rr(puri.params)) {
 /* double route may occure due different IP and port, 
 so force as
  * send interface the one advertise in second Route */
 if 
 (parse_uri(rt-nameaddr.uri.s,rt-nameaddr.uri.len,puri)0) {
 LM_ERR(failed to parse the double route 
 URI\n);
 return RR_ERROR;
 }
 si = grep_sock_info( puri.host, puri.port_no, 
 puri.proto);
 if (si) {
 set_force_socket(_m, si);
 } else {
 if (enable_socket_mismatch_warning)f
 LM_WARN(no socket found for match 
 second RR\n);
 }

 when the disable socket check parameter would have been given either
 globally or per loose_route() call, then this part of the code would not
 be executed at all:

 si = grep_sock_info( puri.host, puri.port_no, 
 puri.proto);
 if (si) {
 set_force_socket(_m, si);
 } else {
 if (enable_socket_mismatch_warning)f
 LM_WARN(no socket found for match 
 second RR\n);
 }


 which would save some cpu cycles.

 -- juha

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Re: [SR-Users] $200 bounty

[Ovidiu Sas]

If you really want to hire someone to configure your kamailio server,
then your starting point should be here:
http://www.kamailio.org/w/business-directory/

Regards,
Ovidiu Sas

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

On Wed, Jun 20, 2012 at 1:47 PM, copycall d...@copycall.com wrote:
 hello,

 install and configure kamailio on a godaddy load-balanced cloud server and
 create a vpn with the customer site.

 the objective is to use the customer's sip provider, which requires a static
 ip address, with a dhcp connection.

 please contact me off-list if you are interested.

 d...@copycall.com

 thank you,
 dave

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Re: [SR-Users] [SIREMIS] send MI Commands to multiple kamailio servers

[Ovidiu Sas]

The example that is provided in the README file is all you need to get
the web management interface up and running.
After that, you can access through the web interface all available rpc
commands.  The syntax is similar with sercmd or kamctl.
If you experience any issues, please report them on the mailing list.

Regards,
Ovidiu Sas

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

On Tue, May 15, 2012 at 12:49 AM, SamyGo govoi...@gmail.com wrote:
 Thanks to both of you Elena and  Ovidius - Since I'm not a php programmer so
 I can't promise a patch at this stage, however, I am planning on spending
 time on it and, like Elena said, can write somewhat dirty coding to dispatch
 the MI commands to as many remote servers as defined and then the replies
 be displayed in order received or something.

 Ovidius I do know about these html,xml rpc modules but I'm kind of person
 who needs some practical examples to understand how things work. I've kept
 this module in memory and if I find any understandable example on
 mailing-list or on internet I will definitely give this a try. Can you
 redirect me to jump-start on this one! current sample code is not enough for
 me to get going.

 BR,
 Sammy


 On Mon, May 14, 2012 at 10:53 PM, Ovidiu Sas o...@voipembedded.com wrote:

 For running mi commands on remote servers you could consider using the
 new xhttp_rpc module:
 http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html

 Regards,
 Ovidiu Sas

 On Mon, May 14, 2012 at 1:45 PM, Elena-Ramona Modroiu ram...@asipto.com
 wrote:
  Hi,
 
  current version of siremis does not support to send MI commands to
  multiple
  kamailio servers. One thing that has to be taken in consideration is
  handling the MI replies. One option, they can be appended one after the
  other and displayed in the text box of siremis page, separated by some
  delimiter with details about the instance that replied (ip
  address/port).
 
  If you submit the patch, it will be committed and released with the next
  version. In case you plan more consistent contributions, once you have
  them
  developed, I can grant write access to sourceforge.net git repository
  for
  siremis, so you will be able to commit yourself.
 
  Regards,
  Ramona
 
 
  On 5/14/12 8:33 AM, SamyGo wrote:
 
  Hello all,
 
  Following through the manual at
  : http://kb.asipto.com/siremis:install32x:mi-commands  I'm able to send
  MI
  commands on external/remote server But next hurdle is how can I define
  multiple remote servers i.e
 
  Remote name=remote0 address=127.0.0.1 port=8033/
  Remote name=remote1 address=192.168.2.156 port=8033/
  Remote name=remote2 address= 192.168.2.150 port=8033/
 
  Is there any already working configuration else I think I may need to
  start
  a surgery on siremisMICommands.php file and make it loop over the remote
  tag
  and disptach commands until loop ends.
  is there any MI-proxy sort of thing available !!?
 
  Regards,
  Sammy.
 
 
 
 
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 --
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 http://www.voipembedded.com

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Re: [SR-Users] [SIREMIS] send MI Commands to multiple kamailio servers

[Ovidiu Sas]

For running mi commands on remote servers you could consider using the
new xhttp_rpc module:
http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html

Regards,
Ovidiu Sas

On Mon, May 14, 2012 at 1:45 PM, Elena-Ramona Modroiu ram...@asipto.com wrote:
 Hi,

 current version of siremis does not support to send MI commands to multiple
 kamailio servers. One thing that has to be taken in consideration is
 handling the MI replies. One option, they can be appended one after the
 other and displayed in the text box of siremis page, separated by some
 delimiter with details about the instance that replied (ip address/port).

