Re: [SR-Users] [sr-dev] Branch 5.0 created

[Alexandru Covalschi]

ed pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
> core/cfg/cfg_ctx.c:1223:5: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>  shm_free(CFG_GROUP_META(block, group)->array);
>  ^
> core/cfg/cfg_ctx.c: In function cfg_add_group_inst:
> core/cfg/cfg_ctx.c:1577:2: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   CFG_GROUP_META(block, group)->array = new_array;
>   ^
> core/cfg/cfg_ctx.c:1578:2: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   CFG_GROUP_META(block, group)->num++;
>   ^
> core/cfg/cfg_ctx.c:1580:2: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   if (CFG_GROUP_META(*cfg_global, group)->array) {
>   ^
> core/cfg/cfg_ctx.c:1589:3: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>replaced[0] = CFG_GROUP_META(*cfg_global, group)->array;
>^
> core/cfg/cfg_ctx.c: In function cfg_del_group_inst:
> core/cfg/cfg_ctx.c:1673:2: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   CFG_GROUP_META(block, group)->array = new_array;
>   ^
> core/cfg/cfg_ctx.c:1674:2: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   CFG_GROUP_META(block, group)->num--;
>   ^
> core/cfg/cfg_ctx.c:1676:2: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   if (CFG_GROUP_META(*cfg_global, group)->array) {
>   ^
> core/cfg/cfg_ctx.c:1687:5: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>  && (*(char **)(group_inst->vars + var->offset) != NULL)
>  ^
> core/cfg/cfg_ctx.c:1705:6: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   && (*(char **)(group_inst->vars + var->offset) != NULL)
>   ^
> core/cfg/cfg_ctx.c:1707:6: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>   replaced[num] = *(char **)(group_inst->vars + var->offset);
>   ^
> core/cfg/cfg_ctx.c:1713:3: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
>replaced[num] = CFG_GROUP_META(*cfg_global, group)->array;
>^
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tel: +37367398493
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[SR-Users] Kamctl stats

[Alexandru Covalschi]

Hello list,

Where can I found any information to completely understand what do values 
returned by 'kamctl stats' represent?

Cheers,
Alex
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Re: [SR-Users] Possible memory leak in mysql driver

[Alexandru Covalschi]

Thanks for response Daniel,

Can you point me to the location where the core dump should be generated? Afaik 
kamailio was compiled in /usr/local, what is the workdir for that setup? 
I can't figure it out from systemd config file. Also, is it really necessary to 
run as root? What if I set enough permissions on workdir?
> 16 нояб. 2016 г., в 11:32, Daniel-Constantin Mierla <mico...@gmail.com> 
> написал(а):
> 
> Hello,
> 
> the plan is to freeze development of v5.0 before the Chirstmas of the
> first week of January, then it will be a 1-1.5 months of testing,
> followed by the release. So expect like 2-3 months till the stable
> release of 5.0.
> 
> As for the issue, it can be a buffer overflow somewhere, not related to
> mysql module code, but triggered by use of it.
> 
> It is important to get the core file for such case, before starting
> kamailio do 'ulimit -c unlimited'. You may need to run kamailio as root
> to be able to write the core file. Enable also the option for one core
> file per process, typically is:
> 
> echo "1" > /proc/sys/kernel/core_uses_pid
> 
> Once you get core files, extract the output of 'bt full' with gdb and
> send it over here.
> 
> Cheers,
> Daniel
> 
> 
> On 15/11/16 22:35, Alexandru Covalschi wrote:
>> Hello list,
>> 
>> We’re using dev version of Kamailio:
>> version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5
>> 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: ff63e5
>> compiled on 15:46:49 May 31 2016 with gcc 4.9.2
>> 
>> Sometimes we encounter such issue:
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR:  
>> [db_row.c:114]: db_allocate_row(): no private memory left
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
>> [km_row.c:57]: db_mysql_convert_row(): could not allocate row
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
>> [km_res.c:188]: db_mysql_convert_rows(): error while converting row #16
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
>> [km_res.c:217]: db_mysql_convert_result(): error while converting rows
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
>> [km_dbase.c:261]: db_mysql_store_result(): error while converting result
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR:  
>> [db_query.c:139]: db_do_query_internal(): error while storing result
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: permissions 
>> [trusted.c:91]: reload_trusted_table(): failed to query database
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR:  
>> [db_row.c:114]: db_allocate_row(): no private memory left
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
>> [km_row.c:57]: db_mysql_convert_row(): could not allocate row
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
>> [km_dbase.c:444]: db_mysql_fetch_result(): error while converting row #15
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: htable 
>> [ht_db.c:234]: ht_db_load_table(): Error while fetching result
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: :  
>> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer 
>> (0x7f5ebda8ae18), called from db_mysql: km_dbase.c: 
>> db_mysql_free_result(305), first free db_mysql: km_dbase.c: 
>> db_mysql_free_result(305) - aborting
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12281]: CRITICAL:  
>> [pass_fd.c:275]: receive_fd(): EOF on 16
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT:  
>> [main.c:739]: handle_sigs(): child process 12276 exited by a signal 6
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT:  
>> [main.c:742]: handle_sigs(): core was not generated
>> Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: INFO:  
>> [main.c:754]: handle_sigs(): terminating due to SIGCHLD
>> 
>> The thing is we heavily use mysql module, but only to update the in-memory 
>> tables by kamcmd. Each N minutes a special script updates the 
>> trusted,address and htable executing kamcmd. Ka

[SR-Users] Possible memory leak in mysql driver

[Alexandru Covalschi]

Hello list,

We’re using dev version of Kamailio:
version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5
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: ff63e5
compiled on 15:46:49 May 31 2016 with gcc 4.9.2

Sometimes we encounter such issue:
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR:  
[db_row.c:114]: db_allocate_row(): no private memory left
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
[km_row.c:57]: db_mysql_convert_row(): could not allocate row
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
[km_res.c:188]: db_mysql_convert_rows(): error while converting row #16
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
[km_res.c:217]: db_mysql_convert_result(): error while converting rows
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
[km_dbase.c:261]: db_mysql_store_result(): error while converting result
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR:  
[db_query.c:139]: db_do_query_internal(): error while storing result
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: permissions 
[trusted.c:91]: reload_trusted_table(): failed to query database
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR:  
[db_row.c:114]: db_allocate_row(): no private memory left
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
[km_row.c:57]: db_mysql_convert_row(): could not allocate row
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql 
[km_dbase.c:444]: db_mysql_fetch_result(): error while converting row #15
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: htable 
[ht_db.c:234]: ht_db_load_table(): Error while fetching result
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: :  
[mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer 
(0x7f5ebda8ae18), called from db_mysql: km_dbase.c: db_mysql_free_result(305), 
first free db_mysql: km_dbase.c: db_mysql_free_result(305) - aborting
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12281]: CRITICAL:  
[pass_fd.c:275]: receive_fd(): EOF on 16
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT:  
[main.c:739]: handle_sigs(): child process 12276 exited by a signal 6
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT:  
[main.c:742]: handle_sigs(): core was not generated
Nov  9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: INFO:  
[main.c:754]: handle_sigs(): terminating due to SIGCHLD

The thing is we heavily use mysql module, but only to update the in-memory 
tables by kamcmd. Each N minutes a special script updates the trusted,address 
and htable executing kamcmd. Kamailio (and kamcmd as well) talks only with 
localhost mysql server.
What I saw when encountered that issue on a live machine is that issue happens 
only with one of child processes, any other are ok.
Interesting thing is that happens at the same time with machines on the same 
«set», I mean that issue happened simultaneously with two our test machines 
which actually didn’t have any load on them. 
The common thing between those machines is that they are in same subnet and 
local mysql databases are filled by scripts which query same external db.
I can’t confirm if there were or there weren’t any networking issues at that 
time with those machines, but as soon as kamcmd queries localhost that 
shouldn’t be the source of the issue.

