from:"Yuriy Gorlichenko"

I know about dispatcher but it not always canbe heplfull. Sometimes i need
my own logic that not implemented at the dispatcher.

For checking kamailio live from asterisk im use qualfy for keepalives, if i
use asterisk only as media server but it not always usefull. Sometimes
scenarios need to use another logic.

2016-09-08 17:50 GMT+03:00 Federico Cabiddu <federico.cabi...@gmail.com>:

> The issue with dispatcher is that, in case of TCP transport, you cannot
> set the sending socket for the same reason.
> Basically, each time a client TCP socket is open by Kamailio, the SO
> select the port, due to the lack of support for SO_REUSEPORT.
>
> Cheers,
>
> Federico
>
> On Thu, Sep 8, 2016 at 4:33 PM, Daniel Tryba <d.tr...@pocos.nl> wrote:
>
>> On Thu, Sep 08, 2016 at 03:38:32PM +0300, Yuriy Gorlichenko wrote:
>> > I didnt thought about keepalive. I suppose it can help.
>>
>> Better than using qualify in asterisk is to use the dispatcher module in
>> kamailio. The idea is the same, but more configurable and it is just 1
>> keepalive mechanisme so the amount of keepalives doesn't scale linear
>> with the number of registered users :)
>>
>>
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>
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Re: [SR-Users] Kamailio send TCP packets from random ports

I know about dispatcher but it not always canbe heplfull. Sometimes i need
my own logic that not implemented at the dispatcher.

For checking kamailio live from asterisk im use qualfy for keepalives, if i
use asterisk only as media server but it not always usefull. Sometimes
scenarios need to use another logic.

2016-09-08 18:24 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:

> I know about dispatcher but it not always canbe heplfull. Sometimes i need
> my own logic that not implemented at the dispatcher.
>
> For checking kamailio live from asterisk im use qualfy for keepalives, if
> i use asterisk only as media server but it not always usefull. Sometimes
> scenarios need to use another logic.
>
> 2016-09-08 17:50 GMT+03:00 Federico Cabiddu <federico.cabi...@gmail.com>:
>
>> The issue with dispatcher is that, in case of TCP transport, you cannot
>> set the sending socket for the same reason.
>> Basically, each time a client TCP socket is open by Kamailio, the SO
>> select the port, due to the lack of support for SO_REUSEPORT.
>>
>> Cheers,
>>
>> Federico
>>
>> On Thu, Sep 8, 2016 at 4:33 PM, Daniel Tryba <d.tr...@pocos.nl> wrote:
>>
>>> On Thu, Sep 08, 2016 at 03:38:32PM +0300, Yuriy Gorlichenko wrote:
>>> > I didnt thought about keepalive. I suppose it can help.
>>>
>>> Better than using qualify in asterisk is to use the dispatcher module in
>>> kamailio. The idea is the same, but more configurable and it is just 1
>>> keepalive mechanisme so the amount of keepalives doesn't scale linear
>>> with the number of registered users :)
>>>
>>>
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>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>
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Re: [SR-Users] Kamailio send TCP packets from random ports

When packet arrives at from the WebSocket its length usually more that 1500
bytes
THat is the problem. sometimes sdp data lost while sending. If asterisk at
the sama machine it uses lo interface and then it is not a problem but for
remote servers it can be.

I didnt thought about keepalive. I suppose it can help.

2016-09-08 15:04 GMT+03:00 Daniel Tryba <d.tr...@pocos.nl>:

> On Thu, Sep 08, 2016 at 02:43:03PM +0300, Yuriy Gorlichenko wrote:
> > yes. Thats will be great because in some system design it must use same
> > port that listening for sendinf like in UDP for example for transcoding
> SIP
> > over  WebSocket to SIP over TCP and masking registration behind
> thanscoder.
> >
> > Like User sends registration, kamailio just Transcoding this request to
> TCP
> > and then resend this registration packet to Asterisk.
> > With this example asteisk must originate all PACKETS to TCP port of
> > kamailio but it tries to send it to port from wich request arrived and if
> > use TCP it will not equal port that kamailio listening for TCP.
>
> And this is a problem? Since all requests from kamailio to that asterisk
> should be send over the same connection it will stay open for some time,
> and enabling qualify for the users on asterisk will keep is open.
>
> But I'd communicate over UDP with asterisk/any backend anyway, so what
> is the reason for TCP?
>
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Re: [SR-Users] Kamailio send TCP packets from random ports

yes. Thats will be great because in some system design it must use same
port that listening for sendinf like in UDP for example for transcoding SIP
over  WebSocket to SIP over TCP and masking registration behind thanscoder.

Like User sends registration, kamailio just Transcoding this request to TCP
and then resend this registration packet to Asterisk.
With this example asteisk must originate all PACKETS to TCP port of
kamailio but it tries to send it to port from wich request arrived and if
use TCP it will not equal port that kamailio listening for TCP.

2016-09-08 13:17 GMT+03:00 Daniel Tryba :

> On Thu, Sep 08, 2016 at 11:16:29AM +0200, Federico Cabiddu wrote:
> > about this subject: linux kernel starting from 3.9 introduced
> SO_REUSEPORT
> > which allows reusing TCP sockets.
> > It could be interesting supporting this in Kamailio. I worked on a patch
> > for this, I can open a PR and start a discussion if you think it's worth.
>
> Interesting, I read about SO_REUSEPORT a couple of weeks ago but it
> didn't dawn on me that it lets you have multiple sender sockets (by just
> not accept()-ing). IMHO it is usefull for setups that use IP based
> authentication with TCP instead of UDP (probably something the OP want
> to do).
>
>
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[SR-Users] Kamailio send TCP packets from random ports

2016-09-07 Thread Yuriy Gorlichenko

Hi. I try to make working kamailio on TCP infront of asterisk

Before to send to asteisk any packet i added
$fs=ip.add.re.ss:port

Also discribed
listen=tcp:ip.add.re.ss:port

But kamailio send outgoing packets from random prot throug TCP
Presume i configured 5060 port but it send from 35410 port.

googling on a core functions didn gives me any answer.

May be i forget something to set at the config?
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Re: [SR-Users] presence modules not send NOTIFY events

2016-09-05 Thread Yuriy Gorlichenko

hi guys. can some one help to understand what there can be wrong?

2016-09-01 20:55 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:

> Hi. I trying to implement full presence server
> I set config like this (part of it)
>
> #!define FLT_DLG 9#!define FLT_DLGINFO 10
>
>
>
> modparam("dialog", "timeout_avp", "$avp(i:10)")
> modparam("dialog", "dlg_flag", 4)
> modparam("dialog", "initial_cbs_inscript", 1)
> modparam("dialog", "profiles_with_value", "caller")
> modparam("dialog", "track_cseq_updates", 1)
> modparam("dialog", "enable_stats", 1)
> modparam("dialog", "dlg_flag", FLT_DLG)
>
> modparam("presence", "db_url", DBURL)
> modparam("presence", "server_address", SIPURI)
> modparam("presence", "send_fast_notify", 1)
> modparam("presence", "db_update_period", 20)
> modparam("presence", "clean_period", 40)
> modparam("presence", "subs_db_mode", 1)
> modparam("presence", "fetch_rows", 1000)
> modparam("presence", "local_log_level", 3)
> # - presence_xml params -
> modparam("presence_xml", "db_url", DBURL)
> modparam("presence_xml", "force_active", 1)
>
> #---presence_dialoginfo params -
> modparam("presence_dialoginfo", "force_dummy_dialog", 1)
> modparam("presence_dialoginfo", "force_single_dialog",1)
>
> # - pua params -
> modparam("pua", "db_url", DBURL)
> modparam("pua", "db_mode", 2)
> modparam("pua", "update_period", 10)
> modparam("pua", "dlginfo_increase_version", 0)
> modparam("pua", "reginfo_increase_version", 0)
> modparam("pua", "check_remote_contact", 1)
> modparam("pua", "fetch_rows", 1000)
>
> # - pua_dialoginfo params -
> modparam("pua_dialoginfo", "include_callid", 1)
> modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO)
> modparam("pua_dialoginfo", "caller_confirmed", 0)
> modparam("pua_dialoginfo", "include_tags", 1)
> modparam("pua_dialoginfo", "override_lifetime", 124)
> modparam("presence", "db_url", DBURL)
> modparam("presence", "server_address", SIPURI)
> modparam("presence", "send_fast_notify", 1)
> modparam("presence", "db_update_period", 20)
> modparam("presence", "clean_period", 40)
> modparam("presence", "subs_db_mode", 1)
> modparam("presence", "fetch_rows", 1000)
> modparam("presence", "local_log_level", 3)
> # - presence_xml params -
> modparam("presence_xml", "db_url", DBURL)
> modparam("presence_xml", "force_active", 1)
>
> #---presence_dialoginfo params -
> modparam("presence_dialoginfo", "force_dummy_dialog", 1)
> modparam("presence_dialoginfo", "force_single_dialog",1)
>
> # - pua params -
> modparam("pua", "db_url", DBURL)
> modparam("pua", "db_mode", 2)
> modparam("pua", "update_period", 10)
> modparam("pua", "dlginfo_increase_version", 0)
> modparam("pua", "reginfo_increase_version", 0)
> modparam("pua", "check_remote_contact", 1)
> modparam("pua", "fetch_rows", 1000)
>
> # - pua_dialoginfo params -
> modparam("pua_dialoginfo", "include_callid", 1)
> modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO)
> modparam("pua_dialoginfo", "caller_confirmed", 0)
> modparam("pua_dialoginfo", "include_tags", 1)
> modparam("pua_dialoginfo", "override_lifetime", 124)
>
> At the config file i made some changes
> route[BLF]
> {
>   # absorb retransmissions
>   if (! t_newtran())
>   {
>   sl_reply_error();
>   exit;
>   };
>
>   if(is_method("PUBLISH"))
>   {
>   if (handle_publish()) {
>xlog("L_INFO","{$rm} handled");
>   };
> t_release();
>   } else if( is_method("SUBSCRIBE")) {
>
>   if (handle_subscribe()) {
>xlog("L_INFO","{$rm} handled");
> };
>   t_release();
>   };
>
>   exit;
> }
>
> Also added for evey call dlg_manage() function
>
> So for now i have issue that no NOTIFY event was sent at the any update of 
> any device
> I see PUBLISH events and SUBSCRIBE events but it looks like
>
> handle_publish()
> and
>
> handle_subscribe()
> not sending any NOTIFY
> But at the dump i see
>
> xlog("L_INFO","{$rm} handled"); where $rm is correct SUBSCRIBE and PUBLISH if 
> it is.
>
> Just dont know what i need to change.
>
> I will be very grateful if someone heps me to resolve this issue.
>
> Thanks
>
>
>
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[SR-Users] presence modules not send NOTIFY events

2016-09-01 Thread Yuriy Gorlichenko

Hi. I trying to implement full presence server
I set config like this (part of it)

#!define FLT_DLG 9#!define FLT_DLGINFO 10



modparam("dialog", "timeout_avp", "$avp(i:10)")
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "initial_cbs_inscript", 1)
modparam("dialog", "profiles_with_value", "caller")
modparam("dialog", "track_cseq_updates", 1)
modparam("dialog", "enable_stats", 1)
modparam("dialog", "dlg_flag", FLT_DLG)

modparam("presence", "db_url", DBURL)
modparam("presence", "server_address", SIPURI)
modparam("presence", "send_fast_notify", 1)
modparam("presence", "db_update_period", 20)
modparam("presence", "clean_period", 40)
modparam("presence", "subs_db_mode", 1)
modparam("presence", "fetch_rows", 1000)
modparam("presence", "local_log_level", 3)
# - presence_xml params -
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)

#---presence_dialoginfo params -
modparam("presence_dialoginfo", "force_dummy_dialog", 1)
modparam("presence_dialoginfo", "force_single_dialog",1)

# - pua params -
modparam("pua", "db_url", DBURL)
modparam("pua", "db_mode", 2)
modparam("pua", "update_period", 10)
modparam("pua", "dlginfo_increase_version", 0)
modparam("pua", "reginfo_increase_version", 0)
modparam("pua", "check_remote_contact", 1)
modparam("pua", "fetch_rows", 1000)

# - pua_dialoginfo params -
modparam("pua_dialoginfo", "include_callid", 1)
modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO)
modparam("pua_dialoginfo", "caller_confirmed", 0)
modparam("pua_dialoginfo", "include_tags", 1)
modparam("pua_dialoginfo", "override_lifetime", 124)
modparam("presence", "db_url", DBURL)
modparam("presence", "server_address", SIPURI)
modparam("presence", "send_fast_notify", 1)
modparam("presence", "db_update_period", 20)
modparam("presence", "clean_period", 40)
modparam("presence", "subs_db_mode", 1)
modparam("presence", "fetch_rows", 1000)
modparam("presence", "local_log_level", 3)
# - presence_xml params -
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)

#---presence_dialoginfo params -
modparam("presence_dialoginfo", "force_dummy_dialog", 1)
modparam("presence_dialoginfo", "force_single_dialog",1)

# - pua params -
modparam("pua", "db_url", DBURL)
modparam("pua", "db_mode", 2)
modparam("pua", "update_period", 10)
modparam("pua", "dlginfo_increase_version", 0)
modparam("pua", "reginfo_increase_version", 0)
modparam("pua", "check_remote_contact", 1)
modparam("pua", "fetch_rows", 1000)

# - pua_dialoginfo params -
modparam("pua_dialoginfo", "include_callid", 1)
modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO)
modparam("pua_dialoginfo", "caller_confirmed", 0)
modparam("pua_dialoginfo", "include_tags", 1)
modparam("pua_dialoginfo", "override_lifetime", 124)

At the config file i made some changes
route[BLF]
{
# absorb retransmissions
if (! t_newtran())
{
sl_reply_error();
exit;
};

if(is_method("PUBLISH"))
{
if (handle_publish()) {
   xlog("L_INFO","{$rm} handled");
};
t_release();
} else if( is_method("SUBSCRIBE")) {

if (handle_subscribe()) {
   xlog("L_INFO","{$rm} handled");
};
t_release();
};

exit;
}

Also added for evey call dlg_manage() function

So for now i have issue that no NOTIFY event was sent at the any
update of any device
I see PUBLISH events and SUBSCRIBE events but it looks like

handle_publish()
and

handle_subscribe()
not sending any NOTIFY
But at the dump i see

xlog("L_INFO","{$rm} handled"); where $rm is correct SUBSCRIBE and
PUBLISH if it is.

Just dont know what i need to change.

I will be very grateful if someone heps me to resolve this issue.

Thanks
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Re: [SR-Users] Fail to register at trunk through UAC

2016-08-02 Thread Yuriy Gorlichenko

Hi. Tried master branch

version: kamailio 5.0.0-dev5 (x86_64/linux) 1f7f96
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 1f7f96
compiled on 19:27:54 Aug  1 2016 with gcc 4.9.2

Still same issue

Added debug=4 output

2016-08-01 15:53 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:

> Hi Daniel. Thanks for answer.
> I will reinstall it today. Ping you ASAP about result
>
> 2016-08-01 13:19 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>:
>
>> Hello,
>>
>> have you tried with latest master branch? It was a large pull request
>> merged lately, just to be sure you use current code...
>>
>> Cheers,
>> Daniel
>>
>> On 01/08/16 10:30, Yuriy Gorlichenko wrote:
>>
>> Hi.Unfortunattely Probles still exists.
>> Will be very grateful if someone will help me to understand what is wrong
>> there...
>> thnk you
>>
>>
>> 2016-07-29 15:23 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:
>>
>>> Also checked all credentians 3 times.
>>> It worked on another platfoms
>>> I tried read sources uac_reg.c
>>> found that it takes form hash credentmans about this trunk but not found
>>> where and what it checks.
>>>
>>> So i suppose it is difference at MD5 but i can not check it.
>>>
>>> 2016-07-29 15:03 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:
>>>
>>>> Hi. All trunks works fine ony one not works
>>>>
>>>> Kamailio sends REGISTER with proxy auth
>>>> provider answers with 407
>>>> Kamailio not send any REGISTER
>>>> Just answers in log
>>>>
>>>> uac_reg_tm_callback(): authentication failed for 
>>>>
>>>> At attachement kamialio debug 3 log and sip log
>>>>
>>>
>>>
>>
>>
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>>
>>
>> --
>> Daniel-Constantin Mierlahttp://www.asipto.com - 
>> http://www.kamailio.orghttp://twitter.com/#!/miconda - 
>> http://www.linkedin.com/in/miconda
>>
>>
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>>
>>
>


trunble with reg.dump
Description: Binary data
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Re: [SR-Users] Fail to register at trunk through UAC

2016-08-01 Thread Yuriy Gorlichenko

Hi Daniel. Thanks for answer.
I will reinstall it today. Ping you ASAP about result

2016-08-01 13:19 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>:

> Hello,
>
> have you tried with latest master branch? It was a large pull request
> merged lately, just to be sure you use current code...
>
> Cheers,
> Daniel
>
> On 01/08/16 10:30, Yuriy Gorlichenko wrote:
>
> Hi.Unfortunattely Probles still exists.
> Will be very grateful if someone will help me to understand what is wrong
> there...
> thnk you
>
>
> 2016-07-29 15:23 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:
>
>> Also checked all credentians 3 times.
>> It worked on another platfoms
>> I tried read sources uac_reg.c
>> found that it takes form hash credentmans about this trunk but not found
>> where and what it checks.
>>
>> So i suppose it is difference at MD5 but i can not check it.
>>
>> 2016-07-29 15:03 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:
>>
>>> Hi. All trunks works fine ony one not works
>>>
>>> Kamailio sends REGISTER with proxy auth
>>> provider answers with 407
>>> Kamailio not send any REGISTER
>>> Just answers in log
>>>
>>> uac_reg_tm_callback(): authentication failed for 
>>>
>>> At attachement kamialio debug 3 log and sip log
>>>
>>
>>
>
>
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>
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> http://www.kamailio.orghttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] Fail to register at trunk through UAC

2016-08-01 Thread Yuriy Gorlichenko

Hi.Unfortunattely Probles still exists.
Will be very grateful if someone will help me to understand what is wrong
there...
thnk you


2016-07-29 15:23 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:

> Also checked all credentians 3 times.
> It worked on another platfoms
> I tried read sources uac_reg.c
> found that it takes form hash credentmans about this trunk but not found
> where and what it checks.
>
> So i suppose it is difference at MD5 but i can not check it.
>
> 2016-07-29 15:03 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:
>
>> Hi. All trunks works fine ony one not works
>>
>> Kamailio sends REGISTER with proxy auth
>> provider answers with 407
>> Kamailio not send any REGISTER
>> Just answers in log
>>
>> uac_reg_tm_callback(): authentication failed for 
>>
>> At attachement kamialio debug 3 log and sip log
>>
>
>
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Re: [SR-Users] Fail to register at trunk through UAC

2016-07-29 Thread Yuriy Gorlichenko

Also checked all credentians 3 times.
It worked on another platfoms
I tried read sources uac_reg.c
found that it takes form hash credentmans about this trunk but not found
where and what it checks.

