[Kamailio-Users] merging users mailing list
Hello, I am starting the merging process of users mailing list. Please ignore any notifications you get, if you get them is by mistake in my config -- none of subscribed users will be affected, the work is done for testing on a temporary new list, therefore everyone is safe. Sorry for any inconvenience! I will announce when it is finished. Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] sr-docs mailing list
Hello, I renamed kamailio-d...@lists.kamailio.org to sr-d...@lists.sip-router.org. Info about it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-docs Lately, several discussions touched the documentation aspects, in a need for a better structuring and updates for the new context of sip router project. The old mailing list was active for a while, but with the start of sip router project was ignored because of more focus on integration work. I hope some of you will join, help to build new documentation and spot mistakes in existing versions. Volunteers that want to lead various documentation efforts are more than welcome! Just say what you need... Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Kamailio Blocking
On 4/9/10 12:07 PM, Klaus Darilion wrote: Am 09.04.2010 06:54, schrieb dotnetdub: kamctl ps: Process:: ID=14 PID=28504 Type=MI DATAGRAM Process:: ID=15 PID=28505 Type=MI DATAGRAM Process:: ID=16 PID=28506 Type=MI DATAGRAM Is it possible to have several MI listeners? I always have only one. for mi_datagram it is possible: http://kamailio.org/docs/modules/stable/modules_k/mi_datagram.html#id2583658 Cheers. Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Kamailio Blocking
Hello, On 4/9/10 10:58 AM, Henning Westerholt wrote: On Friday 09 April 2010, Daniel-Constantin Mierla wrote: I have been running a very stable Kamailio 1.4 install for over a year now with no downtime. From time to time I get a message from the OS telling me that task kamailio: blocked for more than 120 seconds and a dump into syslog. [..] the messages refer to mi_datagram processes. These processes listen on a unixsocket as I could get from the trace, and if there is no mi command, they stay blocked. I haven't seen such messages so far, what is your OS? Hi Daniel, this is a more or less standard behaviour in the linux kernel available since 2.6.26 or so, if i remember correctly. ok, good to know. I think that i saw it a few times on some systems as well, but so far don't remember the cause. Maybe google will reveal something, I will check once I get some spare time... Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Kamailio Blocking
this message. [274198.229641] kamailio D d0e00947 0 28506 28484 [274198.229643]f77f8e60 0082 0002 d0e00947 4292 f77f8fec c4024020 0001 [274198.229646] 0001 0002 f7091398 0001 0001 c01211f2 [274198.229650]f5cba384 f5cba38c f5cba388 f77f8e60 c02c91ec f5cba38c f60add68 f77f8e60 [274198.229653] Call Trace: [274198.229656] [] __wake_up_sync+0x2a/0x3e [274198.229659] [] __mutex_lock_slowpath+0x50/0x7b [274198.229662] [] mutex_lock+0xa/0xb [274198.229664] [] unix_dgram_recvmsg+0x3e/0x231 [274198.229667] [] get_page_from_freelist+0xc1/0x3e9 [274198.229670] [] __rmqueue_smallest+0x83/0xe3 [274198.229673] [] sock_recvmsg+0xde/0xf9 [274198.229677] [] autoremove_wake_function+0x0/0x2d [274198.229681] [] __alloc_pages_internal+0xb5/0x34e [274198.229686] [] sys_recvfrom+0xb4/0x116 [274198.229690] [] cp_new_stat64+0xfc/0x10e [274198.229696] [] do_page_fault+0x4b2/0x8f9 [274198.229699] [] sys_socketcall+0x135/0x19e [274198.229703] [] sysenter_past_esp+0x78/0xb1 [274198.229706] [] xenfb_probe+0xd1/0x35b There was no activity when this happened. ON the 1.4 box there could be about 10 sessions setup when it happens. kamctl ps: Process:: ID=14 PID=28504 Type=MI DATAGRAM Process:: ID=15 PID=28505 Type=MI DATAGRAM Process:: ID=16 PID=28506 Type=MI DATAGRAM Even on the old install this doesn't seem to cause any problem and same here on 3.01 but would like to try and solve it. Any idea? Regards, Stephen ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] fr_inv_timer in kam-3.0
On 4/9/10 6:47 AM, Juha Heinanen wrote: Daniel-Constantin Mierla writes: > > for 3.1 we should get > > rid of them and have only one mode. it also makes writing the docs > > easier, when people can concentrate on one version of the docs instead > > of two or three. > > > I agree we should reduce as much as possible, but as said above, it is > practically just like another global parameter. there has always been changes in config file from one version to another. in 3.1, there should be only one mode and a documented list of changes that are needed from current K or S modes in order to get config working again. Fine with me. IIRC, when we listed the differences, Andrei said that drop behaviour (vs exit) should be made default, the only one that is debatable now is the way branches are handled in serial forking (ie, dropping or not replies of previous branches in serial forking) - this one can get as tm parameter. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] fr_inv_timer in kam-3.0
Hello, On 4/9/10 12:41 AM, Ovidiu Sas wrote: The thing is that the flavor is controlling the behavior of several module as opposed to params that are controlling the behavior of a single module. it is not the flavour that does it. Or at least we use different terms here. The flavour controls the name of binary, compilation flags, what tools are installed. This is the config file compatibility mode which does not depend on flavour - no matter what flavour you have, you can use #!KAMAILIO or #!SER (e.g., compile K flavour and have SER config compatibility). I'm fine with getting rid of script compat directive, but flavours will stay for a while, since there are different db structures and modules for spcific purposes. Cheers, Daniel For the next release, it would be nice to get rid of it and maybe perform the following: - we should switch to ms for all tm timers; - maybe we should replace 'drop' with 'abort' and properly document this (everyone will be forced to update their configs and maybe rethink the logic); - allow fixups for all modules; and so on ... Best thing to do would be to create a wiki page with everything that needs to be done in order to get rid of flavor and get input from the community on how to address each issue. Thanks, Ovidiu On Thu, Apr 8, 2010 at 6:10 PM, Daniel-Constantin Mierla wrote: On 4/8/10 11:06 PM, Alex Balashov wrote: On 04/08/2010 05:06 PM, Ovidiu Sas wrote: I have to agree with Juha here. In the next major release we should get rid of this flavor stuff. Everyone should bite the bullet and make their old scripts compatible with the new architecture. Even I will agree with this, and I am very resistant to change by nature. Three major aspects seem to be controlled by compat mode: - exit vs drop - in K they are distinct (e.g., drop is different in branch and onreply routes), in SER drop==exit - some bits in tm - avp parms format (in K they use PV format, to be coherent with all other modules), auto-correction of timer parmeters that used to be seconds in K and now are milliseconds and auto-dropping of branches for serial forking - modules' functions fixup attempts - in S mode, fixups based on pseudo-variables are not tried If there is a way to make everyone happy with a single mode, then I am all for it. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] fr_inv_timer in kam-3.0
On 4/8/10 11:06 PM, Alex Balashov wrote: On 04/08/2010 05:06 PM, Ovidiu Sas wrote: I have to agree with Juha here. In the next major release we should get rid of this flavor stuff. Everyone should bite the bullet and make their old scripts compatible with the new architecture. Even I will agree with this, and I am very resistant to change by nature. Three major aspects seem to be controlled by compat mode: - exit vs drop - in K they are distinct (e.g., drop is different in branch and onreply routes), in SER drop==exit - some bits in tm - avp parms format (in K they use PV format, to be coherent with all other modules), auto-correction of timer parmeters that used to be seconds in K and now are milliseconds and auto-dropping of branches for serial forking - modules' functions fixup attempts - in S mode, fixups based on pseudo-variables are not tried If there is a way to make everyone happy with a single mode, then I am all for it. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] fr_inv_timer in kam-3.0
On 4/8/10 5:35 PM, Andreas Granig wrote:
Juha,
Juha Heinanen wrote:
for example, this kind of call works for me:
t_set_fr("$avp(i:722)", "@cfg_get.local.phone_timeout");
Thanks. I was doing it wrong, namely without the double-quotes. D'oh.
I committed on git master and kamailio_3.0 a fix that should take the
value of timeout from AVP in seconds. If you can test and tell if works
for you, would be appreciated.
Thanks,
Daniel
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Re: [Kamailio-Users] fr_inv_timer in kam-3.0
On 4/8/10 10:05 PM, Juha Heinanen wrote: Daniel-Constantin Mierla writes: > When #!KAMAILIO is defined and the value of timeout avp is less than > 120, then it is multiplied with 1000 (auto-correction from second to > milisecond), but since it actually second for AVP case, will result in a > very long timeout :-). > > I will fix it. i hope your fix don't break my script, where i have not defined #!KAMAILIO. no, it is not affected if you don't use #!KAMAILIO. Let me try to explain what #!KAMAILIO does, so people understand better what happens inside the code: - when you define it, a global variable is set (similar to a global cfg parameter) - inside the code, there are some IF conditions testing the value of this variable, and if matches kamailio, then some particular behaviour is done - the goal of it is to have kind of profile, for compatibility reasons with behaviour of kamailio 1.5. There are about 5 things controlled by it right now it is VERY bad to have all these different modes. It is like with global parameters, a good documentation should make it easier... for 3.1 we should get rid of them and have only one mode. it also makes writing the docs easier, when people can concentrate on one version of the docs instead of two or three. I agree we should reduce as much as possible, but as said above, it is practically just like another global parameter. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] fr_inv_timer in kam-3.0
On 4/8/10 5:16 PM, Alex Balashov wrote:
I noticed this too.
On 04/08/2010 11:11 AM, Andreas Granig wrote:
Hi,
In the docs at
http://kamailio.org/docs/modules/3.0.x/modules/tm.html#fr_inv_timer it
says that fr_inv_timer_avp should be defined like this, without $ or
$avp:
modparam("tm", "fr_inv_timer_avp", "my_fr_inv_timer")
In kam <= 1.5 I did it like that:
modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)")
if you set config to kamailio compat mode via #!KAMAILIO then it accepts
the kamailio format where all avps are specified in PV format:
$avp(...). Doc needs some update :-) ...
which doesn't give me an error on startup with kam-3.0 either, but the
timer doesn't get fired (I use seconds for that as noted in the docs).
Well, seems to be a bug in code, I thought the timer is set in
miliseconds even for avps. It is a incoherence imo, the fr_timer and
fr_inv_timer module parameters are in miliseconds, but when given via
avps expects seconds, making impossible to have dynamic timeouts less
than 1 sec via avp. There is t_set_fr() but would be easier to have all
timeouts using same unit.
When #!KAMAILIO is defined and the value of timeout avp is less than
120, then it is multiplied with 1000 (auto-correction from second to
milisecond), but since it actually second for AVP case, will result in a
very long timeout :-).
I will fix it.
Thanks,
Daniel
If I change it to
modparam("tm", "fr_inv_timer_avp", "callee_fr_inv_timer")
then I get the error "malformed or non AVP callee_fr_inv_timer AVP
definition", same with setting it to "s:callee_fr_inv_timer".
Anyhow, it seems to be deprecated anyways, so I'm looking to get
t_set_fr() working. I'm just curious how I can use a var or AVP loaded
from DB to set the value on-the-fly? t_set_fr(...) seems to allow only
constants to be set. Couldn't find anything in the docs regarding
that one.
Thanks,
Andreas
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Re: [Kamailio-Users] drop in branch/onsend-route in kam-3.0
Hi,
seems this part was forgotten during the integration process. I attached
a patch that should fix it, please test it (i didn't at all) and let me
know the results.
If all ok I will double check with Andrei to see if breaks something in
the new TM architecture and then push it to git either under #!KAMAILIO
compat mode or enabled all time if everyone agrees.
Thanks,
Daniel
On 4/8/10 3:15 PM, Jon Bonilla (Manwe) wrote:
El Thu, 8 Apr 2010 14:52:15 +0200
Iñaki Baz Castillo escribió:
Any ideas on how to accomplish dropping a specific branch?
If you use it to drop requests going to "unsafe" destinations (like
when a REGISTER contains a spoofed "Contact" URI pointing to a gw or
the proxy itself) then I recommend using blcklist for this purpose.
This is, you set the list of IP's for gateways and proxies (forbidden
addresses into a registration Contact) and enable such blacklist after
doing lookup. If a branch tries to go to one of these addresses it
will be dropped by t_relay (and will return certain code I don't
remember now).
I use it and works well.
This is not the case. It's for dropping some local branches. :(
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diff --git a/modules/tm/t_fwd.c b/modules/tm/t_fwd.c
index c518592..24dfeb1 100644
--- a/modules/tm/t_fwd.c
+++ b/modules/tm/t_fwd.c
@@ -204,6 +204,7 @@ static int prepare_new_uac( struct cell *t, struct sip_msg
*i_req,
snd_flags_t rpl_snd_flags_bak;
struct socket_info *force_send_socket_bak;
struct dest_info *dst;
+ struct run_act_ctx ctx;
shbuf=0;
ret=E_UNSPEC;
@@ -333,7 +334,8 @@ static int prepare_new_uac( struct cell *t, struct sip_msg
*i_req,
/* set the new values */
i_req->fwd_send_flags=snd_flags /* intial value
*/;
set_force_socket(i_req, fsocket);
- if
(run_top_route(branch_rt.rlist[branch_route], i_req, 0) < 0)
+ if
(run_top_route(branch_rt.rlist[branch_route], i_req, &ctx)
+ < 0)
{
LOG(L_ERR, "Error in run_top_route\n");
}
@@ -345,6 +347,13 @@ static int prepare_new_uac( struct cell *t, struct sip_msg
*i_req,
i_req->fwd_send_flags=fwd_snd_flags_bak;
i_req->rpl_send_flags=rpl_snd_flags_bak;
exec_post_script_cb(i_req, BRANCH_CB_TYPE);
+ /* if DROP was called in cfg, don't forward,
jump to end */
+ if (unlikely(ctx.run_flags&DROP_R_F))
+ {
+ tm_ctx_set_branch_index(0);
+ set_route_type(backup_route_type);
+ goto error03;
+ }
}
tm_ctx_set_branch_index(0);
set_route_type(backup_route_type);
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Re: [Kamailio-Users] Understanding memory-leaks by inspecting PKG/SHM memory status at shutdown
On 4/8/10 1:21 PM, Iñaki Baz Castillo wrote:
2010/4/8 Iñaki Baz Castillo:
2010/4/8 Daniel-Constantin Mierla:
it does not look as a dump with memory debugging on.
SOrry, I just applied "MEMDBG=1" in one of the servers and got the
output in the other.
When memdbg is on, you should get something like:
0(17665) 1. N address=0xb5ab2440 frag=0xb5ab2428 size=4 used=1
0(17665) alloc'd from timer.c: init_timer(52)
Notice the "alloc'd ...' line which specifies the place where the memory was
allocated.
A leak is signaled by many occurrences of allocation from same place
(skipping the part of allocation done for config parsing and module
initialization which happen only one, at startup).
Hi again. I already have a kamailio 1.5.4 compiled with mem debugging
(as "kamailio -V" shows DBG_QM_MALLOC flag).
In config file I have:
debug=3
memlog=3 # Same behaviour with 1 or 2 as it equal or less than 'debug'.
Unfortunatelly the ammount of logs it generates makes it unusable for
production environment (~ 10 calls per second). Just restarting
kamailio when memlog is enabled takes really long time (unfortuantelly
I must restart it when adding new entries to 'address' table due to
the issue when performing "fifo address_reload" which completely
freezes kamailio sometimes).
Do I miss something? is it possible to log allocated and freeded
memory without generating so many logs?
the goal is to see the places where the memory was allocated. That will
give the proper hints about the leak.
What you can do is to print pkg status only when you send SIGUSR1 -- I
attached a patch for that.
In this way, a restart does not print pkg and shm status, so it is fast.
At runtime, when you send SIGUSR1 to a pid, the others can work just
fine, so processing should not be affected that much. Use kamctl ps to
spot the pid of an udp worker.
Cheers,
Daniel
If not, I could use "memlog=1" without memory debugging compiled and I
would check periodically the ammounf of PKG memory used. This is, I
get this output:
kamailio[11770]: Memory status (pkg):
kamailio[11770]: fm_status (0x701a40):
kamailio[11770]: heap size= 16777216
kamailio[11770]: used= 190936, used+overhead=250696, free=16526520
kamailio[11770]: max used (+overhead)= 258464
I can check it periodically and inspect if the used memory is
increasing. If so there must be a memleak. Am I right?
Thanks.