So my questions are:
1. Has anyone encountered such thing?
2. Maybe the issue is already localized so it has sense to update? We actually 
use that on production (pls don’t throw too much rocks at me), so maintenance 
should be properly planned and I must be sure update won’t break anything. 
3. If update is proposed - how to do it? I mean - follow the guide 
https://www.kamailio.org/wiki/install/devel/git or there are some other tips? I 
suppose in ideal world I don’t even stop the binary, only restart after make 
all && make install are done, as everything is in-memory. Am I correct?
4. When can we expect stable 5.0 version? (at least tell if it’s months/years)

Thanks in advance!
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Re: [SR-Users] How to understad that user is de-registrating

[Alexandru Covalschi]

Thanks for the info guys, I've fixed my config to suit the correct logic.
Thanks again!

2016-06-23 8:40 GMT+03:00 Daniel-Constantin Mierla <mico...@gmail.com>:

> Hello,
>
> On 19/06/16 19:41, Alexandru Covalschi wrote:
>
> Hello list,
>
> I need to send to an external API events when user is registrated and
> de-registrated.
> As far as I understand standart behaviour is as follows:
>
> If user is not registered, he sends REGISTER and he is registrated (I can
> catch that because I make the auth).
> If user is registered and sends REGISTER he is de-register.
>
>
> This is not always a de-registration. If the expires value in the new
> REGISTER is >0, then it is an update of the registration (or
> re-registration). Only if expires==0, then it is a de-registration.
>
> Cheers,
> Daniel
>
> (Please correct me if I'm wrong.)
> How can I catch that?
> Can I use event_route[usrloc:contact-expired]?
>
> Thanks in advance!
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> tel: +37367398493
> web: http://abriss.solutions/ <http://abs-telecom.com/>
>
>
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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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>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
tel: +37367398493
web: http://abriss.solutions/ <http://abs-telecom.com/>
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Re: [SR-Users] WS to WS calls - No ACK received for 200OK

[Alexandru Covalschi]

The problem may be with record_route header.
Did you set
*advertised_address?*

2016-06-21 12:59 GMT+03:00 Amit Patkar :

> Hi
>
> I am using Kamailio as Websocket proxy.
>
> User 1 & User 2 are registered on Kamailio over WebSocket.
> When User 1 calls User 2, User 2 gets ring and answers the call. 200 OK
> message is received by User 1 but ACK response sent by User 1 does not
> reach User 2. Since User 2 didn't get ACS, after 30 sec timeout it drops
> the call.
> Media is exchanged for 30 seconds, which means ICE is successful.
>
> I am running kamailio 4.3.3 on
>
> Kamailio is behind firewall and running on private IP.
> rtpengine is configured to handle media.
> All ports are forwarded to kamailio server and Kamailio is allowed to send
> data on any public IP.
>
> I used jssip & sipml framework to test. Result was same for both
> frameworks. No ACK received
>
> I can see following errors in log file. What could be the reason which
> indicate ACK was not forwarded
>
> Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6584]: NOTICE: acc
> [acc.c:317]: acc_log_request(): ACC: transaction answered:
> timestamp=1466501141;method=INVITE;from_tag=7926B9NR7U0X99ZfAr1T;to_tag=w3D2JGd2EUFJpobTcFyo;call_id=1edc8e15-f1f1-584d-df81-71c3d59d713d;code=200;reason=OK;src_user=10001;src_domain=
> .com
> ;src_ip=xxx.yyy.zzz.aaa;dst_ouser=10002;dst_user=10002;dst_domain=df7jal23ls0d.invalid
> Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: 
> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no
> corresponding listening socket
> Jun 21 14:55:41 acstemplate /usr/sbin/kamailio[6582]: ERROR: sl
> [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm
> terribly sorry, server error occurred (7/SL)
> Jun 21 14:55:42 acstemplate /usr/sbin/kamailio[6584]: INFO: 

[SR-Users] How to understad that user is de-registrating

[Alexandru Covalschi]

Hello list,

I need to send to an external API events when user is registrated and
de-registrated.
As far as I understand standart behaviour is as follows:

If user is not registered, he sends REGISTER and he is registrated (I can
catch that because I make the auth).
If user is registered and sends REGISTER he is de-register.
(Please correct me if I'm wrong.)
How can I catch that?
Can I use event_route[usrloc:contact-expired]?

Thanks in advance!

-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
tel: +37367398493
web: http://abriss.solutions/ <http://abs-telecom.com/>
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Re: [SR-Users] Kamailio Problem carrying media

[Alexandru Covalschi]

Hi Sirvan,

I suppose you use rtpproxy/rtpproxy-ng/rtpengine as a media-proxy. First of
all check if those services are up and running, probably restarting
media-proxy and Kamailio can solve your problem.
If you don't use any media-proxy you should make a dump of SIP messages
going through your server and check the SDP part.

2016-06-16 21:39 GMT+03:00 Sirvan Paraste <sir...@golden-time.co>:

>
> Hi there
>
> We have got a situation that needed your help to overcome that. We have
> setup a Kamailio with voice, video call and text messaging and even push
> notification ability. it was working and till these days that we have lost
> media (voice and video and calling ability). seems that the clients do not
> see each other but strange is that they can send text messages and even
> push notification is work as well.
>
> Could you please give me an idea how to find the problem, or if you know
> how to solve it?
>
>
> Best regards.
>
> Sirvan Parasteh
>
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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
tel: +37367398493
web: http://abriss.solutions/ <http://abs-telecom.com/>
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Re: [SR-Users] Media server can't return ACK correctly

[Alexandru Covalschi]