 If you submit the patch, it will be committed and released with the next
 version. In case you plan more consistent contributions, once you have them
 developed, I can grant write access to sourceforge.net git repository for
 siremis, so you will be able to commit yourself.

 Regards,
 Ramona


 On 5/14/12 8:33 AM, SamyGo wrote:

 Hello all,

 Following through the manual at
 : http://kb.asipto.com/siremis:install32x:mi-commands  I'm able to send MI
 commands on external/remote server But next hurdle is how can I define
 multiple remote servers i.e

 Remote name=remote0 address=127.0.0.1 port=8033/
 Remote name=remote1 address=192.168.2.156 port=8033/
 Remote name=remote2 address= 192.168.2.150 port=8033/

 Is there any already working configuration else I think I may need to start
 a surgery on siremisMICommands.php file and make it loop over the remote tag
 and disptach commands until loop ends.
 is there any MI-proxy sort of thing available !!?

 Regards,
 Sammy.




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Re: [SR-Users] Limiting registrations to known users without auth

[Ovidiu Sas]

db_text doesn't have sql capabilites (I think it should be mentioned
in the documentation).
Also, it should not crash too (this is a bug that needs to be
addressed - please open a bug and provide a backtrace).

If you don't want to use a truedb, you can try to use sqlite:
http://kamailio.org/docs/modules/stable/modules_k/db_sqlite.html
This should be compatible with sqlops.

Regards,
Ovidiu Sas

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

On Wed, Mar 7, 2012 at 11:18 AM, Pedro Antonio Vico Solano
pvsol...@amper.es wrote:
 Thanks for the clarification, Daniel.

 I've tried the 'sqlops' solution but there is something wrong. Is sqlops
 compatible with the db_text DB?

 I've tried:

 modparam(sqlops,sqlcon,ca=text:///etc/kamailio/dbtext)
 ...
 sql_query(ca, select * from uri, ra);
 xlog(rows: $dbr(ra=rows) cols: $dbr(ra=cols)\n);
 sql_result_free(ra);
 ...

 It gives a segmentation fault.

 Thanks  BR,
 Pedro




 De:        Daniel-Constantin Mierla mico...@gmail.com
 Para:        SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
 Users Mailing List sr-users@lists.sip-router.org
 cc:        Pedro Antonio Vico Solano pvsol...@amper.es
 Fecha:        02/03/2012 10:04
 Asunto:        Re: [SR-Users] Limiting registrations to known users without
 auth
 



 Hello,

 check_to() matches to see if the authenticated user is allowed to use the
 address in the To header, so it has to be called for authenticated requests.

 does_uri_exist() is checking to see if r-uri address is a valid local
 subscriber, but it does not have an option take it from To header.

 What you can do is to use sqlops module to do a query and check if the
 address (or user part of it) in To header is matching a record (username and
 eventually the domain) in subscriber table.

 Cheers,
 Daniel

 On 3/1/12 6:10 PM, Pedro Antonio Vico Solano wrote:
 Hello,

 I'm trying to restrict registrations based on the username/number but
 without authentication. I'm using uri_db module, the URI table and the
 check_to() function. But when a user tries to register Kamailio 3.1.5 says
 the following error:

 0(11832) ERROR: uri_db [checks.c:71]: No authorized credentials found (error
 in scripts)
 0(11832) ERROR: uri_db [checks.c:72]: Call {www,proxy}_authorize before
 calling check_* functions!

 I've read README from uri_db and seems to be possible doing it the way I do.
 I have the following configuration:

 modparam(usrloc|uri_db, db_url, text:///etc/kamailio/dbtext)
 modparam(uri_db, db_table, uri)
 modparam(uri_db, use_uri_table, 1)

 route{
        ...
        check_to()
        ...
 }

 am I doing right?

 Thanks  BR,
 Pedro
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Re: [SR-Users] [newbie] questions

[Ovidiu Sas]

You will need to break down your e-mail into several simpler questions
and you will get some useful replies.
Most of the things that you want to do are possible. To craft a proper
e-mail to all your questions in your original e-mail would take quite
some time ... and time is expensive for all.  I think that's all that
you need to read from all the replies that you got.

Now, to give you a hint on what are you trying to achieve, take a look
at forced socket PV:
http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables#forced_socket
This will put you on the right track in sending SIP request through a
specific interface.


Regards,
Ovidiu Sas

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

On Mon, Jan 30, 2012 at 7:13 PM, Me mojo1...@privatedemail.net wrote:

 It's not a smart-arse reply; it's sincere, earnest advice.

 Really?! Perhaps you could explain to me how exactly is the you should get
 a consultant comment on a routine set of questions I posted on a mailing
 list created for that very purpose - for Kamailio users like myself -
 anything other than a smart-arse reply?