So i suppose it is difference at MD5 but i can not check it.

2016-07-29 15:03 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:

> Hi. All trunks works fine ony one not works
>
> Kamailio sends REGISTER with proxy auth
> provider answers with 407
> Kamailio not send any REGISTER
> Just answers in log
>
> uac_reg_tm_callback(): authentication failed for 
>
> At attachement kamialio debug 3 log and sip log
>
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[SR-Users] Fail to register at trunk through UAC

2016-07-29 Thread Yuriy Gorlichenko

Hi. All trunks works fine ony one not works

Kamailio sends REGISTER with proxy auth
provider answers with 407
Kamailio not send any REGISTER
Just answers in log

uac_reg_tm_callback(): authentication failed for 

At attachement kamialio debug 3 log and sip log


kamailio.dump.reg trouble
Description: Binary data


kamailio.sip.dump.reg trouble
Description: Binary data
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Re: [SR-Users] Kamailio+tls not working with Ubuntu 16.04

2016-07-25 Thread Yuriy Gorlichenko

I have no experience in ssl coding and not deep know algorithms for working
with it.
I will try to make something but i suppose it will be longe than whait you.
Anyway i will try by my side, and let you know if will do it. But if you
will have time - will be great if you start to do it because i suppose you
finish first)

2016-07-25 15:17 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>:

> Hello,
>
> the plan was to dig more in the libssl, because it seems to have the
> memory/locking management functions already set. But with IETF and
> traveling, I didn't have any time for it.
>
> Maybe you can make a very simple c program linking to libssl that just
> prints the memory functions as done by the log message in kamailio and see
> if they are null or not.
> Cheers,
> Daniel
>
>
> On 25/07/16 11:51, Yuriy Gorlichenko wrote:
>
> I also checked archive SR-users.list and found question about same staff.
>
> I installed kamailio from master branch on ubuntu 16.04 with small changes
> at tls module.
> It gives same error but more clear result. (tested with default kamailio
> file - moved tls.so before ALL modules)
>
>  0(27545) ERROR: tls [tls_init.c:493]: tls_pre_init(): Unable to set the
> memory allocation functions
>  0(27545) ERROR: tls [tls_init.c:495]: tls_pre_init(): libssl current mem
> functions - m: 0x7fab2a42d550 r: 0x7fab2a42dc40 f: 0x7fab2a42da70
>  0(27545) ERROR: tls [tls_init.c:497]: tls_pre_init(): Be sure tls module
> is loaded before any other module using libssl (can be loaded first to be
> safe)
>  0(27545) ERROR:  [sr_module.c:607]: load_module():
> /usr/local/lib64/kamailio/modules/tls.so: mod_register failed
>
>
>
>
>
> 2016-07-25 12:23 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:
>
>> Just asking about any progress of this staff
>>
>> THere is a bug descried at the ubuntu bug tracker
>> https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992
>>
>> There is bug that i wrote at the kamailio bug tracker
>> https://github.com/kamailio/kamailio/issues/714
>>
>> Closed it because It is not trouble of kamailio itself. I suppose
>> something wrong at the ubuntu enviroment
>>
>> Any ideas or may be someone resolves it?
>>
>
>
>
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> --
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> http://www.kamailio.orghttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] Kamailio+tls not working with Ubuntu 16.04

2016-07-25 Thread Yuriy Gorlichenko

I also checked archive SR-users.list and found question about same staff.

I installed kamailio from master branch on ubuntu 16.04 with small changes
at tls module.
It gives same error but more clear result. (tested with default kamailio
file - moved tls.so before ALL modules)

 0(27545) ERROR: tls [tls_init.c:493]: tls_pre_init(): Unable to set the
memory allocation functions
 0(27545) ERROR: tls [tls_init.c:495]: tls_pre_init(): libssl current mem
functions - m: 0x7fab2a42d550 r: 0x7fab2a42dc40 f: 0x7fab2a42da70
 0(27545) ERROR: tls [tls_init.c:497]: tls_pre_init(): Be sure tls module
is loaded before any other module using libssl (can be loaded first to be
safe)
 0(27545) ERROR:  [sr_module.c:607]: load_module():
/usr/local/lib64/kamailio/modules/tls.so: mod_register failed





2016-07-25 12:23 GMT+03:00 Yuriy Gorlichenko <ovoshl...@gmail.com>:

> Just asking about any progress of this staff
>
> THere is a bug descried at the ubuntu bug tracker
> https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992
>
> There is bug that i wrote at the kamailio bug tracker
> https://github.com/kamailio/kamailio/issues/714
>
> Closed it because It is not trouble of kamailio itself. I suppose
> something wrong at the ubuntu enviroment
>
> Any ideas or may be someone resolves it?
>
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[SR-Users] Kamailio+tls not working with Ubuntu 16.04

2016-07-25 Thread Yuriy Gorlichenko

Just asking about any progress of this staff

THere is a bug descried at the ubuntu bug tracker
https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992

There is bug that i wrote at the kamailio bug tracker
https://github.com/kamailio/kamailio/issues/714

Closed it because It is not trouble of kamailio itself. I suppose something
wrong at the ubuntu enviroment

Any ideas or may be someone resolves it?
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[SR-Users] uacreg flags field not changing

2016-06-24 Thread Yuriy Gorlichenko

Hi. Im using kamailio 4.4.1

kamailio -v
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 99e1c3
compiled on 13:34:38 Jun 14 2016 with gcc 4.9.2


I using uacreg table for register trunks and want to know their status
For now i see status onlu from
kamcmd uac.reg_dump/info
But if i trying to take flags from uacreg table (flags field) it always 0.
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Re: [SR-Users] Using websocket server for events

2016-05-04 Thread Yuriy Gorlichenko

Hi. websockets module has only handshake and close functions for now.
Im asking the way about send function form server but not SIP messages for
handling it by callback at the client

for example i a some client who was connected to kamailio and want receive
messages from websocket and handle it.

for all events that sends through websocket at kamailio are SIP signallig
staff and it created by core as i see.

I what to know if it is possible to send some other messages (frames,events
as you want) to subscribers that connected at the websocket.

More closr example
Im a some manager and what to connect to Some web app and see active calls
that was made at this web app.
So i as developer set at the kamailio config

send_from_ws("eventname:data")

so for some call positions i set it and send it into subscribed clients.

THen at the client side i will handle it.

Im understand that kamailio is a SIP server. But if it already have some
websocket implementation i just asking it because it will be a cool to
manage this staff from one server instead of sharing data at the redis for
example? setting subscribing at the redis and then use another server for
this staff

2016-05-04 9:48 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>:

> Hello,
>
> what do you mean by websocket events? Can you give a more specific example
> of what you want to do?
>
> Cheers,
> Daniel
>
> On 02/05/16 14:51, Yuriy Gorlichenko wrote:
>
> Hi is there is a valy to send websocket events (not only SIP) on remote
> subscribed client from config file?
>
> I just want make shure that i need to use redis for this staff insead of
> native websocket events
>
>
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> --
> Daniel-Constantin Mierlahttp://www.asipto.comhttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
> Kamailio World Conference, Berlin, May 18-20, 2016 - 
> http://www.kamailioworld.com
>
>
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[SR-Users] Using websocket server for events

2016-05-02 Thread Yuriy Gorlichenko

Hi is there is a valy to send websocket events (not only SIP) on remote
subscribed client from config file?

I just want make shure that i need to use redis for this staff insead of
native websocket events
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Re: [SR-Users] Kamailio Cluster

2016-02-01 Thread Yuriy Gorlichenko

Also one of the most siplies ways to resolve your issue is using queries at
the databases that select needed fields from db with (where
socket="").


2016-02-01 11:43 GMT+03:00 Gholamreza Sabery :

> All right. Thank you so much.
>
> On Mon, Feb 1, 2016 at 11:47 AM, Federico Cabiddu <
> federico.cabi...@gmail.com> wrote:
>
>> Hi,
>> a solution in this case could be making usage of Path header (
>> https://tools.ietf.org/html/rfc3327).
>>
>> You have to load the path module
>>
>> http://www.kamailio.org/docs/modules/devel/modules/path.html
>>
>> and enabled its usage in registrar module
>>
>>
>> http://www.kamailio.org/docs/modules/devel/modules/registrar.html#registrar.p.use_path
>>
>> If you plan to use REGISTER replication through t_replicate() just call
>> add_path() before replicating the message to the other server(s).
>> If you use a shared database with db_mode = 3, call msg_apply_changes (
>> http://www.kamailio.org/docs/modules/devel/modules/textopsx.html#textopsx.f.msg_apply_changes)
>> before saving the contact. In this case also be sure to set
>> path_check_local parametr of the registrar module (
>> http://www.kamailio.org/docs/modules/devel/modules/registrar.html#registrar.p.path_check_local)
>> to 1.
>>
>> Cheers,
>>
>> Federico
>>
>> On Mon, Feb 1, 2016 at 9:10 AM, Jurijs Ivolga 
>> wrote:
>>
>>> Hi Alex,
>>>
>>> You are right. :)
>>>
>>> With kind regards,
>>>
>>> 2016-02-01 10:09 GMT+02:00 Alex Balashov :
>>>
 On 02/01/2016 02:59 AM, Jurijs Ivolga wrote:

 I think you need to make record routing, so it will keep initial(to what
> user registered) Kamailio in signaling path.
>

 Ah, no. Record-Route is for INVITEs only. You're thinking of Path.

 --
 Alex Balashov | Principal | Evariste Systems LLC
 303 Perimeter Center North, Suite 300
 Atlanta, GA 30346
 United States

 Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
 Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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>>>
>>>
>>>
>>> --
>>> Jurijs
>>>
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[SR-Users] checking source domain

2016-01-31 Thread Yuriy Gorlichenko

I need to understand where from packets received. Now I use something like

If $si == "1.2.3.4" {
 xlog("L_INFO","bla bla bla");
}

But I need to check source server not only by IP and PORT, but at Domain too
For example

if (some_pseudovariable=="pbx.server.com"){
 xlog("L_INFO","bla bla bla");
}

I can use $fu for example because for some packets it includes domain name
of kamailio (i think details not important but this situations can be)
Does kamailio have some mechanisms to do that? I searched it at cookbook
but not found anything.

Thank you
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[SR-Users] kamailio 4.4 writes wrong expires parameter at the database

2016-01-27 Thread Yuriy Gorlichenko

I register from sofftphone at my registrar server.
Kamailio must save to location table info about registration
(save("location") uses at reply route)
At my softphone I setted expire as 360 (and see it at packet that errives
at kamailio)
But at the db i only 20 seconds period

| id  | ruid | username | domain| contact
 | received | path |
expires | q | callid   | cseq |
last_modified   | flags | cflags | user_agent | socket
| methods | instance | reg_id | server_id | connection_id | keepalive |
partition |


| 101 | uloc-56a8f5a7-6273-2 | 101  | myserver.com |
sip:101@192.168.1.3:56549;rinstance=73c46a4379766398;transport=udp | NULL
  | NULL | 2016-01-27 10:53:29 | -1.00 | tDQT2I7gI5tA9odtWhdD2w.. |9 |
2016-01-27 10:53:09 | 0 |  0 | n/a| udp:myserver.com |
13279 | NULL |  0 | 0 |-1 | 0 |
0 |
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Re: [SR-Users] kamailio 4.4 writes wrong expires parameter at the database

2016-01-27 Thread Yuriy Gorlichenko

All default. I understand my mistake. I thought that kamailio stores
Expires value for each registration it internal variable. but it seems that
it is not stores and sets default value from registrar module parametr when
registration saves by reply.

2016-01-27 20:29 GMT+03:00 Phil Lavin <phil.la...@synety.com>:

> What is the expiry in the reply from Kamailio to the registration?
>
>
>
> What is the value you have set for registrar module max_expires param?
>
>
>
>
>
> Phil
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *Yuriy Gorlichenko
> *Sent:* 27 January 2016 16:59
> *To:* Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org>
> *Subject:* [SR-Users] kamailio 4.4 writes wrong expires parameter at the
> database
>
>
>
> I register from sofftphone at my registrar server.
> Kamailio must save to location table info about registration
> (save("location") uses at reply route)
> At my softphone I setted expire as 360 (and see it at packet that errives
> at kamailio)
> But at the db i only 20 seconds period
>
> | id  | ruid | username | domain| contact
>| received | path |
> expires | q | callid   | cseq |
> last_modified   | flags | cflags | user_agent | socket
> | methods | instance | reg_id | server_id | connection_id | keepalive |
> partition |
>
>
>
> | 101 | uloc-56a8f5a7-6273-2 | 101  | myserver.com |
> sip:101@192.168.1.3:56549;rinstance=73c46a4379766398;transport=udp | NULL
> | NULL | 2016-01-27 10:53:29 | -1.00 | tDQT2I7gI5tA9odtWhdD2w.. |9
> | 2016-01-27 10:53:09 | 0 |  0 | n/a| udp:myserver.com |
>   13279 | NULL |  0 | 0 |-1 | 0 |
>   0 |
>
>
>
>
>
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[SR-Users] WebRTC no longer supports RTP

2015-12-10 Thread Yuriy Gorlichenko

https://developers.google.com/web/updates/2015/10/chrome-47-webrtc

So at 47 chrome we already have no sound.
What kind of proto we must use and how to handle this with rtpengine?
Do anyone have same problems with it?
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Re: [SR-Users] WebRTC no longer supports RTP

2015-12-10 Thread Yuriy Gorlichenko

I already use DTLS-SRTP (websockets dont works with RTP).

This is my SDP body. And I have no sound at incoming calls
tcpdump shows me that I have no rtp strean fro websocket endpoint

v=0
o=root 1828066564 1828066564 IN IP4 1.1.1.1
s=Cattaxi Media Server
c=IN IP4 1.1.1.1
t=0 0
m=audio 30328 RTP/SAVPF 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
a=rtcp:30329
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:qR051xikD6ric2IviTKsW4ptnIBGRkczoa9zPfQo
a=setup:actpass
a=fingerprint:sha-1
8E:E9:05:41:7B:D6:07:19:2A:CD:AF:73:DC:E6:A3:33:52:B7:87:17
a=ice-ufrag:U0yN8Dop
a=ice-pwd:kn6u9i3uNekfnoRyeLJ70aHU9d
a=candidate:fuazQx0DTYr6GboN 1 UDP 21307064311.1.1.1 30328 typ host
a=candidate:fuazQx0DTYr6GboN 2 UDP 21307064301.1.1.1 30329 typ host

2015-12-10 18:44 GMT+03:00 Vasiliy Ganchev :

> Hi!
> use DTLS-SRTP, to say how to handle it with rtpengine - I think you should
> provide more info about your setup, and call cases
>
> Cheers!
>
>
>
> --
> View this message in context:
> http://sip-router.1086192.n5.nabble.com/WebRTC-no-longer-supports-RTP-tp143834p143836.html
> Sent from the Users mailing list archive at Nabble.com.
>
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[SR-Users] lookup(location) offen dont work with websocket connected endpoints

2015-12-06 Thread Yuriy Gorlichenko

long time ago looked this problem but resolve it by sql queries directly
from confing file to location table.

So while remember

lookup(location) ofen can not find any contacts if its registers from
websocket.

it may work 10-15 times and then fails. So debug shows answer that is no
connections in location table.

Now at production system can not run this issue but at development system
want to do it today or tomorrow.
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[SR-Users] UAC dynamic reg_contact_addr()

2015-12-04 Thread Yuriy Gorlichenko

I have multiple ip addresses at my kamailio. I use uac module for
registration to sip providers. I have one provider but want to register
form different ip addresses used by my server. When register sends it takes
ip address form reg_contact_addr(). But if I want to register from another
interface with different ip i need to get to this parametr another ip
address. So As I understand it imposibleto do with avp. Does kamailio have
some methods to do this?
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Re: [SR-Users] UAC dynamic reg_contact_addr()

2015-12-04 Thread Yuriy Gorlichenko

do I need to recompile kamailio with

make EXTRA_DEFS="-DWITH_EVENT_LOCAL_REQUEST" cfg

?
 Because at my installation adding  event_route[tm:local-request]
fives me syntax error.