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Index: main.c
===
--- main.c (revision 5999)
+++ main.c (working copy)
@@ -457,7 +457,7 @@
while(wait(0) > 0); /* Wait for all the children to
terminate */
signal(SIGALRM, sig_alarm_abort);
- cleanup(1); /* cleanup & show status*/
+ cleanup(0); /* cleanup & show status*/
alarm(0);
signal(SIGALRM, SIG_IGN);
dprint("Thank you for flying " NAME "\n");
@@ -469,8 +469,8 @@
LM_GEN1(memlog, "Memory status (pkg):\n");
pkg_status();
#endif
- LM_GEN1(memlog, "Memory status (shm):\n");
- shm_status();
+ //LM_GEN1(memlog, "Memory status (shm):\n");
+ //shm_status();
break;
case SIGCHLD:
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Re: [Kamailio-Users] [sr-dev] irc devel meeting
OK, since we already are a bunch of people that can (want to) attend, I created a wiki page with some proposed topics, based on what was discussed on mailing lists lately. Please contribute yourself: http://sip-router.org/wiki/devel/irc-meetings/next Thanks, Daniel On 4/8/10 12:07 PM, Elena-Ramona Modroiu wrote: Hi, I'll participate also. Regards, Ramona On 04/08/2010 10:48 AM, Daniel-Constantin Mierla wrote: Hello, I know some of devels are still in Easter vacation, but I hope we can get together soon on irc to sketch the plan for 3.1 release. I propose next week, Wednesday, April 14, 15:00UTC, on #sip-router channel hosted by irc.freenode.net If you are a developer and want to participate, please announce. If it is not possible to attend for major contributors, we can look for another date (eventually start a pool to see the day were most of us are available). Cheers, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] irc devel meeting
Hello, I know some of devels are still in Easter vacation, but I hope we can get together soon on irc to sketch the plan for 3.1 release. I propose next week, Wednesday, April 14, 15:00UTC, on #sip-router channel hosted by irc.freenode.net If you are a developer and want to participate, please announce. If it is not possible to attend for major contributors, we can look for another date (eventually start a pool to see the day were most of us are available). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] [SR-Users] merging users mailing lists
Hello, a final reminder for users mailing lists merging... If nothing else against appears, I will merge users list to sr-users over the weekend. You can continue posting to existing mailing lists addresses, it will work, just that messages will end up on the same place. Thanks, Daniel On 3/28/10 6:52 AM, Jeff Brower wrote: Daniel- On 3/27/10 5:23 PM, Iñaki Baz Castillo wrote: 2010/3/27 Alex Balashov: I am opposed to this. I think there is a large base of Kamailio users that does not wish to get mired in larger discussions about SER-compatible modes of using sip-router and other things of that nature. The merging proposal is good but perhaps it should take place later. ok, I will do it 10 minutes later ;-) The proposal resulted looking at discussions on the mailing lists and feedback I accumulated during last month travelings. We direct new the people looking at our project to three different places for discussions about stable releases and the source code is more or less the same. What is on sr-users is definitely important for k and s users as well. Surprisingly, even for me, the integration done last year had fantastic outcome and the differences between flavours are not radical. I tried to summarize on the page: http://sip-router.org/kamailio-release/ Moreover, the best for our community users is having access to all developers. We share now code that was developed by the other project during 2005-2008 and we tend to stay focused on just one users mailing list, neglecting the others. I think we can sort out better the issues in one mailing list and everyone is sure will get the best answer since all devels and users will have focus in a single place. In addition, the discussions about differences existing now will create the necessary pressure to document properly or find a better solution. Agree. If you want to continue to build critical mass for your software and your community, you should definitely have fewer lists, not more. Serious participants have no trouble to filter out posts that are not of interest or don't affect them (or they don't understand). But serious participants hate to miss things important to them just because it did not appear on "their" list. -Jeff ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] [SR-Users] merging users mailing lists
Hello, a final reminder for users mailing lists merging... If nothing else against appears, I will merge users list to sr-users over the weekend. You can continue posting to existing mailing lists addresses, it will work, just that messages will end up on the same place. Thanks, Daniel On 3/28/10 6:52 AM, Jeff Brower wrote: Daniel- On 3/27/10 5:23 PM, Iñaki Baz Castillo wrote: 2010/3/27 Alex Balashov: I am opposed to this. I think there is a large base of Kamailio users that does not wish to get mired in larger discussions about SER-compatible modes of using sip-router and other things of that nature. The merging proposal is good but perhaps it should take place later. ok, I will do it 10 minutes later ;-) The proposal resulted looking at discussions on the mailing lists and feedback I accumulated during last month travelings. We direct new the people looking at our project to three different places for discussions about stable releases and the source code is more or less the same. What is on sr-users is definitely important for k and s users as well. Surprisingly, even for me, the integration done last year had fantastic outcome and the differences between flavours are not radical. I tried to summarize on the page: http://sip-router.org/kamailio-release/ Moreover, the best for our community users is having access to all developers. We share now code that was developed by the other project during 2005-2008 and we tend to stay focused on just one users mailing list, neglecting the others. I think we can sort out better the issues in one mailing list and everyone is sure will get the best answer since all devels and users will have focus in a single place. In addition, the discussions about differences existing now will create the necessary pressure to document properly or find a better solution. Agree. If you want to continue to build critical mass for your software and your community, you should definitely have fewer lists, not more. Serious participants have no trouble to filter out posts that are not of interest or don't affect them (or they don't understand). But serious participants hate to miss things important to them just because it did not appear on "their" list. -Jeff ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Understanding memory-leaks by inspecting PKG/SHM memory status at shutdown
line which specifies the place where the memory was allocated. A leak is signaled by many occurrences of allocation from same place (skipping the part of allocation done for config parsing and module initialization which happen only one, at startup). Send kamailio -v to see if memory debugging is on. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] PKG memory issues
On 4/8/10 12:07 AM, Iñaki Baz Castillo wrote: 2010/4/7 Iñaki Baz Castillo: Just a question: I usually compile kamailio with "make deb", is it compiled with memory debugging? Autoreply: Yes, DEB packages are compiled with memory debugging built in. I've checked it by setting "memlog=1" and sending a SIGUSR1 to any worker so I get the process PKG status. NOTE: It's SIGUSR1 rather than SIGUSR2 :) ok, I mixed them, it was too late to check the source code. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] PKG memory issues
On 4/7/10 11:37 PM, Iñaki Baz Castillo wrote: 2010/4/7 Daniel-Constantin Mierla: It's Kamailio 1.5.1-notls. Now I've compiled 1.5.4-notls and increased PKG memory to 16. if I spot it right on the svn commit log, there was a fix for a leak related to dst_uri when changed from failure route, nothing else important. Would it affect when use LCR module 'next_gw()' funtion in failure_route? It creates a new branch internally Would be good if you can compile with memory debugging, let it run for a while, then send a sigusr2 to a sip worker process to dump the pkg status -- that can reveal if it is a leak somewhere. Thanks. I cannot do it in the production server as it's very critical now :) But I'll do it in a mirror server generating traffic with SIPp. Just a question: I usually compile kamailio with "make deb", is it compiled with memory debugging? if not, would it be enabled by editing Makefile.vars (MEMDBG=1) and running "make deb"? yes, setting MEMDBG=1 will compile with memory debugging. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] PKG memory issues
On 4/7/10 10:56 PM, Iñaki Baz Castillo wrote: 2010/4/7 Daniel-Constantin Mierla: You haven't mentioned the version yet. Sorry, my fault, I'm forking myself now with different tasks :) It's Kamailio 1.5.1-notls. Now I've compiled 1.5.4-notls and increased PKG memory to 16. if I spot it right on the svn commit log, there was a fix for a leak related to dst_uri when changed from failure route, nothing else important. Would be good if you can compile with memory debugging, let it run for a while, then send a sigusr2 to a sip worker process to dump the pkg status -- that can reveal if it is a leak somewhere. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] PKG memory issues
On 4/7/10 10:47 PM, Iñaki Baz Castillo wrote: 2010/4/7 Daniel-Constantin Mierla: does not look like a very pkg intensive processing. What version are you running? Do you get other frequent error messages (e.g., bad sip message)? No, no errors at all, I check the logs very often (however I set it to INFO level, but wrong messages should be displayed at this level). yes, info level displays everything but 'debug'. You haven't mentioned the version yet. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] PKG memory issues
Hello, On 4/7/10 10:37 PM, Iñaki Baz Castillo wrote: Hi, I've experimented PKG out of memory in a production server. Of course I'm already increasing such value but I would like to know if the current settings and traffic could run into PKG memory issues: - Dual core INTEL XEON 3.00 GHz server. - Kamailio PKG memory = 4 MB. - Kamailio Shared memory = 64 MB. - Kamailio just listens in a single UDP port (8 childrens). - Just INVITE method is handled (no registration, no subscription). The script does the following for each request: - 'permissions' module to match source IP (just ~20 entries in 'address' table). - 2 custom SQL queries (returning a simple value). - 'dialog' memory (just in memory). - 'uac' module to change From header. - There are 10 AVP's set per transaction. - 'lcr' module for routing to two gateways (just 2 entries in 'lcr' table). - 'rtpproxy' is forced for every call. The server has been working properly for months but these days the traffic has been duplicated, having ~400 simultaneous calls in peak hours. Also note that such calls come from callcenters so they are ver "fast". With this environment, is it normal to get into PKG memory issues (4 MB)? I understand that it makes sense, but I would like to hear some opinions. Thanks a lot. does not look like a very pkg intensive processing. What version are you running? Do you get other frequent error messages (e.g., bad sip message)? Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] mc (Midnight Commander) syntax highlighting scheme
Hello, On 4/7/10 7:22 PM, Константин wrote: Hi all! For easy editing kamailio configuration files I developed a syntax highlighting scheme for mc <http://www.midnight-commander.org/>. many thanks! I checked the format and should be easy to update it for 3.0. I will try to find a place in the source tree to host this one along with vim syntax file. Should be easier to maintain for new version and install once sources are downloaded. It works also in mac os x, with mc installed from macports. The difference is that Syntax file is located in /opt/local/etc/mc/. The alternative is to use home directory mc config: ~/.mc/cedit/Syntax (create the file if does not exist). Cheers, Daniel Preview <http://img262.imageshack.us/img262/1834/sshot1r.png> The scheme includes basic syntactic structures of Kamailio 1.5, all constants, variables, operators and standart core functions. To install it, download the file kamailio.syntax from attachment and copy it to /usr/share/mc/syntax (for Debian Etch) Next, add to the file /usr/share/mc/syntax/Syntax including: file ..\*\\.cfg$ Kamailio\sconfig include kamailio.syntax Thats all, I hope its helps someone ;) -- Shpinev Konstantin mailto:kks_m...@inbox.ru ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Version 3
Hi Henning, On 3/31/10 6:17 PM, Henning Westerholt wrote: On Wednesday 31 March 2010, Andreas Heise wrote: thanks for the update, but seems to be that many users wait for 3.1 which was announced as the first real sip-router release and is again a major change which need effort to validate. Hello Andreas, well, 3.1 will have an even better integration, to further blurry the border between the different parts coming from ser and kamailio. So in my opinion, if people wait, its more because of a 'don't trust a .0 release..' opinion. ;-) Is there already a target date proposed for 3.1? I don't think so, Daniel please correct me if i'm wrong. Normally it should be out something roughly six monts after 3.0, this would mean somewhere in July. indeed, no fixed date by now. We should schedule a irc devel meeting soon to sketch the roadmap for 3.1. There is a lot of new features, some still need more polishing to be completed (at least what i started) and the other work is related to merging some duplicated modules. In another email I said maybe summer will be used for testing and release in beginning of autumn so we have approx 8 months cycle. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Version 3
Hi Andreas, On 3/31/10 4:29 PM, Andreas Heise wrote: Hi Daniel, thanks for the update, but seems to be that many users wait for 3.1 which was announced as the first real sip-router release and is again a major change which need effort to validate. I think it was a mistake in communication with the achievements for 3.0 -- we have to work on and fixed it. When we met in Karlsruhe, Nov 2008, we kind of agreed as plan go on to make the core and tm work with kamailio modules or ser modules. The option would have been either at compile time or at startup. However, we succeeded to go one step further and actually be able to run mixed modules at the same time. So we are now, considering the initial goals, pretty much the 3.1: - no need to compile with different flags to run one or the other type of modules - no need to configure what type of modules are running Simply you can mix them, it works. I agree that for 3.1 we will be better from integration point of view, in regards to less duplicated modules (hopefully sl, domain, pdt, ... will be merged). However some of the modules will stay as they are now for longer time. Here are auth_db, usrloc, and the other modules that differ as backend database structure (e.g., user profiles tables, location). There are public (siremis, serweb) and private tools, povisioning and monitoring systems that cannot be dropped easily. So we will have flavours packaging for a while. Otherwise, 3.1 won't have other integration work for core and tm, that work is finished. What comes in those parts of code for 3.1 are pure brand new features. Kamailio 3.0 just enables some features by default, sets different default behavior which can be tuned by parameters anyhow. Here I tried to collect more details lately: http://sip-router.org/releases/ http://sip-router.org/kamailio-release/ There is no patch that has to be applied in kamailio 3.0 branch order to compile ser flavour out of it. Is there already a target date proposed for 3.1? Not clearly decided, but the usual 6-8 months is still in place, that means testing should start beginning of summer, which will result in release maybe beginning of autumn (not to do it from the beach :-) in vacation). Cheers, Daniel 2010/3/31 Daniel-Constantin Mierla <mailto:mico...@gmail.com>> On 3/24/10 11:10 AM, Alex Balashov wrote: Using it in several production environments; awesome reliability! Thanks Alex! And you know, this is not by chance... During last month I have been traveling a lot, I met people that use 3.0 in production or they passed the internal testing phase, being just to switch it to production. They were pleasantly surprised by the results. Since this question pops up from time to time, I tried to collect the facts and procedures during past year and demystify why 3.0.x is very stable. Probably I forgot to mention other people or companies that substantially contributed to 3.0.x (drop me an email to fix it) ... anyhow, here is the link: http://www.kamailio.org/w/2010/03/remarks-about-v3-0-x-strong-stability/ Cheers, Daniel Wouldn't go back to 1.5.x for the world. -- Sent from mobile device On Mar 24, 2010, at 4:09 AM, dotnetdub mailto:dotnet...@gmail.com>> wrote: Hi List, Anybody using this in production yet? If so what kind of volume and how is reliability? Looking to move to this platform, looks very good, interested to hear some experiences. Thanks, Stephen ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo
Re: [Kamailio-Users] Version 3
On 3/24/10 11:10 AM, Alex Balashov wrote: Using it in several production environments; awesome reliability! Thanks Alex! And you know, this is not by chance... During last month I have been traveling a lot, I met people that use 3.0 in production or they passed the internal testing phase, being just to switch it to production. They were pleasantly surprised by the results. Since this question pops up from time to time, I tried to collect the facts and procedures during past year and demystify why 3.0.x is very stable. Probably I forgot to mention other people or companies that substantially contributed to 3.0.x (drop me an email to fix it) ... anyhow, here is the link: http://www.kamailio.org/w/2010/03/remarks-about-v3-0-x-strong-stability/ Cheers, Daniel Wouldn't go back to 1.5.x for the world. -- Sent from mobile device On Mar 24, 2010, at 4:09 AM, dotnetdub wrote: Hi List, Anybody using this in production yet? If so what kind of volume and how is reliability? Looking to move to this platform, looks very good, interested to hear some experiences. Thanks, Stephen ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] more type conversion wonders
On 3/31/10 8:40 AM, Juha Heinanen wrote:
i played a bit more with selects and found that statement
if ($sht(auth=>foo::count)> @cfg_get.local.gw_timeout) {
xlog("foo");
}
I guess selects have types, while pseudo-variables are kind of type
agnostic. Any PV has a string representation of the value, comparison is
done as integer if both PVs are integers, otherwise is done as string.
The safe side is to use PV representation of selects when comparing with
another PV.
PV and selects is another point for the future coherence.
Cheers,
Daniel
produces error
0(9213) : [cfg.y:3379]: parse error in config file
/etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch:
str instead of int at (507,55)
but the error goes away if i either make explicit conversion
if ($sht(auth=>foo::count)> (int)@cfg_get.local.gw_timeout) {
xlog("foo");
}
or use $sel
if ($sht(auth=>foo::count)> $sel(cfg_get.local.gw_timeout)) {
xlog("foo");
}
this is thus exactly opposite than in my t_set_fr ordeal where i got
conversion error when i used $sel instead of @.
in my opinion, no (int) conversion should be needed in the firs case,
because the value of the cfg variable is int.
in summary, writing statements that include selects is very difficult
and error prone.
-- juha
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Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
On 3/30/10 8:29 PM, Ovidiu Sas wrote: If you need this in 1.5, try to fix the contact header and then forward the REGISTER back to yourself. The loopback REGISTER should have the fixed Contact header and now you can call save(). You will need to tweak a a little bit the config to get everything right into the usrloc (the loopback REGISTER will come from the server IP) and it might work. it works for no nat scenario, otherwise make sure you will get the looped message on the same socket and you add before looping a header with source ip and port (corresponding nat box pinhole) that you store in received avp. Cheers, Daniel Regards, Ovidiu Sas On Tue, Mar 30, 2010 at 11:36 AM, NeoTel Lists wrote: Hello Everybody! Is there any way to save(domain, 0x02) the contact after it has been changed somehow in Kamailio<= 1.5? Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? Or: Can I set the to be saved q value somehow? br Walter ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
On 3/30/10 7:27 PM, Alex Balashov wrote: On 03/30/2010 01:17 PM, Daniel-Constantin Mierla wrote: No, r-uri is the contact address from REGISTER. Otherwise, how do you get far-end NAT traversal for incoming calls (to the registrants) to work? dst_uri field is set to received ip:port, so the nat box is used as outbound proxy. Hmm. I guess I have been doing it wrong for a long time on 1.x by using fix_nated_contact() on REGISTER processing where NAT is detected, it is harmless, since it does update to contact header, the changes are visible only when forwarding. instead of fix_nated_register() and separation into contact and received as you describe. But in my case it works... for natted cases the important thing is to get in location the ip and port of nat box. then depends on sip phone how it accepts incoming requests. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
On 3/30/10 7:14 PM, Alex Balashov wrote: On 03/30/2010 01:10 PM, Daniel-Constantin Mierla wrote: On 3/30/10 5:58 PM, Alex Balashov wrote: On 03/30/2010 11:36 AM, NeoTel Lists wrote: Hello Everybody! Is there any way to save(domain, 0x02) the contact after it has been changed somehow in Kamailio <= 1.5? If it were not possible, how would nathelper:fix_nated_contact() work? this is slightly different thing, that is intended for forwarded requests, patching the contact address, which becomes effecting before forwarding. For save in location, still the contact from header is used, since most of the phones won't accept requests not matching their registered contact address. In the first version, in location was saved the ip and port of NAT box as contact address, but phones rejected calls. Now, the contact address from REGISTER is saved and along with it the source ip and port in received column. Yes, but if NAT flag is set, then RURI will contain the received ip:port in the domain portion upon lookup(), right? No, r-uri is the contact address from REGISTER. Otherwise, how do you get far-end NAT traversal for incoming calls (to the registrants) to work? dst_uri field is set to received ip:port, so the nat box is used as outbound proxy. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue
Hello,
for <= 1.5 is not easy to get to it, unless you use db mode only and
play directly to database.