 Does it answer any of my queries? No!
 Is it helping me in any way, shape or form? No (do you seriously think I
 haven't really thought of getting a consultant before posting my queries
 on this mailing list?)!
 Does it provide any insight or fresh ideas on either what I want to achieve
 or the difficulties I am facing, given the problems I described earlier? No!
 Does it contribute anything to the discussion on this mailing list, apart
 from wasting my own time and bandwidth so that I have to enlighten smart
 sparks like the previous poster as well as yourself? No!


  There's only so much of a massive conceptual nexus that people can
 reasonably traverse on a mailing list.  For the most part, mailing lists
 exist to answer specific questions, not provide broad, fundamental guidance
 or extensive pedagogical surveys.


 I didn't ask for fundamental guidance or pedagogical surveys. I asked to
 be pointed in the right direction where I could get (further) help. I did
 not force anyone to answer, let alone come up with gems like the one I
 commented on in my previous post.

 I'd say it again though - this thread is for me to seek
 answers/advice/help/guidance - if you, the previous poster, or anyone else
 for that matter is unwilling or unable to provide one, then just move along
 - there is nothing to see here. Smart-arse comments like get a consultant
 isn't what I am looking for, nor is the reason for starting this thread on
 this mailing list.


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Re: [SR-Users] Kamailio : Listening on multiple ports behind NAT

[Ovidiu Sas]

When you route through usrloc, there is a PV that should be set - forced socket:
http://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#forced_socket
You can check the socket via 'kamctl ul show' command.
If the PV is not populated, check the send attributes:
http://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#send_address_attributes
Based on that, you should know through which interface the INVITE
should be sent and therefore you should be able to set the proper
Record-Route header.

Regards,
Ovidiu Sas

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


On Tue, Jan 17, 2012 at 3:01 PM, Reda Aouad reda.ao...@gmail.com wrote:
 I just tried the record_route_advertised_address(public_ip).
 It doesn't add the port number of the outgoing socket.

 Any suggestions?

 RA



 On Mon, Jan 16, 2012 at 15:57, Reda Aouad reda.ao...@gmail.com wrote:

 I know about record_route_advertised_address(ip:port) function. If I
 understood correctly, it inserts a top-most RR header with the public IP if
 double RR is enabled. But that doesn't solve the multiple ports problem. I
 would get in the SIP header :

 Record-Route: public_ip;lr=on
 Record-Route: private_ip:port;lr=on

 If user B sees the first Record-Route header, it remembers port=5060 for
 future requests.
 I cannot manually set the port in the config file since it depends on
 which port user B is registered, which I don't have a way to find it.

 RA



 On Mon, Jan 16, 2012 at 15:51, Andrew Pogrebennyk
 apogreben...@sipwise.com wrote:

 Hi,

 On 01/16/2012 03:41 PM, Reda Aouad wrote:
  I suggest that the function record_route( ) takes a public IP address
  as
  a parameter, still doing what it does (correct record routing and
  cookie
  addition did=xxx and loose route lr=on), but only replacing the private
  IP address on which Kamailio listens with a public IP address. Or that
  the record_route( ) function uses the advertised_address to construct
  the RR header.

 maybe you are looking for the function record_route_advertised_address()
 which is available in git master:
 http://web.archiveorange.com/archive/v/jZFTGE0yjPqCTTcAkzuf


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Re: [SR-Users] usrloc, timer process and cache cleanup

[Ovidiu Sas]

The replace solution will mask the real issue.
The flag that is in the usrloc should switch between update or insert
and that is the real fix.

Regards,
Ovidiu Sas

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

On Thu, Dec 22, 2011 at 12:58 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Hello,


 On 12/22/11 6:19 PM, Andreas Granig wrote:

 Got the cause of the issue.

 What happens is that there's an AOR which registers ever 120 seconds.
 For some reason, the location entry is in usrloc cache, but not in db.
 What happens now is that usrloc tries an update query in the db,
 because it still assumes that the entry is there, which obviously fails.

 If you remove the entry from usrloc (kamctl ul rm aor), then on the
 next re-registration it's both inserted into the cache and into the db.

 Wondering how it could happen to get out of sync, and how we could
 improve this. Maybe using a replace into instead of update, at least
 for mysql? Suggestions?

 is the timer interval parameter of usrloc higher than 120sec?

 http://kamailio.org/docs/modules/3.2.x/modules_k/usrloc.html#id2494575

 IIRC, there should be anyhow a flag to mark if the record is in db or not,
 and based on that do insert or update, maybe something is lost there. If you
 do 'kamctl ul show __aor__', what are the values for flags fields?

 Another option, perhaps more portable, but with two db hits is: update and
 if fails then insert -- considering that these should be corner cases, maybe
 the performance is not affected much. A blended version is even better, if
 the db driver supports replace, do replace instead of update (I don't know
 if replace is faster/slower than update).

 Cheers,
 Daniel


 Andreas

 On 12/22/2011 05:12 PM, Andreas Granig wrote:

 Hi,

 Could you please tell me which of the three timer processes (timer,
 slow timer or timer nh) is responsible for cleaning up the internal
 usrloc cache?