Thanks everyone

2016-02-25 19:41 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Ok guys. The issue was in my misunderstanding of RFC and
> advertised_address variable.
> Removing advertised_address solved the issue.
>
> 2016-02-25 17:49 GMT+02:00 Alberto Sagredo <alberto.sagr...@avanzada7.com>
> :
>
>> :) Great
>>
>> So you will have maybe now something as this
>>
>> Record-Route: <sip:PUBLIC_IP;r2=on;lr=on;ftag=as2c0c55b9>
>> Record-Route: <sip:PRIVATE_IP;r2=on;lr=on;ftag=as2c0c55b9>
>>
>> And ACKS will go to right place..
>>
>>
>> 2016-02-25 16:43 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> force_send_socket is a good idea - thanx!
>>> traces are in initial message
>>>
>>> 2016-02-25 17:02 GMT+02:00 Alberto Sagredo <
>>> alberto.sagr...@avanzada7.com>:
>>>
>>>> HI Alexandru i talk about something like this maybe in your RELAY route
>>>> or similar.
>>>>
>>>> I think you would have issues with ACKs until you would have
>>>> Record-Route: doubled
>>>>
>>>>   if (dst_ip==LOCALIPNETWORK/24) {
>>>>
>>>>   xlog("Using socket: LOCALIP:5060");
>>>>
>>>>  force_send_socket(udp:LOCALIP:5060);
>>>>
>>>>   } else {
>>>>
>>>>xlog("Using socket: PUBLICIP:5060");
>>>>
>>>> force_send_socket(udp:PUBLICIP:5060);
>>>>
>>>>}
>>>>
>>>> Hope this helps you
>>>> Use record_route() as well.
>>>>
>>>> Anyway show me a trace that goes to FreeSwitch from Kamailio.
>>>>
>>>>
>>>>
>>>> 2016-02-25 10:55 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>>>>
>>>>> No other rr params defined so double rr is default - enabled.
>>>>> What do you mean by "force traffic" - how to do that? Every other
>>>>> request (excep BYE - same problem with it) flows OK.
>>>>>
>>>>> 2016-02-25 11:49 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>>>>
>>>>>> Hi, thanks for answer
>>>>>>
>>>>>> Here's configuration:
>>>>>>
>>>>>> listen=udp:MY_EXT_IP_ADDR:5060
>>>>>> listen=tcp:MY_EXT_IP_ADDR:5060
>>>>>> listen=udp:MY_INT_IP_ADDR:5060
>>>>>> listen=TCP:MY_INT_IP_ADDR:5060
>>>>>> listen=MY_WS_ADDR
>>>>>> advertised_address = MY_EXT_IP_ADDR
>>>>>> alias = MY_INT_IP_ADDR
>>>>>> alias = MY_DOMAIN
>>>>>>
>>>>>> #!ifdef WITH_TLS
>>>>>> listen=MY_WSS_ADDR
>>>>>> #!endif
>>>>>>
>>>>>> port=5060
>>>>>>
>>>>>> ...
>>>>>>
>>>>>> # - rr params -
>>>>>> modparam("rr", "enable_full_lr", 1)
>>>>>> modparam("rr", "append_fromtag", 1)
>>>>>>
>>>>>>
>>>>>> 2016-02-25 8:47 GMT+02:00 Alberto Sagredo <
>>>>>> alberto.sagr...@avanzada7.com>:
>>>>>>
>>>>>>> Hi Alexandru
>>>>>>>
>>>>>>> How is your configuration about Public IP and Private IP?
>>>>>>>
>>>>>>> Do you use advertise?
>>>>>>>
>>>>>>> Maybe you need to force Outbound traffic to Public IP Socket and
>>>>>>> inside traffic to Private IP .
>>>>>>>
>>>>>>> Do you have double record routing?
>>>>>>>
>>>>>>> BR
>>>>>>>
>>>>>>> 2016-02-25 1:24 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>>>>>>>
>>>>>>>> Hello everyone.
>>>>>>>>
>>>>>>>> The setup is:
>>>>>>>> Carrier ip is CARRIER_IP
>>>>>>>> Public network Kamailio IP will be PUBLIC_IP
>>>>>>>> Private network Kamailio IP will be KAMAILIO_PRIVATE_IP
>>>>>>>> Private network Freeswitch IP is FREESWITCH_PRIVATE_IP
>>>>>>>>
>>>>>>>> ACK
>>>>>>>> CARRIER_IP -> PU

Re: [SR-Users] Media server can't return ACK correctly

[Alexandru Covalschi]

Ok guys. The issue was in my misunderstanding of RFC and advertised_address
variable.
Removing advertised_address solved the issue.

2016-02-25 17:49 GMT+02:00 Alberto Sagredo <alberto.sagr...@avanzada7.com>:

> :) Great
>
> So you will have maybe now something as this
>
> Record-Route: <sip:PUBLIC_IP;r2=on;lr=on;ftag=as2c0c55b9>
> Record-Route: <sip:PRIVATE_IP;r2=on;lr=on;ftag=as2c0c55b9>
>
> And ACKS will go to right place..
>
>
> 2016-02-25 16:43 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>
>> force_send_socket is a good idea - thanx!
>> traces are in initial message
>>
>> 2016-02-25 17:02 GMT+02:00 Alberto Sagredo <alberto.sagr...@avanzada7.com
>> >:
>>
>>> HI Alexandru i talk about something like this maybe in your RELAY route
>>> or similar.
>>>
>>> I think you would have issues with ACKs until you would have
>>> Record-Route: doubled
>>>
>>>   if (dst_ip==LOCALIPNETWORK/24) {
>>>
>>>   xlog("Using socket: LOCALIP:5060");
>>>
>>>  force_send_socket(udp:LOCALIP:5060);
>>>
>>>   } else {
>>>
>>>xlog("Using socket: PUBLICIP:5060");
>>>
>>> force_send_socket(udp:PUBLICIP:5060);
>>>
>>>}
>>>
>>> Hope this helps you
>>> Use record_route() as well.
>>>
>>> Anyway show me a trace that goes to FreeSwitch from Kamailio.
>>>
>>>
>>>
>>> 2016-02-25 10:55 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>>>
>>>> No other rr params defined so double rr is default - enabled.
>>>> What do you mean by "force traffic" - how to do that? Every other
>>>> request (excep BYE - same problem with it) flows OK.
>>>>
>>>> 2016-02-25 11:49 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>>>
>>>>> Hi, thanks for answer
>>>>>
>>>>> Here's configuration:
>>>>>
>>>>> listen=udp:MY_EXT_IP_ADDR:5060
>>>>> listen=tcp:MY_EXT_IP_ADDR:5060
>>>>> listen=udp:MY_INT_IP_ADDR:5060
>>>>> listen=TCP:MY_INT_IP_ADDR:5060
>>>>> listen=MY_WS_ADDR
>>>>> advertised_address = MY_EXT_IP_ADDR
>>>>> alias = MY_INT_IP_ADDR
>>>>> alias = MY_DOMAIN
>>>>>
>>>>> #!ifdef WITH_TLS
>>>>> listen=MY_WSS_ADDR
>>>>> #!endif
>>>>>
>>>>> port=5060
>>>>>
>>>>> ...
>>>>>
>>>>> # - rr params -
>>>>> modparam("rr", "enable_full_lr", 1)
>>>>> modparam("rr", "append_fromtag", 1)
>>>>>
>>>>>
>>>>> 2016-02-25 8:47 GMT+02:00 Alberto Sagredo <
>>>>> alberto.sagr...@avanzada7.com>:
>>>>>
>>>>>> Hi Alexandru
>>>>>>
>>>>>> How is your configuration about Public IP and Private IP?
>>>>>>
>>>>>> Do you use advertise?
>>>>>>
>>>>>> Maybe you need to force Outbound traffic to Public IP Socket and
>>>>>> inside traffic to Private IP .
>>>>>>
>>>>>> Do you have double record routing?
>>>>>>
>>>>>> BR
>>>>>>
>>>>>> 2016-02-25 1:24 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>>>>>>
>>>>>>> Hello everyone.
>>>>>>>
>>>>>>> The setup is:
>>>>>>> Carrier ip is CARRIER_IP
>>>>>>> Public network Kamailio IP will be PUBLIC_IP
>>>>>>> Private network Kamailio IP will be KAMAILIO_PRIVATE_IP
>>>>>>> Private network Freeswitch IP is FREESWITCH_PRIVATE_IP
>>>>>>>
>>>>>>> ACK
>>>>>>> CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP
>>>>>>>
>>>>>>> And freeswitch tries to actually send ACK back to PUBLIC_IP which he
>>>>>>> can't access.
>>>>>>>
>>>>>>> Kamailio trace: http://pastebin.com/raw/1W1sXuUa
>>>>>>> Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ
>>>>>>>
>>>>>>> request_route: http://pastebin.com/raw/Y17pXUGY
>>>>>>> NATMANAGE route: http://pastebin

Re: [SR-Users] Media server can't return ACK correctly

[Alexandru Covalschi]

No other rr params defined so double rr is default - enabled.
What do you mean by "force traffic" - how to do that? Every other request
(excep BYE - same problem with it) flows OK.