In 3.0+ you can change the contact header and then use msg_apply_changes():
http://kamailio.org/docs/modules/stable/modules_k/textops.html#id2749047
then do save to location.
Cheers,
Daniel
On 3/30/10 7:03 PM, NeoTel Lists wrote:
Again sorry, too much trial&error makes a lot of noise ...
if (avp_subst("$(avp(contact))",
"/;phone-context=q([01]\.[0-9]+)(.*>)?([^>]*)/\2\3;q=\1/i")) {
# just to have it handy int that var(Q)
$var(Q) = $(avp(contact){s.select,1,>}{param.value,q});
remove_hf("Contact");
append_hf("Contact: $(avp(contact))\r\n");# must not add q= twice
}
-> Same result: q=-1 in Database.
-Ursprüngliche Nachricht-
Von: users-boun...@lists.kamailio.org [mailto:users-boun...@lists.kamailio.org]
Im Auftrag von NeoTel Lists
Gesendet: Dienstag, 30. März 2010 18:56
An: Alex Balashov; users@lists.kamailio.org
Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue
Sorry for being unclear: I meant: Modify the contact in the script, then do
save().
E.g.
$(avp(contact)) = $ct;
# Those Patton can't send the q value ... Misuse their ;phone-context param
if (avp_subst("$(avp(contact))",
"/;phone-context=q([01]\.[0-9])(.*>)?([^>]*)/\2\3;q=\1/i")) {
# just to have it handy int that var(Q)
$var(Q) = $(avp(contact){s.select,1,>}{param.value,q});
remove_hf("Contact");
append_hf("Contact: $(avp(contact));q=$var(Q)r\n");
# subst() does the same, also not save()ed
}
...
setbflag(1);
fix_nated_register();
save("contactimpl_nat", "0x02");
Br
Walter
-Ursprüngliche Nachricht-
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Im Auftrag von Alex Balashov
Gesendet: Dienstag, 30. März 2010 17:58
An: users@lists.kamailio.org
Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
On 03/30/2010 11:36 AM, NeoTel Lists wrote:
Hello Everybody!
Is there any way to save(domain, 0x02) the contact after it has been
changed somehow in Kamailio<= 1.5?
If it were not possible, how would nathelper:fix_nated_contact() work?
--
Daniel-Constantin Mierla * http://www.asipto.com/ *
http://twitter.com/miconda *
http://www.linkedin.com/in/danielconstantinmierla
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Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue
Hello,
for <= 1.5 is not easy to get to it, unless you use db mode only and
play directly to database.
In 3.0+ you can change the contact header and then use msg_apply_changes():
http://kamailio.org/docs/modules/stable/modules_k/textops.html#id2749047
then do save to location.
Cheers,
Daniel
On 3/30/10 7:03 PM, NeoTel Lists wrote:
Again sorry, too much trial&error makes a lot of noise ...
if (avp_subst("$(avp(contact))",
"/;phone-context=q([01]\.[0-9]+)(.*>)?([^>]*)/\2\3;q=\1/i")) {
# just to have it handy int that var(Q)
$var(Q) = $(avp(contact){s.select,1,>}{param.value,q});
remove_hf("Contact");
append_hf("Contact: $(avp(contact))\r\n");# must not add q= twice
}
-> Same result: q=-1 in Database.
-Ursprüngliche Nachricht-
Von: users-boun...@lists.kamailio.org [mailto:users-boun...@lists.kamailio.org]
Im Auftrag von NeoTel Lists
Gesendet: Dienstag, 30. März 2010 18:56
An: Alex Balashov; users@lists.kamailio.org
Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue
Sorry for being unclear: I meant: Modify the contact in the script, then do
save().
E.g.
$(avp(contact)) = $ct;
# Those Patton can't send the q value ... Misuse their ;phone-context param
if (avp_subst("$(avp(contact))",
"/;phone-context=q([01]\.[0-9])(.*>)?([^>]*)/\2\3;q=\1/i")) {
# just to have it handy int that var(Q)
$var(Q) = $(avp(contact){s.select,1,>}{param.value,q});
remove_hf("Contact");
append_hf("Contact: $(avp(contact));q=$var(Q)r\n");
# subst() does the same, also not save()ed
}
...
setbflag(1);
fix_nated_register();
save("contactimpl_nat", "0x02");
Br
Walter
-Ursprüngliche Nachricht-
Von: users-boun...@lists.kamailio.org [mailto:users-boun...@lists.kamailio.org]
Im Auftrag von Alex Balashov
Gesendet: Dienstag, 30. März 2010 17:58
An: users@lists.kamailio.org
Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
On 03/30/2010 11:36 AM, NeoTel Lists wrote:
Hello Everybody!
Is there any way to save(domain, 0x02) the contact after it has been
changed somehow in Kamailio<= 1.5?
If it were not possible, how would nathelper:fix_nated_contact() work?
--
Daniel-Constantin Mierla * http://www.asipto.com/ *
http://twitter.com/miconda *
http://www.linkedin.com/in/danielconstantinmierla
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Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
On 3/30/10 5:58 PM, Alex Balashov wrote: On 03/30/2010 11:36 AM, NeoTel Lists wrote: Hello Everybody! Is there any way to save(domain, 0x02) the contact after it has been changed somehow in Kamailio <= 1.5? If it were not possible, how would nathelper:fix_nated_contact() work? this is slightly different thing, that is intended for forwarded requests, patching the contact address, which becomes effecting before forwarding. For save in location, still the contact from header is used, since most of the phones won't accept requests not matching their registered contact address. In the first version, in location was saved the ip and port of NAT box as contact address, but phones rejected calls. Now, the contact address from REGISTER is saved and along with it the source ip and port in received column. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Quick REGEX to match Private IP
On 3/26/10 4:25 PM, Uriel Rozenbaum wrote:
Done,
I had a lot of errors so I'll just show the final version that works OK.
=~ "192\.168\.([0-9]{1,3})\.([0-9]{1,3})
The only drawback is that I could pass as valid 192.168.999.999 but as
these IPs come from a DNS query, I assume they'll be fine.
I think you can reduce it to be just: "^192\.168\."
Since it comes from dns server or socket attributes, then you are safe.
The other option is to convert it to integer (with transformations) and
check it as net mask via bit wise operators.
Cheers,
Daniel
Cheers,
Uriel
On Fri, Mar 26, 2010 at 11:29 AM, Uriel Rozenbaum
mailto:uriel.rozenb...@gmail.com>> wrote:
Hi Alex,
Actually what I'm trying to do is check the IPs on a request on a
Kamailio+RTPProxy acting as border of our network.
So I have the ingress IP and egress IP and need to check if I have
to bridge ii, ei, ie or ee.
I managed to obtain all IPs in AVPs, but now I have to check if
they are public or private.
So far our network uses only 192.168.x.x class for private servers.
Thanks for the quick reply
Uriel
On Fri, Mar 26, 2010 at 10:56 AM, Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:
172.16.0.0/12 <http://172.16.0.0/12> does not line up on octet
boundaries. You will need to do something other than a
regular expression. Fortunately, 'src_ip' is a composite that
supports comparisons against subnets in shorthand CIDR notation.
It might also be that whatever you are trying to accomplish
can be done better some other way, but since you did not pose
the question in terms of the objective, I cannot speak to that.
--
Sent from mobile device
On Mar 26, 2010, at 9:46 AM, Uriel Rozenbaum
mailto:uriel.rozenb...@gmail.com>>
wrote:
Hi guys,
Does anyone have a REGEX syntax to match a private IP on
the 192.168.x.x range?
I'm trying with:
if($avp(s:ip_origen)=~"192.168(\.([1]?\d{1,2}|2[0-4]{1}\d{1}|25[0-5]{1})){2}"
)
But all IPs pass as private, even public ones.
Thanks!
Uriel
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Re: [Kamailio-Users] Kammailio as proxy and clustered asterisk as registrar
On 3/27/10 7:04 PM, Vic Jolin wrote: Hi, I would like to seek help from the list regarding my setup. I have kamailio setup to send any sip messages to load balance asterisk clusters. The registration part is successful. The calling part is not, calls get dropped, I believe it's rtp issue or maybe nat. I'm very new to kamailio and would like to ask for help on this. If asterisk does not support PATH extension and you deal with natted clients then you cannot have kamailio as load balancer and asterisk as registrars. Calls must go back to phones behind nat via kamailio, otherwise symmetric NATs drop the sip traffic coming directly from asterisk. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] kam 1.5 sqlops sql_query timeout value
On 3/29/10 3:04 PM, Alex Balashov wrote:
Robert,
Such parameters are specific to the database backend module used by
sqlops, if they exist at all. Thus, you should consult the db_mysql
documentation; sqlops is an abstraction layer over the underlying
database connection facility.
It sounds like this is what you are looking for:
http://www.kamailio.org/docs/modules/1.5.x/db_mysql.html#id2452828
For some reason the readme for 3.0 was not built on web site, should be
fixed. The parameters are the same as for 1.5.
Daniel
Cheers,
-- Alex
On 03/29/2010 08:58 AM, Robert McGilvray wrote:
Hello,
I’m using the sqlops module for some custom queries against mysql.
Sometimes during a cluster failure the mysqld nodes will hang there
waiting for the backend to finish up whatever it’s doing, it still
accepts the connection and the query but doesn’t return results. I have
a pair of F5 load balancers in front of the two sql nodes, so there is
redundancy as long as the cluster is operational. I’d prefer to keep the
cross-site failover in kamailio.
I looked through the docs on sqlops and I can’t find any reference to a
timeout value. I’d like to implement a failover in the script to my
other database cluster but if kam waits for a long time before returning
a negative it may not work very well. Consider this code in my script
for 911 services in my US offices: (I rewrite the rpid/pai and ruri
based on IP address then send it to my provider(s))
modparam("sqlops","sqlcon","gokam=>mysql://*:**...@172.20.180.21/sip_gokam")
if (!sql_query("gokam", "select
location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as
network,inet_ntoa(netmask) \
as netmask from e911 where (inet_aton('$si') & netmask) = network",
"result")) {
sl_send_reply("500", "Database error");
exit;
}
What I’d like to do is:
modparam("sqlops","sqlcon","gokam=>mysql://:*...@172.20.180.21/sip_gokam")
modparam("sqlops","sqlcon","gokam_site2=>mysql://*:**...@172.23.180.21/sip_gokam")
if (!sql_query("gokam", "select
location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as
network,inet_ntoa(netmask) \
as netmask from e911 where (inet_aton('$si') & netmask) = network",
"result")) {
xlog(“L_CRIT”, “Primary database failure, using alternate\n”);
if (!sql_query(“gokam_site2”, “select
location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as
network,inet_ntoa(netmask) \
as netmask from e911 where (inet_aton('$si') & netmask) = network",
"result")) {
sl_send_reply(“500”, Database error”);
exit;
}
What is the default timeout for sql_query before it returns a negative,
is it configurable?
Thanks!
Bob
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Re: [Kamailio-Users] get warning when sip client is not online
Hello, On 3/30/10 11:06 AM, s.maert...@telenet.be wrote: Hello, I have a kamailio setup running for several SIP devices. I would like to be warned whenever one of those devices is not reacheable anymore. That means when they go offline or are not anymore registered to the kamailio server. (timeout, unregister, ...). I am using the location table so we would be able to write a script that just at specific times checks the difference between the subscriber table and the location table but maybe there is a much better and more realtime solution. So basically the question is : how to execute a script whenever a host gets removed fom the location table. I am running Kamailio 3.0.1 All hints are very much appreciated :) one option is to use pua_usrloc that publishes online/offline when location cache record is updated in some way (offline when is removed). You can intercept that in config and do what you want there. Another option could be triggers in mysql, when a record is deleted from location db table. Easiest way, imo, is to add an event route to be executed when a record goes offline. Requires some c coding, but afterwards is clean -- this features was listed as to-be-done in the future when event route was introduced, but no time yet, should be there in 3.1.0: http://lists.kamailio.org/pipermail/users/2009-May/023270.html Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] GSoC 2010 Student Registration is Open
Hello, if you are a student (or know one) interested in working with SIP and Presence over the summer, earn some money and experience, you have the choice of Kamailio - SIP Router project within SIP Communicator organization: http://www.kamailio.org/w/gsoc-2010/ Registration is now open, up to April 9, more details in above link. If you have further questions, drop an email. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] R: R: Fast lock loop with arm
Hello,
On 3/29/10 8:53 AM, Zappasodi Daniele wrote:
Hello,
I have tried a lot of times with different processes, but backtrace
shows always only this.
this is really strange. do you use mi_fifo? if yes, when you start
openser run 'openserctl ps'
Spot a udp worker process, the mi fifo process, the main process and the
timer process. when the locking happens, attach to each of them and get
the backtrace.
Thanks,
Daniel
Daniele
-Messaggio originale-
*Da:* Daniel-Constantin Mierla [mailto:mico...@gmail.com]
*Inviato:* venerdì 26 marzo 2010 19.40
*A:* Zappasodi Daniele
*Cc:* users@lists.kamailio.org; sr-dev
*Oggetto:* Re: R: [Kamailio-Users] Fast lock loop with arm
Hello,
On 3/26/10 4:13 PM, Zappasodi Daniele wrote:
Hello,
this is what I get with gdb:
(gdb) bt full
#0 0x40137e54 in sched_yield () from /usr/local/lib2/libc.so.6
No symbol table info available.
I don't think that it can help, but I am not able to load the
symbol table for openser on the server.
hmm, strange. Did you try with many processes? Sam result in the
backtrace?
Cheers,
Daniel
thanks,
Daniele
-Messaggio originale-
*Da:* Daniel-Constantin Mierla [mailto:mico...@gmail.com]
*Inviato:* mercoledì 24 marzo 2010 16.47
*A:* Zappasodi Daniele
*Cc:* users@lists.kamailio.org; sr-dev
*Oggetto:* Re: [Kamailio-Users] Fast lock loop with arm
Hello,
what version of kamailio do you use?
Can you grab a backtrace with gdb once you get this high cpu
usage? Spot one of the processes, take the pid and do:
gdb /path/to/kamailio _pid_
Then:
bt
You should see the bactrace of executed function getting to
deadloc. Make sure you get at least one SIP worker process
backtrace (do kamctl ps to see the type of kamailio process).
Cheers,
Daniel
On 3/24/10 4:29 PM, Zappasodi Daniele wrote:
Hi,
I have a big problem with fast lock mutex on arm CPU:
sometimes I find one or more children that go in an infinite
loop, in the while loop of get_lock function.
They remain in Run for long time (sometimes hours) and
openser keeps 100% CPU.
Now I have changed the functions get_lock and tsl in order
to obtain more info and
I observe that (if and) when the process finish the cycle,
it have done a big amount of cycles.
This the log with my added info:
Mar 26 20:29:08 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1
(cycle)
Mar 26 20:29:51 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1
(cycle)
Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1
(cycle)
Mar 26 20:29:55 SAM-IP openser[1645]:
NOTICE:tm:t_retransmit_reply: MYTM lock
[process in loop]
Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:_lock: ret
59374917 (cycle)
[after 4 minutes and 59374917 cycles, this is an example
with a "short" loop]
Mar 26 20:33:46 SAM-IP openser[1645]:
NOTICE:tm:t_retransmit_reply: MYTM lock done
I'm not able to recognize the call flow that generates the
loop (I know only that it always happens when
t_retransmit_reply calls LOCK_REPLIES),
but I urgently need a work around to escape from the loop.
This is the get_lock function with my counter j:
inline static int get_lock(fl_lock_t* lock)
{
#ifdef ADAPTIVE_WAIT
int i=ADAPTIVE_WAIT_LOOPS;
int j=1;/* my
change /
#endif
while(tsl(lock)){
#ifdef BUSY_WAIT
#elif defined ADAPTIVE_WAIT
j++;/* my change /
if (i>0) i--;
else sched_yield();
#else
sched_yield();
#endif
}
return(j);
}
Can I break the lock when my counter reaches a big value?