 Looks like every now and then the cleanup of the internal location cache
 is starting to fail. Funny thing is that expired locations are removed
 from the mysql backend, but not from the internal cache. We're running
 kamailio 3.1.5, are there any known issues fixed since that version?

 In the meanwhile we're trying to pin the issue down, but maybe someone
 has a clue...

 Andreas

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Re: [SR-Users] usrloc, timer process and cache cleanup

[Ovidiu Sas]

 Right, that has to be done, but there are some cases when db can become
 inconsistent, due to database unavailability, and then some trick have to be
 done at db layer, example:

 - db is unavailable, phone unregisters, contact deleted from memory but not
 from database
 - phone register again, usrloc will try insert and will fail - in this case
 it should be update if insert fails (or replace)

If the phone registers again, it should be a brand new entry
(different Call-ID, CSeq and so on).
The leftover entry on the db will be cleanup on a server restart.

 The other way around could happen when mistakenly deleting/changing records 
 in db, which should not happen, but Murphy says opposite.
If someone is messing with the db, kamailio shouldn't try to correct
admin mistakes.

I think that the replace solution should be a last resort.
If implemented, should be configurable.  I would rather see the
original issue instead of being masked and let it trigger other issues
later on which would be more difficult to debug.


Regards,
Ovidiu Sas

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Re: [SR-Users] using xhttp_rpc to send MESSAGE to SIP user from web server

[Ovidiu Sas]

It is rather difficult to properly craft SIP messages in a single line
(to properly pass all parameters).
Therefore, the xhttp_rpc module does not support this kind of functionality.
Asynchronous commands are not implemented by the xhttp_rpc module:
http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html#id2521422

It is better to use the xmlrpc module for this kind of functionality.
http://kamailio.org/docs/modules/stable/modules/xmlrpc.html

Regards,
Ovidiu Sas

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

On Sat, Dec 17, 2011 at 10:37 AM, Krishna Kurapati kkura...@gmail.com wrote:
 Dear list members,

 I have seen examples of using mi_fifo and mi_xmlrpc modules to send a
 MESSAGE or INVITE
 from a webserver.

 Since I am using xhttp to control presence policies from webserver, I would
 like to use xhttp_rpc
 module to send MESSAGE and INVITE from webserver.

 Are there any examples of using xhttp_rpc to achieve the function?

 Thanks

 Krish Kura

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Re: [SR-Users] Problem to verify contact header

[Ovidiu Sas]

You would want to have:
if(_msg-contact!= NULL  _msg-contact-body.s!= NULL){


Regards,
Ovidiu Sas

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

On Wed, Dec 7, 2011 at 11:24 AM, Bruno Bresciani
bruno.bresci...@gmail.com wrote:
 Hi All,

 Kamailio generate a core at line below

 if(_msg-contact!= NULL || _msg-contact-body.s!= NULL){

 _msg is a sip_msg struct that my module receive from kamailio. I want verify
 if on that request messagem have a contact header, but a core is being
 generated when contact header isn't present on message.

 Someone knows why this is happening?

 Cheers

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Re: [SR-Users] rate limit module

[Ovidiu Sas]

Take a look at the rl_set_dbg command:
http://www.kamailio.org/docs/modules/1.5.x/ratelimit.html#id2506196
Enable debug mode and you will see in the logs what ratelimit is doing
internally.

Regards,
Ovidiu Sas


--
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http://www.voipembedded.com

On Fri, Dec 2, 2011 at 2:56 PM, Fabian Borot fbo...@hotmail.com wrote:

 Hello,

 I am trying to use the rate limit module using Kamailio 1.5.2. I feel that I 
 got it but I would
  like some pointers and recommendations.

 These are my settings: (this is a lab of course), I used 1 on the 
 timer_interval because I am
  generating the calls manually and wanted to see it in action quickly:


 #  ratelimit ---
 modparam(ratelimit, timer_interval, 1)
 modparam(ratelimit, reply_code, 506)
 modparam(ratelimit, reply_reason, Rejecting due to high load)
 modparam(ratelimit, queue, 0:INVITE)
 modparam(ratelimit, pipe, 0:TAILDROP:1)


  then in the route section:


     if (method==INVITE) {
     xlog(L_INFO,mylog: RL found INVITE.\n);
     if (!rl_check()) {
     xlog(L_INFO,mylog: RL dropped message.\n);
     rl_drop();
     exit;
     };
     xlog(L_INFO,mylog: RL found INVITE but did not drop 
 it.\n);
     };


 The TAILDROP algorithm seems to work better than the RED, based on what I 
 expected of course (with 1 sec timer interval and 1 calls/sec on the pipe).
 Making manual calls one right after the other almost always triggered the 
 protecting when there was another call on the same second.