2016-02-25 11:49 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Hi, thanks for answer
>
> Here's configuration:
>
> listen=udp:MY_EXT_IP_ADDR:5060
> listen=tcp:MY_EXT_IP_ADDR:5060
> listen=udp:MY_INT_IP_ADDR:5060
> listen=TCP:MY_INT_IP_ADDR:5060
> listen=MY_WS_ADDR
> advertised_address = MY_EXT_IP_ADDR
> alias = MY_INT_IP_ADDR
> alias = MY_DOMAIN
>
> #!ifdef WITH_TLS
> listen=MY_WSS_ADDR
> #!endif
>
> port=5060
>
> ...
>
> # - rr params -
> modparam("rr", "enable_full_lr", 1)
> modparam("rr", "append_fromtag", 1)
>
>
> 2016-02-25 8:47 GMT+02:00 Alberto Sagredo <alberto.sagr...@avanzada7.com>:
>
>> Hi Alexandru
>>
>> How is your configuration about Public IP and Private IP?
>>
>> Do you use advertise?
>>
>> Maybe you need to force Outbound traffic to Public IP Socket and inside
>> traffic to Private IP .
>>
>> Do you have double record routing?
>>
>> BR
>>
>> 2016-02-25 1:24 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> Hello everyone.
>>>
>>> The setup is:
>>> Carrier ip is CARRIER_IP
>>> Public network Kamailio IP will be PUBLIC_IP
>>> Private network Kamailio IP will be KAMAILIO_PRIVATE_IP
>>> Private network Freeswitch IP is FREESWITCH_PRIVATE_IP
>>>
>>> ACK
>>> CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP
>>>
>>> And freeswitch tries to actually send ACK back to PUBLIC_IP which he
>>> can't access.
>>>
>>> Kamailio trace: http://pastebin.com/raw/1W1sXuUa
>>> Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ
>>>
>>> request_route: http://pastebin.com/raw/Y17pXUGY
>>> NATMANAGE route: http://pastebin.com/raw/0BpPDjN0
>>> WITHINDLG route: http://pastebin.com/raw/5LpwSigF
>>>
>>> I'm seeking help with that - what parameter I need to change/add to
>>> solve that?
>>> Maybe it's a networking problem - but why then ACK reaches Freeswitch
>>> and all other requests flow OK?
>>>
>>> Thanks in advance, Alex
>>>
>>> --
>>> Alexandru Covalschi
>>> ABRISS-Solutions
>>> VoIP engineer and system administrator
>>> phone: +37367398493
>>> web: http://abs-telecom.com/
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> ___
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Media server can't return ACK correctly

[Alexandru Covalschi]

Hi, thanks for answer

Here's configuration:

listen=udp:MY_EXT_IP_ADDR:5060
listen=tcp:MY_EXT_IP_ADDR:5060
listen=udp:MY_INT_IP_ADDR:5060
listen=TCP:MY_INT_IP_ADDR:5060
listen=MY_WS_ADDR
advertised_address = MY_EXT_IP_ADDR
alias = MY_INT_IP_ADDR
alias = MY_DOMAIN

#!ifdef WITH_TLS
listen=MY_WSS_ADDR
#!endif

port=5060

...

# - rr params -
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 1)


2016-02-25 8:47 GMT+02:00 Alberto Sagredo <alberto.sagr...@avanzada7.com>:

> Hi Alexandru
>
> How is your configuration about Public IP and Private IP?
>
> Do you use advertise?
>
> Maybe you need to force Outbound traffic to Public IP Socket and inside
> traffic to Private IP .
>
> Do you have double record routing?
>
> BR
>
> 2016-02-25 1:24 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Hello everyone.
>>
>> The setup is:
>> Carrier ip is CARRIER_IP
>> Public network Kamailio IP will be PUBLIC_IP
>> Private network Kamailio IP will be KAMAILIO_PRIVATE_IP
>> Private network Freeswitch IP is FREESWITCH_PRIVATE_IP
>>
>> ACK
>> CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP
>>
>> And freeswitch tries to actually send ACK back to PUBLIC_IP which he
>> can't access.
>>
>> Kamailio trace: http://pastebin.com/raw/1W1sXuUa
>> Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ
>>
>> request_route: http://pastebin.com/raw/Y17pXUGY
>> NATMANAGE route: http://pastebin.com/raw/0BpPDjN0
>> WITHINDLG route: http://pastebin.com/raw/5LpwSigF
>>
>> I'm seeking help with that - what parameter I need to change/add to solve
>> that?
>> Maybe it's a networking problem - but why then ACK reaches Freeswitch and
>> all other requests flow OK?
>>
>> Thanks in advance, Alex
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493
>> web: http://abs-telecom.com/
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] Media server can't return ACK correctly

[Alexandru Covalschi]

Hello everyone.

The setup is:
Carrier ip is CARRIER_IP
Public network Kamailio IP will be PUBLIC_IP
Private network Kamailio IP will be KAMAILIO_PRIVATE_IP
Private network Freeswitch IP is FREESWITCH_PRIVATE_IP

ACK
CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP

And freeswitch tries to actually send ACK back to PUBLIC_IP which he can't
access.

Kamailio trace: http://pastebin.com/raw/1W1sXuUa
Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ

request_route: http://pastebin.com/raw/Y17pXUGY
NATMANAGE route: http://pastebin.com/raw/0BpPDjN0
WITHINDLG route: http://pastebin.com/raw/5LpwSigF

I'm seeking help with that - what parameter I need to change/add to solve
that?
Maybe it's a networking problem - but why then ACK reaches Freeswitch and
all other requests flow OK?