Should I call the unlock function after the break?
which value can be considered too big?
Thanks,
Daniele
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Re: [Kamailio-Users] [SR-Users] merging users mailing lists
On 3/27/10 5:23 PM, Iñaki Baz Castillo wrote: 2010/3/27 Alex Balashov: I am opposed to this. I think there is a large base of Kamailio users that does not wish to get mired in larger discussions about SER-compatible modes of using sip-router and other things of that nature. The merging proposal is good but perhaps it should take place later. ok, I will do it 10 minutes later ;-) The proposal resulted looking at discussions on the mailing lists and feedback I accumulated during last month travelings. We direct new the people looking at our project to three different places for discussions about stable releases and the source code is more or less the same. What is on sr-users is definitely important for k and s users as well. Surprisingly, even for me, the integration done last year had fantastic outcome and the differences between flavours are not radical. I tried to summarize on the page: http://sip-router.org/kamailio-release/ Moreover, the best for our community users is having access to all developers. We share now code that was developed by the other project during 2005-2008 and we tend to stay focused on just one users mailing list, neglecting the others. I think we can sort out better the issues in one mailing list and everyone is sure will get the best answer since all devels and users will have focus in a single place. In addition, the discussions about differences existing now will create the necessary pressure to document properly or find a better solution. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] merging users mailing lists
Hello, I propose to merge the users mailing lists, most of the traffic these days is about 3.0 and even there are 2 stables branches now, they are sync'ed, so same code more or less. For 3.1 will be one stable branch, falvor selection will be only a matter of make command. Lately common useful topics are discussed on those different mailing lists, notifications and knowledge base building require cross-posting, lot of overhead imo. Any other opinion? Like with devel mailing lists, existing email addresses for users ML can still be used, just that end on same ML. Natural choice will be to have us...@kamailio and serus...@iptel to be directed to sr-us...@lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] R: Fast lock loop with arm
Hello,
On 3/26/10 4:13 PM, Zappasodi Daniele wrote:
Hello,
this is what I get with gdb:
(gdb) bt full
#0 0x40137e54 in sched_yield () from /usr/local/lib2/libc.so.6
No symbol table info available.
I don't think that it can help, but I am not able to load the symbol
table for openser on the server.
hmm, strange. Did you try with many processes? Sam result in the backtrace?
Cheers,
Daniel
thanks,
Daniele
-Messaggio originale-
*Da:* Daniel-Constantin Mierla [mailto:mico...@gmail.com]
*Inviato:* mercoledì 24 marzo 2010 16.47
*A:* Zappasodi Daniele
*Cc:* users@lists.kamailio.org; sr-dev
*Oggetto:* Re: [Kamailio-Users] Fast lock loop with arm
Hello,
what version of kamailio do you use?
Can you grab a backtrace with gdb once you get this high cpu
usage? Spot one of the processes, take the pid and do:
gdb /path/to/kamailio _pid_
Then:
bt
You should see the bactrace of executed function getting to
deadloc. Make sure you get at least one SIP worker process
backtrace (do kamctl ps to see the type of kamailio process).
Cheers,
Daniel
On 3/24/10 4:29 PM, Zappasodi Daniele wrote:
Hi,
I have a big problem with fast lock mutex on arm CPU:
sometimes I find one or more children that go in an infinite
loop, in the while loop of get_lock function.
They remain in Run for long time (sometimes hours) and openser
keeps 100% CPU.
Now I have changed the functions get_lock and tsl in order to
obtain more info and
I observe that (if and) when the process finish the cycle, it
have done a big amount of cycles.
This the log with my added info:
Mar 26 20:29:08 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle)
Mar 26 20:29:51 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle)
Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle)
Mar 26 20:29:55 SAM-IP openser[1645]:
NOTICE:tm:t_retransmit_reply: MYTM lock
[process in loop]
Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:_lock: ret
59374917 (cycle)
[after 4 minutes and 59374917 cycles, this is an example with a
"short" loop]
Mar 26 20:33:46 SAM-IP openser[1645]:
NOTICE:tm:t_retransmit_reply: MYTM lock done
I'm not able to recognize the call flow that generates the loop
(I know only that it always happens when t_retransmit_reply calls
LOCK_REPLIES),
but I urgently need a work around to escape from the loop.
This is the get_lock function with my counter j:
inline static int get_lock(fl_lock_t* lock)
{
#ifdef ADAPTIVE_WAIT
int i=ADAPTIVE_WAIT_LOOPS;
int j=1;/* my change
/
#endif
while(tsl(lock)){
#ifdef BUSY_WAIT
#elif defined ADAPTIVE_WAIT
j++;/* my change /
if (i>0) i--;
else sched_yield();
#else
sched_yield();
#endif
}
return(j);
}
Can I break the lock when my counter reaches a big value?
Should I call the unlock function after the break?
which value can be considered too big?
Thanks,
Daniele
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Re: [Kamailio-Users] Fast lock loop with arm
Hello,
what version of kamailio do you use?
Can you grab a backtrace with gdb once you get this high cpu usage? Spot
one of the processes, take the pid and do:
gdb /path/to/kamailio _pid_
Then:
bt
You should see the bactrace of executed function getting to deadloc.
Make sure you get at least one SIP worker process backtrace (do kamctl
ps to see the type of kamailio process).
Cheers,
Daniel
On 3/24/10 4:29 PM, Zappasodi Daniele wrote:
Hi,
I have a big problem with fast lock mutex on arm CPU:
sometimes I find one or more children that go in an infinite loop, in
the while loop of get_lock function.
They remain in Run for long time (sometimes hours) and openser keeps
100% CPU.
Now I have changed the functions get_lock and tsl in order to obtain
more info and
I observe that (if and) when the process finish the cycle, it have
done a big amount of cycles.
This the log with my added info:
Mar 26 20:29:08 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle)
Mar 26 20:29:51 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle)
Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle)
Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:t_retransmit_reply:
MYTM lock
[process in loop]
Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 59374917
(cycle)
[after 4 minutes and 59374917 cycles, this is an example with a
"short" loop]
Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:t_retransmit_reply:
MYTM lock done
I'm not able to recognize the call flow that generates the loop (I
know only that it always happens when t_retransmit_reply calls
LOCK_REPLIES),
but I urgently need a work around to escape from the loop.
This is the get_lock function with my counter j:
inline static int get_lock(fl_lock_t* lock)
{
#ifdef ADAPTIVE_WAIT
int i=ADAPTIVE_WAIT_LOOPS;
int j=1;/* my change /
#endif
while(tsl(lock)){
#ifdef BUSY_WAIT
#elif defined ADAPTIVE_WAIT
j++;/* my change /
if (i>0) i--;
else sched_yield();
#else
sched_yield();
#endif
}
return(j);
}
Can I break the lock when my counter reaches a big value?
Should I call the unlock function after the break?
which value can be considered too big?
Thanks,
Daniele
**
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privileged. It is intended solely for the addressee. Access to this
message by anyone else is unauthorized. If you are not the intended
recipient, any disclosure, copying, or distribution of the message, or
any action or omission taken by you in reliance on it, is prohibited
and may be unlawful. Please immediately contact the sender if you have
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[Kamailio-Users] dinner in berlin, thursday, march 25, 19:00
Hello, couple of developers from SIP and VoIP area are in Berlin this week and plan to meet for dinner and beer Thursday, March 25, 19:00 at the traditional place by now: Lemke Brauhaus Luisenplatz 1, 10585 Berlin Across the corner with Charlottenburg Castle http://www.brauhaus-lemke.com/index.php?area=4 If you are around, just pop up. You can write me if you need more details about how to get there. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] VUC Podcast: The SIP Router Project
Hello, the recording of the VUC session last Friday about SIP Router Project is now available online at: http://www.voipusersconference.org/2010/kamailio3/ Here you find some details about who was in the call: http://www.kamailio.org/w/2010/03/vuc-listen-the-sip-router-project-podcast/ Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] GSoC 2010 - Call for students
Hello, if you are a student (or you know a student) interested in working with SIP Router over the summer, earning some money as well, please apply for GSoC Conferencing Presence support (or forward this email). The description of the project is done at: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 If you have technical details about the project please ask them on sr-...@lists.sip-router.org. For GSoC related questions, please address to: g...@sip-communicator.dev.java.net. A good FAQ for applicants is available at: http://www.sip-communicator.org/index.php/GSOC2010/HowToApply Application must be done directly to the google site, link provided in the FAQ. I will be in charge of mentoring this particular project, helping you as much as possible to understand the current presence server architecture and the core API of sip router. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Kamdbctl not installing and Install error
On 3/22/10 3:00 AM, Nathaniel L Keeling III wrote: I have successfully compliled and installed Kamailio 3.0 from git but I still do not have kamdbctl or kamctlrc installed in my install directory. Have they been replaced? no, the fix to makefile for opensolaris introduced a c&p issue in installing the utilities. Do a: git pull origin in source code tree and try again. Let me know if it ok now. Thanks, Daniel Thanks Nathaniel Daniel-Constantin Mierla wrote: Hello, On 3/21/10 12:23 AM, Nathaniel L Keeling wrote: I have installed Kamailio 3.0 but without db support. When I tried to add support for postgres, the install errors and the kamdbctl does not install in order to create the database tables. I am installing on Solaris 10 and have included the error from the install: Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/xlog.so ginstall -m 755 xlog.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' # other configs /bin/sh: syntax error at line 1: `;' unexpected gmake: *** [install-cfg] Error 2 did you installed from tarball or git? There was a fix for makefile system for solaris done after 3.0.0 release, installing from git should get it: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Next minor release will have it as well. Cheers, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Monitoring Upstream Carrier Health
Hello,
On 3/19/10 7:28 PM, Alex Balashov wrote:
You can log negative replies from a failure route, or put them in a
database, or issue an HTTP request.
On 03/19/2010 01:53 PM, Geoffrey Mina wrote:
Hello,
I am wondering if anyone has a clever way to remotely monitor a Kamailio
1.5 server. I am not looking for the standard monitoring, what I am
looking to achieve is catching situations where my upstream carrier is
having problems. We have a certain level of 404, 500, 503 errors
throughout the day which are not indicative of a major carrier problem.
I want to be able to monitor the ratio of properly setup calls to failed
setups - so I can know when a carrier is having issues and is responding
with many 503 errors.
Any push in the right direction would be greatly appreciated.
if you have a fixed number of carriers and use snmp, then you can define
some statistics.
I use htable with 3.0 and dump them via mi/rpc command. I do not need to
have persistence, otherwise I would use database.
Usually at the beginning of production I catch every reply in
onreply_route and have classes of replies per carrier in hashtable, kind of
if(status=~"4[0-9][0-9]")
{
$sht(ht=>$si::4xx) = $sht(ht=>$si::4xx) + 1;
}
Dumping the content via rpc from time to time for analysis, which is
good for checking after restarts.
Cheers,
Daniel
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Re: [Kamailio-Users] Quick RTPproxy question
Hello,
On 3/19/10 10:12 PM, Uriel Rozenbaum wrote:
Hi guys,
I have some easy doubt about nathelper functions using RTPProxy.
I'm trying to bridge from an external IP to an internal IP.
The start-line for rtpproxy is: rtpproxy -l PUBLIC_IP/PRIVATE_IP -s
udp:127.0.0.1:7999 <http://127.0.0.1:7999> -F
It starts OK and I see it when kamailio starts.
I'm going to use something like Daniel showed on some other mail:
if(dst_ip==private)
force_rtp_proxy("ocfaei");
else
force_rtp_proxy("ocfaei");
if i have the invite from a public IP to someone in private on the
request I'll run
force_rtp_proxy("ocfaei");
then the reply will be from private to public... should I run
"force_rtp_proxy("ocfaei");"? or should it be the same?
I can see only same series of flags in all your calls. You need to use
..ie in some cases. IIRC, for invite and reply you should use same
sequence. Good part is that you have only two options :-) .
Cheers,
Daniel
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Re: [Kamailio-Users] Kamdbctl not installing and Install error
Hello, On 3/21/10 12:23 AM, Nathaniel L Keeling wrote: I have installed Kamailio 3.0 but without db support. When I tried to add support for postgres, the install errors and the kamdbctl does not install in order to create the database tables. I am installing on Solaris 10 and have included the error from the install: Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/xlog.so ginstall -m 755 xlog.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' # other configs /bin/sh: syntax error at line 1: `;' unexpected gmake: *** [install-cfg] Error 2 did you installed from tarball or git? There was a fix for makefile system for solaris done after 3.0.0 release, installing from git should get it: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Next minor release will have it as well. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] Reminder: VUC Today, SIP Router Project
Hello, a kind reminder for today's audio conference scheduled for 16:00GMT. Check the start time for your zone at: http://vuc.me/next After presenting the achievements so far within SIP Router, with what is new in 3.0.0 release and development version, you have an unique opportunity to ask questions to: - Andrei Pelinescu-Onciul, the creator of SIP Express Router (SER), the architect behind the core (transport layers, memory management, asynchronous processing, timers, etc), who will be also able to answer anything from project's history started in 2001 - Alex Balashov, Kamailio management team member, experienced consultant in building large SIP platforms - myself, as co-founder of Kamailio (OpenSER) You can join via irc on #vuc (note that some of us started to hang out on #sip-router) channel at irc.freenode.net. Dialing to VUC is possible via sip, skype, pstn, web page and more, see: http://vuc.me Five hours to go right now, hear you then! Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] Kamailio - SIP Router on GSoC 2010
Hello, I have the pleasure to announce that our project has an accepted application for Google Summer of Code 2010. The proposed application was done within SIP Communicator Organization and is related to extending the presence server for conference calls notifications. See more: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 http://www.sip-communicator.org/index.php/Development/Gsoc2010 The friends from SEMS project will be part as well, with separate application, implementing the audio mixer part. More details will be published soon. Start thinking about good candidates (they must be students (college or university)), we need them to get the project accepted for implementation. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] uac_replace_from unexpected behavior
Hello,
On 03/16/2010 03:05 PM, Brandon Armstead wrote:
Daniel,
I did do a check and "" quotes are printing in xlog for $fn.
can you send the sip trace for such case? I tested with options and ""
is removed.
Cheers,
Daniel
Thanks!
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 7:32 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 01:22 PM, Brandon Armstead wrote:
Daniel,
So if I am to set it to "none" it should give me the desired
affects, and not alter back to the original From header upon
transmission of an ACK?
auto mode should do everything (update/restore From (or To)) for
within dialog requests, if you used uac_replace_from() for initial
INVITE, therefore this is the best mode. However, it adds an extra
parameter (pretty long) to RR header and some UA strips it when
building the reply.
If you know you are in a SIP2.0 (rfc3261) compatible environment,
then you can use other modes.
In sip 2.0 a dialog is identified by call-id, from-tag and to-tag,
which are not affected by From updates. However, in previous
version of sip, From URI and To URI were used to identify the sip
dialog, therefore, in order to be compatible with sip 1.0 then you
should not change From/To.
In auto mode, the From/To are restored to be safe with sip 1.0
devices.
Btw, if you have time, can you please print the $fn in xlog for
ACK and send it here? Will show if quotes are considered part of
display name. If not, I will look later in sources.
Cheers,
Daniel
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 7:05 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
On 03/16/2010 01:03 PM, Brandon Armstead wrote:
Value of uac_restore_mode is not set so "auto".
but if it is not set to something else, this is the default
value.
Cheers,
Daniel
Thanks!
On Tue, Mar 16, 2010 at 7:00 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 12:52 PM, Brandon Armstead wrote:
Daniel,
This is 1.5 and there is nothing between the quotes "".
the question is whether the display name attribute in
From header structure includes the quotes or not -- this
is to be revealed by code.
The last time I tried to remove_hf, and then
append_hf(From) or To header, it seemed to break call
flow completely?
It can break in case you have non-RFC3261 compliant devices.
What is the value of uac module parameter from_restore_mode?
If it is auto or not set, then it is not the same
behavior as with remove_hf/append_hf.
I will give it another go, however if you have any
further thoughts it is much appreciated, thanks!
Going to check the sources and come back with more details.
Cheers,
Daniel
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 6:44 AM, Daniel-Constantin
Mierla mailto:mico...@gmail.com>>
wrote:
Hello,
On 03/16/2010 02:30 AM, Brandon Armstead wrote:
Hello,
As always thank you ahead of time for your help
and input!
I am currently calling uac_replace_from("", "")
in effort to "leave uri" and "toss away display
name"
Which does seem to work... for the initial
INVITE
However upon receiving an ACK with an empty
display, however "" <- quotations, it does not
clear the display "" which is causing issues
with one of my upstream vendors.
Example / Scenario:
From: ""
Expected Result upon uac_replace_from("",""):
From:
Current Result: From: ""
As you can see it is not stripping the "" empty
display quotes.
Any thoughts / ideas / suggestions to get my
desired affect?
could be that display name is set to empty string
(what is between double quotes) and in this case is
nothing to replace -- I have to doublecheck the
sources. Is it 1.5 or 3.0?