2015-12-05 0:04 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>:

> Hello,
>
> On 04/12/15 14:13, Yuriy Gorlichenko wrote:
> > I have multiple ip addresses at my kamailio. I use uac module for
> > registration to sip providers. I have one provider but want to
> > register form different ip addresses used by my server. When register
> > sends it takes ip address form reg_contact_addr(). But if I want to
> > register from another interface with different ip i need to get to
> > this parametr another ip address. So As I understand it imposibleto do
> > with avp. Does kamailio have some methods to do this?
> >
> perhaps makes sense to add some extra columns to be able to specify the
> local socket and contact addresses.
>
> For now, I expect to work by using event_route[tm:local-request] where
> you force the socket with $fs.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> http://miconda.eu
>
>
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[SR-Users] Kamailio can not read to_tag at BYE request

2015-11-09 Thread Yuriy Gorlichenko

Hello. I have some one provider that sends me BYE request with to_tag that
kamailio can not parse

IP my.pro.vi.der.5060 > my.ser.vi.ce.5060: UDP, length 406
E...x...S.0X_..L..8.BYE sip:0987654...@my.ser.vi.ce SIP/2.0
Via: SIP/2.0/UDP
my.pro.vi.der:5060;branch=z9hG4bKsj7bv820c0s18k8a15q1sd000ag33.1
Call-ID: 00137fb21f65f19a3094c38b30d80f5c
CSeq: 1479 BYE
From: ;tag=SDi9r0899-5ci36vojn0
To: ;tag=as6635bbe1
Reason: Q.850;cause=16;text="Normal call clearing"
Max-Forwards: 69
Content-Length: 0

So I guess that problem in provider (may be ";" with another code)
But may be we have some instruments to get tag (without parsing strings in
a config file or something like that )
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[SR-Users] Dialog module issue

2015-09-09 Thread Yuriy Gorlichenko

Hello. I try to manage by dialog module every signaling session, that goes
through my proxy.
I added newx mod params

modparam("dialog", "timeout_avp", "$avp(i:10)")
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "initial_cbs_inscript", 1)
modparam("dialog", "profiles_with_value", "caller")
modparam("dialog", "track_cseq_updates", 1)
modparam("dialog", "ka_timer", 10)
modparam("dialog", "ka_interval", 30


And at request route I added this for all invite methods

if (is_method("INVITE")){
$dlg_ctx(timeout_route) = "DIALOG_END";
$avp(i:10)=43200;
$dlg_ctx(timeout_bye) = 1;
dlg_set_property("ka-src");
dlg_set_property("ka-dst");
dlg_manage();
xlog("L_INFO","Dialog manage is {$ct}\n");
...
}

So after this As I think every session must be dialog managed session and
every leg of call must be checked by keepalive OPTIONS packets but no one
OPTIONS request generated  wen session goes through my proxy.

thanks
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Re: [SR-Users] Kamailio no recieved 480/486 messages

2015-07-01 Thread Yuriy Gorlichenko

Thanks for answer. Main problem is that 1[08][03] and 200 replies resend
correctly but 480/486 can not do it at same way...
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[SR-Users] Fwd: websocket devices keepalive not working

2015-07-01 Thread Yuriy Gorlichenko

As I read at README to Nathelper it may not work with usrloc table, Is it
write?

now I use NAT traversal module and Websocket module to keepalive
mechanisms, that uses UDP or WS.

with UDP it is wery simple method - OPTIONS replies.

With WebSocket module keepalive mechanism uses thomething else. Not a SIP
methods, I see at TCPDUMP packets that marks with modparam(websocket,
ping_application_data, WebSockets rock)

I need to handle replies of keepaliwe mechanism. How I may handle replies
form this mechanism?

2015-06-30 18:37 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 thnks for response. I thing - nathelper is what I need. If I resolve issue
 above (why it not work with setbflag(7)  ) I thing this questions will be
 closed.

 P.S may be it help - my kamailio version is

 version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53
 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 1dbd53
 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4

 2015-06-30 18:18 GMT+03:00 Camille Oudot camille.ou...@orange.com:

 Le Tue, 30 Jun 2015 15:49:26 +0300,
 Yuriy Gorlichenko ovoshl...@gmail.com a écrit :

  4) I tried to do it with tcpops
 
  Only for ws and tcp (WS is use TCP as i know)
  tcp_keepalive_enable(15, 5, 15);
 
  same result...

 Hi Yuriy,

 the tcp_keepalive_enable() function will send keepalive packets at TCP
 level, not at application level (SIP OPTIONS). It can be used to check
 if the remote peer connection is still valid, and close the TCP
 connection on the server side if it is no more valid. If you enable the
 TCP keepalives, you should see these packets using tcpdump or wireshark.

 Along with handle_lost_tcp option on usrloc, this can be used to
 maintain the usrloc table as consistent as possible.

 --
 Camille



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[SR-Users] websocket devices keepalive not working

2015-06-30 Thread Yuriy Gorlichenko

I try to send keepalive requests (options) to clients at usrloc

I tried 4 mechanisms
1) nathelper

modparam(nathelper, sipping_method, OPTIONS)
modparam(nathelper, natping_interval, 15)
modparam(nathelper, ping_nated_only, 0)
modparam(nathelper, sipping_bflag, 7)
modparam(nathelper, sipping_from, sip:pinger@mydomain)

but it didn't send keepalives to noone client

2)nat_traversal
I tried to trigger keepalives with nat_keepalive() when REGISTER coming to
my service

It works fine with UDP packets

modparam(nat_traversal, keepalive_interval, 15)
modparam(nat_traversal, keepalive_method, OPTIONS)
modparam(nat_traversal, keepalive_from, sip:pinger@mydomain)
modparam(nat_traversal, keepalive_state_file,
/var/run/keepalive_state)

3)So I also tried to do it with WebSocket keepalive mechanism, but it still
not work for (off course I checked only WS devices)

modparam(websocket, keepalive_mechanism, 15)
modparam(websocket, keepalive_timeout, 15)
modparam(websocket, keepalive_interval, 15)

4) I tried to do it with tcpops

Only for ws and tcp (WS is use TCP as i know)
tcp_keepalive_enable(15, 5, 15);

same result...

So I need mechanism to check all devices for keepalive. I think nathelper
works with all protos (As I read at docs)

Can somebody explain to me  what I doing wrong?
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Re: [SR-Users] websocket devices keepalive not working

2015-06-30 Thread Yuriy Gorlichenko

Now did this.
Still no Options

if (!save(location)){
sl_reply_error();
}
else
{
setbflag(7);
...

2015-06-30 17:05 GMT+03:00 Daniel Tryba d.tr...@pocos.nl:

 On Tuesday 30 June 2015 15:49:26 Yuriy Gorlichenko wrote:
  1) nathelper
 
  modparam(nathelper, sipping_method, OPTIONS)
  modparam(nathelper, natping_interval, 15)
  modparam(nathelper, ping_nated_only, 0)
  modparam(nathelper, sipping_bflag, 7)
  modparam(nathelper, sipping_from, sip:pinger@mydomain)
 
  but it didn't send keepalives to noone client

 This isn't enough, you need to set the sipping_bflag on REGISTER, eg:
 if (!save(location))
 {
  sl_reply_error();
 }
 else
 {
  setbflag(7);
 }

 Did you do this and still got not OPTIONS send to the clients?

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[SR-Users] Kamailio no recieved 480/486 messages

2015-06-30 Thread Yuriy Gorlichenko

At my confiig I have route like this

onreply_route[REPLY_FROM_WS] {


if(status=~[12][0-9][0-9]) {
xlog(L_INFO, Manage_Reply from webrtc client {$si:$sp} for method {$rm}:
$rs);
rtpengine_manage(force trust-address replace-origin
replace-session-connection DTLS=passive ICE=remove RTP/AVP);
route(NATMANAGE);
}
 }

This route successfully handling replies 180 183 and 200 and then (after
natmanage route) resend to mediaserver.

When callee client reject call  it send 480 ot 486 message that not
resended by kamailio nowhere.

First of all I thought that 480 and 486 replies are failure replies but at
log I see its at onreply route.

So Ok . I added this to onreply_route
 if (status=~48[06]){
xlog(L_INFO, Manage_Reply from webrtc client {$si:$sp} for method {$rm}:
$rs);
rtpengine_manage(force trust-address replace-origin
replace-session-connection DTLS=passive ICE=remove RTP/AVP);
route(NATMANAGE);
}

But message just going through natmanage route and goes nowhere

My NATMANAGE route is

   xlog(L_INFO,NATMANAGE reply  {$rm});
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
add_contact_alias();
}
xlog(L_INFO,reply  {$rm});

}

I see at my log only first NATMANAGE xlog... no more...

thnks for answer
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[SR-Users] Kannot start kamailio 4.3

Try to start kamailio on Ubuntu 14.04.02

Get this errors
ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock:
bind: No such file or directory [2]
Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]:
mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed
Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: core
[sr_module.c:945]: init_mod(): Error while initializing module ctl
(/usr/local/lib64/kamailio/modules/ctl.so)

my params for ctl

# - ctl params -
modparam(ctl, binrpc, unix:/var/run/kamailio/kamailio_ctl)


version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 1dbd53
compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4
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Re: [SR-Users] Kannot start kamailio 4.3

]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 26, -1, 0x0)
fd_no=18 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 27
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 14, pid 4139 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 27, -1, 0x0)
fd_no=17 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 29
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 15, pid 4140 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 29, -1, 0x0)
fd_no=16 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 30
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 16, pid 4141 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 30, -1, 0x0)
fd_no=15 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 32
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 17, pid 4148 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 32, -1, 0x0)
fd_no=14 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 34
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 18, pid 4149 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 34, -1, 0x0)
fd_no=13 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 36
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 19, pid 4150 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 36, -1, 0x0)
fd_no=12 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: CRITICAL: core
[pass_fd.c:275]: receive_fd(): EOF on 38
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3448]: handle_ser_child(): dead child 20, pid 4153 (shutting
down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 38, -1, 0x0)
fd_no=11 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3216]: handle_tcp_child(): dead tcp child 0 (pid 4140, no 15)
(shutting down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 31, -1, 0x0)
fd_no=10 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3216]: handle_tcp_child(): dead tcp child 1 (pid 4141, no 16)
(shutting down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 33, -1, 0x0)
fd_no=9 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3216]: handle_tcp_child(): dead tcp child 2 (pid 4148, no 17)
(shutting down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 35, -1, 0x0)
fd_no=8 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3216]: handle_tcp_child(): dead tcp child 3 (pid 4149, no 18)
(shutting down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 37, -1, 0x0)
fd_no=7 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3216]: handle_tcp_child(): dead tcp child 4 (pid 4150, no 19)
(shutting down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 39, -1, 0x0)
fd_no=6 called
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[tcp_main.c:3216]: handle_tcp_child(): dead tcp child 5 (pid 4153, no 20)
(shutting down?)
Jun 23 09:56:22 kamailio-test kamailio[4155]: DEBUG: core
[io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa02120, 41, -1, 0x0)
fd_no=5 called


2015-06-23 17:00 GMT+03:00 Roberto Fichera ker...@tekno-soft.it:

  On 06/23/2015 03:57 PM, Yuriy Gorlichenko wrote:

 Hi Yuriy,

  done this.
 now kamailio fails when try to fork...

 Jun 23 09:56:22 kamailio-test kamailio[4122]: ALERT: core [main.c:725]:
 handle_sigs(): child process 4133 exited normally, status=255
 Jun 23 09:56:22 kamailio-test kamailio[4122]: INFO: core [main.c

Re: [SR-Users] Kannot start kamailio 4.3

done this.
now kamailio fails when try to fork...

Jun 23 09:56:22 kamailio-test kamailio[4122]: ALERT: core [main.c:725]:
handle_sigs(): child process 4133 exited normally, status=255
Jun 23 09:56:22 kamailio-test kamailio[4122]: INFO: core [main.c:743]:
handle_sigs(): terminating due to SIGCHLD


2015-06-23 16:08 GMT+03:00 Roberto Fichera ker...@tekno-soft.it:

  On 06/23/2015 03:03 PM, Roberto Fichera wrote:

 On 06/23/2015 02:58 PM, Yuriy Gorlichenko wrote:

 Hi,

  Try to start kamailio on Ubuntu 14.04.02

 Get this errors
 ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock:
 bind: No such file or directory [2]
 Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]:
 mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed
 Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: core
 [sr_module.c:945]: init_mod(): Error while initializing module ctl
 (/usr/local/lib64/kamailio/modules/ctl.so)

 my params for ctl

 # - ctl params -
 modparam(ctl, binrpc, unix:/var/run/kamailio/kamailio_ctl)


 mkdir -p /var/run/kamailio
 chmod kamailio:kamailio /var/run/kamailio


 sorry

 chown kamailio:kamailio /var/run/kamailio


 and restart it



 version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53
 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 1dbd53
 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4



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Re: [SR-Users] Kannot start kamailio 4.3

 kamailio[1348]: DEBUG: debugger
[debugger_mod.c:198]: child_init(): rank is (11)
Jun 23 10:15:20 kamailio-test kamailio[1347]: DEBUG: core [db.c:319]:
db_do_init2(): connection 0x7f975c151c90 found in pool
Jun 23 10:15:20 kamailio-test kamailio[1348]: DEBUG: core
[local_timer.c:61]: init_local_timer(): timer_list between 0xa46588 and
0xa8a588
Jun 23 10:15:20 kamailio-test kamailio[1347]: DEBUG: core
[sr_module.c:897]: init_mod_child(): rank 10: avpops
Jun 23 10:15:20 kamailio-test kamailio[1347]: DEBUG: core
[sr_module.c:897]: init_mod_child(): rank 10: ndb_redis
Jun 23 10:15:20 kamailio-test kamailio[1313]: ALERT: core [main.c:725]:
handle_sigs(): child process 1330 exited normally, status=255
Jun 23 10:15:20 kamailio-test kamailio[1349]: DEBUG: core
[sr_module.c:897]: init_mod_child(): rank 12: uac
Jun 23 10:15:20 kamailio-test kamailio[1349]: DEBUG: core
[sr_module.c:897]: init_mod_child(): rank 12: dialog
Jun 23 10:15:20 kamailio-test kamailio[1353]: DEBUG: core
[sr_module.c:897]: init_mod_child(): rank -4: websocket
Jun 23 10:15:20 kamailio-test kamailio[1313]: INFO: core [main.c:743]:
handle_sigs(): terminating due to SIGCHLD
Jun 23 10:15:20 kamailio-test kamailio[1349]: DEBUG: core
[sr_module.c:897]: init_mod_child(): rank 12: auth_db
Jun 23 10:15:20 kamailio-test kamailio[1343]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1340]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1338]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1335]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1334]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1333]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1328]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1324]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1323]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1322]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1321]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1318]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1317]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1316]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1315]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1342]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1347]: INFO: core [main.c:794]:
sig_usr(): signal 15 received
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: ndb_redis
[ndb_redis_mod.c:130]: mod_destroy(): cleaning up
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: core
[db_pool.c:100]: pool_remove(): removing connection from the pool
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: tm [t_funcs.c:86]:
tm_shutdown(): DEBUG: tm_shutdown : start
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: tm [t_funcs.c:89]:
tm_shutdown(): DEBUG: tm_shutdown : emptying hash table
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: tm [t_funcs.c:91]:
tm_shutdown(): DEBUG: tm_shutdown : removing semaphores
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: tm [t_funcs.c:93]:
tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: tm [t_funcs.c:96]:
tm_shutdown(): DEBUG: tm_shutdown : done
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: tls [tls_init.c:720]:
destroy_tls_h(): tls module final tls destroy
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: core
[mem/shm_mem.c:232]: shm_mem_destroy(): shm_mem_destroy
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: core
[mem/shm_mem.c:235]: shm_mem_destroy(): destroying the shared memory lock
Jun 23 10:15:20 kamailio-test kamailio[1313]: DEBUG: core [main.c:747]:
handle_sigs(): terminating due to SIGCHLD


2015-06-23 17:08 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 un 23 09:56:22 kamailio-test kamailio[4150]: DEBUG: core [db.c:205]:
 db_bind_mod(): using db bind api for db_mysql
 Jun 23 09:56:22 kamailio-test kamailio[4150]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7fcefca07c90 found in pool
 Jun 23 09:56:22 kamailio-test kamailio[4150]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 11: avpops
 Jun 23 09:56:22 kamailio-test kamailio[4150]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 11: ndb_redis
 Jun 23 09:56:22 kamailio-test kamailio[4150]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 11: uac
 Jun 23 09:56:22 kamailio-test kamailio[4150

Re: [SR-Users] Kannot start kamailio 4.3

At last log i Found function that gives me this alert

ALERT: core [main.c:725]: handle_sigs(): child process 1408 exited
normally, status=255


 kamailio-test kamailio[1408]: CRITICAL: mi_fifo [mi_fifo.c:251]:
fifo_process(): The function mi_init_fifo_server returned with error!!!

fifo file was wrong configured


2015-06-23 17:17 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 now restart host. Last log is here

 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core [sruid.c:106]:
 sruid_init(): root for sruid is [uloc-558969f8-53e-] (0 / 18)
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: mi_rpc
 [mi_rpc_mod.c:104]: child_init(): initializing child[9] for rpc handling
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: ctl
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7f975c150488 found in pool
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: mi_rpc
 Jun 23 10:15:20 kamailio-test kamailio[1340]: DEBUG: core
 [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa463c0, 32, 1, (nil)),
 fd_no=0
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core [sruid.c:106]:
 sruid_init(): root for sruid is [ulcx-558969f8-53f-] (0 / 18)
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: mi_rpc
 [mi_rpc_mod.c:104]: child_init(): initializing child[8] for rpc handling
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: sqlops
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7f975c150488 found in pool
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:672]: find_mod_export_record(): find_export_record: found
 db_bind_api in module db_mysql
 [/usr/local/lib64/kamailio/modules/db_mysql.so]
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core [sruid.c:106]:
 sruid_init(): root for sruid is [ulcx-558969f8-53e-] (0 / 18)
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: sqlops
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core [db.c:205]:
 db_bind_mod(): using db bind api for db_mysql
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:672]: find_mod_export_record(): find_export_record: found
 db_bind_api in module db_mysql
 [/usr/local/lib64/kamailio/modules/db_mysql.so]
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7f975c151c90 found in pool
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: avpops
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: ndb_redis
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core [db.c:205]:
 db_bind_mod(): using db bind api for db_mysql
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7f975c151c90 found in pool
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: avpops
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: ndb_redis
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: uac
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: dialog
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: auth_db
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7f975c151c90 found in pool
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: uac
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: nathelper
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: dialog
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: auth_db
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 9: rtpengine
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core [db.c:319]:
 db_do_init2(): connection 0x7f975c151c90 found in pool
 Jun 23 10:15:20 kamailio-test kamailio[1343]: DEBUG: rtpengine
 [rtpengine.c:489]: bind_force_send_ip(): force_send_ip_str not specified in
 .cfg file!
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: nathelper
 Jun 23 10:15:20 kamailio-test kamailio[1342]: DEBUG: core
 [sr_module.c:897]: init_mod_child(): rank 8: rtpengine
 Jun 23 10:15:20 kamailio-test rtpengine[1302]: Received command 'ping'
 from 127.0.0.1:50860
 Jun 23 10:15:20 kamailio-test rtpengine[1302

Re: [SR-Users] kamailio fails after start

2015-06-22 Thread Yuriy Gorlichenko

git
OpenVZ have an issue with linux headers. When I run uname -r it give me
heades 2.6 that works at the host server. Ubuntu have 3.13.
I think it may be main reason of this issue or simphtom af another issue at
OpenVZ systems...