Thanks in advance, Alex

-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Kamailio and NAT

[Alexandru Covalschi]

AFAIK bye is usually sent to the address stored in record_route. Try
setting changing record_route() to
record_route_preset("PUBLICIP:5060;nat=yes:)

2015-12-23 16:28 GMT+02:00 Nelson Migliaro <eng.migli...@gmail.com>:

>
> Hello,
>
> I am running Kamailio behind NAT.
>
> Kanailio has a private IP and I am relaying NAT to internet router.
>
> I am using:
>
> - #!define WITH_NAT
> - listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060
>
> - Patched RTP proxy including the advertise option
>
> And everything goes fine. I can make calls and have two way audio.
>
> The problem begins when the callee ends the call. BYE is not received in
> Kamailio (caller)
>
> I included the public IP using "add_contact_alias" because
> "set_contact_alias" was not adding the public IP. I included this in in
> NATDETECT (pre loaded router)
>
> if(is_first_hop()) {
> xlog("L_NOTICE","Metodo: $rm \n");
> xlog("L_NOTICE","is first hop\n");
> #set_contact_alias();
>  if (!add_contact_alias("PUBLIC-IP", "$Rp", "udp")) {
>  xlog("L_ERR", "Error in aliasing contact $ct\n");
> send_reply("400", "Bad request");
> exit;
> }
> }
>
> I think the problem is related to destination that BYE is sent by the
> vendor. From what I see IP and port is taken from advertised in contact
> (PUBLIC-IP and 5060).
> The problem is that internet router changes the source port.
>
> Contact: <sip:9@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1>
>
> --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in
> order to received new transactions or should I follow a different
> procedure???
>
> Thank you
>
>
>
>
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>
>


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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] IMC module

[Alexandru Covalschi]

Hello again
First of all I wanted to ask if someone ever implemented that
http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC
Second question is - I don't understand the logic. In description is said:

Handles Message method.It detects if the body of the message is a
conference command.If so it executes it, otherwise it sends the message to
all the members in the room.

But why in example (well however it has broken syntax) to IMC manager are
sent only messages from chat-rooms? How message from client can possibly
reach imc-manager then?

Also - when I send message with body "111" from user 1001 and
imc_manager catches it -  I receive 500 command error. Why? :/

All that is working on top of sipjs-demo.

-- 
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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] IMC module

[Alexandru Covalschi]

upd: Let me describe my use case. I need conference chats. Clients are
1001-1...@domain.name
Conference is 3...@domain.name
Messages are sent from 1001-1...@domain.name to 3...@domain.name
Every user joins conference room chat-3500. Join is successful - but
nothing more. Messages are not relayed anywhere and I don't want to relay
them - I just need to understand how this module works. My current
configuration is:
if (is_method("MESSAGE") && $fU != "chat-3500")
{
if(imc_manager())
sl_send_reply("200", "ok");
else
sl_send_reply("500", "command error");
exit;
}
This allows me to receive system messages - but I can't get any messages
from clients.

2015-12-15 18:43 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Hello again
> First of all I wanted to ask if someone ever implemented that
> http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC
> Second question is - I don't understand the logic. In description is said:
>
> Handles Message method.It detects if the body of the message is a
> conference command.If so it executes it, otherwise it sends the message to
> all the members in the room.
>
> But why in example (well however it has broken syntax) to IMC manager are
> sent only messages from chat-rooms? How message from client can possibly
> reach imc-manager then?
>
> Also - when I send message with body "111" from user 1001 and
> imc_manager catches it -  I receive 500 command error. Why? :/
>
> All that is working on top of sipjs-demo.
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] IMC module

[Alexandru Covalschi]

Ok I got it - endpoint must send MESSAGE to sip:conference-n...@domain.name,
but initially it sends to conference-name.
So as I understood - if (is_method("MESSAGE") && !(starts_with("$fU",
"chat"))) should make everything work correct.
Sorry for emotions :)

2015-12-15 19:00 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> upd: Let me describe my use case. I need conference chats. Clients are
> 1001-1...@domain.name
> Conference is 3...@domain.name
> Messages are sent from 1001-1...@domain.name to 3...@domain.name
> Every user joins conference room chat-3500. Join is successful - but
> nothing more. Messages are not relayed anywhere and I don't want to relay
> them - I just need to understand how this module works. My current
> configuration is:
> if (is_method("MESSAGE") && $fU != "chat-3500")
> {
> if(imc_manager())
> sl_send_reply("200", "ok");
> else
> sl_send_reply("500", "command error");
> exit;
> }
> This allows me to receive system messages - but I can't get any messages
> from clients.
>
> 2015-12-15 18:43 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Hello again
>> First of all I wanted to ask if someone ever implemented that
>> http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC
>> Second question is - I don't understand the logic. In description is
>> said:
>>
>> Handles Message method.It detects if the body of the message is a
>> conference command.If so it executes it, otherwise it sends the message to
>> all the members in the room.
>>
>> But why in example (well however it has broken syntax) to IMC manager are
>> sent only messages from chat-rooms? How message from client can possibly
>> reach imc-manager then?
>>
>> Also - when I send message with body "111" from user 1001 and
>> imc_manager catches it -  I receive 500 command error. Why? :/
>>
>> All that is working on top of sipjs-demo.
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493
>> web: http://abs-telecom.com/
>>
>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] Relaying failure codes back to initial server

[Alexandru Covalschi]

Hello everyone!
I need to relay 486/408/... other failure codes back to initial INVITE
server. Here
http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is
recommended just to exit failure_route, but that didn't work for me. I need
that to let Freeswitch know which cause has ended the call. Now Kamailio
just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me
how to achieve that?
Thanks in advance
-- 
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Re: [SR-Users] Relaying failure codes back to initial server

[Alexandru Covalschi]

I use sngrep to track view the flow and I'm pretty sure it's accurate
enough to tell me that.
Here's relay route:
route[RELAY] {

# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route"))
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}

and here's reply routes

# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}

# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}

# Manage failure routing cases
failure_route[MANAGE_FAILURE] {

if (t_check_status("486")) {
exit;
}
route(NATMANAGE);

if (t_is_canceled()) {
exit;
}

}


However when endpoint replies with 486 BUSY I can't see that on FS,
Kamailio just sends 408 REQ TERM after some amount of time

2015-12-15 13:34 GMT+02:00 Alex Balashov <abalas...@evaristesys.com>:

> From what you describe, the reply should be going back to the sender. Are
> you absolutely sure that it's not? If so, are there any other actions you
> could be taking somewhere to drop it, such as in an onreply_route?
>
> ACKs to negative final replies are hop-by-hop, so the ACK you're seeing
> directly from the proxy to the UAS is normal.
>
> --
> 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/
>
> Sent from my BlackBerry.
> *From: *Alexandru Covalschi
> *Sent: *Tuesday, December 15, 2015 05:03
> *To: *Kamailio (SER) - Users Mailing List
> *Reply To: *Kamailio (SER) - Users Mailing List
> *Subject: *[SR-Users] Relaying failure codes back to initial server
>
> Hello everyone!
> I need to relay 486/408/... other failure codes back to initial INVITE
> server. Here
> http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html
> is recommended just to exit failure_route, but that didn't work for me. I
> need that to let Freeswitch know which cause has ended the call. Now
> Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly
> tell me how to achieve that?
> Thanks in advance
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Relaying failure codes back to initial server

[Alexandru Covalschi]

Yes, sometimes there are more than one INVITE (custom platform specific
behavior), in case of 1 INVITE per-dialogue it works nice. Can you suggest
something to cover such cases?