Are you using From auto-replacing mode? If not, a
Re: [Kamailio-Users] Rtpproxy/Kamailio modification to support highcapacity encryption, transcoding
Hi Jeff, On 03/17/2010 09:04 PM, Jeff Brower wrote: [...] We're making initial modifications to rtpproxy to support high channel capacity transcoding and encryption. At this point we want to get some general idea of the scope of changes needed for rtpproxy and Kamailio... so we're starting with small steps. We've been studying rtpproxy source and our current thinking is to add a sub-structure to the existing rtpp_session structure (defined in rtpp_session.h). The new sub-structure would include: -encryption options (type, key length, salt size, type of key mgt protocol, etc) -encode / decode options (type, VAD/CNG, VIF size, etc) Any comments or advice on this approach appreciated. Not being a rtpproxy developer at all, I do not see a problem with the approach. Also note that rtpproxy is a single process application (or used to be in case last version changed), take that in consideration when designing. Not sure whether to start a separate thread, but also there is the issue of what changes are necessary to Kamailio to support sending updated commands to rtpproxy. Would modifying Nathelper alone be sufficient? Just updating the nathelper is sufficient in kamailio. Another idea I was playing with in the past, but time was limited, was to enhance sems (sip express media server) to support communication via nathelper-rtpproxy protocol. sems is lightweight sip media server, supporting already transcoding. Not being a rtp/audio guy, my plan was to use an existing audio mixer. Yes SEMS looks good. We've been talking with Stefan Sayer. Right now i use routing via a media server when needing transconding, so call flow is: [caller] [kamailio] -- [media server: asterisk/freeswitch/sems] - [kamailio] -- [caller] With the above scenario, the issue for us is channel capacity -- Asterisk can't do a lot of transcoding, and even if the TC400B card is used, it turns into a "PCIe bus slugfest" as all speech/packet data has to go back and forth. Asterisk + Linux kernel still have to "touch" every RTP packet. Also the TC400B can't do encryption, conferencing, etc. Other hardware can do 100s or even 1000s of channels, so it seems to make sense to enhance rtpproxy, at least at this point. For an Asterisk-centric approach, one way may be to enhance native bridging (canreinvite=yes). We're looking at ways to spoof SDP negotiation so both ends think they have the same media (codec) capabilities, even if not actually the case. Then exchange RTP through a card with its own GbE that does the transcoding (or other required functionality). The advantage in this case would be "keep it simple": add a card to Asterisk, some external software, and capacity is enhanced. Advantages such as "included with the chip" Texas Inst licensing might also be useful. when coming to transconding, capacity is indeed a problem. I agree that some of the existing solutions are consuming resources and would be good to have a lighweight application that takes care only of this job, removing the overhead of dealing with signaling, allocating channels, etc. I can help you assisting with kamailio code, which is nathelper module related. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] SIP Router Project of VUC, March 19
Hello, please note an update to the starting time of VUC conference - it is 16:00 GMT (09:00 US Pacific, 17:00 Berlin, Paris). The reference is 09:00 US Pacific and USA entered the daylight saving time zone, resulting in starting time being one hour earlier for some time zones (for Europe at least) than what was advertised so far on VUC and project sites. We updated the web pages as well. You can get the time for your zone following the link: http://vuc.me/next Hear you tomorrow on VUC! Daniel On 03/16/2010 02:39 PM, Daniel-Constantin Mierla wrote: Hello, this Friday, March 19, late afternoon, the weekly VoIP User Conference is hosting a session about SIP Router project. My goals are to present the achievements so far within SIP Router projects, what is new in Kamailio 3.0 release and plans for the future. More details can be found at: http://www.kamailio.org/w/vuc-the-sip-router-project/ You can join the audio conference via sip, skype, pstn line or other several options presented on http://vuc.me site. There is a irc channel available for it: #vuc on irc.freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] kamailio logo artwork
Hello, courtesy of Elio Rojano (http://www.sinologic.net), very nice logo artworks are available now, using openser or sip-router thematics. You can browse at: http://www.asipto.com/gallery/v/ksr-artwork/ Available for download as well at: http://www.kamailio.org/pub/ksr-artwork/ If you are good at graphic design and have new ideas, please submit. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Is Timestamp supported in kamailio?
Hello, On 03/16/2010 07:03 PM, Iñaki Baz Castillo wrote: 2010/3/16 Alex Balashov: Sure. What I meant is that apart from $Ts in K>= 3.0.x, there aren't really any ways to compute the delay on that level of time resolution. Are there? Yes, right. I did something similar long time ago and just got seconds precision :) one of the old function in textops is append_time() which adds a date header, with complete date and time, still up to second precision. http://kamailio.org/docs/modules/3.0.x/modules_k/textops.html#id2494947 It is true that before 3.0.0 there was no script variable returning better time precision than second, devel has it as timeval variable: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#timeval But there are options to do it in case you really need better precision, using sql query, exec or perl. Not only those, because benchmark has a nicer way to get the difference of time for config execution: http://kamailio.org/docs/modules/stable/modules_k/benchmark.html#id2521780 Then adding a header to a local generated reply is easy. Cheers, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Rtpproxy/Kamailio modification to support high capacity encryption, transcoding
Hello, On 03/16/2010 10:33 PM, Vikram Ragukumar wrote: Hello, We're making initial modifications to rtpproxy to support high channel capacity transcoding and encryption. At this point we want to get some general idea of the scope of changes needed for rtpproxy and Kamailio... so we're starting with small steps. We've been studying rtpproxy source and our current thinking is to add a sub-structure to the existing rtpp_session structure (defined in rtpp_session.h). The new sub-structure would include: -encryption options (type, key length, salt size, type of key mgt protocol, etc) -encode / decode options (type, VAD/CNG, VIF size, etc) Any comments or advice on this approach appreciated. Not being a rtpproxy developer at all, I do not see a problem with the approach. Also note that rtpproxy is a single process application (or used to be in case last version changed), take that in consideration when designing. Not sure whether to start a separate thread, but also there is the issue of what changes are necessary to Kamailio to support sending updated commands to rtpproxy. Would modifying Nathelper alone be sufficient? Just updating the nathelper is sufficient in kamailio. Another idea I was playing with in the past, but time was limited, was to enhance sems (sip express media server) to support communication via nathelper-rtpproxy protocol. sems is lightweight sip media server, supporting already transcoding. Not being a rtp/audio guy, my plan was to use an existing audio mixer. Right now i use routing via a media server when needing transconding, so call flow is: [caller] [kamailio] -- [media server: asterisk/freeswitch/sems] - [kamailio] -- [caller] rtpproxy is no longer used, since media servers support comedia extension. But a kamailio-mediaserver control protocol will reduce the signaling path. Cheers, Daniel Thanks and Regards, Vikram. PS : I'm posting on Kamailio's mailing list because it seems that both Kamailio and rtpproxy developers closely follow this list. ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] [OT] MSN presence flow
On 03/17/2010 01:40 PM, Iñaki Baz Castillo wrote: 2010/3/17 Daniel-Constantin Mierla: Hello, On 03/17/2010 11:19 AM, Iñaki Baz Castillo wrote: Hi, does somebody have a MSN protocol flow related to presence rules or buddies management? This is, I would like to know how MSN protocol imlements some tasks as: - Adding a buddy. - Blocking a buddy for presence. - Blocking a contact (not a buddy) for presence. no diagram. In the past (well, 2003-2004) I checked and not much was out. Now seems that wikipwdia has good resources: http://en.wikipedia.org/wiki/Microsoft_Notification_Protocol From there I got to: http://www.hypothetic.org/docs/msn/notification/presence.php Great, I'll take a look at it. btw, maybe libpurple (used by pidgin) has some docs as well. If not, then the source code is a last resort. Cheers, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] kamailio crashes
Hello Panagiotis, getting the at least the call-ind in headers of sip reply would be good, do it by chunks (I do not know other option), in gdb: $ frame 7 $ print *(buf+100) $ print *(buf+200) $ print *(buf+300) $ print *(buf+400) Then more or less same to get the request: $ frame 2 $ print *(req->buf) $ print *(req->buf+100) $ print *(req->buf+200) $ print *(req->buf+300) $ print *(req->buf+400) $ frame 1 $ print *hf Make sure you pick a core that gives a backtrace like: (gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x0047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x2ad23fe3e525 in run_trans_callbacks (type=out>, trans=, req=0x2ad2432dcf58, rpl=0x772d28, code=) at sip_msg.h:54 #3 0x2ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=) at t_lookup.c:888 #4 0x2ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x79c016bc) at t_lookup.c:964 #5 0x2ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x0041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x00445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x79c017a0) at receive.c:212 #8 0x004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x0042760e in main (argc=3, argv=0x79c019b8) at main.c:774 Would be good if we get on irc together, if you have time, to do a more realtime debugging -- will be faster. Let me know if you can do it today. I am on #kamailio or #sip-router channels on irc.freenode.net with id miconda. Thanks, Daniel On 03/17/2010 11:56 AM, Panagiotis Skoulikaritis wrote: Hello Daniel I do have quite a few core files, please send me the gdb commands. Regards Panagiotis. ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] [OT] MSN presence flow
Hello, On 03/17/2010 11:19 AM, Iñaki Baz Castillo wrote: Hi, does somebody have a MSN protocol flow related to presence rules or buddies management? This is, I would like to know how MSN protocol imlements some tasks as: - Adding a buddy. - Blocking a buddy for presence. - Blocking a contact (not a buddy) for presence. no diagram. In the past (well, 2003-2004) I checked and not much was out. Now seems that wikipwdia has good resources: http://en.wikipedia.org/wiki/Microsoft_Notification_Protocol From there I got to: http://www.hypothetic.org/docs/msn/notification/presence.php I haven't looked further, though. Cheers, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] SIP Router Project of VUC, March 19
Hello, this Friday, March 19, late afternoon, the weekly VoIP User Conference is hosting a session about SIP Router project. My goals are to present the achievements so far within SIP Router projects, what is new in Kamailio 3.0 release and plans for the future. More details can be found at: http://www.kamailio.org/w/vuc-the-sip-router-project/ You can join the audio conference via sip, skype, pstn line or other several options presented on http://vuc.me site. There is a irc channel available for it: #vuc on irc.freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] [dialog module] dialog is not terminated when no ACK is received
On 03/16/2010 01:36 PM, Iñaki Baz Castillo wrote: 2010/3/16 Daniel-Constantin Mierla: I agree that state 3 should stay no longer than 32sec, normally this should be clear by caller sending BYE due to non-ACK. You said you are not doing record routing, how the bye gets then to the proxy so the dialog is cleared from memory? Humm no, I didn't mean it, I'm using record route (I just said that without record route dialog module makes no sense). sorry, I misunderstood. The issues are two: 1) INVITE, 200 but no ACK received. The dialog remains in state 3 for dialog module default_timeout value (long time usually). IMHO as no ACK is received the dialog should be deleted after 32 seconds (the time the TM module waits for the ACK). But isn't there a BYE coming from callee after 32sec? Callee should end the dialog in its side if no ACK is received. 2) When the INVITE transaction is terminated by a final [3456]XX response, the dialog remains in memory in state 5 for ~4 seconds. I've inspected the code and couldn't find a timer or whatever that could make the dialog information to be kept for such time. Not in dialog module, but it is tm module. IIRC, the last reference to such dialog is destroyed when the transaction is deleted from memory. That should happen aprox 2sec after the transaction is completed and reply sent. To be sure it is this one, you can play with tm parameter delete_timer or so. I am going to dig in more as well. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] uac_replace_from unexpected behavior
Hello,
On 03/16/2010 01:22 PM, Brandon Armstead wrote:
Daniel,
So if I am to set it to "none" it should give me the desired
affects, and not alter back to the original From header upon
transmission of an ACK?
auto mode should do everything (update/restore From (or To)) for within
dialog requests, if you used uac_replace_from() for initial INVITE,
therefore this is the best mode. However, it adds an extra parameter
(pretty long) to RR header and some UA strips it when building the reply.
If you know you are in a SIP2.0 (rfc3261) compatible environment, then
you can use other modes.
In sip 2.0 a dialog is identified by call-id, from-tag and to-tag, which
are not affected by From updates. However, in previous version of sip,
From URI and To URI were used to identify the sip dialog, therefore, in
order to be compatible with sip 1.0 then you should not change From/To.
In auto mode, the From/To are restored to be safe with sip 1.0 devices.
Btw, if you have time, can you please print the $fn in xlog for ACK and
send it here? Will show if quotes are considered part of display name.
If not, I will look later in sources.
Cheers,
Daniel
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 7:05 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
On 03/16/2010 01:03 PM, Brandon Armstead wrote:
Value of uac_restore_mode is not set so "auto".
but if it is not set to something else, this is the default value.
Cheers,
Daniel
Thanks!
On Tue, Mar 16, 2010 at 7:00 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 12:52 PM, Brandon Armstead wrote:
Daniel,
This is 1.5 and there is nothing between the quotes "".
the question is whether the display name attribute in From
header structure includes the quotes or not -- this is to be
revealed by code.
The last time I tried to remove_hf, and then append_hf(From)
or To header, it seemed to break call flow completely?
It can break in case you have non-RFC3261 compliant devices.
What is the value of uac module parameter from_restore_mode?
If it is auto or not set, then it is not the same behavior as
with remove_hf/append_hf.
I will give it another go, however if you have any further
thoughts it is much appreciated, thanks!
Going to check the sources and come back with more details.
Cheers,
Daniel
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 6:44 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 02:30 AM, Brandon Armstead wrote:
Hello,
As always thank you ahead of time for your help and
input!
I am currently calling uac_replace_from("", "") in
effort to "leave uri" and "toss away display name"
Which does seem to work... for the initial INVITE
However upon receiving an ACK with an empty display,
however "" <- quotations, it does not clear the
display "" which is causing issues with one of my
upstream vendors.
Example / Scenario:
From: ""
Expected Result upon uac_replace_from("",""):
From:
Current Result: From: ""
As you can see it is not stripping the "" empty
display quotes.
Any thoughts / ideas / suggestions to get my desired
affect?
could be that display name is set to empty string (what
is between double quotes) and in this case is nothing to
replace -- I have to doublecheck the sources. Is it 1.5
or 3.0?
Are you using From auto-replacing mode? If not, a
solution for now is to do From update using header
manipulation functions:
remove_hf("From");
append_hf("From: <$fu>;tag=$ft\r\n", "From");
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
*http://www.asipto.com/index.php/sip-router-masterclass/
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
*http://www.asipto.com/index.php/sip-router
Re: [Kamailio-Users] [dialog module] dialog is not terminated when no ACK is received
Hello Inaki, On 03/04/2010 04:51 PM, Iñaki Baz Castillo wrote: El Jueves, 4 de Marzo de 2010, Daniel-Constantin Mierla escribió: Hello, On 03/04/2010 04:17 PM, Iñaki Baz Castillo wrote: El Miércoles, 3 de Marzo de 2010, Iñaki Baz Castillo escribió: Hi, using Kamailio 1.5.4. I use dlg_manage() for an INVITE. 200 Ok is replied by the callee but the UAC doesn't send the ACK (due to a crash). The dialog remains in Kamailio dialog memory/table in state 3 and would expire after default_timeout (which usually is 3600 seconds or more, unsuitable for this case). Yes, it could occur that the proxy is not doing loose_routing, but in that case it doesn't make sense to use dialog module, so shouldn't dialog module expire dialogs in state 3 after ~32 seconds? I'm using profiles_with_value to limit the number of calls per user, so this issue is a bit important, as a UAC not sending the ACK for a 200 means one less available channel for this user during dialog module default_timeout. Another issue: when a call is cancelled the dialog remains in memory for ~4 seconds more. This is: - INVITE received and any provisional response => dialog in state 2. - CANCEL received => dialog in state 5 for ~4 seconds. Is there any reason for that? there might be a detele timer delay. I've checked and the same occurs when the INVITE transaction is terminated with a final [3456]XX response. In any case it remains in memory in state 5 for ~4 seconds. took me a bit to get back to this one. From what you describe this seems to be related to transaction life in delete state. If the dialog is not answered, it is not inserted in dialogs list. It is kept attached to invite transaction. I agree that state 3 should stay no longer than 32sec, normally this should be clear by caller sending BYE due to non-ACK. You said you are not doing record routing, how the bye gets then to the proxy so the dialog is cleared from memory? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] uac_replace_from unexpected behavior
On 03/16/2010 01:03 PM, Brandon Armstead wrote:
Value of uac_restore_mode is not set so "auto".
but if it is not set to something else, this is the default value.
Cheers,
Daniel
Thanks!
On Tue, Mar 16, 2010 at 7:00 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 12:52 PM, Brandon Armstead wrote:
Daniel,
This is 1.5 and there is nothing between the quotes "".
the question is whether the display name attribute in From header
structure includes the quotes or not -- this is to be revealed by
code.
The last time I tried to remove_hf, and then append_hf(From) or
To header, it seemed to break call flow completely?
It can break in case you have non-RFC3261 compliant devices.
What is the value of uac module parameter from_restore_mode?
If it is auto or not set, then it is not the same behavior as with
remove_hf/append_hf.
I will give it another go, however if you have any further
thoughts it is much appreciated, thanks!
Going to check the sources and come back with more details.
Cheers,
Daniel
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 6:44 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 02:30 AM, Brandon Armstead wrote:
Hello,
As always thank you ahead of time for your help and input!
I am currently calling uac_replace_from("", "") in effort
to "leave uri" and "toss away display name"
Which does seem to work... for the initial INVITE
However upon receiving an ACK with an empty display,
however "" <- quotations, it does not clear the display
"" which is causing issues with one of my upstream vendors.