 But these lines (1.6.3. pipe) on the doc got me kind of confused:

  When specifying a limit, the unit depends on the algorithm used and
 doesn't need to be specified also (eg, for TAILDROP or RED, limit means 
 packets/sec, whereas with the FEEDBACK
 algorithm, it means [CPU] load factor).

 For these 2 lines below, does this mean that the interval =10 will be 
 overridden by the 100 calls/sec on the TAILDROP algorithm?

 modparam(ratelimit, timer_interval, 10)
 modparam(ratelimit, pipe, 0:TAILDROP:100)

 I made a quick test and with timer_interval = 10 and TAILDROP:1, it looks 
 like the protection kicks in almost every 10 secs: (Dec 2 19:44:07  and Dec  
 2 19:43:57),

  tail -f proxy.log | grep RL
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:54 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:54 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:55 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:55 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL dropped message.
 Dec  2 19:43:58 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:58 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:59 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:44:01 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:44:01 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE but 
 did not drop it.
 Dec  2 19:44:02 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:44:02

Re: [SR-Users] rate limit module

[Ovidiu Sas]

It is allow up to 50 calls/30sec.
You can verify that by observing the debug logs.

Regards,
Ovidiu Sas

On Fri, Dec 2, 2011 at 4:11 PM, Fabian Borot fbo...@hotmail.com wrote:

 thank you Ovidiu, the debug it really helps.
 please help me out with these questions though:

 modparam(ratelimit, timer_interval, 30)
  modparam(ratelimit, pipe, 0:TAILDROP:50)

 does this mean allow up to 50 calls/sec or allow up to 50 calls/30sec

 or is the logic similar to:

 every 30 seconds get a count of messages, if algorithm's threshold is set to
 50calls/sec, then after the timer elapses (30 secs) if there are more than
 50 * 30 calls/transactions during the past 30 secs then drop the next call?

  with these settings should be no more than 50 calls every second

 modparam(ratelimit, timer_interval, 1)
  modparam(ratelimit, pipe, 0:TAILDROP:50)

 I really appreciate your assitance here.
 txs a lot in advance
 fborot

 
 From: fbo...@hotmail.com
 To: us...@lists.kamailio.org
 Subject: rate limit module
 Date: Fri, 2 Dec 2011 14:56:28 -0500


 Hello,

 I am trying to use the rate limit module using Kamailio 1.5.2. I feel that I
 got it but I would
  like some pointers and recommendations.

 These are my settings: (this is a lab of course), I used 1 on the
 timer_interval because I am
  generating the calls manually and wanted to see it in action quickly:


 #  ratelimit ---
 modparam(ratelimit, timer_interval, 1)
 modparam(ratelimit, reply_code, 506)
 modparam(ratelimit, reply_reason, Rejecting due to high load)
 modparam(ratelimit, queue, 0:INVITE)
 modparam(ratelimit, pipe, 0:TAILDROP:1)


  then in the route section:


     if (method==INVITE) {
     xlog(L_INFO,mylog: RL found INVITE.\n);
     if (!rl_check()) {
     xlog(L_INFO,mylog: RL dropped message.\n);
     rl_drop();
     exit;
     };
     xlog(L_INFO,mylog: RL found INVITE but did not drop
 it.\n);
     };


 The TAILDROP algorithm seems to work better than the RED, based on what I
 expected of course (with 1 sec timer interval and 1 calls/sec on the pipe).
 Making manual calls one right after the other almost always triggered the
 protecting when there was another call on the same second.


 But these lines (1.6.3. pipe) on the doc got me kind of confused:

  When specifying a limit, the unit depends on the algorithm used and
 doesn't need to be specified also (eg, for TAILDROP or RED, limit means
 packets/sec, whereas with the FEEDBACK
 algorithm, it means [CPU] load factor).

 For these 2 lines below, does this mean that the interval =10 will be
 overridden by the 100 calls/sec on the TAILDROP algorithm?

 modparam(ratelimit, timer_interval, 10)
 modparam(ratelimit, pipe, 0:TAILDROP:100)

 I made a quick test and with timer_interval = 10 and TAILDROP:1, it looks
 like the protection kicks in almost every 10 secs: (Dec 2 19:44:07  and Dec
 2 19:43:57),

  tail -f proxy.log | grep RL
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:53 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:54 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:54 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:55 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:55 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:56 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17450]: mylog: RL found INVITE
 but did not drop it.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found INVITE.
 Dec  2 19:43:57 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL dropped
 message.
 Dec  2 19:43:58 AW-ATL-PROXY-01 ./kamailio[17449]: mylog: RL found

Re: [SR-Users] undefined symbol: dprint_crit

[Ovidiu Sas]

Probably you are having a linker issue.
For which platform are you trying to build?
In the trunk, there are a couple of fixes for cross compiling but the
work is not 100% completed.