2015-12-15 14:12 GMT+02:00 Daniel-Constantin Mierla <mico...@gmail.com>:

> Maybe there is a parallel forking and one branch gets timed out (in this
> case 408 is selected against 486). How many INVITE requests do you see
> being sent out? Or you can eventually make the sip trace available for
> viewing on this mailing list or some web site/pastebin out there.
>
> Cheers,
> Daniel
>
>
> On 15/12/15 12:54, Alexandru Covalschi wrote:
>
> I use sngrep to track view the flow and I'm pretty sure it's accurate
> enough to tell me that.
> Here's relay route:
> route[RELAY] {
>
> # enable additional event routes for forwarded requests
> # - serial forking, RTP relaying handling, a.s.o.
> if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
> if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
> }
> if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
> if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
> }
> if (is_method("INVITE")) {
> if(!t_is_set("failure_route"))
> t_on_failure("MANAGE_FAILURE");
> }
> if (!t_relay()) {
> sl_reply_error();
> }
> exit;
> }
>
> and here's reply routes
>
> # Manage outgoing branches
> branch_route[MANAGE_BRANCH] {
> xdbg("new branch [$T_branch_idx] to $ru\n");
> route(NATMANAGE);
> }
>
> # Manage incoming replies
> onreply_route[MANAGE_REPLY] {
> xdbg("incoming reply\n");
> if(status=~"[12][0-9][0-9]")
> route(NATMANAGE);
> }
>
> # Manage failure routing cases
> failure_route[MANAGE_FAILURE] {
>
> if (t_check_status("486")) {
> exit;
> }
> route(NATMANAGE);
>
> if (t_is_canceled()) {
> exit;
> }
>
> }
>
>
> However when endpoint replies with 486 BUSY I can't see that on FS,
> Kamailio just sends 408 REQ TERM after some amount of time
>
> 2015-12-15 13:34 GMT+02:00 Alex Balashov <abalas...@evaristesys.com>:
>
>> From what you describe, the reply should be going back to the sender. Are
>> you absolutely sure that it's not? If so, are there any other actions you
>> could be taking somewhere to drop it, such as in an onreply_route?
>>
>> ACKs to negative final replies are hop-by-hop, so the ACK you're seeing
>> directly from the proxy to the UAS is normal.
>>
>> --
>> 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/>
>> http://www.csrpswitch.com/
>>
>> Sent from my BlackBerry.
>> *From: *Alexandru Covalschi
>> *Sent: *Tuesday, December 15, 2015 05:03
>> *To: *Kamailio (SER) - Users Mailing List
>> *Reply To: *Kamailio (SER) - Users Mailing List
>> *Subject: *[SR-Users] Relaying failure codes back to initial server
>>
>> Hello everyone!
>> I need to relay 486/408/... other failure codes back to initial INVITE
>> server. Here
>> <http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html>
>> http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html
>> is recommended just to exit failure_route, but that didn't work for me. I
>> need that to let Freeswitch know which cause has ended the call. Now
>> Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly
>> tell me how to achieve that?
>> Thanks in advance
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493
>> web: http://abs-telecom.com/
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>
>
> ___
> SIP Express Route

[SR-Users] Kamailio as UAC

[Alexandru Covalschi]

Hello everyone!

Subj is an ethernal theme, I know, but odds are so that I just need to do
that.

I've configured UAC auth to successfully register and my route[PSTN] looks
like that

# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not
defined\n");
return;
}
if !ds_is_from_list()
return;
# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN
routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;
if (strempty($sel(cfg_get.pstn.gw_port))) {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
} else {
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
+ $sel(cfg_get.pstn.gw_port);
}

remove_hf("To");
insert_hf("To: <sip:$r...@sipprovider.com>\r\n", "Call-ID");
uac_replace_from("","sip:u...@sipprovider.com");
route(RELAY);
exit;

On INVITE's I get 407 PROXY-AUTH, which are transfered to backend FS.
If I try to put
if ($T_reply_code == 401 or $T_reply_code == 407) {
xlog("L_NOTICE", "Remote asked for authentication");
    uac_auth();
}
to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start.

Is that even possible?

-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Kamailio as UAC

[Alexandru Covalschi]

I saw that, but
1. It doesn't work in failure_route (MANAGE_FAILURE from std. config) either
2. My question was more general - is it even possible to do what I need
with Kamailio

2015-11-27 13:24 GMT+02:00 Daniel Tryba <d.tr...@pocos.nl>:

> On Friday 27 November 2015 13:01:19 Alexandru Covalschi wrote:
> > If I try to put
> > if ($T_reply_code == 401 or $T_reply_code == 407) {
> > xlog("L_NOTICE", "Remote asked for authentication");
> > uac_auth();
> > }
> > to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start.
> >
> > Is that even possible?
>
>
> Why/What is the error reported?
>
> Have you read:
>
> http://www.kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth%28%29
> "
> 4.7.  uac_auth()
>
> This function can be called only from failure route and will build the
> authentication response header and insert it into the request without
> sending
> anything.
>
> This function can be used from FAILURE_ROUTE.
> "
>
> So is the error that you are using a function in the reply route that isn't
> being allowed in that router?
>
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>



-- 
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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Kamailio as UAC

[Alexandru Covalschi]

Well I tried but didn't work for me :( however problem is solved using
other voip provider. Thanks for help!
27 нояб. 2015 г. 14:09 пользователь "Daniel Tryba" <d.tr...@pocos.nl>
написал:

> On Friday 27 November 2015 13:50:36 Alexandru Covalschi wrote:
> > I saw that, but
> > 1. It doesn't work in failure_route (MANAGE_FAILURE from std. config)
> either
> > 2. My question was more general - is it even possible to do what I need
> > with Kamailio
>
> Yes it is possible. I tried the exact example from
>
> http://www.kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth%28%29
> once upon a time and it worked.
>
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

Well, reopening that thread seaking for some help again :(
The solution is working pretty nice, and my config looks like that
# authenticate requests
if has_credentials(""){
$var(y) = @msg.header.Authorization;
xlog("$var(y)");
exec_avp("/etc/kamailio/login.py '$var(y)' '$rm'",
"$avp(s:test)");
xlog("$avp(s:test)");
}

if ($avp(s:test) != "1") {
www_challenge(", "1");
exit;
}

login.py returns 1 if creds are OK and 0 if no.
Now I'm seeking help with such question - as I understand, currently anyone
can register or auth his requests by using same Authorization header for
all purposes. So, I mean, someone can grab Auth header from the user's
packet and just use it to dig in the server.
How to avoid that? As I understood it's implemented in Kamailio. Can you
please tell me? Or give a link to RFC/doc where this is described? As I
understood, I'll need to implement that in my script, or maybe I can use
some built-it functions?