2015-06-22 22:33 GMT+03:00 Victor Seva linuxman...@torreviejawireless.org:

 On 06/22/2015 03:05 PM, Yuriy Gorlichenko wrote:
  Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual
  systemOpenVZ.

 git or deb?

  after starting kamailio I see that it runs ok with
  kamailio start
  or
  kamctl start

 If deb:
   I would say that you should use /etc/init.d/kamailio start
   check /etc/defaults/kamailio too


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[SR-Users] kamailio fails after start

2015-06-22 Thread Yuriy Gorlichenko

Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual
systemOpenVZ.

after starting kamailio I see that it runs ok with
kamailio start
or
kamctl start

at ps -ax I see working processes.
But after one minute of working kamailio fails with

kamailio: ERROR: core [daemonize.c:315]: daemonize(): Main process exited
before writing to pipe

I see nothing more at debug file. No errors. (I run with debug=4)

my cfg parameters at start

mhomed=1
memdbg=5
memlog=5

fork=yes
children=6


version: kamailio 4.2.4 (x86_64/linux) ac3682
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: ac3682
compiled on 04:27:04 Apr 10 2015 with gcc 4.8.2
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Re: [SR-Users] UAC Module

2015-05-02 Thread Yuriy Gorlichenko

One more thing may be useful for you. If you will get an error with cseq
numder when provider send 401/407 message- usedialog module. It resole an
issuevwith cseq( read documentation)
30.04.2015 18:23 пользователь SamyGo govoi...@gmail.com написал:

 I'd like you to google around, there is a function available from another
module which will apply the changes in SIP Message.


 On Thu, Apr 30, 2015 at 9:51 AM, Ali Jibran alijib...@vividtech.io
wrote:

 Perfect. Yeah got the working.

 Just one last issue. I don’t think this is rewriting the header. When I
log the header again after the changes it still shows me the old values.



 From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf
Of SamyGo
 Sent: Thursday, April 30, 2015 6:50 PM


 To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] UAC Module



 t_on_failure(F_VOIP) to be used before t_relay();

 That will arm the call to go to F_VOIP on failure responses.



 On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran alijib...@vividtech.io
wrote:



 #!ifdef WITH_FREESWITCH

 if(is_method(INVITE)  route(FROMFREESWITCH))) {

 xlog(L_INFO ,[$fU/$tU@$si:$sp]{$rm} Call from
FreeSWITCH needs to be sent TOVOIP \n);

 route(TOVOIP);

 t_on_failure(F_VOIP);

 exit;

 }



 #!endif







 route[TOVOIP] {

 xlog(L_INFO,ALERT: $fu to $tu  );

 $fU=XX;

 $td=sip.voipfone.net;

 $du=sip:...@sip.voipfone.net;

 t_relay();



 }





 failure_route[F_VOIP] {

 uac_auth();

 xlog(L_INFO,ALERT: IN FAIL);

}





 I tried this but it never makes it to the failure branch. Im a newbie
to kamailio and still working around the scripting. Can you please help me
out here to where I am making the mistake?



 From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf
Of SamyGo
 Sent: Thursday, April 30, 2015 9:18 AM
 To: Kamailio (SER) - Users Mailing List
 Subject: Re: [SR-Users] UAC Module



 Hi Jibran,



 Here is an old thread as reference:


 http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html



 I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE
with username/password on a Provider for huge number of calls..imagine
sending thousands of call to that provider and for each call going through
the trouble of exchanging authentication.

 Thats why its usually recommended to go with IP-Authentication only.
Send INVITE and Provider says Lets do this call,simple and easy.



 From the configuration perspective this is my idea of still using UAC.



 - Call coming from FS on kamailio

 - Rewrite the from-uri  (so the provider receives calls from the
registered username)

 - modify the to-domain part to contain the IP address of the provider

 - set the $du to ip of the provider, and t_relay() the call.

 - Most likely the Provider would say Proxy-Auth required..that can be
caught in failure_route[]

 - There you can call the uac_auth() function to have username.password
attached to the response of above.
http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()

 - once this function is successful send the INVITE again to the
provider.



 Last three steps can be the following snippet of code(reference from
here):



 failure_route[2] {

  if (t_check_status(40[17])) {

 xlog(got challenged \n);

 if (uac_auth()) {

 xlog(auth was succesful \n);

 t_relay(udp:ip_addr:5060); //provider's IP_ADDR

 }

 }





 I hope you get IP Auth from the provider, and find the reply useful.



 Regards,







 On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran alijib...@vividtech.io
wrote:


 Hi all.
 I have this setup.
 Trunk---KamailioFreeSWITCH

 I have a trunk from a sip provided and registered successfully with
the UAC module. Incoming is working fine. I need to make out going through
kamailio too.

 I have it in the dialplan to forward the invite to kamailio from
FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I
make the call via the trunk?

 Basically this is what I'm trying to workout
 FSkamailiotrunk.


 Any help will be much appreciated. Thanks.
 AJ
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Re: [SR-Users] UAC not answer with new register after 401 reply from porvider

2015-04-30 Thread Yuriy Gorlichenko

This happens only with one trunk. We also have plivo trunks and it works
fine.

Syslog show nothing when this message comes

Started with debug mode and saw that realms didn't mach. Sorry for stupid
questions. All works fine.

2015-04-30 13:24 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 does that happen in all cases or just for some records? Can you rung with
 debug=3 and check the syslog messages for what happens at that moment when
 401 is processed?

 Cheers,
 Daniel


 On 30/04/15 11:37, Yuriy Gorlichenko wrote:

 Hello. We have an issue with REGISTER to Provider. When Provider answers
 401 Kamailio don't send any REGISTER with digest auth

 IP ourservice.com.5068  provider.dev.5060: UDP, length 468
 E...U...@.3'
 ..AREGISTER sip:provider.dev SIP/2.0
 Via: SIP/2.0/UDP ourservice.com:5068
 ;branch=z9hG4bK4238.45a4a626.0
 To: sip:lo...@provider.dev
 From: sip:lo...@provider.dev;tag=6a39af2190f8b2c99cc63bfeb3a3959c-1ff8
 CSeq: 10 REGISTER
 Call-ID: 1b3f9f5d5822eca2-16...@ourservice.com
 Max-Forwards: 70
 Content-Length: 0
 User-Agent: kamailio (4.3.0-dev5 (x86_64/linux))
 Contact: sip:lo...@ourservice.com:5068
 Expires: 360


  IP provider.dev.5060  ourservice.com.5068: UDP, length 572
 E..XC...-.W.
 DkJSIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP ourservice.com:5068
 ;branch=z9hG4bK4238.45a4a626.0;received=23.100.24.250
 From: sip:lo...@provider.dev;tag=6a39af2190f8b2c99cc63bfeb3a3959c-1ff8
 To: sip:lo...@provider.dev;tag=as6ad61d4a
 Call-ID: 1b3f9f5d5822eca2-16...@ourservice.com
 CSeq: 10 REGISTER
 Server: FastTel SoftSwitch
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces
 WWW-Authenticate: Digest algorithm=MD5, realm=provider.dev,
 nonce=63fc6951
 Content-Length: 0



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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


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[SR-Users] UAC not answer with new register after 401 reply from porvider

2015-04-30 Thread Yuriy Gorlichenko

Hello. We have an issue with REGISTER to Provider. When Provider answers
401 Kamailio don't send any REGISTER with digest auth

IP ourservice.com.5068  provider.dev.5060: UDP, length 468
E...U...@.3'
..AREGISTER sip:provider.dev SIP/2.0
Via: SIP/2.0/UDP ourservice.com:5068
;branch=z9hG4bK4238.45a4a626.0
To: sip:lo...@provider.dev
From: sip:lo...@provider.dev;tag=6a39af2190f8b2c99cc63bfeb3a3959c-1ff8
CSeq: 10 REGISTER
Call-ID: 1b3f9f5d5822eca2-16...@ourservice.com
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (4.3.0-dev5 (x86_64/linux))
Contact: sip:lo...@ourservice.com:5068
Expires: 360

IP provider.dev.5060  ourservice.com.5068: UDP, length 572
E..XC...-.W.
DkJSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ourservice.com:5068
;branch=z9hG4bK4238.45a4a626.0;received=23.100.24.250
From: sip:lo...@provider.dev;tag=6a39af2190f8b2c99cc63bfeb3a3959c-1ff8
To: sip:lo...@provider.dev;tag=as6ad61d4a
Call-ID: 1b3f9f5d5822eca2-16...@ourservice.com
CSeq: 10 REGISTER
Server: FastTel SoftSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=provider.dev,
nonce=63fc6951
Content-Length: 0
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[SR-Users] append_branch. How to disable original URI

2015-04-17 Thread Yuriy Gorlichenko

Hello. I thry to integrate redis for location module and first at all that
I do - dublicate location to redis.

First At all I create analog of lookup procedure that use location but from
redis. I take values from location and create branches by mannualy. All
works good but branch route create dublicate of first branch. We talk about
it already and I cnow that branch route creates first original Request and
then create branches. So my question is how to disable creation of original
URI?

My cfg part of creation branches is:
First of all I create massive of needed endpoints and then create branches
as bellow.

I create it with different сucles  while because websockets not blocked
when creates dublicate INVITE, but some UDP endpoints can not take call
because answer to kamailio 482 reply and CANCELs call.

It works fine when I logged on with wesocket device and UNP at one time.
But when I logget with 2 UDP devices only this algorithm not worked.

Thanks for help.



$var(k)=0;
xlog(L_INFO, request URI is $ru);
while ($var(k)= $var(j)){
 if ($(avp(device_contact)[$var(k)])=~device){
xlog(L_INFO, This is a classic UDP call to endpoint);
if ($(avp(device_received)[$var(k)])==){
xlog(L_INFO, Received string is EMPTY);
$du=sip:+$(avp(device_contact[$var(k)]){s.select,1,@});
}
else
{
xlog(L_INFO, Received string is {$avp(device_received)[$var(k)]});
$du=$(avp(device_received)[$var(k)]);
}
$var(UDP_contact)=sip:+$(avp(device_contact[$var(k)]){s.select,1,@});
 append_branch(sip:$tU@$(du{s.select,1,:}),0.3);
 xlog(L_INFO,Classic Destination URI is
{$(avp(device_contact[$var(k)]){s.select,1,@})} for {$tU}}. Destination is
{$du}\n);
}
$var(k) = $var(k) + 1;
 }



$var(k)=0;
xlog(L_INFO, request URI is $ru);
while ($var(k)=$var(j)){
 if ($(avp(device_contact)[$var(k)])=~transport=ws){
xlog(L_INFO, This is a classic UDP call to endpoint);
 xlog(L_INFO, Received string is {$avp(device_received)[$var(k)]});
$du=$(avp(device_received)[$var(k)]);
 append_branch(sip:$tU@$(du{s.select,1,:}),0.7);
 xlog(L_INFO,Classic Destination URI is
{$(avp(device_contact[$var(k)]){s.select,1,@})} for {$tU}}. Destination is
{$du}\n);
}
$var(k) = $var(k) + 1;
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Re: [SR-Users] seturi analog for Kamailio 4.3

2015-03-16 Thread Yuriy Gorlichenko

I mean that algorithm of creating new headers of creating INVOTE message is
the same for branch_branch route and single client. Only one difference i
that I use $ru=sip:+$tU+@+$(du{s.select,1,:} vs append_branch(sip:$tU@
$(du{s.select,1,:}))

This is full algo of my reqest
the first is I check at location table num of endpoints for this peer and
then at the while do this

sql_pvquery(ca, select
received from location where
contact='$dbr(ra=[$var(i),0])',$var(recieved));
$du=$var(recieved);
xlog(L_INFO,SQL query return recieved {$var(recieved)} for {$tU}.
Destination is {$du}\n);
# if only one client used it means that I use $ru
if ($dbr(ra=rows)==1)
{
 xlog(L_INFO,Single ANGENT);
 xlog(L_INFO,WS Branch is {$du)} for {$tU}\n);
 rtpengine_manage(force trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF);
 t_on_reply(REPLY_FROM_WS);
 $ru=sip:+$tU+@+$(du{s.select,1,:});
 route(FINAL_RELAY);
}
else
{
append_branch(sip:$tU@$(du{s.select,1,:}),0.7);
}

#after while ended if more that one endpoint used

t_on_branch(1);
return;

branch_route[1]{

if($du=~transport=ws){
xlog(L_INFO,Websocket Branch is {$du} for {$tU}\n);
rtpengine_manage(force trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF);
t_on_reply(REPLY_FROM_WS);
 }
else{
xlog(L_INFO,UDP Branch is {$du)} for {$tU}\n);
rtpengine_manage(replace-origin replace-session-connection ICE=remove
RTP/AVP);
t_on_reply(MANAGE_CLASSIC_REPLY);
}
}

route[FINAL_RELAY]
{

if (!t_relay()) {
sl_reply_error();
}
return;
}



2015-03-16 15:46 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 request_route

 2015-03-16 15:43 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Is this used in request_route or in failure_route or other routing
 block?

 Cheers,
 Daniel


 On 16/03/15 12:44, Yuriy Gorlichenko wrote:

 If I use

 $ru=sip:+$tU+@+$(du{s.select,1,:});

  if (!t_relay()) {
  sl_reply_error();
 }

  I see

 t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding
 Mar 16 11:36:04 Kamailio kamailio[4335]: ERROR: sl [sl_funcs.c:363]:
 sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server
 error occurred (6/SL)

   if I use append_branch()
 for single client - thats is ok but 2 invites going to client.
 So client may pickup call and it successfully established.

 2015-03-16 13:24 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 instead of seturi use:

 $ru = sip: + $tU + @ + $(du{s.select,1,:});

 Cheers,
 Daniel


 On 16/03/15 05:44, Yuriy Gorlichenko wrote:

  Now. when I use

 seturi(sip:$tU@$(du{s.select,1,:}));

 I see error at my log

  ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
  ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
  ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
  ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
  ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error
 used: Regretfully, we were not able to process the URI (479/SL)
  ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error
 used: Regretfully, we were not able to process the URI (479/SL)

 As i see error generate twice maby because I ure t_on branch() route



 2015-03-16 7:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Hello. I try to call multi[ple endpoints from my server using
 append_branch. It works fine but when I have only one endpoint - kamailio
 generate 2 INVITE requests to it.
 As I understand it is original request and the next one is branch.
 I used seturi() before for sending original reqest to destination, but
 I can not see this function at kamailio 4.3.
 Kamailio 4.3 uses send() dunnction? but it works at stateless mode. I
 need analog of it or seturi that works at statefull mode.





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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


 ___
 SIP Express Router (SER) and Kamailio (OpenSER

Re: [SR-Users] seturi analog for Kamailio 4.3

2015-03-16 Thread Yuriy Gorlichenko

If I use

$ru=sip:+$tU+@+$(du{s.select,1,:});

if (!t_relay()) {
sl_reply_error();
}

I see

t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding
Mar 16 11:36:04 Kamailio kamailio[4335]: ERROR: sl [sl_funcs.c:363]:
sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server
error occurred (6/SL)

 if I use append_branch()
for single client - thats is ok but 2 invites going to client.
So client may pickup call and it successfully established.

2015-03-16 13:24 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 instead of seturi use:

 $ru = sip: + $tU + @ + $(du{s.select,1,:});

 Cheers,
 Daniel


 On 16/03/15 05:44, Yuriy Gorlichenko wrote:

 Now. when I use

 seturi(sip:$tU@$(du{s.select,1,:}));

 I see error at my log

  ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
  ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
  ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
  ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
  ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used:
 Regretfully, we were not able to process the URI (479/SL)
  ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used:
 Regretfully, we were not able to process the URI (479/SL)

 As i see error generate twice maby because I ure t_on branch() route



 2015-03-16 7:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Hello. I try to call multi[ple endpoints from my server using
 append_branch. It works fine but when I have only one endpoint - kamailio
 generate 2 INVITE requests to it.
 As I understand it is original request and the next one is branch.
 I used seturi() before for sending original reqest to destination, but I
 can not see this function at kamailio 4.3.
 Kamailio 4.3 uses send() dunnction? but it works at stateless mode. I
 need analog of it or seturi that works at statefull mode.





 ___
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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


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Re: [SR-Users] seturi analog for Kamailio 4.3

2015-03-16 Thread Yuriy Gorlichenko

request_route

2015-03-16 15:43 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Is this used in request_route or in failure_route or other routing block?

 Cheers,
 Daniel


 On 16/03/15 12:44, Yuriy Gorlichenko wrote:

 If I use

 $ru=sip:+$tU+@+$(du{s.select,1,:});

  if (!t_relay()) {
  sl_reply_error();
 }

  I see

 t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding
 Mar 16 11:36:04 Kamailio kamailio[4335]: ERROR: sl [sl_funcs.c:363]:
 sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server
 error occurred (6/SL)

   if I use append_branch()
 for single client - thats is ok but 2 invites going to client.
 So client may pickup call and it successfully established.

 2015-03-16 13:24 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 instead of seturi use:

 $ru = sip: + $tU + @ + $(du{s.select,1,:});

 Cheers,
 Daniel


 On 16/03/15 05:44, Yuriy Gorlichenko wrote:

  Now. when I use

 seturi(sip:$tU@$(du{s.select,1,:}));

 I see error at my log

  ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
  ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
  ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
  ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
  ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error
 used: Regretfully, we were not able to process the URI (479/SL)
  ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error
 used: Regretfully, we were not able to process the URI (479/SL)

 As i see error generate twice maby because I ure t_on branch() route



 2015-03-16 7:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Hello. I try to call multi[ple endpoints from my server using
 append_branch. It works fine but when I have only one endpoint - kamailio
 generate 2 INVITE requests to it.
 As I understand it is original request and the next one is branch.
 I used seturi() before for sending original reqest to destination, but I
 can not see this function at kamailio 4.3.
 Kamailio 4.3 uses send() dunnction? but it works at stateless mode. I
 need analog of it or seturi that works at statefull mode.