Regards,
Ovidiu Sas

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

On Mon, Nov 28, 2011 at 1:55 PM, Ämin Baumeler ze...@zitune.ch wrote:
 Hi everybody

 I cross compiled kamailio for ARM. However, when I try to start kamailio on 
 the ARM machine, it tells me that there are a lot of undefined symbols. E.g. 
 libkcore.so does not find the symbol dprint_crit. Having a look at 
 libkcore.so:
 $ nm -C -D lib/kcore/libkcore.so|grep dprint
         U dprint_crit

 All deployed .so files do not contain dprint_crit. However I found 
 dprint_crit in an object file, that was used for compilation. Namely in 
 dprint.o:
 $ nm -C  dprint.o |grep print
  B dprint_crit


 I used following commands to cross compile kamailio:
 export GCC_ARM_HOME=/opt/arm-2009q1/arm-none-linux-gnueabi/
 export TOOL_PREFIX=/opt/arm-2009q1/bin/arm-none-linux-gnueabi
 export CXX=$TOOL_PREFIX-g++
 export AR=$TOOL_PREFIX-ar
 export RANLIB=$TOOL_PREFIX-ranlib
 export CC=$TOOL_PREFIX-gcc
 export LD=$TOOL_PREFIX-ld
 export ARM_TARGET_LIB=/opt/arm-2009q1/arm-none-linux-gnueabi/libc
 make proper
 make cfg FLAVOUR=kamailio PREFIX=/root/kamailio/local include_modules=auth 
 auth_db db_text registrar ARCH=arm TARGET=arm-none-linux-gnueabi
 make all  make install

 Thanks
 Amin

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Re: [SR-Users] parse 503 message

[Ovidiu Sas]

You need to use t_check_status inside a failure_route.Take a look at
the default config to see how a failure_route is enabled.
Regards,Ovidiu Sas
On Tue, Nov 22, 2011 at 3:20 PM, Robert R rob1...@gmail.com wrote:
 Hi,

 How can I set a filter for receiving 5xx messages, i.e. how can I parse 503
 messages received by the proxy?

 I have tried the following and none works:

 t_check_status(503)
 is_method(503)

 Thanks,
 R


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Re: [SR-Users] [Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file

[Ovidiu Sas]

I ran into a similar issue and while searching for a solution, I found
a few threads describing the same problem but no solution.It turns out
that the sox command must have the proper arguments in order to
properly decode raw files.Feeding sox with an u-law stream and asking
to be decoded as an a-law stream will create garbled sound.
I posted a more descriptive procedure on how to extract audio from
rtpproxy pcap files
here:http://voipembedded.wordpress.com/extracting-audio-from-calls-recorded-with-rtpproxy/
Regards,Ovidiu Sas
On Thu, Jan 14, 2010 at 5:37 PM, Vikram Ragukumar
vraguku...@signalogic.com wrote:
 Hello,

 I would also like to mention that the rtpproxy recorded .rtp files were
 generated using the following command

 rtpproxy -l listen_ip_address -s udp:127.0.0.1:7722 -a -P -F -r
 record_directory_path -S spool_directory_path

 where -P indicates that files would be recorded in the pcap format.

 Thanks and Regards,
 Vikram.

 Vikram Ragukumar wrote:

 Carsten,

 Thank you for your reply.

 Carsten Bock wrote:

 Hi,

 1)
 Did you try to post your problem on the RTP-Proxy-Users' List?
 http://lists.rtpproxy.org/mailman/listinfo/users
 Probably, you might get more help there

 Yes i have posted on rtpproxy users list.

 2)
 Did you try to extract the Audio with Wireshark? If Wireshark can play
 the Audio (and i assume it is correctly implemented), then the recorded
 stream as such is correct. Then you can check, if the bug lies in rtpbreak
 (or as a next step: sox). Just to limit the possible sources of the
 problem...

 I have not tried Wireshark to extract the audio. I have been using
 rtpbreak to generate the .raw file. Subsequent to which i use sox to convert
 the .raw to a .wav file. By importing the the .raw file into Hypersignal
 software, we found that the .raw file doesnt entirely seem to be composed of
 speech samples, so there could be a problem at the rtpbreak step.

 This is the command i used to convert rtpproxy's capture file to .raw
 format

 rtpbreak -W -r capturefile.rtp

 Am i missing something here ?

 Thanks and Regards,
 Vikram.

 Carsten

 2010/1/14 Vikram Ragukumar vraguku...@signalogic.com
 mailto:vraguku...@signalogic.com

    Hello,

    An update,

    I tried using sox to convert the two .raw files into 2 mono channel
    wave files. The command line i used is below :
    sox -r 8k -b -c 1 -u rtp.0.0.raw rtp0.wav
    sox -r 8k -b -c 1 -u rtp.1.0.raw rtp1.wav

    When i listen to the .wav files, i hear speech but it is buried in a
    lot of noise. During blank periods (periods of no speech) there is a
    constant volume high pitched noise. Also during periods of speech,
    there seems to be bursts of noise in the background.

    The other engineer i work with and i, think that it is possibly
    because non-speech data is being interpreted as speech.

    What switch options should i change while invoking sox from the
    command line to get rid of the noise?

    Thanks and Regards,
    Vikram.