2015-11-13 19:52 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Many thanks for you help Sebastian!
>
> 2015-11-13 19:13 GMT+02:00 Sebastian Damm <d...@sipgate.de>:
>
>>
>> On Fri, Nov 13, 2015 at 3:43 PM, Alexandru Covalschi <568...@gmail.com>
>> wrote:
>>
>>> What if I don't need a plaintext password on Kamailio? I mean, I don't
>>> want to user pv_www_authenticate or other auth functions again - I need to
>>> fully control AUTH on API. Is it ok to just send 200 OK to client if API
>>> tells me that password is ok?
>>>
>>
>> You don't need to use pv_*_authenticate. This is just an internal
>> function which tells you, whether your user supplied correct credentials or
>> not. You can replace it by checking the return code or output of the script
>> and then continue in your dialplan. You could then call save() from the
>> registrar module, which automatically sends a 200 OK to your user (unless
>> you explicitly prevent it from doing so).
>>
>> Sebastian
>>
>> _______
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Authorization selects

[Alexandru Covalschi]

UPD: proxy_auth doesn't work either, however I'm sure I have WWW-Auth, not
Proxy-Auth :)

2015-11-16 16:23 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Hello everyone!
>
> I need to extract values from authentication header, but
> 408. @authorization["string"]
> 409. @authorization["string"].username
> 410. @authorization["string"].username.user
> 411. @authorization["string"].username.domain
> 412. @authorization["string"].realm
> 413. @authorization["string"].nonce
> 414. @authorization["string"].uri
> 415. @authorization["string"].cnonce
> 416. @authorization["string"].nc
> 417. @authorization["string"].response
> 418. @authorization["string"].opaque
> 419. @authorization["string"].algorithm
> 420. @authorization["string"].qop
>
> these values doesn't seem to work for me.
> Used
> https://github.com/kamailio/kamailio/blob/master/doc/select_list/select_core.txt
> doc.
>
> $var(z) = @authorization["string"].realm;
> xlog("Realm $var(z)");
> Using that simple check I get nothing in $var(z)
>
> version: kamailio 4.3.3 (x86_64/linux)
> 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,
> DBG_QM_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: unknown
> compiled with gcc 4.9.2
>
> Can anyone check that please? Or maybe syntax is wrong and I need newer
> selects doc?
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] Authorization selects

[Alexandru Covalschi]

Hello everyone!

I need to extract values from authentication header, but
408. @authorization["string"]
409. @authorization["string"].username
410. @authorization["string"].username.user
411. @authorization["string"].username.domain
412. @authorization["string"].realm
413. @authorization["string"].nonce
414. @authorization["string"].uri
415. @authorization["string"].cnonce
416. @authorization["string"].nc
417. @authorization["string"].response
418. @authorization["string"].opaque
419. @authorization["string"].algorithm
420. @authorization["string"].qop

these values doesn't seem to work for me.
Used
https://github.com/kamailio/kamailio/blob/master/doc/select_list/select_core.txt
doc.

$var(z) = @authorization["string"].realm;
xlog("Realm $var(z)");
Using that simple check I get nothing in $var(z)

version: kamailio 4.3.3 (x86_64/linux)
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,
DBG_QM_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: unknown
compiled with gcc 4.9.2

Can anyone check that please? Or maybe syntax is wrong and I need newer
selects doc?
-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

Many thanks for you help Sebastian!

2015-11-13 19:13 GMT+02:00 Sebastian Damm <d...@sipgate.de>:

>
> On Fri, Nov 13, 2015 at 3:43 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> What if I don't need a plaintext password on Kamailio? I mean, I don't
>> want to user pv_www_authenticate or other auth functions again - I need to
>> fully control AUTH on API. Is it ok to just send 200 OK to client if API
>> tells me that password is ok?
>>
>
> You don't need to use pv_*_authenticate. This is just an internal function
> which tells you, whether your user supplied correct credentials or not. You
> can replace it by checking the return code or output of the script and then
> continue in your dialplan. You could then call save() from the registrar
> module, which automatically sends a 200 OK to your user (unless you
> explicitly prevent it from doing so).
>
> Sebastian
>
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>
>


-- 
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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

So it should be like

...
if (!has_credentials("myrealm")) {
www_challenge("$td", "1");
}

else {

 if (!my_script()){

 sl_send_reply("401", "Not Authorized");
     }

}

...

2015-11-13 16:13 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> simple send_reply("200", "OK");, sorry
>
> 2015-11-13 16:02 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>
>> Thanks for your reply! But the problem is - I need to provide to API
>> user's login and password. Kamailio doesn't know them. So my idea was to
>> transmit to API the salt and encrypted password. Would that work? I see it
>> that way
>> 1. User sends register request.
>> 2. Kamailio sends to API salt and ecnr.passwd
>> 3. API recalculates MD5 on its side and compares with encr.passwd
>> 4. Sends OK if it's ok, huh
>> 5. I receive OK from API and send simple 200 OK to user
>>
>> Do you see any logical mistakes here? Do I need some speacial 200 OK to
>> approve registration, or simple send_reply("401", "OK"); is enough?
>>
>>
>> 2015-11-13 15:21 GMT+02:00 Sebastian Damm <d...@sipgate.de>:
>>
>>> Hello,
>>>
>>> if your script can return the password for the user to Kamailio, you
>>> could use the pv_*_authenticate functions. You can pass the password to
>>> check against to these functions in a pseudo variable.
>>>
>>>
>>> http://www.kamailio.net/docs/modules/4.3.x/modules/auth.html#auth.f.pv_www_authenticate
>>>
>>> Best Regards,
>>> Sebastian
>>>
>>> On Fri, Nov 13, 2015 at 2:14 PM, Alexandru Covalschi <568...@gmail.com>
>>> wrote:
>>>
>>>> UPD: If upper method is possible - I assume I can check if message has
>>>> Auth header using
>>>>
>>>> if (has_credentials("myrealm")) {
>>>> ...
>>>> }
>>>> Can you please specify how to grab it?
>>>>
>>>>
>>>> 2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>>>
>>>>> Hello!
>>>>> My problem is I need to do users authentication through API. So I need
>>>>> to replace
>>>>>
>>>>> if (!www_authenticate("$td", "subscriber")) {
>>>>>   www_challenge("$td", "1");
>>>>> }
>>>>>
>>>>> With
>>>>>
>>>>> if (!my_auth_script()) {
>>>>>   www_challenge("$td", "1");
>>>>> }
>>>>>
>>>>> The main problem is - how can I grab or compare users password? I know
>>>>> nonce, which I understand is MD5 salt. Can I, for example, grab users
>>>>> password from API, then grab the MD5 string and the nonce user sent me,
>>>>> calculate MD5 on base of API password and nonce - and then compare MD5
>>>>> strings sent by user and calculated?
>>>>>
>>>>> --
>>>>> Alexandru Covalschi
>>>>> ABRISS-Solutions
>>>>> VoIP engineer and system administrator
>>>>> phone: +37367398493
>>>>> web: http://abs-telecom.com/
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Alexandru Covalschi
>>>> ABRISS-Solutions
>>>> VoIP engineer and system administrator
>>>> phone: +37367398493
>>>> web: http://abs-telecom.com/
>>>>
>>>> ___
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users@lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>> ___
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>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493
>> web: http://abs-telecom.com/
>>
>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

What if I don't need a plaintext password on Kamailio? I mean, I don't want
to user pv_www_authenticate or other auth functions again - I need to fully
control AUTH on API. Is it ok to just send 200 OK to client if API tells me
that password is ok?