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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


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[SR-Users] seturi analog for Kamailio 4.3

2015-03-15 Thread Yuriy Gorlichenko

Hello. I try to call multi[ple endpoints from my server using
append_branch. It works fine but when I have only one endpoint - kamailio
generate 2 INVITE requests to it.
As I understand it is original request and the next one is branch.
I used seturi() before for sending original reqest to destination, but I
can not see this function at kamailio 4.3.
Kamailio 4.3 uses send() dunnction? but it works at stateless mode. I need
analog of it or seturi that works at statefull mode.
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Re: [SR-Users] seturi analog for Kamailio 4.3

2015-03-15 Thread Yuriy Gorlichenko

Now. when I use

seturi(sip:$tU@$(du{s.select,1,:}));

I see error at my log

 ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
 ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
 ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
 ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR: t_newtran: new_t failed
 ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used:
Regretfully, we were not able to process the URI (479/SL)
 ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used:
Regretfully, we were not able to process the URI (479/SL)

As i see error generate twice maby because I ure t_on branch() route



2015-03-16 7:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Hello. I try to call multi[ple endpoints from my server using
 append_branch. It works fine but when I have only one endpoint - kamailio
 generate 2 INVITE requests to it.
 As I understand it is original request and the next one is branch.
 I used seturi() before for sending original reqest to destination, but I
 can not see this function at kamailio 4.3.
 Kamailio 4.3 uses send() dunnction? but it works at stateless mode. I need
 analog of it or seturi that works at statefull mode.



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Re: [SR-Users] ndb_redis set null reply to EXISTS query

2015-03-06 Thread Yuriy Gorlichenko

modparam(ndb_redis, server,
name=srv1;addr=non_local_serv;port=6379;db=4;pass=mypass)

[TOASTERISK]
redis_cmd(srv1, EXISTS $si, s);
xlog(L_INFO,ASTERISK with ip $si is {$redis(s=value)});
 $var(setid)=0;
if ($redis(s=value) == 0) {
xlog(L_INFO,Request {$rm} from $si != {$var(dest)} It means call NOT
from ASTRISK);
}

Ok/ I will try to do it with debug

2015-03-06 10:51 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

 Unless you send the relevant part of your configuration and of the
 restults the tests I proposed it is really diificult to provide more
 detailed advice. For instance, is your redis instance in the same machine
 as Kamailio? Are you using unix sockets or tcp sockets? Can you run
 kamailio in debig mode to see any potentially helpful message?

 Javi
 On 06/03/15 08:04, Yuriy Gorlichenko wrote:

 arrives kamailio stil disconnects from redis.  Haw can I debug this
 deeper?



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Re: [SR-Users] ndb_redis set null reply to EXISTS query

2015-03-05 Thread Yuriy Gorlichenko

Hello All. I still have problems with disconnect kamailio to redis.

All tests above  are correct but when INVITE arrives kamailio stil
disconnects from redis.  Haw can I debug this deeper?

2015-02-27 16:48 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello

 I ignore your setup, a few tips that can help you to narrow down the
 problem:

 -Make sure your kamailio is connected to redis: 'redis-cli info | grep
 connected_clients' should show a number bigger than the number of kamailio
 children

 -Print with xlog the value of $si before calling redis_cmd

 -At the redis prompt (redis-cli) type the command EXISTS x, with x being
 the value of $si. If that gives you a 1 it means that the key exists,
 otherwise you need to create it.

 -Pritn the values of $redis(s=type), $redis(s=value) after calling
 redis_cmd

 -With redis-cli monitor you are goign to see all the commands sent to redis

 Javi

 On 27/02/15 14:30, Yuriy Gorlichenko wrote:

 What type of info can I provide for deeper analys of this situation?

 2015-02-27 14:34 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello
 On 27/02/15 11:58, Yuriy Gorlichenko wrote:

 at the monitor I see nothing about this request

  It's difficult to say without further information, but for some reason
 your command is not hitting redis, probably because the connection is down.

 Javi


 2015-02-27 13:21 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Now I see that null values recieved after I see this at kamailio log

 redisc_exec(): Redis error: Server closed the connection


 2015-02-27 12:45 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello,
 check out http://redis.io/commands/monitor
 It will help you to determine the exact command that kamailio is
 delivering to the server when you execute the redis_cmd(...) function
 inside the script.
 For a non null reply, yo will need a key in redis with the same value
 as $si.

 Javi

 On 27/02/15 10:04, Yuriy Gorlichenko wrote:

 I will try. I new at redis. Does cli monitor get resul of kamailio
 request at the cli?

 2015-02-27 11:38 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello

 can you check with redis-cli monitor what is the command sent to
 Redis in that case?

 Javi
 On 27/02/15 09:15, Yuriy Gorlichenko wrote:

  Hello I try to get some replies from redis. Time after time redis
 request give me null result. But redis bs not disconnected.

 This happens only with websocket endpoints. My queries is:

 redis_cmd(srv1, EXISTS $si, s);

 So at xLOG i see that $si correctly sended, but result is null. At db
 I keep key IP with value TIMESTAMP

 Intresting that route, that givesme this result idependend of WS or
 UDP endpoint. IT does not know about it anything.


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Re: [SR-Users] REmove Header line with tags

2015-03-03 Thread Yuriy Gorlichenko

I addes msg_apply_changes() but now kamailio dont send ACK message. I see
that it only arrievs to kamailio.

2015-03-03 9:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 remove_hf() removes the entire header, including its parameters.

 The issue here seems to be related to the fact that you remove Via header
 and the core adds rport and received parameters -- this happens due to
 requirements from SIP RFCs.

 What you need to do is to apply changes once the Via is removed so the
 core doesn't see it anymore, see:

   -
 http://kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers_or

 Cheers,
 Daniel


 On 03/03/15 01:28, Yuriy Gorlichenko wrote:

 I need to remove all header line witht tags but remove_hf() removes only
 Header:value
 All tags after ; stay at the packet as garbage and next Header moves up
 As there

 ACK sip:12345678...@phone.provider.com SIP/2.0
 Via: SIP/2.0/UDP sip.server.com:5068
 ;branch=z9hG4bKcb7a.9f70ce5990dd153cd0eaa6f1762c059c.0.cs102
 *There var another Via header that I removed with remove_hf_re but tags
 is stiil there and Route header mover 
 up*;rport=1578;received=23.101.134.111Route:
 sip:54.241.2.206;lr=on;ftag=as38f20f7b;did=ce8.10a
 Max-Forwards: 70
 From: New User sip:00987654...@phone.provider.com;tag=as38f20f7b
 To: sip:12345678...@phone.provider.com;tag=B4r1B569001DB
 Contact: sip:contact@10.0.1.17:50600;alias=23.101.134.111~1578~1
 Call-ID: 473fc8a64d7d96aa515780dc6237a6d5@10.0.1.17:50600
 CSeq: 103 ACK
 User-Agent: Some Device
 Content-Length: 0



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 http://www.linkedin.com/in/miconda
 Kamailio World Conference, May 27-29, 2015
 Berlin, Germany - http://www.kamailioworld.com


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[SR-Users] REmove Header line with tags

2015-03-02 Thread Yuriy Gorlichenko

I need to remove all header line witht tags but remove_hf() removes only
Header:value
All tags after ; stay at the packet as garbage and next Header moves up
As there

ACK sip:12345678...@phone.provider.com SIP/2.0
Via: SIP/2.0/UDP sip.server.com:5068
;branch=z9hG4bKcb7a.9f70ce5990dd153cd0eaa6f1762c059c.0.cs102
*There var another Via header that I removed with remove_hf_re but tags is
stiil there and Route header mover
up*;rport=1578;received=23.101.134.111Route:
sip:54.241.2.206;lr=on;ftag=as38f20f7b;did=ce8.10a
Max-Forwards: 70
From: New User sip:00987654...@phone.provider.com;tag=as38f20f7b
To: sip:12345678...@phone.provider.com;tag=B4r1B569001DB
Contact: sip:contact@10.0.1.17:50600;alias=23.101.134.111~1578~1
Call-ID: 473fc8a64d7d96aa515780dc6237a6d5@10.0.1.17:50600
CSeq: 103 ACK
User-Agent: Some Device
Content-Length: 0
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Re: [SR-Users] ndb_redis set null reply to EXISTS query

What type of info can I provide for deeper analys of this situation?

2015-02-27 14:34 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello
 On 27/02/15 11:58, Yuriy Gorlichenko wrote:

 at the monitor I see nothing about this request

 It's difficult to say without further information, but for some reason
 your command is not hitting redis, probably because the connection is down.

 Javi


 2015-02-27 13:21 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Now I see that null values recieved after I see this at kamailio log

 redisc_exec(): Redis error: Server closed the connection


 2015-02-27 12:45 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello,
 check out http://redis.io/commands/monitor
 It will help you to determine the exact command that kamailio is
 delivering to the server when you execute the redis_cmd(...) function
 inside the script.
 For a non null reply, yo will need a key in redis with the same value as
 $si.

 Javi

 On 27/02/15 10:04, Yuriy Gorlichenko wrote:

 I will try. I new at redis. Does cli monitor get resul of kamailio
 request at the cli?

 2015-02-27 11:38 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello

 can you check with redis-cli monitor what is the command sent to
 Redis in that case?

 Javi
 On 27/02/15 09:15, Yuriy Gorlichenko wrote:

  Hello I try to get some replies from redis. Time after time redis
 request give me null result. But redis bs not disconnected.

 This happens only with websocket endpoints. My queries is:

 redis_cmd(srv1, EXISTS $si, s);

 So at xLOG i see that $si correctly sended, but result is null. At db I
 keep key IP with value TIMESTAMP

 Intresting that route, that givesme this result idependend of WS or UDP
 endpoint. IT does not know about it anything.


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[SR-Users] ndb_redis set null reply to EXISTS query

Hello I try to get some replies from redis. Time after time redis request
give me null result. But redis bs not disconnected.

This happens only with websocket endpoints. My queries is:

redis_cmd(srv1, EXISTS $si, s);

So at xLOG i see that $si correctly sended, but result is null. At db I
keep key IP with value TIMESTAMP

Intresting that route, that givesme this result idependend of WS or UDP
endpoint. IT does not know about it anything.
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Re: [SR-Users] ndb_redis set null reply to EXISTS query

I will try. I new at redis. Does cli monitor get resul of kamailio request
at the cli?

2015-02-27 11:38 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello

 can you check with redis-cli monitor what is the command sent to Redis
 in that case?

 Javi
 On 27/02/15 09:15, Yuriy Gorlichenko wrote:

 Hello I try to get some replies from redis. Time after time redis request
 give me null result. But redis bs not disconnected.

 This happens only with websocket endpoints. My queries is:

 redis_cmd(srv1, EXISTS $si, s);

 So at xLOG i see that $si correctly sended, but result is null. At db I
 keep key IP with value TIMESTAMP

 Intresting that route, that givesme this result idependend of WS or UDP
 endpoint. IT does not know about it anything.


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Re: [SR-Users] ndb_redis set null reply to EXISTS query

at the monitor I see nothing about this request

2015-02-27 13:21 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Now I see that null values recieved after I see this at kamailio log

 redisc_exec(): Redis error: Server closed the connection


 2015-02-27 12:45 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello,
 check out http://redis.io/commands/monitor
 It will help you to determine the exact command that kamailio is
 delivering to the server when you execute the redis_cmd(...) function
 inside the script.
 For a non null reply, yo will need a key in redis with the same value as
 $si.

 Javi

 On 27/02/15 10:04, Yuriy Gorlichenko wrote:

 I will try. I new at redis. Does cli monitor get resul of kamailio
 request at the cli?

 2015-02-27 11:38 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

  Hello

 can you check with redis-cli monitor what is the command sent to Redis
 in that case?

 Javi
 On 27/02/15 09:15, Yuriy Gorlichenko wrote:

  Hello I try to get some replies from redis. Time after time redis
 request give me null result. But redis bs not disconnected.

 This happens only with websocket endpoints. My queries is:

 redis_cmd(srv1, EXISTS $si, s);

 So at xLOG i see that $si correctly sended, but result is null. At db I
 keep key IP with value TIMESTAMP

 Intresting that route, that givesme this result idependend of WS or UDP
 endpoint. IT does not know about it anything.


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Re: [SR-Users] ndb_redis set null reply to EXISTS query

Now I see that null values recieved after I see this at kamailio log

redisc_exec(): Redis error: Server closed the connection

2015-02-27 12:45 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

Hello,
check out http://redis.io/commands/monitor
It will help you to determine the exact command that kamailio is
delivering to the server when you execute the redis_cmd(...) function
inside the script.
For a non null reply, yo will need a key in redis with the same value as
$si.

Javi

On 27/02/15 10:04, Yuriy Gorlichenko wrote:

I will try. I new at redis. Does cli monitor get resul of kamailio request
at the cli?

2015-02-27 11:38 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:

Hello

can you check with redis-cli monitor what is the command sent to Redis
in that case?

Javi
On 27/02/15 09:15, Yuriy Gorlichenko wrote:

Hello I try to get some replies from redis. Time after time redis
request give me null result. But redis bs not disconnected.

This happens only with websocket endpoints. My queries is:

redis_cmd(srv1, EXISTS $si, s);

So at xLOG i see that $si correctly sended, but result is null. At db I
keep key IP with value TIMESTAMP

Intresting that route, that givesme this result idependend of WS or UDP
endpoint. IT does not know about it anything.

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[SR-Users] usrloc table to redis

2015-02-26 Thread Yuriy Gorlichenko

Hello. We try to use redis for maximum features of kamailio.
We already realise dispatcher (not as module, but I want to do it at the
future), and now we want to relocate usrloc to redis. Does anyone do this
with Redis?
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[SR-Users] Kamailio appends firs inserted branch twice

2015-02-10 Thread Yuriy Gorlichenko

Hello I use this version of kamailio

 kamailio -v
version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 8cdbe7
compiled on 17:54:50 Jan 20 2015 with gcc 4.8.2

I hav an issue with append branches to branch route when I need fork call
to one endpoint woth different destionations.

I use my own algorithm for call to this devices because with
lookup(location) I can not use RTPENGINE for different types of endpoints
(web endoints and standart UDP endpoints)

My alg is here:

{

sql_query(ca, select contact from location where
username='$tU', ra);
xlog(L_INFO,rows: $dbr(ra=rows) cols: $dbr(ra=cols)\n);
if($dbr(ra=rows)0){
$var(i)=0;
 while($var(i)$dbr(ra=rows)){

xlog(L_INFO,SQL query return contact
{$dbr(ra=[$var(i),0])} for {$tU} at step {$var(i)}\n);

if ($dbr(ra=[$var(i),0])=~transport=ws){
xlog(L_INFO, This is a Websocket call to endpoint);
sql_pvquery(ca, select received from
location where contact='$dbr(ra=[$var(i),0])',$var(recieved));

$du=$var(recieved);
xlog(L_INFO,SQL query return recieved
{$var(recieved)} for {$tU}. Destination is {$du}\n);


append_branch(sip:$tU@$(du{s.select,1,:}));


}

else
{

xlog(L_INFO, This is a classic UDP call to
endpoint);
$var(recieved)='';
sql_pvquery(ca, select received from
location where contact='$dbr(ra=[$var(i),0])',$var(recieved));
xlog(L_INFO, SQL query return RECIEVED
{$var(recieved)});
if ($var(recieved)==0){
xlog(L_INFO, Recieved string is EMPTY);
$du=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});
}
else {
xlog(L_INFO, Recieved string is
{$var(recieved)});
$du=$var(recieved);
}

$var(UDP_contact)=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});

append_branch(sip:$tU@$(du{s.select,1,:}));

xlog(L_INFO,Classic Destination URI is
{$dbr(ra=[$var(i),0])} for {$tU}}. Destination is {$du}\n);
}
$var(i) = $var(i) + 1;

}
}
else{
exit;
}
t_on_branch(1);
return;

}
}

}


branch_route[1]{

if($du=~transport=ws){
xlog(L_INFO,Websocket Branch is {$du} for {$tU}\n);

rtpengine_manage(internal extenal force trust-address
replace-origin replace-session-connection ICE=force RTP/SAVPF);
t_on_reply(REPLY_FROM_WS);

}
else{
xlog(L_INFO,UDP Branch is {$du)} for {$tU}\n);

rtpengine_manage(replace-origin
replace-session-connection ICE=remove RTP/AVP);
t_on_reply(MANAGE_CLASSIC_REPLY);
}
}

So as you see I choose array of devices and then set its to branches.

At the kamailio console I see output

This is output of alg that above.

I have 3 devices: 2 websocket devices and 1 udp
But when you look at dump where rtpengine make changes with packets you can
see that it runs for 4 devices and 1 and 4 devices is the same.
You can see start of every branch from words of log Websocket Branch is
or UDP Branch is

So this bug gives a mistake because some of hardphones can not handlie
double INVITE and kamilio response for this devices cancel with 200 cause
because hardphone set 482 reply to kamialio.
(you can see it reply at the log)

Feb  8 23:04:43 Kamailio2 kamailio[59406]: INFO: script: rows: 3 cols: 1
Feb  8 23:04:43 Kamailio2 kamailio[59406]: INFO: script: SQL query
return contact {sip:5ue13vsh@2t4iielj109h.invalid;transport=ws} for
{123455678} at step {0}
Feb  8 23:04:43 Kamailio2 kamailio[59406]: INFO: script: This is a
Websocket call to endpoint
Feb  8 23:04:43 Kamailio2 kamailio[59406]: INFO: script: SQL query
return recieved {sip:85.6.14.8:4328;transport=WS} for {123455678}.
Destination is {sip:85.6.14.8:4328;transport=WS}
Feb  8 23:04:43 Kamailio2 kamailio[59406]: INFO: script: SQL query
return contact {sip:a7uo9vsq@7c0oo5kvc71e.invalid;transport=ws} for
{123455678} at step {1}
Feb  8 23:04:43 Kamailio2 kamailio[59406]: INFO: script: This is a
Websocket call

Re: [SR-Users] Kamailio appends firs inserted branch twice

2015-02-10 Thread Yuriy Gorlichenko

I need to handle every route with rtpengine depending  websocket it or not
(branch_route[1] at my first message). Lookup keep handling close with
branch route. So I can not understand how send to branch_route with lookup
function.