    Vikram Ragukumar wrote:

        Hello,

        I used Kamailio+rtpproxy to record a session and rtpproxy
        outputs the following files

        long_file_name.a.rtp, long_file_name.a.rtcp,
        long_file_name.o.rtp, long_file_name.o.rtcp

        http://www.rtpproxy.org/wiki/RTPproxy/FAQ
         From the Rtpproxy FAQ above, i tried to extract the audio using
        rtpbreak and sox.

        rtpbreak -W -r long_file_name.a.rtp
        rtpbreak -W -r long_file_name.o.rtp

        The above commands generate rtp.0.0.raw, rtp.1.0.raw.

        Then when i run sox using
        sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t
        wavpcm -s out.wav i get the following errors :

        sox: invalid option -- -
        sox: -c must be given a number

        Is there a switch/anything else that i am missing ?

        Thanks in advance,
        Regards,
        Vikram.




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 --
 Carsten Bock
 Schomburgstr. 80
 22767 Hamburg
 Germany

 Mobile +49 179 2021244
 Home +49 40 34927217
 Fax +49 40 34927218
 mailto:cars...@bock.info mailto:cars...@bock.info

 Mike Ditka http://www.brainyquote.com/quotes/authors/m/mike_ditka.html
  - If God had wanted man to play soccer, he wouldn't have given us arms.


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Re: [SR-Users] Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?

[Ovidiu Sas]

Take a look at path module:
http://kamailio.org/docs/modules/stable/modules_k/path.html
Check if your softswitch has support for path.  That will be the
simpler approach.

Regards,
Ovidiu Sas

On Mon, Nov 14, 2011 at 11:21 AM, edson.gomes.leme
edson.gomes.l...@uol.com.br wrote:
 Hi

 I am using the following tutorial: “Kamailio 3.1.x and FreeSWITCH 1.0.6+ for
 Media Services and SBC”
 Site: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc

 I am trying to configure SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) for
 Upper Registration. This feature forwards REGISTER requests sent from
 clients to another server. Example:

 Client phone  SBC ( Kamailio 3.1.x and FreeSWITCH 1.0.6)
 -- mysipswitch

 How to enable Upper Registration no Kamailio?
 Any idea?

 Best regards,
 Edson Gomes Leme


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[SR-Users] New module: xhttp_rpc

[Ovidiu Sas]

Hello all,


A new module providing a web interface to kamailio/sip-router RPC
interface is available in trunk:
http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html
http://sip-router.org/docbook/sip-router/branch/master/modules/xhttp_rpc/xhttp_rpc.html
The module is using the embedded web server provided by xhttp module:
http://kamailio.org/docs/modules/devel/modules/xhttp.html

A simple config that will enable the web interface is provide on the
README file.
Simply point your browser to
http://server_IP:TCP_port/xhttp_rpc_root and start browsing
available RPC commands.
For more info about the module please see the README file.


Regards,
Ovidiu Sas

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

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Re: [SR-Users] Directory missing compiling dbtext module

[Ovidiu Sas]

If modules_s/dbtext is no longer maintained, then it should be dropped
and db_text from modules_k should be moved to modules.

Regards,
Ovidiu Sas

On Sat, Nov 5, 2011 at 7:18 AM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
 Hello,

 I am not sure the dbtext module from modules_s is actually maintained. You
 should use the one from modules_k/ which is named db_text -- it should be
 the same, but coming from the kamailio branch and there were people using it
 lately.

 Also, maybe db_sqlite is another option for your need, it is also in
 modules_k/ folder. Note that some modules from modules_s may not work, if
 they require DB API v2, modules_k uses DB API v1.

 Cheers,
 Daniel

 On 11/4/11 12:43 PM, Pedro Antonio Vico Solano wrote:

 Hello,

 I have an issue compiling dbtext modude (SER flavour) on SIP-Router 3.1.2.
 Seems that /db directory is missing.

 CC (/opt/eldk/usr/bin/ppc_82xx-gcc) [M dbtext.so]               dbt_api.o
 In file included from dbt_api.c:42:
 dbt_res.h:41:28: ../../db/db_op.h: No such file or directory
 In file included from dbt_res.h:44,
                  from dbt_api.c:42:
 dbt_lib.h:42:29: ../../db/db_val.h: No such file or directory
 In file included from dbt_res.h:44,
                  from dbt_api.c:42:
 dbt_lib.h:73: error: parse error before dbt_val_t

 I use the tarball ser-3.1.2_src_2011-03-31_2fe4d6.tar.gz but cannot find it
 on GIT either.

 Can anyone help me?

 BR,
 Pedro
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 --
 Daniel-Constantin Mierla -- http://www.asipto.com
 Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
 http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Force signalling??

[Ovidiu Sas]

You really need to read RFC 3261 (see the loose-routing behaviour -
Route and Record-Route headers).

Regards,
Ovidiu Sas

On Thu, Oct 20, 2011 at 9:29 PM, Skyler skchopper...@gmail.com wrote:
 Hi,

 On Thu, 2011-10-20 at 17:58 -0400, Alex Balashov wrote:
 This is where record_route() comes in - see 'rr' module docs.