Example / Scenario:
From: ""
Expected Result upon uac_replace_from("",""):
From:
Current Result: From: ""
As you can see it is not stripping the "" empty display
quotes.
Any thoughts / ideas / suggestions to get my desired affect?
could be that display name is set to empty string (what is
between double quotes) and in this case is nothing to replace
-- I have to doublecheck the sources. Is it 1.5 or 3.0?
Are you using From auto-replacing mode? If not, a solution
for now is to do From update using header manipulation functions:
remove_hf("From");
append_hf("From: <$fu>;tag=$ft\r\n", "From");
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
*http://www.asipto.com/index.php/sip-router-masterclass/
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
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Re: [Kamailio-Users] uac_replace_from unexpected behavior
Hello,
On 03/16/2010 12:52 PM, Brandon Armstead wrote:
Daniel,
This is 1.5 and there is nothing between the quotes "".
the question is whether the display name attribute in From header
structure includes the quotes or not -- this is to be revealed by code.
The last time I tried to remove_hf, and then append_hf(From) or To
header, it seemed to break call flow completely?
It can break in case you have non-RFC3261 compliant devices.
What is the value of uac module parameter from_restore_mode?
If it is auto or not set, then it is not the same behavior as with
remove_hf/append_hf.
I will give it another go, however if you have any further thoughts
it is much appreciated, thanks!
Going to check the sources and come back with more details.
Cheers,
Daniel
Sincerely,
Brandon Armstead
On Tue, Mar 16, 2010 at 6:44 AM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
On 03/16/2010 02:30 AM, Brandon Armstead wrote:
Hello,
As always thank you ahead of time for your help and input!
I am currently calling uac_replace_from("", "") in effort to
"leave uri" and "toss away display name"
Which does seem to work... for the initial INVITE
However upon receiving an ACK with an empty display, however
"" <- quotations, it does not clear the display "" which is
causing issues with one of my upstream vendors.
Example / Scenario:
From: ""
Expected Result upon uac_replace_from("",""):From:
Current Result: From: ""
As you can see it is not stripping the "" empty display quotes.
Any thoughts / ideas / suggestions to get my desired affect?
could be that display name is set to empty string (what is between
double quotes) and in this case is nothing to replace -- I have to
doublecheck the sources. Is it 1.5 or 3.0?
Are you using From auto-replacing mode? If not, a solution for now
is to do From update using header manipulation functions:
remove_hf("From");
append_hf("From: <$fu>;tag=$ft\r\n", "From");
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
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Re: [Kamailio-Users] uac_replace_from unexpected behavior
Hello,
On 03/16/2010 02:30 AM, Brandon Armstead wrote:
Hello,
As always thank you ahead of time for your help and input!
I am currently calling uac_replace_from("", "") in effort to "leave
uri" and "toss away display name"
Which does seem to work... for the initial INVITE
However upon receiving an ACK with an empty display, however "" <-
quotations, it does not clear the display "" which is causing issues
with one of my upstream vendors.
Example / Scenario:
From: ""
Expected Result upon uac_replace_from("",""):From:
Current Result: From: ""
As you can see it is not stripping the "" empty display quotes.
Any thoughts / ideas / suggestions to get my desired affect?
could be that display name is set to empty string (what is between
double quotes) and in this case is nothing to replace -- I have to
doublecheck the sources. Is it 1.5 or 3.0?
Are you using From auto-replacing mode? If not, a solution for now is to
do From update using header manipulation functions:
remove_hf("From");
append_hf("From: <$fu>;tag=$ft\r\n", "From");
Cheers,
Daniel
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Re: [Kamailio-Users] re : openserctl statistics..
Hello, On 03/13/2010 07:08 PM, Jignesh Gandhi wrote: Hello, What does the following stat mean when you run the " openserctl moni " sl: received_ACKs = I did some digging and I got the following explanation... But am not sure if this is an error in operser or an error on a message received by openser or what? " sl: received_ACKs - number of received ACKs due sending negative replies. " when you send a negative reply (code >=300), the calling party sends an ACK. This ack is filtered by sl module (does not get in config file). This statistic counts them. Typical case is authentication of calls or registration. The auth modules use sl to send back the challenge (401 or 408 replies). Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] maintenance this afternoon
Hello, On 03/15/2010 11:52 AM, Daniel-Constantin Mierla wrote: Hello, some small maintenance tasks will be performed afternoon today to the server hosting the mailing lists and kamailio.org site, if you notice some downtime, try again a bit later. hopefully now mailing lists are back online completely. exim4 update proved to be longer than expected due to config incompatibility. If you discover problems with mailing lists, please write me. Many thanks, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] maintenance this afternoon
Hello, some small maintenance tasks will be performed afternoon today to the server hosting the mailing lists and kamailio.org site, if you notice some downtime, try again a bit later. Thanks, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] kamailio crashes
sername so we can use different
prefixes
$avp(s:user) = $rU;
# select destination from first group
if(ds_select_domain("$avp(s:dstgrp)",
"4"))
{
if($(ru{uri.param,prefix})!=null)
{
$ru =
"sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd;
} else {
$ru =
"sip:" + $avp(s:user) + "@" + $rd;
}
}
$avp(s:dstgrp) = null;
xlog("alx --- The final
RURI is $ru --- ");
if($avp(s:port_translation) == 1)
{
rewriteport("5061");
}
t_on_failure("3");
t_relay();
exit;
}
}
}
Attached is the trace
Regards.
P.
marius zbihlei wrote:
Panagiotis Skoulikaritis wrote:
Hello Daniel
the kamailio version is 1.5.3
Regards
P.
Hello,
Can you give us more details like the sip message that generates the
coredump (or if every sip message received generates the core), if
your config does something more out of the ordinary(let's say
exotic). Can we reproduce it ?
It would also be helpful if you specify the list of modules you have
loaded.
Cheers,
Marius
Daniel-Constantin Mierla wrote:
Hello,
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
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Re: [Kamailio-Users] Loadbalancer/outbound_proxy and ACK problem
Hello, On 03/13/2010 10:11 AM, Pavel Miskov wrote: Hello Daniel, that was it. I was calling fix_nated_contact in LB but also in onreply_route in REG+PROXY. Thank you very much! great you fixed, welcome! Daniel Pavel On Fri, Mar 12, 2010 at 4:21 PM, Daniel-Constantin Mierla wrote: Hello, very likely you use fix natted contact in reg+proxy. You have to use that in load balancer, since that is the instance that sees the right public ip address for UA. reg+proxy will see IP address of LB as being source IP address. Cheers, Daniel On 03/12/2010 12:15 PM, Pavel Miskov wrote: Hello Inaki, thanks for replying and here is more readable form taken from LB: Pavel # U +0.00 UAC_A_PUB_IP:31488 ->LB_IP:5678 INVITE sip:ua...@test.com SIP/2.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-a9127351d94a483a-1---d8754z-;rport. Max-Forwards: 70. Contact:. To: "UAC_B". From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="UAC_A",realm="test.com",nonce="4b9a102f00030feed796808fa4eb1df2de3c9a2b1034",uri="sip:ua...@test.com",response="2f023b974dd4cde6f29387681168de8f",algorithm=MD5. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 368. # U +0.004000 LB_IP:5678 ->UAC_A_PUB_IP:31488 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-a9127351d94a483a-1---d8754z-;rport=31488;received=UACs_PUB_IP. To: "UAC_B". From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 INVITE. Server: Kamailio (1.5.1-tls (x86_64/linux)). Content-Length: 0. . # U +0.00 LB_IP:5678 ->REG_PROXY_IP:5166 INVITE sip:ua...@test.com SIP/2.0. Record-Route:. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.a8d68282.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-a9127351d94a483a-1---d8754z-;rport=31488. Max-Forwards: 69. Contact:. To: "UAC_B". From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="UAC_A",realm="test.com",nonce="4b9a102f00030feed796808fa4eb1df2de3c9a2b1034",uri="sip:ua...@test.com",response="2f023b974dd4cde6f29387681168de8f",algorithm=MD5. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 368. Path:. . # U +0.00 REG_PROXY_IP:5166 ->LB_IP:5678 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.a8d68282.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-a9127351d94a483a-1---d8754z-;rport=31488. To: "UAC_B". From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 INVITE. Server: Kamailio (1.5.1-tls (x86_64/linux)). Content-Length: 0. . # U +0.00 REG_PROXY_IP:5166 ->LB_IP:5678 INVITE sip:ua...@uac_b_priv_ip:31468;rinstance=06b43c2b0e1ae81a SIP/2.0. Record-Route:. Record-Route:. Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bK3ae9.ff046b92.0. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.a8d68282.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-a9127351d94a483a-1---d8754z-;rport=31488. Route:. Max-Forwards: 68. Contact:. To: "UAC_B". From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 384. Path:. . # U +0.032000 LB_IP:5678 ->REG_PROXY_IP:5166 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bK3ae9.ff046b92.0;rport=5166;received=REG_PROXY_IP. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.a8d68282.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-a9127351d94a483a-1---d8754z-;rport=31488. To: "UAC_B". From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 INVITE. Server: Kamailio (1.5.1-tls (x86_64/linux)). Content-Length: 0. . # U +0.00 LB_IP:5678 ->UAC_B_PUB_IP:31468 INVITE sip:ua...@uac_b_priv_ip:31468;rinstance=06b43c2b0e1ae81a SIP/2.0. Record-Route:. Record-Route:. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.b8d68282.0. Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bK3ae9.ff046b92.0. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.a8d68282.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9h
Re: [Kamailio-Users] What is the best module for routing INVITE based on DID in R-URI?
On 03/13/2010 02:30 AM, Geoffrey Mina wrote:
Thanks, that worked perfectly. The PDT module seems to be a solid
solution for what I'm trying to accomplish. I just need a mapping for
inbound DIDs... i suppose I could have used htable, but this seems a
little cleaner.
pdt uses internally a tree which is better to index dids (very fast
lookup). htable can be used, but is more appropriate when the key is
alphanumeric.
Cheers,
Daniel
On Fri, Mar 12, 2010 at 5:40 PM, Daniel-Constantin Mierla
wrote:
Hello,
On 03/12/2010 10:36 PM, Geoffrey Mina wrote:
Sorry to bring up an old thread here... but I have finally gotten
around to implementing your PDT suggestion. I have a table which
looks like:
source | prefix | domain |
* | 551212 | 192.168.200.1
* | 551213 | 192.168.200.1
* | 551214 | 192.168.200.1
The calls are coming in as: RURI=sip:551...@192.168.200.0:5060 and
I am using the pdt module by calling with prefixdomain("2","0") so
that we aren't actually stripping out any of the URI, I am just
matching to a domain and rewriting. In my logs I am seeing the
following and I am unable to route to anything but the first.
Mar 12 16:29:13 cust-sipgateway1 /usr/local/sbin/kamailio[9157]:
ERROR:pdt:pdt_load_db: sdomain [*]: prefix [551213] or domain
<192.168.200.1>duplicated
Mar 12 16:29:13 cust-sipgateway1 /usr/local/sbin/kamailio[9157]:
ERROR:pdt:pdt_load_db: sdomain [*]: prefix [551214] or domain
<192.168.200.1>duplicated
Any ideas? It would be a shame if I had to scrap this plan... as it
works so nicely with only a single prefix/domain! :)
just disable domain duplication checking:
http://kamailio.org/docs/modules/stable/modules_k/pdt.html#id2533886
First version of this module had the constraint of one prefix-domain
relation (like with unique prefix for each country) and the parameter
controls the backward compatibility.
Cheers,
Daniel
On Thu, Jan 21, 2010 at 3:58 AM, Daniel-Constantin Mierla
wrote:
On 1/20/10 12:54 AM, Geoffrey Mina wrote:
Thanks for the idea. Ill have lots of these, so If you wouldn't mind,
could you elaborate a bit on using ENUM in kamailio.
P.s. I'm on 1.5
enum implementation is pretty mature, without relevant changes since 1.3
or
so.
Regarding enum, practically is about storing relation between numbers and
sip addresses in DNS server and you query the DNS server each time you
get a
call. Is good if you are familiar with dns servers. For more, you can
start
from here:
http://en.wikipedia.org/wiki/Telephone_Number_Mapping
Cheers,
Daniel
Thanks
On 1/19/10, Andreas Sikkema wrote:
On Jan 19, 2010, at 8:34 PM, Geoffrey Mina wrote:
I am putting up a Kamailio server which will do nothing but route
INVITE requests from my upstream carrier to individual offices on my
side. The office locations will NOT be registered SIP UAs, but other
Kamailio proxy servers. What I want to have is a database of DIDs
associated with a forwarding IP:Port and/or SRV records.
551212 ==> 1.2.3.4:5060
551213 ==> 1.2.3.5:5060
If I had a reasonable amount of relations like this so that maintenance
by
hand would be an issue I'd try to find an ENUM setup that was easily
manageable. Point an ENUM address to a trusted peer and you're done.
--
Andreas
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Re: [Kamailio-Users] need help to use the Kamailio register to remote sip proxy
Hello, On 03/09/2010 03:45 PM, Cucku Cucku wrote: Hi Marius its very helpfull for me so i have to wait for the next release of sip-router/kamailio :) you can use it straightaway. I don't recall any change that should affect it in core or other modules (of if it is should be minor), so you can just copy the module in 3.0 tree, create the table, add records and roll the service. Cheers, Daniel Thank you Từ: marius zbihlei Chủ đề: Re: [Kamailio-Users] need help to use the Kamailio register to remote sip proxy Đến: "Cucku Cucku" Cc: users@lists.kamailio.org Ngày: Thứ Ba, 9 tháng 3, 2010, 10:01 Cucku Cucku wrote: > Hi all > Hello > i found the Kamailio supports register to remote sip proxy: > http://www.mail-archive.com/users@lists.kamailio.org/msg07451.html > it mentions to use : > SQL to create the mysql table is in utils/kamctl/mysql/uac-create.sql > > Quote from the mail you referred : "if you follow the sr-dev mailing list, you may have noticed some new features added in master branch (for the 3.1.0). I will send more details about each, now: remote user registration." So it looks like this is a 3.1.0 feature not a 3.0.1 feature. Indeed in the master branch the uac_create.sql file exists. mar...@marius:~/dev/sip-router$ git checkout master Switched to branch 'master' mar...@marius:~/dev/sip-router$ ls utils/kamctl/mysql/uac-create.sql utils/kamctl/mysql/uac-create.sql Hope this helps Marius > but i didnt find the sql script > > My sip proxy version : > sercmd> core.version > Server: kamailio (3.0.1 (i386/linux)) 679736 > > my util folder : > [r...@localhost kamailio-3.0.1]# ls utils/kamctl/mysql/ > acc-create.sql domainpolicy-create.sql purple-create.sql > alias_db-create.sql drouting-create.sql registrar-create.sql > auth_db-create.sql group-create.sql rls-create.sql > avpops-create.sqlhtable-create.sql siptrace-create.sql > carrierroute-create.sql imc-create.sql speeddial-create.sql > cpl-create.sql lcr-create.sql standard-create.sql > dialog-create.sqlmsilo-create.sql uri_db-create.sql > dialplan-create.sql pdt-create.sql userblacklist-create.sql > dispatcher-create.sqlpermissions-create.sql usrloc-create.sql > domain-create.sqlpresence-create.sql > > please help > > Thank you > > > ___ > Kamailio (OpenSER) - Users mailing list > Users@lists.kamailio.org <http://vn.mc762.mail.yahoo.com/mc/compose?to=us...@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users Truy cập Mail nhanh hơn! <http://us.lrd.yahoo.com/_ylc=X3oDMTFnYXY0dG83BHRtX2RtZWNoA1RleHQgTGluawR0bV9sbmsDVTExMDM0NjgEdG1fbmV0A1lhaG9vIQ--/SIG=11ka86nha/**http%3A//downloads.yahoo.com/vn/internetexplorer/> Yahoo! khuyến khích bạn nâng cấp trình duyệt lên Internet Explorer 8 mới, tối ưu hóa cho Yahoo!. Tải tại đây! (Miễn phí) <http://us.lrd.yahoo.com/_ylc=X3oDMTFnYXY0dG83BHRtX2RtZWNoA1RleHQgTGluawR0bV9sbmsDVTExMDM0NjgEdG1fbmV0A1lhaG9vIQ--/SIG=11ka86nha/**http%3A//downloads.yahoo.com/vn/internetexplorer/> ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] What is the best module for routing INVITE based on DID in R-URI?
Hello,
On 03/12/2010 10:36 PM, Geoffrey Mina wrote:
Sorry to bring up an old thread here... but I have finally gotten
around to implementing your PDT suggestion. I have a table which
looks like:
source | prefix | domain |
* | 551212 | 192.168.200.1
* | 551213 | 192.168.200.1
* | 551214 | 192.168.200.1
The calls are coming in as: RURI=sip:551...@192.168.200.0:5060 and
I am using the pdt module by calling with prefixdomain("2","0") so
that we aren't actually stripping out any of the URI, I am just
matching to a domain and rewriting. In my logs I am seeing the
following and I am unable to route to anything but the first.