2015-02-10 14:17 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 r-uri itself is a branch. Practically, an append_branch() in request_route
 is adding a new destination in addition to r-uri address.

 Maybe you can explain why lookup(location) is not working for you, as it
 is updates the branch for r-uri as well as adds the other contacts as extra
 branches.

 You may want to set a branch flag when processing the REGISTER to know
 that it is a websocket. Then, you can test the same branch flag in
 branch_route to discover if the destination is over websocket or not.

 Cheers,
 Daniel


 On 10/02/15 12:01, Yuriy Gorlichenko wrote:

  Hello I use this version of kamailio

  kamailio -v
 version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7
 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, 
 USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, 
 DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, 
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 8cdbe7
 compiled on 17:54:50 Jan 20 2015 with gcc 4.8.2

 I hav an issue with append branches to branch route when I need fork call
 to one endpoint woth different destionations.

 I use my own algorithm for call to this devices because with
 lookup(location) I can not use RTPENGINE for different types of endpoints
 (web endoints and standart UDP endpoints)

 My alg is here:

 {

 sql_query(ca, select contact from location where 
 username='$tU', ra);
 xlog(L_INFO,rows: $dbr(ra=rows) cols: $dbr(ra=cols)\n);
 if($dbr(ra=rows)0){
 $var(i)=0;
  while($var(i)$dbr(ra=rows)){

 xlog(L_INFO,SQL query return contact 
 {$dbr(ra=[$var(i),0])} for {$tU} at step {$var(i)}\n);

 if ($dbr(ra=[$var(i),0])=~transport=ws){
 xlog(L_INFO, This is a Websocket call to 
 endpoint);
 sql_pvquery(ca, select received from location 
 where contact='$dbr(ra=[$var(i),0])',$var(recieved));

 $du=$var(recieved);
 xlog(L_INFO,SQL query return recieved 
 {$var(recieved)} for {$tU}. Destination is {$du}\n);


 append_branch(sip:$tU@$(du{s.select,1,:}));


 }

 else
 {

 xlog(L_INFO, This is a classic UDP call to 
 endpoint);
 $var(recieved)='';
 sql_pvquery(ca, select received from location 
 where contact='$dbr(ra=[$var(i),0])',$var(recieved));
 xlog(L_INFO, SQL query return RECIEVED 
 {$var(recieved)});
 if ($var(recieved)==0){
 xlog(L_INFO, Recieved string is EMPTY);
 $du=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});
 }
 else {
 xlog(L_INFO, Recieved string is 
 {$var(recieved)});
 $du=$var(recieved);
 }
 
 $var(UDP_contact)=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});

 append_branch(sip:$tU@$(du{s.select,1,:}));

 xlog(L_INFO,Classic Destination URI is 
 {$dbr(ra=[$var(i),0])} for {$tU}}. Destination is {$du}\n);
 }
 $var(i) = $var(i) + 1;

 }
 }
 else{
 exit;
 }
 t_on_branch(1);
 return;

 }
 }

 }


 branch_route[1]{

 if($du=~transport=ws){
 xlog(L_INFO,Websocket Branch is {$du} for {$tU}\n);

 rtpengine_manage(internal extenal force trust-address 
 replace-origin replace-session-connection ICE=force RTP/SAVPF);
 t_on_reply(REPLY_FROM_WS);

 }
 else{
 xlog(L_INFO,UDP Branch is {$du)} for {$tU}\n);

 rtpengine_manage(replace-origin replace-session-connection 
 ICE=remove RTP/AVP);
 t_on_reply(MANAGE_CLASSIC_REPLY);
 }
 }

 So as you see I choose array of devices and then set its to branches.

 At the kamailio console I see output

 This is output of alg that above.

 I have 3 devices: 2 websocket devices and 1 udp
 But when you look at dump where rtpengine make changes with packets you
 can see that it runs for 4 devices and 1 and 4 devices is the same.
 You can see start of every branch from

Re: [SR-Users] Kamailio load balansing 2 servers issues

2015-02-06 Thread Yuriy Gorlichenko

We solved this issue with another way because enpoints can registar at only
one kam (at location table we see socket field). So we need not register at
both servers one endpoint- wee ned call all servers for calling endpoints
with same username from both servers. This philosophy right for load
balansers.

2015-02-03 22:30 GMT+03:00 Juha Heinanen j...@tutpro.com:

 Victor V. Kustov writes:

  We use failover such way:
 
  2 K. servers with 1 shared IP (CARP) and all actions replicate via DMQ.
  This way usable for load balancing too.

 by all actions, to do you mean also transaction state so that reply to a
 request can come to different server than the one that set the request?

 -- juha

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[SR-Users] NDB_REDIS password to REDIS remote DB

2015-01-29 Thread Yuriy Gorlichenko

Hello. I try to use NDB_REDIS with remote REDIS DB and can not to connect
because remote DB use password, but Kamailio module have no any variable or
attr of modparam that implenets password for DB.

How I can connect to my REDIS?
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Re: [SR-Users] Dispatcher weight dont work

2015-01-29 Thread Yuriy Gorlichenko

Will, thanks for your answer. I already implemented my own lgorithm without
using dispatcher module. It works better for me. Now I try to implement it
with REDIS for faster results.

2015-01-29 1:35 GMT+03:00 Will Ferrer will.fer...@switchsoft.com:

 Hi Yuri

 I shared your issue with my business partner who works on the configs with
 me. I had remembered he had some similar issue he pinged me about some time
 back.

 His response to your issue follows:

 Looks like he is missing flags 8 and the trailing ; after weight in attrs.
 CREATE TABLE `dispatcher` (
 `id`   `setid` `destination` `flags` `priority` `attrs`  `description`
 1 1 sip:10.0.0.1 8 1 weight=50;
 2 1 sip:10.0.0.2 8 2 weight=50;

 To see if it changes is being used by Kamailio run:
 kamcmd dispatcher.list

 BODY: weight=50  -- this is the raw attrs
 WEIGHT: 50 -- this show that the attribute has been processed.

 {
 NRSETS: 1
 RECORDS: {
 SET: {
 ID: 1
 TARGETS: {
 DEST: {
 URI: sip:10.0.0.1
 FLAGS: AP
 PRIORITY: 1
 ATTRS: {
 BODY: weight=50
 DUID:
 MAXLOAD: 0
 WEIGHT: 50
 }
 }
 DEST: {
 URI: sip:10.0.0.2
 FLAGS: AP
 PRIORITY: 1
 ATTRS: {
 BODY: weight=50
 DUID:
 MAXLOAD: 0
 WEIGHT: 50
 }
 }
 }
 }
 }
 }

 I hope that helps.

 All the best.

 Will

 On Tue, Jan 27, 2015 at 3:12 AM, Yuriy Gorlichenko ovoshl...@gmail.com
 wrote:

 Hello I use dipatcher  algorithm 8 that works with weight. I added  2
 Asterisks and try to call its with my kam.We use 4.3 version.

 Tthis config select needed dst from database with my scenario.

 if(!ds_select_dst($var(setid), 8))

 $var(setid)- is variable for setting setid that i get from database with
 my own scenario. IT does not matter.

 When running asterisk with weight 90 - all calls goes through it. When I
 starting asterisk with weight 10 -calls going through asterisk 90. When I
 shut down asterisk with weight 90 -calls goes through asterisk 10? but when
 i start asterisk weight 90 all calls goes through sterisk 10 until I shut
 down it.

 root@Kamailio:~# kamailio -v
 version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7
 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
 DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
 F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
 USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
 id: 8cdbe7
 compiled on 01:17:56 Jan 21 2015 with gcc 4.8.2


 id setid   destination  flags priority attrs
 1   2   sip:34.25.123.45:506000  0 weight=10



 2   2   sip:10.0.1.6:506000  0weight=90

 modparam(dispatcher, db_url,DBURL)
 modparam(dispatcher, table_name, dispatcher)
 modparam(dispatcher, setid_col, setid)
 modparam(dispatcher, destination_col, destination)
 modparam(dispatcher, force_dst, 1)
 modparam(dispatcher, flags, 3)
 modparam(dispatcher, dst_avp, $avp(i:271))
 modparam(dispatcher, grp_avp, $avp(i:272))
 modparam(dispatcher, cnt_avp, $avp(i:273))
 modparam(dispatcher, ds_ping_from, sip:proxy@10.0.1.1)
 modparam(dispatcher, ds_ping_interval,15)
 modparam(dispatcher, ds_probing_mode, 1)
 modparam(dispatcher, ds_ping_reply_codes,
 class=2;code=403;code=404;code=484;class=3)
 modparam(tm, reparse_on_dns_failover, 0)


 Thanks

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[SR-Users] Dispatcher weight dont work

2015-01-27 Thread Yuriy Gorlichenko

Hello I use dipatcher  algorithm 8 that works with weight. I added  2
Asterisks and try to call its with my kam.We use 4.3 version.

Tthis config select needed dst from database with my scenario.

if(!ds_select_dst($var(setid), 8))

$var(setid)- is variable for setting setid that i get from database with my
own scenario. IT does not matter.

When running asterisk with weight 90 - all calls goes through it. When I
starting asterisk with weight 10 -calls going through asterisk 90. When I
shut down asterisk with weight 90 -calls goes through asterisk 10? but when
i start asterisk weight 90 all calls goes through sterisk 10 until I shut
down it.

root@Kamailio:~# kamailio -v
version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 8cdbe7
compiled on 01:17:56 Jan 21 2015 with gcc 4.8.2


id setid   destination  flags priority attrs
1   2   sip:34.25.123.45:506000  0 weight=10



2   2   sip:10.0.1.6:506000  0weight=90

modparam(dispatcher, db_url,DBURL)
modparam(dispatcher, table_name, dispatcher)
modparam(dispatcher, setid_col, setid)
modparam(dispatcher, destination_col, destination)
modparam(dispatcher, force_dst, 1)
modparam(dispatcher, flags, 3)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, grp_avp, $avp(i:272))
modparam(dispatcher, cnt_avp, $avp(i:273))
modparam(dispatcher, ds_ping_from, sip:proxy@10.0.1.1)
modparam(dispatcher, ds_ping_interval,15)
modparam(dispatcher, ds_probing_mode, 1)
modparam(dispatcher, ds_ping_reply_codes,
class=2;code=403;code=404;code=484;class=3)
modparam(tm, reparse_on_dns_failover, 0)


Thanks
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[SR-Users] lookup_branches and rtpengine_manage

2015-01-26 Thread Yuriy Gorlichenko

Hello. I need parallel forking calls with the  same username. (Call to all
contacts with name for example User123), my endpoints may be  WebSocket
based and standart UDP endpoints. And I use rtpengine_manage for nmanaging
calls wor webphones and standart softh/hard phones.

I get all contacts manually and than at the branch route set
rtpengine_manage settings for every call.

It works fine but it works for one kamailio server.

When I use 2 kamailio servers as load balansers Server that handle call get
all endpoints from location but call to only one, that registred only at
ths server

for example I call user123
I have 3 contacts
user123@1.2.3.4 - was registered at kamailio 1
user123@3.2.1.4 - was registered at kamailio 2
user123@4.3.2.1 - was registered at kamailio 1

So if call goes through kamailio 1 it call only to user123@1.2.3.4 and
user123@4.3.2.1

I use this settings for usrloc  at 2 kamailios to share all table between 2
servers

modparam(usrloc, db_url, DBURL)
modparam(usrloc, db_mode, 3)
modparam(usrloc, user_column, username)
modparam(usrloc, contact_column, contact)
modparam(usrloc, expires_column, expires)
modparam(usrloc, q_column, q)
modparam(usrloc, callid_column, callid)
modparam(usrloc, cseq_column, cseq)
modparam(usrloc, methods_column, methods)
modparam(usrloc, cflags_column, cflags)
modparam(usrloc, user_agent_column, user_agent)
modparam(usrloc, received_column, received)
modparam(usrloc, socket_column, socket)
modparam(usrloc, path_column, path)
modparam(usrloc, ruid_column, ruid)
modparam(usrloc, instance_column, instance)
modparam(usrloc, use_domain, 1)

and this code for calling them

[GET_CONTACTS]
{
sql_query(ca, select contact from location where username='$tU', ra);
xlog(rows: $dbr(ra=rows) cols: $dbr(ra=cols)\n);
if($dbr(ra=rows)0){
 $var(i)=0;
  while($var(i)$dbr(ra=rows)){
 xlog(L_INFO,SQL query return contact {$dbr(ra=[$var(i),0])} for {$tU}
at step {$var(i)}\n);
  if ($dbr(ra=[$var(i),0])=~transport=ws){
 xlog(L_INFO, This is a Websocket call to endpoint);
 sql_pvquery(ca, select received from location where
contact='$dbr(ra=[$var(i),0])',$var(recieved));
  $du=$var(recieved);
 xlog(L_INFO,SQL query return recieved {$var(recieved)} for {$tU}.
Destination is {$du}\n);
 append_branch(sip:$tU@$(du{s.select,1,:}));
   }
 else
 {
 xlog(L_INFO, This is a classic UDP call to endpoint);
 $var(recieved)='';
 sql_pvquery(ca, select received from location where
contact='$dbr(ra=[$var(i),0])',$var(recieved));
 xlog(L_INFO, SQL query return RECIEVED {$var(recieved)});
 if ($var(recieved)==0){
 xlog(L_INFO, Recieved string is EMPTY);
 $du=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});
 }
 else {
 xlog(L_INFO, Recieved string is {$var(recieved)});
 $du=$var(recieved);
 }
 $var(UDP_contact)=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});
 append_branch(sip:$tU@$(du{s.select,1,:}));
  xlog(L_INFO,Classic Destination URI is {$dbr(ra=[$var(i),0])} for
{$tU}}. Destination is {$du}\n);
 }
 $var(i) = $var(i) + 1;
  }
 }
t_on_branch(1);
return;
 }
}

}


branch_route[1]{

if($du=~transport=ws){
xlog(L_INFO,Websocket Branch is {$du} for {$tU}\n);
rtpengine_manage(internal extenal force trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF);
t_on_reply(REPLY_FROM_WS);
 }
else{
xlog(L_INFO,UDP Branch is {$du)} for {$tU}\n);
rtpengine_manage(replace-origin replace-session-connection ICE=remove
RTP/AVP);
t_on_reply(MANAGE_CLASSIC_REPLY);
}
}


When it try to branch endpoint without registration at server that handle
call I get errors that tm module can not build Via header

**via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be
 found*
 *ERROR: core [msg_translator.c:1725]: build_req_buf_from_sip_req():
 could not create Via header*
 *ERROR: core [forward.c:607]: forward_request(): ERROR:
 forward_request: building failed*
*

UDP calls get errors something like above (sorry than can not share
error code, This situation not often).

So I think I have this trouble because I use manually handling call
and tried to substitute to lookup_branches function. but I have no
Idea how to set rtpengine_manage paraments for each endpoint depending
this is websocket or standart call.

IF there is write problem for callings thhrough 2 kamailios as
loadbalansers pleas let me know about how to set rtpengine_manage
parametrs wor endpoints for every fork.  If not- can you tell me how I
can call to all endpoints endepending of registration server (kamailio
1 or 2).

Thanks.
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[SR-Users] lookup_branches and rtpengine_manage

2015-01-25 Thread Yuriy Gorlichenko

Hello. I need parallel forking calls with the  same username. (Call to all
contacts with name for example User123), my endpoints may be  WebSocket
based and standart UDP endpoints. And I use rtpengine_manage for nmanaging
calls wor webphones and standart softh/hard phones.

I get all contacts manually and than at the branch route set
rtpengine_manage settings for every call.

It works fine but it works for one kamailio server.