  Do you mean I need to store the original $ru $fu $tu  $ct info into rr
 and restore those within an on_reply_route?

  or did I misunderstand?

 Skyler


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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways

[Ovidiu Sas]

It seems that you are on the right path now.

Regards,
Ovidiu Sas

On Wed, Sep 7, 2011 at 5:22 AM, Sarat C. Vemuri sarat.vem...@fthco.com wrote:
 Ovidiu,

 Thanks for your time.  The fixes I pulled in is the latest rr module only. 
 I didn't see anything in that diff to make this work.  Could you elaborate a 
 little?

 However, I was able to get Outbound working properly by adding ;r2=on to  
 the record_route_preset IPs.  In going through the source code, I noticed 
 that this is what regular record_route uses to enable double rr and it would 
 automatically take both headers out if it sees that.  That seems to have 
 worked.  Any pitfals with this?

 I am still working on Inbound.  For some reason my carrier GW keeps 
 resending invites even after receiving ACK.  I need to see if it is an issue 
 with the carrier.

 Thanks
 SV.


 -Original Message-
 From: Ovidiu Sas [mailto:o...@voipembedded.com]
 Sent: Tuesday, September 06, 2011 9:50 AM
 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
 Mailing List
 Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier 
 gateways

 It seems that something is miss configured on your server.  The fixes that I 
 made in the trunk (and you pulled in your local 3.1 repo) were designed to 
 handle the scenario that you are trying to implement.
 The ACK should be handled properly and routed to the upstream carrier 
 (following the same path as the initial INVITE).


 Regards,
 Ovidiu Sas

 On Mon, Sep 5, 2011 at 5:07 PM, Sarat C. Vemuri sarat.vem...@fthco.com 
 wrote:
 Again, I apologize for this clumsy way of replying.

 Ovidiu,

 Thanks for the pointer on set_advertised_address.  I had to patch rtpproxy 
 module (and rr module for the two parameters to request_route_preset) since 
 I am running 3.1.

 However, I still have a problem with ACKs after following what you suggested.

 INVITE from Internal to Carrier routes properly (two Request-Route headers, 
 one internal IP and other public IP).  On 200 OK, the carrier GW properly 
 copies the route set in to Route header.  Now the route contains two 
 entries, the public IP and the private IP of Kamailio.  The Internal UAC 
 then sends the ACK back to Kamailio.  Everything is fine till this point.  
 Now, Kamailio removes the top entry, which is the private IP and then 
 promptly sends the ACK to the public IP of itself!.  Of course, that doesn't 
 go anywhere.

 How do I remove the public IP entry from the route set before forwarding 
 the reply to Internal UAC?  Is there another way to deal with this?  I've 
 tried to set an alias= core parameter with the public IP, but doesn't seem 
 to have any effect. The public IP is not reachable from internal network.

 Thanks for your help

 SV.
 --

 Message: 3
 Date: Sat, 3 Sep 2011 16:44:22 -0700
 From: Ovidiu Sas o...@voipembedded.com
 Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple
        carrier gateways - some via Firewall/NAT
 To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
        Users   Mailing List sr-users@lists.sip-router.org
 Message-ID:

 CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=lyrzb1wfx3on32...@mail.gmail.com
 Content-Type: text/plain; charset=windows-1252

 It is feasable to what you want: kamailio behind NAT proxying traffic
 from/to public internet to/from private network.
 You will need to properly craft the INVITE and use proper record route 
 headers.
 Use set_advertised_address when needed:
 http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertis
 ed_address Also, use record_route_preset (note that there are two
 parameters):
 http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566

 That should do it.  You don't need any patches for rtpproxy.
 Just use force_rtpp_proxy (and force the IP address):
 http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034


 Note: Make sure that you understand how in-dialog requests are routed
 by a proxy in order to know how to properly handle the initial INVITE.


 Regards,
 Ovidiu Sas

 On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri sarat.vem...@fthco.com 
 wrote:
 We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy?
 where it acts as the front for various carrier gateways so that
 internal UACs and UASs are unaware of the carrier gateways.



 Let me try to present a clear picture of our setup.

 1.?? Kamailio has several NICs (physical or vlan).? Each on a
 different subnet. One subnet is internal/has routes for internal.?
 Other subnets are private connections to carriers or a ?route to public 
 Internet.

 2.?? All of these subnets are non-routable from Internet. In
 addition , the carrier private connections are not routable internally.

 3.?? Connection to public internet is via a FW/NAT (one-to-one
 NAT), which maps to one of the interfaces.

 4.?? All internal? UAC/UAS connect on the internal subnet.

 5.?? We are using RTPProxy

Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.

[Ovidiu Sas]

only trunk

On Thu, Jul 7, 2011 at 2:17 AM, MingHon gming...@gmail.com wrote:
 Hi,
 Thanks!
 do you think willl any other version will work?
 like version 3.0.x?
 Cheers,
 MingHon


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