Mar 12 16:29:13 cust-sipgateway1 /usr/local/sbin/kamailio[9157]:
ERROR:pdt:pdt_load_db: sdomain [*]: prefix [551213] or domain
<192.168.200.1> duplicated
Mar 12 16:29:13 cust-sipgateway1 /usr/local/sbin/kamailio[9157]:
ERROR:pdt:pdt_load_db: sdomain [*]: prefix [551214] or domain
<192.168.200.1> duplicated
Any ideas? It would be a shame if I had to scrap this plan... as it
works so nicely with only a single prefix/domain! :)
just disable domain duplication checking:
http://kamailio.org/docs/modules/stable/modules_k/pdt.html#id2533886
First version of this module had the constraint of one prefix-domain
relation (like with unique prefix for each country) and the parameter
controls the backward compatibility.
Cheers,
Daniel
On Thu, Jan 21, 2010 at 3:58 AM, Daniel-Constantin Mierla
wrote:
On 1/20/10 12:54 AM, Geoffrey Mina wrote:
Thanks for the idea. Ill have lots of these, so If you wouldn't mind,
could you elaborate a bit on using ENUM in kamailio.
P.s. I'm on 1.5
enum implementation is pretty mature, without relevant changes since 1.3 or
so.
Regarding enum, practically is about storing relation between numbers and
sip addresses in DNS server and you query the DNS server each time you get a
call. Is good if you are familiar with dns servers. For more, you can start
from here:
http://en.wikipedia.org/wiki/Telephone_Number_Mapping
Cheers,
Daniel
Thanks
On 1/19/10, Andreas Sikkemawrote:
On Jan 19, 2010, at 8:34 PM, Geoffrey Mina wrote:
I am putting up a Kamailio server which will do nothing but route
INVITE requests from my upstream carrier to individual offices on my
side. The office locations will NOT be registered SIP UAs, but other
Kamailio proxy servers. What I want to have is a database of DIDs
associated with a forwarding IP:Port and/or SRV records.
551212 ==>1.2.3.4:5060
551213 ==>1.2.3.5:5060
If I had a reasonable amount of relations like this so that maintenance
by
hand would be an issue I'd try to find an ENUM setup that was easily
manageable. Point an ENUM address to a trusted peer and you're done.
--
Andreas
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--
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* http://www.asipto.com/
--
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* http://www.asipto.com/index.php/sip-router-masterclass/
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Re: [Kamailio-Users] Relaying NOTIFY UDP messages over TCP
Hello, On 03/11/2010 05:58 PM, Iñaki Baz Castillo wrote: 2010/3/11 Pascal Maugeri: Does such NOTIFY go to a TCP registered user? Of course if there is not an existing TCP connection between Kamailio and the final natted user then it's not possible to send such NOTIFY. Do you mean that the user is sending "transport=tcp" in his Contact header ? This must be present in the initial SUBSCRIBE. However if the client is behind NAT and uses TCP it's required some way to mantain the keepalive in the router, if not a future NOTIFY could not arrive. A common approach is the client sending some TCP data through the existing connection (i.e. as defined in defat-oubound, now RFC ). I have seen clients sending registration over UDP requiring to be contacted via TCP. To be sure it registers via TCP check the configuration of the phone and watch the sip traffic with ngrep (or ethereal) to see the transport layer protocol. Connecting from server to a client behind nat is possible only if you have port forwarding on your nat box to phone IP address. Therefore, if the phone connects via tcp it must keep the connection open. If for some reason it closes, it must re-open it. Otherwise it becomes unreachable. In the server side there are lot of tcp options to tune the behavior and optimize. I do suggest using version 3.0 for a much improved TCP architecture and implementation (including asynchronous tcp -- in case you deal with lot of tcp connections, then this saves you). http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#tcp_parameters Worth to mention as well that you can change the value of tcp parameters at runtime without need to restart (e.g., connecting timeout, send timeout, etc) using sercmd. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Stateless failover
Hello, On 03/11/2010 10:11 PM, Iñaki Baz Castillo wrote: 2010/3/11 Santiago Soares: says I can't use forward(), for failover, I have to use t_relay, which means that the server must be statefull. But the thing is, i wouldn't like to maintain transactions state in the server, due to the high memory usage. Is it true? Can't I have failover support with forward()? If yoou want Kamailio to perform failover you need it to be transaction stateful, if not Kamailio has no information about hte request when it receives a 500/503 so can not dispatch it to other server (failover). Load balancing is possible in stateless mode as it just involves sending the requests to different servers randomly. Also, are you sure you need it to be stateless? TM performance is very good and it's mostly used. just to confirm what Inaki says that you need tm for doing failover and give you a bit more insights to understand why. In stateless mode, the sip message is received, processed via config (e.g., in your case select a destination), relayed and completely forgotten. Reply comes, the sip router takes the Via header stack and routes it to origin. Nothing exists about the SIP request in sip router when the reply is processed. In stateful mode, the initial request is saved in memory, when the reply comes it, tm module matches what is the corresponding request. If the reply code was negative, via failure_route you can get the initial request back for processing and re-send it to new destination. The performances of tm are very good, with latest 3.0 one more time improved a lot. Also note that transaction means "request to final reply", not "beginning of call to end of call". So memory is used from the moment the initial invite comes in until the call is answered, canceled or time-out. During the time call is active, no memory is consumed. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Advanced Call Scenario / One Way Audio
Hello, On 03/12/2010 10:03 AM, Klaus Darilion wrote: Am 10.03.2010 21:33, schrieb Brandon Armstead: REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Having 3 media relays is a bit strange. Only one should be enough (e.g. Asterisk). Use a packet sniffer and verify who is sending RTP packets, and where the RTP flow stop. Then analyze the SDPs seen by the component where the RTP stream stops. having 2 rtp relays in a chain may create a deadlock if the rtpproxy is in used in learning mode. There is a flag (r) that can be passed to rtpproxy to trust the address in sdp: http://kamailio.org/docs/modules/stable/modules_k/nathelper.html#id2493375 Cheers, Daniel Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Loadbalancer/outbound_proxy and ACK problem
as. Min-SE: 90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: timer. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 381. . # U +0.144000 UAC_A_PUB_IP:31488 -> LB_IP:5678 ACK sip:ua...@lb_ip:5678;rinstance=06b43c2b0e1ae81a SIP/2.0. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-5f2152497c1dcb39-1---d8754z-;rport. Max-Forwards: 70. Route:. Route:. Contact:. To: "UAC_B";tag=d6775b48. From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 ACK. Proxy-Authorization: Digest username="UAC_A",realm="test.com",nonce="4b9a102f00030feed796808fa4eb1df2de3c9a2b1034",uri="sip:ua...@test.com",response="2f023b974dd4cde6f29387681168de8f",algorithm=MD5. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 0. . # U +0.00 LB_IP:5678 -> REG_PROXY_IP:5166 ACK sip:REG_PROXY_IP:5166;lr;nat=yes SIP/2.0. Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bK3ae9.a8d68282.2. Via: SIP/2.0/UDP LB_IP:5678;rport=5678;received=LB_IP;branch=z9hG4bK3ae9.a8d68282.2. Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-5f2152497c1dcb39-1---d8754z-;rport=31488. Max-Forwards: 68. Contact:. To: "UAC_B";tag=d6775b48. From: "UAC_A";tag=406aba65. Call-ID: YjEyNjJhYzczOTIyYWYyMjkwZGZiOTFmMjdkNmNmODk.. CSeq: 2 ACK. Proxy-Authorization: Digest username="UAC_A",realm="test.com",nonce="4b9a102f00030feed796808fa4eb1df2de3c9a2b1034",uri="sip:ua...@test.com",response="2f023b974dd4cde6f29387681168de8f",algorithm=MD5. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 0. P-hint: rr-enforced. P-hint: rr-enforced. . On Fri, Mar 12, 2010 at 11:04 AM, Iñaki Baz Castillo wrote: 2010/3/12 Pavel Miskov: Hello list, let me first show my scenario: UAC_A ---> LB ---> | | PROXY+REG #1 | or | PROXY+REG #2 UAC_B<--- LB<--- | Could you please repeat the trace but in a easier format: ngrep -d eth0 -W byline -T port 5060 -- Iñaki Baz Castillo ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Kamailio 3.0.0 Install Error
Hello, On Thu, Mar 11, 2010 at 3:32 AM, Nathaniel L Keeling wrote: > I am trying to install the new release of Kamailio 3.0.0 on Solaris 10. > When I perform the install, I am getting this error. Here is the command > that I am entering for the install: > > make prefix=/usr/local/kamailio-3.0.0 INSTALL=install > group_include="standard standard-dep postgres" CPU=ultrasparc install > > install mode > make[2]: Entering directory `/usr/local/src/kamailio-3.0.0/lib/kcore' > make[2]: Nothing to be done for `install-if-newer'. > make[2]: Leaving directory `/usr/local/src/kamailio-3.0.0/lib/kcore' > touch > /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/speeddial.so > install -m 755 speeddial.so > /usr/local/kamailio-3.0.0/lib/kamailio/modules_k > make[1]: Leaving directory > `/usr/local/src/kamailio-3.0.0/modules_k/speeddial' > mkdir -p /usr/local/kamailio-3.0.0/etc/kamailio/ > # other configs > /bin/sh: syntax error at line 1: `;' unexpected > make: *** [install-cfg] Error 2 > > I am new to Kamailio and would appreciate any help. > are you using gmake? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] 400 Bad Request
Can you register with another phone on the public interface and then call the Genexis? The you see if the invite coming from asterisk has something bad in it or is just something that Genexis cannot cope with. First idea of mine was the double record routing, some devices still cannot cope with more than one. But if it works the other way around, which should have two record-routes as well. Cheers, Daniel On Tue, Mar 9, 2010 at 10:17 AM, Iñaki Baz Castillo wrote: > 2010/3/9 Ernest Mavrel : > > but I always get 400 Bad Request from Genexis OCQ118. > > Have I malformed message INVITE or what could be wrong? > > Some gateways/softswitches uses 400 to reject a request when the > origin IP or calling number is not allowed to call through such > gateway. > > > -- > Iñaki Baz Castillo > > > ___ > Kamailio (OpenSER) - Users mailing list > Users@lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla http://www.asipto.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] cdr_id not registered in acc
Hello, On Wed, Mar 10, 2010 at 2:58 PM, Anders wrote: > Hi, > > (Kamailio 1.5 + freeradius + cdrtool) > > After a call is finished, it's INVITE and BYE are registered just fine > in the acc table, but the records in acc never receive a cdr_id, and > (therefore) nothing is inserted the cdrs table. > > Ideas on where to start looking? > isn't it about siremis here? Sorry if not, but the name of column and table match: http://siremis.asipto.com/install-accounting/ Cheers, Daniel > > Thanks, > Anders > > ___ > Kamailio (OpenSER) - Users mailing list > Users@lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] DEFINE question
Hello,
On Wed, Mar 10, 2010 at 7:57 PM, Uriel Rozenbaum
wrote:
> Daniel,
>
> I got the idea, but maybe I used a lame example. I'll need the same method
> to activate or deactivate some custom features like prepending some prefix
> and stuff.
>
> Maybe I'll just set some variable and ask for that. Do you agree?
>
if activation/deactivation is needed at runtime, then custom global
parameters is clearly the way to go. You can update via sercmd cli or
siremis web interface the value at runtime.
Cheers,
Daniel
> Uriel
>
>
> On Wed, Mar 10, 2010 at 3:35 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> On Wed, Mar 10, 2010 at 7:05 PM, Uriel Rozenbaum <
>> uriel.rozenb...@gmail.com> wrote:
>>
>>> Thanks guys, I'm using 1.5.3
>>>
>>> So I can use
>>>
>>> define(`SHOULD_AUTH', 1)
>>> ...
>>> if(SHOULD_AUTH)
>>> {
>>> route(5); #Auth
>>> }
>>>
>>> within my cfg file?
>>>
>>
>> you cannot have that for now. It is for controlling which parts of config
>> is loaded, like:
>>
>> #!define AUTH
>>
>> #ifdef AUTH
>>route[AUTH);
>> #!endif
>>
>> The default kamailio config in 3.0 use it to provide auth, nat, presence,
>> etc. See it online at:
>>
>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=etc/kamailio.cfg;hb=kamailio_3.0
>>
>> You can achieve similar functionality as you described above with custom
>> cfg global parameters:
>>
>>
>> http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x#custom_cfg_file_parameters
>>
>> auth.enabled = 1
>> ...
>> if($sel(cfg_get.auth.enabled))
>> {
>> route(5); #Auth
>> }
>>
>> The extra benefit is that you can change the value at runtime without
>> restart.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>>> On Wed, Mar 10, 2010 at 1:48 PM, Henning Westerholt <
>>> henning.westerh...@1und1.de> wrote:
>>>
>>>> On Wednesday 10 March 2010, Uriel Rozenbaum wrote:
>>>> > The question's simple, is there any pre-processor command to DEFINE
>>>> > constants?
>>>>
>>>> Hi Uriel,
>>>>
>>>> in kamailio 3.0 there is also the #define directive, which works more or
>>>> less
>>>> like the one in other languages.
>>>>
>>>> http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-2-
>>>> define.html<http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-2-%0Adefine.html>
>>>>
>>>> Cheers,
>>>>
>>>> Henning
>>>>
>>>
>>>
>>> ___
>>> Kamailio (OpenSER) - Users mailing list
>>> Users@lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>>
>> --
>> Daniel-Constantin Mierla
>> http://www.asipto.com
>>
>
>
--
Daniel-Constantin Mierla
http://www.asipto.com
___
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Re: [Kamailio-Users] DEFINE question
Hello,
On Wed, Mar 10, 2010 at 7:05 PM, Uriel Rozenbaum
wrote:
> Thanks guys, I'm using 1.5.3
>
> So I can use
>
> define(`SHOULD_AUTH', 1)
> ...
> if(SHOULD_AUTH)
> {
> route(5); #Auth
> }
>
> within my cfg file?
>
you cannot have that for now. It is for controlling which parts of config is
loaded, like:
#!define AUTH
#ifdef AUTH
route[AUTH);
#!endif
The default kamailio config in 3.0 use it to provide auth, nat, presence,
etc. See it online at:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=etc/kamailio.cfg;hb=kamailio_3.0
You can achieve similar functionality as you described above with custom cfg
global parameters:
http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x#custom_cfg_file_parameters
auth.enabled = 1
...
if($sel(cfg_get.auth.enabled))
{
route(5); #Auth
}
The extra benefit is that you can change the value at runtime without
restart.
Cheers,
Daniel
> On Wed, Mar 10, 2010 at 1:48 PM, Henning Westerholt <
> henning.westerh...@1und1.de> wrote:
>
>> On Wednesday 10 March 2010, Uriel Rozenbaum wrote:
>> > The question's simple, is there any pre-processor command to DEFINE
>> > constants?
>>
>> Hi Uriel,
>>
>> in kamailio 3.0 there is also the #define directive, which works more or
>> less
>> like the one in other languages.