When I use 2 kamailio servers as load balansers Server that handle call get
all endpoints from location but call to only one, that registred only at
ths server

for example I call user123
I have 3 contacts
user123@1.2.3.4 - was registered at kamailio 1
user123@3.2.1.4 - was registered at kamailio 2
user123@4.3.2.1 - was registered at kamailio 1

So if call goes through kamailio 1 it call only to user123@1.2.3.4 and
user123@4.3.2.1

I use this settings for usrloc  at 2 kamailios to share all table between 2
servers

modparam(usrloc, db_url, DBURL)
modparam(usrloc, db_mode, 3)
modparam(usrloc, user_column, username)
modparam(usrloc, contact_column, contact)
modparam(usrloc, expires_column, expires)
modparam(usrloc, q_column, q)
modparam(usrloc, callid_column, callid)
modparam(usrloc, cseq_column, cseq)
modparam(usrloc, methods_column, methods)
modparam(usrloc, cflags_column, cflags)
modparam(usrloc, user_agent_column, user_agent)
modparam(usrloc, received_column, received)
modparam(usrloc, socket_column, socket)
modparam(usrloc, path_column, path)
modparam(usrloc, ruid_column, ruid)
modparam(usrloc, instance_column, instance)
modparam(usrloc, use_domain, 1)

and this code for calling them

[GET_CONTACTS]
{
sql_query(ca, select contact from location where username='$tU', ra);
xlog(rows: $dbr(ra=rows) cols: $dbr(ra=cols)\n);
if($dbr(ra=rows)0){
 $var(i)=0;
 while($var(i)$dbr(ra=rows)){
 xlog(L_INFO,SQL query return contact {$dbr(ra=[$var(i),0])} for {$tU}
at step {$var(i)}\n);
  if ($dbr(ra=[$var(i),0])=~transport=ws){
 xlog(L_INFO, This is a Websocket call to endpoint);
 sql_pvquery(ca, select received from location where
contact='$dbr(ra=[$var(i),0])',$var(recieved));
  $du=$var(recieved);
 xlog(L_INFO,SQL query return recieved {$var(recieved)} for {$tU}.
Destination is {$du}\n);
 append_branch(sip:$tU@$(du{s.select,1,:}));
   }
 else
 {
 xlog(L_INFO, This is a classic UDP call to endpoint);
 $var(recieved)='';
 sql_pvquery(ca, select received from location where
contact='$dbr(ra=[$var(i),0])',$var(recieved));
 xlog(L_INFO, SQL query return RECIEVED {$var(recieved)});
 if ($var(recieved)==0){
 xlog(L_INFO, Recieved string is EMPTY);
 $du=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});
 }
 else {
 xlog(L_INFO, Recieved string is {$var(recieved)});
 $du=$var(recieved);
 }
 $var(UDP_contact)=sip:+$(dbr(ra=[$var(i),0]){s.select,1,@});
 append_branch(sip:$tU@$(du{s.select,1,:}));
  xlog(L_INFO,Classic Destination URI is {$dbr(ra=[$var(i),0])} for
{$tU}}. Destination is {$du}\n);
 }
 $var(i) = $var(i) + 1;
  }
 }
t_on_branch(1);
return;
 }
}

}


branch_route[1]{

if($du=~transport=ws){
xlog(L_INFO,Websocket Branch is {$du} for {$tU}\n);
rtpengine_manage(internal extenal force trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF);
t_on_reply(REPLY_FROM_WS);
 }
else{
xlog(L_INFO,UDP Branch is {$du)} for {$tU}\n);
rtpengine_manage(replace-origin replace-session-connection ICE=remove
RTP/AVP);
t_on_reply(MANAGE_CLASSIC_REPLY);
}
}


When it try to branch endpoint without registration at server that handle
call I get errors that tm module can not build Via header

**via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be
 found*
 *ERROR: core [msg_translator.c:1725]: build_req_buf_from_sip_req():
 could not create Via header*
 *ERROR: core [forward.c:607]: forward_request(): ERROR:
 forward_request: building failed*
*

UDP calls get errors something like above (sorry than can not share
error code, This situation not often).

So I think I have this trouble because I use manually handling call
and tried to substitute to lookup_branches function. but I have no
Idea how to set rtpengine_manage paraments for each endpoint depending
this is websocket or standart call.

IF there is write problem for callings thhrough 2 kamailios as
loadbalansers pleas let me know about how to set rtpengine_manage
parametrs wor endpoints for every fork.  If not- can you tell me how I
can call to all endpoints endepending of registration server (kamailio
1 or 2).

Thanks.
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[SR-Users] Kamailio load balansing 2 servers issues

2015-01-23 Thread Yuriy Gorlichenko

Hello. We use 2 kamailio server 4.3 master brancher for load balansing
cluster. We have some problems with this deplooiment

fist of all we have REgstering issue: Witth one server all works fine (we
have some endpoints with same creditians) all endpoints reinging well. But
at 2 servers we hawe problem with this- call ringing only at one endpoint.
We use Database for USRLOC module

modparam(usrloc, db_url, DBURL)
modparam(usrloc, db_mode, 3)
modparam(usrloc, use_domain, MULTIDOMAIN)

As i think it mean al connections to servers ives at database. We have one
database for 2 servers and this is means it only one table LOCATION for its.

The second problen id that when loadbalanser sends bye to kamailio from the
outside- bye is not recieves to endpoint. It say that no transaction with
this CallID. I think it is because transaction goues through another
kamailio and kamiaio that recieves BYE does not know about it transaction.

So my first qestion- Does anybody have some information about loafd
balansiong between kamailios?

And second question - how to resolve issue with LOCATION table?

Thanks
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Re: [SR-Users] Dispatcher 8 algorithm based on priority strange works

2015-01-14 Thread Yuriy Gorlichenko

About servers - I see at asterisk successfull option requests from kamailio
and 200 reply o this requests. This meants as I understand, that servers
know about each other.

Now about behaivor:
Todays morning I try to down server with highest priority. Kam successfully
takes server with low priority an calls works through it. After that I try
to up server with highest rpiority. Then I take a call ans it goes through
server with lowes priority. I goes down server with lowest priority. Take
call. Call goes through highest priority server.
After it all I up server with low priority and then calls goes through it
(!!!) not as yesterday when call staing at highest priority server. It is
not normal as I think. and strange...

I can give you ssh to our servers to see this. if it needed. or some more
info but tomorrow morning. At this time servers busy with testion other
functions.


2015-01-14 12:31 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:


 On 13/01/15 16:52, Yuriy Gorlichenko wrote:

 Daniel.  I added 8 algorithm to our server and it works with 2 asterisk
 now but it works strange because:
 While works server with priority 1 - all ok. When this server goes down
 dispatcher choose next server with lowes priority. But when server with
 highest priority waking up dispatcher use server with lowes priority until
 this server not goes down.


 Perhaps it is a mistake in the part 'until this server not goes down'? Can
 you explain again the observed behaviour? Also, wrote in a previous email
 to a thread that seems to expose same issue, check the state of dispatcher
 addresses in memory via mi or rpc commands.

 Cheers,
 Daniel


 here id my config for dispatcher

 modparam(dispatcher, db_url,DBURL)
 modparam(dispatcher, table_name, dispatcher)
 modparam(dispatcher, setid_col, setid)
 modparam(dispatcher, priority_col, priority)
 modparam(dispatcher, destination_col, destination)
 modparam(dispatcher, force_dst, 1)
 modparam(dispatcher, flags, 3)
 modparam(dispatcher, dst_avp, $avp(i:271))
 modparam(dispatcher, grp_avp, $avp(i:272))
 modparam(dispatcher, cnt_avp, $avp(i:273))
 modparam(dispatcher, ds_ping_from, sip:kamailio1@10.0.1.12)
 modparam(dispatcher, ds_ping_interval,15)
 modparam(dispatcher, ds_probing_mode, 1)
 modparam(dispatcher, ds_ping_reply_codes,
 class=2;code=403;code=404;code=484;class=3)
 modparam(tm, reparse_on_dns_failover, 0)


 (!ds_select_dst($var(setid), 8)){

 sl_send_reply(500, Service Unavailable);
 xlog(L_INFO,{$rm} from [$fU@$si:$sp] But NO destinations
 available for $rd \n);
  t_on_failure(DISPATCHER_ROLLOVER);

 }

 if (!t_relay()) {
  sl_reply_error();

 we use 4.3 master branch kamailio


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


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Re: [SR-Users] Dispatcher Failover algorithm

2015-01-13 Thread Yuriy Gorlichenko

Daniel.  I added 8 algorithm to our server and it works with 2 servers now
but it works strange because:
While works server with priority 1 - all ok. When this server goes down
dispatcher choose next server with lowes priority. But when server with
highest priority waking up dispatcher use server with lowes priority until
this server not goes down.

2015-01-12 15:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Daniel. Hello. I see changes at documentation about algorithms at 4.3
 documentation for dispatcher. Now I see than 8 algo use priority. Not I set
 this algorithm to my servers but priority still not worked.

 my dispatcher settings Are


 modparam(dispatcher, db_url,DBURL)
 modparam(dispatcher, table_name, dispatcher)
 modparam(dispatcher, setid_col, setid)
 modparam(dispatcher, priority_col, priority)
 modparam(dispatcher, destination_col, destination)
 modparam(dispatcher, force_dst, 1)
 modparam(dispatcher, flags, 3)
 modparam(dispatcher, dst_avp, $avp(i:271))
 modparam(dispatcher, grp_avp, $avp(i:272))
 modparam(dispatcher, cnt_avp, $avp(i:273))
 modparam(dispatcher, ds_ping_from, sip:kamailio1@10.0.1.12)
 modparam(dispatcher, ds_ping_interval,15)
 modparam(dispatcher, ds_probing_mode, 1)
 modparam(dispatcher, ds_ping_reply_codes,
 class=2;code=403;code=404;code=484;class=3)
 modparam(tm, reparse_on_dns_failover, 0)



 (!ds_select_dst($var(setid), 8)){

 sl_send_reply(500, Service Unavailable);
 xlog(L_INFO,{$rm} from [$fU@$si:$sp] But NO destinations
 available for $rd \n);
 t_on_failure(DISPATCHER_ROLLOVER);

 }

 if (!t_relay()) {
 sl_reply_error();


 2015-01-10 2:31 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Priority bassed? I've read about all algorithms of disatcher and can not
 find that anone use priority...


-

“0” - hash over callid
-

“1” - hash over from URI.
-

“2” - hash over to URI.
-

“3” - hash over request-URI.
-

“4” - round-robin (next destination).
-

“5” - hash over authorization-username (Proxy-Authorization or
normal authorization). If no username is found, round robin is used.
-

“6” - random (using rand()).
-

“7” - hash over the content of PVs string. Note: This works only when
the parameter hash_pvar is set.
-

“8” - use first destination (good for failover).
-

“9” - use weight based load distribution. You have to set the
attribute 'weight' per each address in destination set.
-

“10” - use call load distribution. You have to set the attribute
'duid' (as an unique string id) per each address in destination set. Also,
you must set parameters 'dstid_avp' and 'ds_hash_size'.

The algorithm can be used even with stateless proxy mode, there is no
SIP dialog tracking depending on other modules, just an internal
lightweight call tracking by Call-Id, thus is fast and suitable even for
embedded systems.

The first destination selected by this algorithm is the one that has
the least number of calls associated. The rest of the destination list is
taken in order of the entries in set - anyhow, until a re-route to next
destination happens, the load on each address can change.

This algorithm can be used only for dispatching INVITE requests as it
is the only SIP method creating a SIP call.
-

“X” - if the algorithm is not implemented, the first entry in set is
chosen.


 2015-01-09 20:23 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  You probably look for priority based routing -- see the readme of
 dispatcher module.

 Cheers,
 Daniel


 On 09/01/15 17:52, Yuriy Gorlichenko wrote:

 I as wrote before - we find dispatcher algorithm than can do mechanism
 something like this:
 Try call to fist server with max priority or weight. OIf this server
 unavailible then call second server with less weight and etc.

 Does anyone know what ling of algorithm we can use for this?


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


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[SR-Users] Dispatcher 8 algorithm based on priority strange works

2015-01-13 Thread Yuriy Gorlichenko

Daniel.  I added 8 algorithm to our server and it works with 2 asterisk now
but it works strange because:
While works server with priority 1 - all ok. When this server goes down
dispatcher choose next server with lowes priority. But when server with
highest priority waking up dispatcher use server with lowes priority until
this server not goes down.

here id my config for dispatcher

modparam(dispatcher, db_url,DBURL)
modparam(dispatcher, table_name, dispatcher)
modparam(dispatcher, setid_col, setid)
modparam(dispatcher, priority_col, priority)
modparam(dispatcher, destination_col, destination)
modparam(dispatcher, force_dst, 1)
modparam(dispatcher, flags, 3)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, grp_avp, $avp(i:272))
modparam(dispatcher, cnt_avp, $avp(i:273))
modparam(dispatcher, ds_ping_from, sip:kamailio1@10.0.1.12)
modparam(dispatcher, ds_ping_interval,15)
modparam(dispatcher, ds_probing_mode, 1)
modparam(dispatcher, ds_ping_reply_codes,
class=2;code=403;code=404;code=484;class=3)
modparam(tm, reparse_on_dns_failover, 0)


(!ds_select_dst($var(setid), 8)){

sl_send_reply(500, Service Unavailable);
xlog(L_INFO,{$rm} from [$fU@$si:$sp] But NO destinations
available for $rd \n);
t_on_failure(DISPATCHER_ROLLOVER);

}

if (!t_relay()) {
sl_reply_error();

we use 4.3 master branch kamailio
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Re: [SR-Users] Dispatcher Failover algorithm

2015-01-12 Thread Yuriy Gorlichenko

Daniel. Hello. I see changes at documentation about algorithms at 4.3
documentation for dispatcher. Now I see than 8 algo use priority. Not I set
this algorithm to my servers but priority still not worked.

my dispatcher settings Are


modparam(dispatcher, db_url,DBURL)
modparam(dispatcher, table_name, dispatcher)
modparam(dispatcher, setid_col, setid)
modparam(dispatcher, priority_col, priority)
modparam(dispatcher, destination_col, destination)
modparam(dispatcher, force_dst, 1)
modparam(dispatcher, flags, 3)
modparam(dispatcher, dst_avp, $avp(i:271))
modparam(dispatcher, grp_avp, $avp(i:272))
modparam(dispatcher, cnt_avp, $avp(i:273))
modparam(dispatcher, ds_ping_from, sip:kamailio1@10.0.1.12)
modparam(dispatcher, ds_ping_interval,15)
modparam(dispatcher, ds_probing_mode, 1)
modparam(dispatcher, ds_ping_reply_codes,
class=2;code=403;code=404;code=484;class=3)
modparam(tm, reparse_on_dns_failover, 0)



(!ds_select_dst($var(setid), 8)){

sl_send_reply(500, Service Unavailable);
xlog(L_INFO,{$rm} from [$fU@$si:$sp] But NO destinations
available for $rd \n);
t_on_failure(DISPATCHER_ROLLOVER);

}

if (!t_relay()) {
sl_reply_error();


2015-01-10 2:31 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Priority bassed? I've read about all algorithms of disatcher and can not
 find that anone use priority...


-

“0” - hash over callid
-

“1” - hash over from URI.
-

“2” - hash over to URI.
-

“3” - hash over request-URI.
-

“4” - round-robin (next destination).
-

“5” - hash over authorization-username (Proxy-Authorization or
normal authorization). If no username is found, round robin is used.
-

“6” - random (using rand()).
-

“7” - hash over the content of PVs string. Note: This works only when
the parameter hash_pvar is set.
-

“8” - use first destination (good for failover).
-

“9” - use weight based load distribution. You have to set the
attribute 'weight' per each address in destination set.
-

“10” - use call load distribution. You have to set the attribute
'duid' (as an unique string id) per each address in destination set. Also,
you must set parameters 'dstid_avp' and 'ds_hash_size'.

The algorithm can be used even with stateless proxy mode, there is no
SIP dialog tracking depending on other modules, just an internal
lightweight call tracking by Call-Id, thus is fast and suitable even for
embedded systems.

The first destination selected by this algorithm is the one that has
the least number of calls associated. The rest of the destination list is
taken in order of the entries in set - anyhow, until a re-route to next
destination happens, the load on each address can change.

This algorithm can be used only for dispatching INVITE requests as it
is the only SIP method creating a SIP call.
-

“X” - if the algorithm is not implemented, the first entry in set is
chosen.


 2015-01-09 20:23 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  You probably look for priority based routing -- see the readme of
 dispatcher module.

 Cheers,
 Daniel


 On 09/01/15 17:52, Yuriy Gorlichenko wrote:

 I as wrote before - we find dispatcher algorithm than can do mechanism
 something like this:
 Try call to fist server with max priority or weight. OIf this server
 unavailible then call second server with less weight and etc.

 Does anyone know what ling of algorithm we can use for this?


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
 listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


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


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[SR-Users] event on table location Duplicate entry for key 'ruid_idx'

2015-01-12 Thread Yuriy Gorlichenko

Hello. We use 2 kamailio servers  cluster  and we have porblems with db.
Database failed pecause of error:

Could not execute Write_rows_v1 event on table production.location;
Duplicate entry 'uloc-54aae947-86d-a67' for key 'ruid_idx', Error_code:
1062; handler error HA_ERR_FOUND_DUPP_KEY; the event's master log FIRST,
end_log_pos 380, Internal MariaDB error code: 1062

But a location table no row 'ruid_idx' and no entry uloc-54aae947-86d-a67.
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Re: [SR-Users] Dispatcher Failover algorithm

2015-01-09 Thread Yuriy Gorlichenko

Priority bassed? I've read about all algorithms of disatcher and can not
find that anone use priority...

“0” - hash over callid
-

“1” - hash over from URI.
-

“2” - hash over to URI.
-

“3” - hash over request-URI.
-

“4” - round-robin (next destination).
-

“5” - hash over authorization-username (Proxy-Authorization or normal
authorization). If no username is found, round robin is used.
-

“6” - random (using rand()).
-

“7” - hash over the content of PVs string. Note: This works only when
the parameter hash_pvar is set.
-

“8” - use first destination (good for failover).
-

“9” - use weight based load distribution. You have to set the attribute
'weight' per each address in destination set.
-

“10” - use call load distribution. You have to set the attribute 'duid'
(as an unique string id) per each address in destination set. Also, you
must set parameters 'dstid_avp' and 'ds_hash_size'.

The algorithm can be used even with stateless proxy mode, there is no
SIP dialog tracking depending on other modules, just an internal
lightweight call tracking by Call-Id, thus is fast and suitable even for
embedded systems.

The first destination selected by this algorithm is the one that has the
least number of calls associated. The rest of the destination list is taken
in order of the entries in set - anyhow, until a re-route to next
destination happens, the load on each address can change.

This algorithm can be used only for dispatching INVITE requests as it is
the only SIP method creating a SIP call.
-

“X” - if the algorithm is not implemented, the first entry in set is
chosen.

2015-01-09 20:23 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

You probably look for priority based routing -- see the readme of
dispatcher module.

Cheers,
Daniel

On 09/01/15 17:52, Yuriy Gorlichenko wrote:

I as wrote before - we find dispatcher algorithm than can do mechanism
something like this:
Try call to fist server with max priority or weight. OIf this server
unavailible then call second server with less weight and etc.

Does anyone know what ling of algorithm we can use for this?

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

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[SR-Users] Dispatcher Failover algorithm

2015-01-09 Thread Yuriy Gorlichenko

I as wrote before - we find dispatcher algorithm than can do mechanism
something like this:
Try call to fist server with max priority or weight. OIf this server
unavailible then call second server with less weight and etc.

Does anyone know what ling of algorithm we can use for this?
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Re: [SR-Users] dispatcher weight dont work

2015-01-08 Thread Yuriy Gorlichenko

We use 4.3 version.

if(!ds_select_dst($var(setid), 9))

$var(setid)- is variable for setting setid that i get from database with my
own scenario. IT does not matter. You may change it to something like 1,2,
or anthing else.
So as you see setid 2 have 2 servers. I have an issue with this config

id setid   destination  flags priority attrs
1   2   sip:34.25.123.45:506000  0 weight=10
2   1   sip:10.0.1.6:506000  0weight=503   2   sip:10.0.1.6:506000
0weight=50

2015-01-08 13:36 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 what version are you using? Can you paste here the records you have for
 the destination set (you can replace the ip addresses, I am interested in
 attributes) and the ds_select_dst() or ds_select_domain() lines from your
 config?

 Cheers,
 Daniel


 On 08/01/15 04:07, Yuriy Gorlichenko wrote:

 Hello I use dipatcher  algorithm that works with weight. I added  2
 Asterisks and try to call its with my kam, but this still works like 4
 algorithm. Weight does not work.
 How I must configure dispatchr for working with weight?
 My configuration now is

 modparam(dispatcher, db_url,DBURL)
 modparam(dispatcher, table_name, dispatcher)
 modparam(dispatcher, setid_col, setid)
 modparam(dispatcher, destination_col, destination)
 modparam(dispatcher, force_dst, 1)
 modparam(dispatcher, flags, 3)
 modparam(dispatcher, dst_avp, $avp(i:271))
 modparam(dispatcher, grp_avp, $avp(i:272))
 modparam(dispatcher, cnt_avp, $avp(i:273))
 modparam(dispatcher, ds_ping_from, sip:proxy@10.0.1.1)
 modparam(dispatcher, ds_ping_interval,15)
 modparam(dispatcher, ds_probing_mode, 1)
 modparam(dispatcher, ds_ping_reply_codes,
 class=2;code=403;code=404;code=484;class=3)
 modparam(tm, reparse_on_dns_failover, 0)


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


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[SR-Users] dispatcher html-rpc

2015-01-07 Thread Yuriy Gorlichenko

Hello. How I must use this function for dynamic reload dispatcher without
restarting me server?
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[SR-Users] Dispatcher weight nor working

2014-12-24 Thread Yuriy Gorlichenko

Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db
instanse for dispatcher and at attrs column for our backend servers set
WEIGHT=40 and 60 for first and second server, but packets sended only at
first server ignoring weight
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[SR-Users] ATT and UAC cancel after 200 OK

2014-11-07 Thread Yuriy Gorlichenko

Hello. I use UAC module for trunks and have trouble when call to mobile
endpoint of ATT provider.

When endpoint pickup call ans 200 OK reply come into kamailio, kamailio
send CANCEL. so at the endpoint I see than call is dropped and at kamailio
client I hear voicemail speech form ATT endpoint.

Provider say that his trace is OK...

My sip trace bellow:

IP 10.0.2.4.5068  34.43.4.23.5060: UDP, length 1129
E...@.l.
...6q..INVITE sip:98765432...@phone.myprovider.com SIP/2.0
Record-Route: sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: John Connor sip:12345...@phone.myprovider.com;tag=as288f6857
To: sip:98765432...@phone.myprovider.com
Contact:sip:12345...@sip.myservice.info:5068
Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031

IP 34.43.4.23.5060  10.0.2.4.5068: UDP, length 651
E...*.Z.6...
.}.SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
From: John Connor sip:12345...@phone.myprovider.com;tag=as288f6857
To: sip:98765432...@phone.myprovider.com
;tag=04e2a294d0728b89107e081a19babab4.43a7
Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=phone.myprovider.com,
nonce=VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq, qop=auth
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0


IP 10.0.2.4.5068  34.43.4.23.5060: UDP, length 410
E...@.o.
...6.j%ACK sip:98765432...@phone.myprovider.com SIP/2.0
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0
Max-Forwards: 70
From: John Connor sip:12345...@phone.myprovider.com;tag=as288f6857
To: sip:98765432...@phone.myprovider.com
;tag=04e2a294d0728b89107e081a19babab4.43a7
Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600
CSeq: 102 ACK
Content-Length: 0


IP 10.0.2.4.5068  34.43.4.23.5060: UDP, length 1402
E...@.k.
...6..`INVITE sip:98765432...@phone.myprovider.com SIP/2.0
Record-Route: sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: John Connor sip:12345...@phone.myprovider.com;tag=as288f6857
To: sip:98765432...@phone.myprovider.com
Contact:sip:12345...@sip.myservice.info:5068
Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600
CSeq: 103 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
Proxy-Authorization: Digest username=12345678, realm=phone.myprovider.com,
nonce=VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq, uri=
sip:98765432...@phone.myprovider.com, qop=auth, nc=0001,
cnonce=2107612791, response=51db231dedb4a74396447729a192a8d3,
algorithm=MD5

v=0
o=root 1363870848 1363870848 IN IP4 68.34.22.11
s=Asterisk PBX 12.5.0
c=IN IP4 68.34.22.11
t=0 0
m=audio 30030 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30031

IP 10.0.2.4.5068  34.43.4.23.5060: UDP, length 1402
E...@.k.
...6.._INVITE sip:98765432...@phone.myprovider.com SIP/2.0
Record-Route: sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on
Via: SIP/2.0/UDP sip.myservice.info:5068
;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102
Via: SIP/2.0/UDP 10.0.1.41:50600
;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600
Max-Forwards: 70
From: John Connor sip:12345...@phone.myprovider.com;tag=as288f6857
To: sip:98765432...@phone.myprovider.com
Contact:sip:12345...@sip.myservice.info:5068
Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600
CSeq: 104 INVITE
User-Agent: Asterisk PBX 12.5.0
Date: Mon, 03 Nov 2014 20:21:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type:

[SR-Users] UAC error at reply : t_should_relay_response:

2014-11-05 Thread Yuriy Gorlichenko

ERROR: t_should_relay_response: status rewrite by UAS: stored: 491,
received: 200


Hello. I use UAC module for trunks form 4.2 master branch. I have an
error when endpoit at the trunk picked up. OK reply never  replied to
caller because I see next error:


ERROR: t_should_relay_response: status rewrite by UAS: stored: 491,
received: 200

So as I see before reply 200 sended to kamailio (from callee) I see packet
with 491 SIP code. It is strange because at the another server (same
configuration) OK successfully sended to caller and session is Ok...
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Re: [SR-Users] Help with dialog $dlg_var(cseq_diff)

2014-11-03 Thread Yuriy Gorlichenko

Great! I will waiting for answer. If it needed I may make some tests. We
building new system and want to use this technology insread of classic
gateway. We will happy to cooperate with you for findinf issues and solve
it as faster as we may. Thanks!

2014-11-03 20:03 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 $dlg_var(cseq_diff) is incremented after sending the invite out from
 failure route, being done when forwarding callback in dialog detects that
 the cseq value has to be incremented.

 I am going to test and see if there is an issue -- uac_auth() should set
 some internal flag to tell dialog to increment cseq.

 Cheers,
 Daniel

  On 01/11/14 16:29, Yuriy Gorlichenko wrote:

 Hello. I need to increment CSeq value for INVITE with Auth params when use
 UAC_AUTH for outgoing calls to provider.

 Kamailio 4.2 may increment this using dialog module


 http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html

 Now I experements with this and var $dlg_var(cseq_diff) and at transaction
 way int have NULL value. I can not understand why

 My config is:

  # - dialog params
 modparam(dialog, db_url,DBURL)
 modparam(dialog, db_mode, 1)
 modparam(dialog,table_name,dialog)
 modparam(dialog, dlg_flag, 4)
 modparam(dialog, initial_cbs_inscript, 1)
 modparam(dialog, profiles_with_value, caller)
 modparam(dialog, default_timeout, 60)
 modparam(dialog, track_cseq_updates, 1)


 route config is

 at request route

 if(is_method(INVITE)  !has_totag()){
  $dlg_ctx(timeout_route) = DIALOG_END;
  $dlg_ctx(timeout_bye) = 1;
  dlg_manage();
  xlog(L_INFO,Dialog manage is {$ct}\n);
  } } t_relay();

  next I handle failure reply because 407 reply is recieved 
 failure_route[MANAGE_FAILURE]
 { route(NATMANAGE); if (t_check_status(401|407)){ xlog(L_INFO, Reply
 from provider on failure: $tU); xlog(L_ERR,401/407 - Unauthorized. ($ci
 .) ($rm) from ($fu) (IP:$si:$sp) to ($Ri:$Rp). Must be authorized with
 digest Auth.); avp_print(); xlog(L_INFO, CSeq diff:
 $dlg_var(cseq_diff)); uac_auth(); xlog(L_INFO, UAC_AUTH(): $tU);
 append_branch(); t_relay(); } if (t_is_canceled()) { exit; } } As you may
 see i logging $dlg_var(cseq_diff) value and now it NULL. So I can not
 understand why? What wrong I do? Thanks for advice.


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 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda
 Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com


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[SR-Users] Help with dialog $dlg_var(cseq_diff)

2014-11-01 Thread Yuriy Gorlichenko

Hello. I need to increment CSeq value for INVITE with Auth params when use
UAC_AUTH for outgoing calls to provider.

Kamailio 4.2 may increment this using dialog module

http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html

Now I experements with this and var $dlg_var(cseq_diff) and at transaction
way int have NULL value. I can not understand why

My config is:

 # - dialog params
modparam(dialog, db_url,DBURL)
modparam(dialog, db_mode, 1)
modparam(dialog,table_name,dialog)
modparam(dialog, dlg_flag, 4)
modparam(dialog, initial_cbs_inscript, 1)
modparam(dialog, profiles_with_value, caller)
modparam(dialog, default_timeout, 60)
modparam(dialog, track_cseq_updates, 1)


route config is

at request route

if(is_method(INVITE)  !has_totag()){
$dlg_ctx(timeout_route) = DIALOG_END;
$dlg_ctx(timeout_bye) = 1;
dlg_manage();
xlog(L_INFO,Dialog manage is {$ct}\n);
} } t_relay();

next I handle failure reply because 407 reply is recieved
failure_route[MANAGE_FAILURE]
{ route(NATMANAGE); if (t_check_status(401|407)){ xlog(L_INFO, Reply
from provider on failure: $tU); xlog(L_ERR,401/407 - Unauthorized. ($ci
.) ($rm) from ($fu) (IP:$si:$sp) to ($Ri:$Rp). Must be authorized with
digest Auth.); avp_print(); xlog(L_INFO, CSeq diff:
$dlg_var(cseq_diff)); uac_auth(); xlog(L_INFO, UAC_AUTH(): $tU);
append_branch(); t_relay(); } if (t_is_canceled()) { exit; } } As you may
see i logging $dlg_var(cseq_diff) value and now it NULL. So I can not
understand why? What wrong I do? Thanks for advice.
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[SR-Users] cSeq increasing

Does it possible increase cSeq manually (for example remove  and then
append headers?) for UAC module when send INVITE messages with Auth, or
kamailio have pseudovar for this header?
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Re: [SR-Users] cSeq increasing

As I understand UAC module can not be used at production as module
foroutgoing calls from kamailio to provider with this limitations?

2014-10-30 18:24 GMT+04:00 Pavel Eremin eremina@gmail.com:

 No way. Use sems or b2b.
 30.10.2014 19:59 пользователь Yuriy Gorlichenko ovoshl...@gmail.com
 написал:

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

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Re: [SR-Users] cSeq increasing

Thanks for answer. Now will insttall it for tests.

2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:

  This feature (increasing/decreasing cseq for calls authenticated to the
 next hop by kamailio) is available with 4.2.0, by using dialog and uac
 modules.

 See more details at:
   -
 http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html

 Let me know if works ok for you, as I did not test it yet extensively.

 Cheers,
 Daniel


 On 30/10/14 16:11, Yuriy Gorlichenko wrote:

 As I understand UAC module can not be used at production as module
 foroutgoing calls from kamailio to provider with this limitations?

 2014-10-30 18:24 GMT+04:00 Pavel Eremin eremina@gmail.com:

 No way. Use sems or b2b.
 30.10.2014 19:59 пользователь Yuriy Gorlichenko ovoshl...@gmail.com
 написал:

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

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


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Re: [SR-Users] cSeq increasing

,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 314
Authorization: Digest username=gw2, realm=sip.myprovider.com,
nonce=7d150eae, uri=sip:89176270...@sip.myprovider.com,
response=68af82b65cbbcd29a27873c7288a246f, algorithm=MD5

v=0
o=root 1822659339 1822659339 IN IP4 2.10.4.20
s=Asterisk PBX 12.6.1
c=IN IP4 2.10.4.20
t=0 0
m=audio 30162 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30163



IP 21.47.2.3.5060  10.0.1.12.5068: UDP, length 671
e@..3.ca
...SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP sip.myservice.com:5068
;branch=z9hG4bK02b5.9eca1752d440937103c7e9bfc226bc94.1;received=2.10.4.20;rport=5068
Via: SIP/2.0/UDP 17.6.43.24:50600
;received=17.6.43.24;branch=z9hG4bK4203f70a;rport=50600
From: sip:g...@sip.myprovider.com;tag=as5255aaa8
To: sip:89176270...@sip.myprovider.com;tag=as2ce5c2f5
Call-ID: 1b74d0a5402e76fb249fe8dc427ce99c@17.6.43.24:50600
CSeq: 102 INVITE
Server: FastTel SoftSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=sip.myprovider.com,
nonce=5f11cf69
Content-Length: 0

2014-10-30 20:26 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com:

 Thanks for answer. Now will insttall it for tests.

 2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:

  This feature (increasing/decreasing cseq for calls authenticated to the
 next hop by kamailio) is available with 4.2.0, by using dialog and uac
 modules.

 See more details at:
   -
 http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html

 Let me know if works ok for you, as I did not test it yet extensively.

 Cheers,
 Daniel


 On 30/10/14 16:11, Yuriy Gorlichenko wrote:

 As I understand UAC module can not be used at production as module
 foroutgoing calls from kamailio to provider with this limitations?

 2014-10-30 18:24 GMT+04:00 Pavel Eremin eremina@gmail.com:

 No way. Use sems or b2b.
 30.10.2014 19:59 пользователь Yuriy Gorlichenko ovoshl...@gmail.com
 написал:

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

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 --
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Re: [SR-Users] UAC INVITE to Provider

Thanks for answer. Contact is Ok. It is just literal mistake at dump. This
error happens because CSeq not incremented by kamailio. Already talking
about this with Daniel at the another List.

2014-10-31 1:37 GMT+04:00 Gonzalo Gasca gascagonz...@gmail.com:

 What about the Contact header,
 Contact:sip:vebinar-...@sip.myservice.com:5068
 Can you verify is a valid one.




 On Wed, Oct 29, 2014 at 3:56 PM, Yuriy Gorlichenko ovoshl...@gmail.com
 wrote:

 Hello. I use kamailio for calling to porvider. My providr seccefuully
 registered from UAC module, but when I try to call through it? it back 401
 Unauthorised. I send second try with Digest Auth header at INVITE and it
 receive me 401 too...

 I register this provider from asterisk and call succesfully Ok. So i get
 dump from asterisk This is successfull INVITE:

 INVITE sip:89126975...@sip.provider.com SIP/2.0
 Via: SIP/2.0/UDP 17.4.28.7:50600;branch=z9hG4bK5f118681;rport
 Max-Forwards: 70
 From: sip:gw2@17.4.28.7:50600;tag=as33192a38
 To: sip:89126975...@sip.provider.com
 Contact: sip:gw2@17.4.28.7:50600
 Call-ID: 021088c360a8dbf023bf35560a9daf1e@17.4.28.7:50600
 CSeq: 103 INVITE
 User-Agent: Asterisk PBX 12.6.1
 Authorization: Digest username=gw2, realm=provider.com,
 algorithm=MD5, uri=sip:89126975...@sip.provider.com, nonce=014d80ca,
 response=67bad8a0c97afc2b6747b471a56bca9f
 Date: Wed, 29 Oct 2014 18:50:50 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH, MESSAGE
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 253

 v=0
 o=root 1098729670 1098729671 IN IP4 17.4.28.7
 s=Asterisk PBX 12.6.1
 c=IN IP4 17.4.28.7
 t=0 0
 m=audio 10088 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=maxptime:150
 a=sendrecv


 Then I get dump from my kamailio (unsuccessfull INVITE)

 INVITE sip:89126975...@sip.provider.com  SIP/2.0
 Record-Route: sip:sip.myservice.com:5068;nat=yes;ftag=as4684d4b9;lr=on
 Via: SIP/2.0/UDP sip.myservice.com:5068
 ;branch=z9hG4bK600b.1d5ff0fd59d4f3d2a1a06d722c0daa92.2
 Via: SIP/2.0/UDP my.aterisk:50600;branch=z9hG4bK2b8d9b09;rport=50600
 Max-Forwards: 70
 From: sip:g...@sip.myservice.com:5068;tag=as4684d4b9
 To: sip:89126975...@sip.provider.com 
 Contact:sip:vebinar-...@sip.myservice.com:5068
 Call-ID: 445a7b884aeeab125d91886210c9b...@sip.myservice.com:50600
 CSeq: 102 INVITE
 User-Agent: SoftSwitch
 Date: Wed, 29 Oct 2014 22:32:32 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH, MESSAGE
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 312
 Authorization: Digest username=gw2, realm=provider.com,
 nonce=10129bde, uri=sip:89126975...@sip.provider.com ,
 response=6d3411b24cbb57ad72271790ec01b453, algorithm=MD5

 v=0
 o=root 468654998 468654998 IN IP4 1.2.3.4
 s=SoftSwitch
 c=IN IP4 1.2.3.4
 t=0 0
 m=audio 30104 RTP/AVP 8 3 0 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=maxptime:150
 a=sendrecv
 a=rtcp:30105


 I see difference between packetts only at SDP (not inportant things) and
 at VIA and request route Headers. All other fields identical.

 So -why Asterisk call successull and Kamailio kall unsuccessfull? What
 the differense?

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Re: [SR-Users] cSeq increasing