>>
>> http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-2-
>> define.html<http://by-miconda.blogspot.com/2009/12/best-of-new-in-kamailio-300-2-%0Adefine.html>
>>
>> Cheers,
>>
>> Henning
>>
>
>
> ___
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> Users@lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
--
Daniel-Constantin Mierla
http://www.asipto.com
___
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Re: [Kamailio-Users] kamailio crashes
Hello, what version do you have? If it is for 3.0, please register a bug at: http://sip-router.org/tracker/ In 3.0 the crash is at: 186 case HDR_REFER_TO_T: 187 free_to(hf->parsed); 188 break; I am out of the office without my linux box these days to be able to check more. Maybe some other devel can look a bit at it. Thanks, Daniel On Wed, Mar 10, 2010 at 9:17 AM, Panagiotis Skoulikaritis wrote: > Dear fellow kaimailio users > > We have a kamailio server which crashes. > below is the backtrace from the core files > any idea why the kamailio is crashing > > Regards > > Panagiotis > > core.29568 Mar 10 09:27 > > #0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 > #1 0x00423d5c in sig_usr (signo=15) at main.c:563 > #2 > #3 0x0037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 > #4 0x004790cc in udp_rcv_loop () at udp_server.c:408 > #5 0x0042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 > > > > > > core.19350 Mar 10 09:05 > > (gdb) backtrace > #0 free_to (tb=0x775c00) at parser/parse_to.c:79 > #1 0x0047fd42 in clean_hdr_field (hf=0x2ad2432de100) at > parser/hf.c:187 > #2 0x2ad23fe3e525 in run_trans_callbacks (type=, > trans=, req=0x2ad2432dcf58, > rpl=0x772d28, code=) at sip_msg.h:54 > #3 0x2ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch= optimized out>) at t_lookup.c:888 > #4 0x2ad23fe47fa2 in t_check (p_msg=0x772d28, > param_branch=0x79c016bc) at t_lookup.c:964 > #5 0x2ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 > #6 0x0041eebc in forward_reply (msg=0x772d28) at forward.c:521 > #7 0x00445313 in receive_msg ( > buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", > len=920, rcv_info=0x79c017a0) at receive.c:212 > #8 0x004794ae in udp_rcv_loop () at udp_server.c:449 > #9 0x0042760e in main (argc=3, argv=0x79c019b8) at main.c:774 > (gdb) > > > > > core.29567 Mar 10 09:27 > > gdb) backtrace > #0 free_to (tb=0x776460) at parser/parse_to.c:79 > #1 0x0047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at > parser/hf.c:187 > #2 0x2ad17d15a525 in run_trans_callbacks (type=, > trans=, req=0x2ad1805f8f58, > rpl=0x771920, code=) at sip_msg.h:54 > #3 0x2ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch= optimized out>) at t_lookup.c:888 > #4 0x2ad17d163fa2 in t_check (p_msg=0x771920, > param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 > #5 0x2ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 > #6 0x0041eebc in forward_reply (msg=0x771920) at forward.c:521 > #7 0x00445313 in receive_msg ( > buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: > SIP/2.0/UDP > 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: >at receive.c:212 > #8 0x004794ae in udp_rcv_loop () at udp_server.c:449 > #9 0x0042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 > (gdb) > > > core.5326 Mar 10 09:02 > > (gdb) backtrace > #0 free_to (tb=0x772c48) at parser/parse_to.c:79 > #1 0x0047fd42 in clean_hdr_field (hf=0x2b2578e22560) at > parser/hf.c:187 > #2 0x2b257597b525 in run_trans_callbacks (type=, > trans=, req=0x2b2578e213b8, > rpl=0x771b50, code=) at sip_msg.h:54 > #3 0x2b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch= optimized out>) at t_lookup.c:888 > #4 0x2b2575984fa2 in t_check (p_msg=0x771b50, > param_branch=0x7fff7cdc63ac) at t_lookup.c:964 > #5 0x2b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 > #6 0x0041eebc in forward_reply (msg=0x771b50) at forward.c:521 > #7 0x00445313 in receive_msg ( > buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP > 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: > SIP/2.0/UDP > 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: >at receive.c:212 > #8 0x004794ae in udp_rcv_loop () at udp_server.c:449 > #9 0x0042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 > (gdb) quit > > ___ > Kamailio (OpenSER) - Users mailing list > Users@lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Kamailio (OpenSER) v3.0.1 Released
On 03/08/2010 11:58 PM, Alex Balashov wrote: Does this include the patch to ut.c/h to fix the 'acc' db_extra PV typing issue? yes. Fix commited Feb 2, 229496c7170bcc85f517a4985f7ab4bad553c8d3 Cheers, Daniel On 03/08/2010 05:47 PM, Daniel-Constantin Mierla wrote: Hello, the first patch release for 3.0 series is out as version 3.0.1. It includes the fixes to issues discovered since 3.0.0. Database structure and configuration file compatibility are preserved so the upgrade from 3.0.0 is straightforward. Links and more details are available at: http://www.kamailio.org/w/2010/03/kamailio-v3-0-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] Kamailio (OpenSER) v3.0.1 Released
Hello, the first patch release for 3.0 series is out as version 3.0.1. It includes the fixes to issues discovered since 3.0.0. Database structure and configuration file compatibility are preserved so the upgrade from 3.0.0 is straightforward. Links and more details are available at: http://www.kamailio.org/w/2010/03/kamailio-v3-0-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Get all the values for dialogs with a specific profile
On 03/08/2010 01:49 PM, Iñaki Baz Castillo wrote:
El Lunes 08 Marzo 2010, Daniel-Constantin Mierla escribió:
Hello,
On 03/08/2010 01:29 PM, Iñaki Baz Castillo wrote:
Hi, by running "kamctl fifo profile_get_size togw" I can see the number
of dialog tagged as "togw".
Also if I run "kamctl fifo profile_get_size togw 999888777" I can see
the number of dialogs tagged as "togw" with value "999888777".
Is there any way to display all current values for dialogs with a
specific profile/tag (i.e. "togw")?
by all current values you mean attributes from dialog structure (from,
to, ...)?
No, I mean values assigned in script to dialog profiles, i.e:
set_dlg_profile("togw","$fU");
I would like to display all the current values ("$fU") assigned to profile
"togw". This is: I'd like to know who is currently calling (based on $fU, $au
or whatever).
Then, with such information I could call "profile_get_size togw USER" (being
"USER" each value previously got).
ahh, ok, I see now. Probably you have to extend the mi command (or add a
new one). Should not be difficult. There is a personal to-do list I have
for dialog, you can add a feature request on tracker not to forget about
this one.
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
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Re: [Kamailio-Users] Get all the values for dialogs with a specific profile
Hello, On 03/08/2010 01:29 PM, Iñaki Baz Castillo wrote: Hi, by running "kamctl fifo profile_get_size togw" I can see the number of dialog tagged as "togw". Also if I run "kamctl fifo profile_get_size togw 999888777" I can see the number of dialogs tagged as "togw" with value "999888777". Is there any way to display all current values for dialogs with a specific profile/tag (i.e. "togw")? by all current values you mean attributes from dialog structure (from, to, ...)? Cheers, Daniel The idea is building a simple script to display the current number of calls per user (having that the value applied to dialog profiles means the calling user). Thanks for any suggestion. -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] new website look
Hello, http://www.kamailio.org has a new look! The former site was built with a cms system that discontinued the development, providing no longer updates for newer technologies and nicer design. We hope the new design helps to locate project resources easier and makes site navigation more straightforward via comprehensive list of links in the right sidebar. Credits go to Elio Rojano for the logo artwork and to developers of GPL Wordpress theme Cordobo Green Park which was used as start for the new design. If you are skilled in web design and can build a more personalized look on top of wordpress, please contact us. Cheers, Daniel PS. Do not forget, if you are tomorrow in London, UK, consider attending Present and Future of SIP Routing to get latest update about the project: http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/ -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Dispatcher ds_list results in flag=P
Hello Geoffrey, On 03/05/2010 03:16 PM, Geoffrey Mina wrote: Well, just for kicks, i added the listen=208.X.X.X:5060 listen=208.X.X.X:5060 listen=10.3.200.202:5060 Into my config to explicitly listen on my three IP addresses... and magically everything started working again. interesting. This one needs to be investigated a bit. I see no difference in specifying the list of sockets manually or being built from the system by auto-discovery. Please fill an issue on sip-router.org tracker: http://sip-router.org/tracker/ Thanks, Daniel Should I file this as a bug, or is there something I'm missing here and this was really a 'feature' and not a 'bug'? :) Either way, my problem is resolved now. Thanks everyone (especially Daniel, helpful as usual!) On Fri, Mar 5, 2010 at 8:54 AM, Henning Westerholt wrote: On Friday 05 March 2010, Geoffrey Mina wrote: So why would not having any listen= parameters cause this to become a problem? I am guessing that is the problem... Also, I am a little concerned about the mhomed parameter, specifically this statement: "By default is off (0) - it is rather time consuming." Hi Geoffrey, this is a bit outdated, Marius did recently here some optimisations, so the performance impact should be much smaller in 1.5.4 and upcoming 3.1. We'll fix the documentation. Also, why wouldn't Kamailio just be forwarding the request on the socket which received the incoming request? That would work fine as it's being received on the public IP and I want the forwarding to be sent on the public IP. Normally kamailio should work this way, if you not use mhomed mode or force the send socket in the cfg. Cheers, Henning ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] [OT] BYE sip:SIMPLE&x...@ietf.org SIP/1.0
Hi Inaki, On 03/05/2010 05:09 PM, Iñaki Baz Castillo wrote: Hi, if somebody is bored and has 5-10 minutes of spare time I suggest to read the mail I've sent to sip-implementors maillist, full of hate and fury: https://lists.cs.columbia.edu/pipermail/sip-implementors/2010- March/024529.html sad, but true! Your conclusions are right. I think we have to find ourselves a way in SIP to go the xmpp approach, which means: first implement then specify. IM & Presence are basic services in IP communication, they simply must work. I would impose a rule in IETF that each rfc must have one prototype/reference implementation before approval. Otherwise they will keep pushing generic and complex service architectures. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Users Digest, Vol 57, Issue 85
Done! If you would have read the info you included in the email, unsubscription could have been done by yourself. Cheers, Daniel On 03/06/2010 06:10 AM, Zhe wrote: Help. Please remove me from this mail list. Thanks在2010-02-27 19:00:01，users-requ...@lists.kamailio.org 写道： Send Users mailing list submissions to users@lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-requ...@lists.kamailio.org You can reach the person managing the list at users-ow...@lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. remove parameter (Elgar Onel) 2. Exotic case (but real) in which RtpProxy doesn't work (I?aki Baz Castillo) 3. Re: Exotic case (but real) in which RtpProxy doesn't work (Daniel-Constantin Mierla) 4. Re: Exotic case (but real) in which RtpProxy doesn't work (I?aki Baz Castillo) 5. Re: Exotic case (but real) in which RtpProxy doesn't work (Daniel-Constantin Mierla) 6. Re: Exotic case (but real) in which RtpProxy doesn't work (I?aki Baz Castillo) -- Message: 1 Date: Fri, 26 Feb 2010 03:00:25 -0800 (PST) From: Elgar Onel Subject: [Kamailio-Users] remove parameter To: users@lists.kamailio.org Message-ID:<444271.31134...@web114207.mail.gq1.yahoo.com> Content-Type: text/plain; charset="us-ascii" Dear friends, how to remove parameters from INVITE sip address? tia, e. -- next part -- An HTML attachment was scrubbed... URL:<http://lists.kamailio.org/pipermail/users/attachments/20100226/bb3d7c60/attachment.html> -- Message: 2 Date: Fri, 26 Feb 2010 14:08:36 +0100 From: I?aki Baz Castillo Subject: [Kamailio-Users] Exotic case (but real) in which RtpProxy doesn't work To: users@lists.kamailio.org Message-ID:<201002261408.36778@aliax.net> Content-Type: text/plain; charset="utf-8" Hi, I've a problem with an Alcatel PBX in the way it performs transference: - Alcatel (behind NAT) sends INVITE to 111 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 1000. - Alcatel sends INVITE to 222 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 2000. Then Alcatel performs the transference as follows: - It sends a re-INVITE for 111 to Kamailio by setting the SDP to the IP or RtpProxy and port 2000. - It also sends a re-INVITE for 222 to Kamailio by setting the SDP to the IP or RtpProxy and port 1000. This is, Alcatel wants that the provider (me) sends the RTP to itself, while mantaining the original SIP dialogs established (so it's ok at signalling level, but at RTP level it cannot work as RtpProxy shoud send RTP to itself). I'm thinking on how to solve it but find no solution. Any suggestion? Thanks. -- I?aki Baz Castillo -- Message: 3 Date: Fri, 26 Feb 2010 15:04:36 +0100 From: Daniel-Constantin Mierla Subject: Re: [Kamailio-Users] Exotic case (but real) in which RtpProxy doesn't work To: I?aki Baz Castillo Cc: users@lists.kamailio.org Message-ID:<4b87d4f4.9090...@gmail.com> Content-Type: text/plain; charset=UTF-8; format=flowed Hello, On 02/26/2010 02:08 PM, I?aki Baz Castillo wrote: Hi, I've a problem with an Alcatel PBX in the way it performs transference: - Alcatel (behind NAT) sends INVITE to 111 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 1000. - Alcatel sends INVITE to 222 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 2000. Then Alcatel performs the transference as follows: - It sends a re-INVITE for 111 to Kamailio by setting the SDP to the IP or RtpProxy and port 2000. - It also sends a re-INVITE for 222 to Kamailio by setting the SDP to the IP or RtpProxy and port 1000. This is, Alcatel wants that the provider (me) sends the RTP to itself, while mantaining the original SIP dialogs established (so it's ok at signalling level, but at RTP level it cannot work as RtpProxy shoud send RTP to itself). I'm thinking on how to solve it but find no solution. Any suggestion? Do you re-engage rtpproxy for re-INVITEs? Also, have you played with force rtp proxy flags to trust public addresses? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ -- Message: 4 Date: Fri, 26 Feb 2010 15:09:08 +0100 From: I?aki Baz Castillo Subject: Re: [Kamailio-Users] E
Re: [Kamailio-Users] 3.0 Modules
Hello, On 03/06/2010 03:48 PM, dotnetdub wrote: Hi List, I see two modules in particular that I could really do with running on 1.5.3 - I have started testing 3.0 but will not put into production just yet. Call control and topoh Would it be possible to backport these to 1.53 ? Would there be much to it.? I can talk for topoh, it requires changes in the core. There are not big, but you need to be familiar with how messages are handled and how changes to sip message are applied. Then it is a core event framework that needs to be backported as well. . Call Control in particular is something that is very useful for us. I don't think this one has many changes since 1.5, it uses the openser module interface. Probably just few files were relocated or function prototypes changed. We will eventually move to 3.0 of course as it is amazing product with lots of really nice new features just been burned badly before by moving to something so new. Workaround for topoh would be to run 3.0 on the same server but different port, just do record-routing and relay. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] OpenSER+Asterisk+BLF
Hello, On 03/07/2010 01:23 AM, Mark Sayer wrote: We are needing to modify the configure of a currently operating OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that are sent between Asterisk and a phone that supports BLF (like the Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 & Asterisk 1.4.17 in the same box. OpenSER performs all registration, authentication and NAT. Asterisk handles the media and the accounting. openser 1.2.x is too old and does not have the blf features. You need at least kamailio (openser) 1.5.x, better 3.0.0. It might work with latest Asterisk to send the subscribe to it and get NOTIFYs even phone is not registered to Asterisk, never tried though. In a pure Asterisk environment a "hint" would be setup in the Asterisk extensions.conf file and the phone (UA) would SUBSCRIBE to that HINT. Once Asterisk has registered that UA to the HINT then it sends SIP-NOTIFY messages to the UA as the status of the channel changes (available, ringing, busy). Our current openser.cfg file makes no mention of either SUBSCRIBE or NOTIFY which is an obvious reason that my Asterisk installation never registers the UA to the HINT. Is anyone interested in getting paid to fix this for us (we're too stupid to do it ourselves) or to offer another solution for controlling BLF in this setup. Can help only with 1.5.x/3.0.0. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Dispatcher ds_list results in flag=P
Hello, On 03/05/2010 12:33 AM, Geoffrey Mina wrote: No, I am not seeing any options messages. It doesn't appear to be doing anything while in probing mode. How often should an OPTIONS message go out? One thing to note, i have two public IP addresses on eht0, so I have eht0 and eth0:1. I also have eth1 which is a private network interface. I have nothing specified in the kamailio.cfg "listen", so Kamailio just listens on all IP addresses. I do not believe this happened before I added the 2nd IP to eth0. At that time, i also had explicit listen parameters specifying the internal and external IP addreses. When the configuration was in it's previous state, this never happened. One additional bit of info, the servers that are in "P" state are on a different public subnet. The servers in "A" state are all in the same public subnet as the Kamailio. do you have mhomed turned on? http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#mhomed You need it if there is no routing bridging between the two IP interfaces. Cheers, Daniel On Thu, Mar 4, 2010 at 11:28 AM, Daniel-Constantin Mierla wrote: Hello, On 03/04/2010 07:33 AM, Geoffrey Mina wrote: As a side note, i have enabled sip tracing on the machines which are being flagged as "P", and no SIP packets are ever arriving. I am positive there is no firewall or other device in the middle which would be stopping the flow of traffic. On Wed, Mar 3, 2010 at 10:58 PM, Geoffrey Mina wrote: Hello, I have a weird issue with Dispatcher module. When I do a ds_list I get some of my destinations returned with flag=P. It appears that when they are in this state, they are not included in the list of destinations to send traffic to. I have tried restarting Kamailio, and these certain servers always return to flag=P. I have confirmed the servers are up and running fine. They are communicating without issue at the SIP and network level. Any idea what could cause this? P is probing mode and destination should go in this state if one call couldn't be routed via it. Do you see OPTIONS messages sent to those gateways while in P mode? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] 10 years sip express router (ser)
Hello, yesterday at CeBIT I've run into Fraunhofer booth and a banner attracted my attention, photo at: http://www.asipto.com/gallery/v/cebit2010/P3030721.JPG.html?g2_imageViewsIndex=1 According to the middle right bubble, 2000 was the first year FhG presented SER at CeBIT. Given that, means the project should celebrate (at least) 10 years these days?!? I was among FhG staffers at CeBIT 2002, presenting latest SER at that time. According to CVS repository on BerliOS, files such as Makefile and main.c are dated 2001. Officially, SER was GPL-ed and open sourced in 2002 (year when ser project was registered to berlios.de software forge). Andrei, more insights from non-open-source era? I got into the project in January 2002, would be interesting to trace the birthday... Cheers, Daniel ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Re: [Kamailio-Users] Dispatcher ds_list results in flag=P
Hello, On 03/04/2010 07:33 AM, Geoffrey Mina wrote: As a side note, i have enabled sip tracing on the machines which are being flagged as "P", and no SIP packets are ever arriving. I am positive there is no firewall or other device in the middle which would be stopping the flow of traffic. On Wed, Mar 3, 2010 at 10:58 PM, Geoffrey Mina wrote: Hello, I have a weird issue with Dispatcher module. When I do a ds_list I get some of my destinations returned with flag=P. It appears that when they are in this state, they are not included in the list of destinations to send traffic to. I have tried restarting Kamailio, and these certain servers always return to flag=P. I have confirmed the servers are up and running fine. They are communicating without issue at the SIP and network level. Any idea what could cause this? P is probing mode and destination should go in this state if one call couldn't be routed via it. Do you see OPTIONS messages sent to those gateways while in P mode? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/ ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