2015-11-13 16:39 GMT+02:00 Sebastian Damm <d...@sipgate.de>:

> Hello,
>
> it't been a while since I worked with external scripts, but you can exec
> external scripts. See:
> http://www.kamailio.net/docs/modules/4.3.x/modules/exec.html
>
> The documentation says, you can access header fields of the packet via
> environment variables. So you can get the WWW-Authorize header into your
> script, extract the needed fields and send them to the API. The API then
> should be able to calculate the response again according to the Digest
> Authentication rules with the supplied information and the plain password.
>
> Best Regards,
> Sebastian
>
>
>
> On Fri, Nov 13, 2015 at 3:13 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> simple send_reply("200", "OK");, sorry
>>
>> 2015-11-13 16:02 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> Thanks for your reply! But the problem is - I need to provide to API
>>> user's login and password. Kamailio doesn't know them. So my idea was to
>>> transmit to API the salt and encrypted password. Would that work? I see it
>>> that way
>>> 1. User sends register request.
>>> 2. Kamailio sends to API salt and ecnr.passwd
>>> 3. API recalculates MD5 on its side and compares with encr.passwd
>>> 4. Sends OK if it's ok, huh
>>> 5. I receive OK from API and send simple 200 OK to user
>>>
>>> Do you see any logical mistakes here? Do I need some speacial 200 OK to
>>> approve registration, or simple send_reply("401", "OK"); is enough?
>>>
>>>
>>> 2015-11-13 15:21 GMT+02:00 Sebastian Damm <d...@sipgate.de>:
>>>
>>>> Hello,
>>>>
>>>> if your script can return the password for the user to Kamailio, you
>>>> could use the pv_*_authenticate functions. You can pass the password to
>>>> check against to these functions in a pseudo variable.
>>>>
>>>>
>>>> http://www.kamailio.net/docs/modules/4.3.x/modules/auth.html#auth.f.pv_www_authenticate
>>>>
>>>> Best Regards,
>>>> Sebastian
>>>>
>>>> On Fri, Nov 13, 2015 at 2:14 PM, Alexandru Covalschi <568...@gmail.com>
>>>> wrote:
>>>>
>>>>> UPD: If upper method is possible - I assume I can check if message has
>>>>> Auth header using
>>>>>
>>>>> if (has_credentials("myrealm")) {
>>>>> ...
>>>>> }
>>>>> Can you please specify how to grab it?
>>>>>
>>>>>
>>>>> 2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>>>>
>>>>>> Hello!
>>>>>> My problem is I need to do users authentication through API. So I
>>>>>> need to replace
>>>>>>
>>>>>> if (!www_authenticate("$td", "subscriber")) {
>>>>>>  www_challenge("$td", "1");
>>>>>> }
>>>>>>
>>>>>> With
>>>>>>
>>>>>> if (!my_auth_script()) {
>>>>>>  www_challenge("$td", "1");
>>>>>> }
>>>>>>
>>>>>> The main problem is - how can I grab or compare users password? I
>>>>>> know nonce, which I understand is MD5 salt. Can I, for example, grab 
>>>>>> users
>>>>>> password from API, then grab the MD5 string and the nonce user sent me,
>>>>>> calculate MD5 on base of API password and nonce - and then compare MD5
>>>>>> strings sent by user and calculated?
>>>>>>
>>>>>>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

Thanks for your reply! But the problem is - I need to provide to API user's
login and password. Kamailio doesn't know them. So my idea was to transmit
to API the salt and encrypted password. Would that work? I see it that way
1. User sends register request.
2. Kamailio sends to API salt and ecnr.passwd
3. API recalculates MD5 on its side and compares with encr.passwd
4. Sends OK if it's ok, huh
5. I receive OK from API and send simple 200 OK to user

Do you see any logical mistakes here? Do I need some speacial 200 OK to
approve registration, or simple send_reply("401", "OK"); is enough?


2015-11-13 15:21 GMT+02:00 Sebastian Damm <d...@sipgate.de>:

> Hello,
>
> if your script can return the password for the user to Kamailio, you could
> use the pv_*_authenticate functions. You can pass the password to check
> against to these functions in a pseudo variable.
>
>
> http://www.kamailio.net/docs/modules/4.3.x/modules/auth.html#auth.f.pv_www_authenticate
>
> Best Regards,
> Sebastian
>
> On Fri, Nov 13, 2015 at 2:14 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> UPD: If upper method is possible - I assume I can check if message has
>> Auth header using
>>
>> if (has_credentials("myrealm")) {
>> ...
>> }
>> Can you please specify how to grab it?
>>
>>
>> 2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>
>>> Hello!
>>> My problem is I need to do users authentication through API. So I need
>>> to replace
>>>
>>> if (!www_authenticate("$td", "subscriber")) {
>>> www_challenge("$td", "1");
>>> }
>>>
>>> With
>>>
>>> if (!my_auth_script()) {
>>> www_challenge("$td", "1");
>>> }
>>>
>>> The main problem is - how can I grab or compare users password? I know
>>> nonce, which I understand is MD5 salt. Can I, for example, grab users
>>> password from API, then grab the MD5 string and the nonce user sent me,
>>> calculate MD5 on base of API password and nonce - and then compare MD5
>>> strings sent by user and calculated?
>>>
>>> --
>>> Alexandru Covalschi
>>> ABRISS-Solutions
>>> VoIP engineer and system administrator
>>> phone: +37367398493
>>> web: http://abs-telecom.com/
>>>
>>
>>
>>
>> --
>> Alexandru Covalschi
>> ABRISS-Solutions
>> VoIP engineer and system administrator
>> phone: +37367398493
>> web: http://abs-telecom.com/
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

simple send_reply("200", "OK");, sorry

2015-11-13 16:02 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Thanks for your reply! But the problem is - I need to provide to API
> user's login and password. Kamailio doesn't know them. So my idea was to
> transmit to API the salt and encrypted password. Would that work? I see it
> that way
> 1. User sends register request.
> 2. Kamailio sends to API salt and ecnr.passwd
> 3. API recalculates MD5 on its side and compares with encr.passwd
> 4. Sends OK if it's ok, huh
> 5. I receive OK from API and send simple 200 OK to user
>
> Do you see any logical mistakes here? Do I need some speacial 200 OK to
> approve registration, or simple send_reply("401", "OK"); is enough?
>
>
> 2015-11-13 15:21 GMT+02:00 Sebastian Damm <d...@sipgate.de>:
>
>> Hello,
>>
>> if your script can return the password for the user to Kamailio, you
>> could use the pv_*_authenticate functions. You can pass the password to
>> check against to these functions in a pseudo variable.
>>
>>
>> http://www.kamailio.net/docs/modules/4.3.x/modules/auth.html#auth.f.pv_www_authenticate
>>
>> Best Regards,
>> Sebastian
>>
>> On Fri, Nov 13, 2015 at 2:14 PM, Alexandru Covalschi <568...@gmail.com>
>> wrote:
>>
>>> UPD: If upper method is possible - I assume I can check if message has
>>> Auth header using
>>>
>>> if (has_credentials("myrealm")) {
>>> ...
>>> }
>>> Can you please specify how to grab it?
>>>
>>>
>>> 2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
>>>
>>>> Hello!
>>>> My problem is I need to do users authentication through API. So I need
>>>> to replace
>>>>
>>>> if (!www_authenticate("$td", "subscriber")) {
>>>>www_challenge("$td", "1");
>>>> }
>>>>
>>>> With
>>>>
>>>> if (!my_auth_script()) {
>>>>www_challenge("$td", "1");
>>>> }
>>>>
>>>> The main problem is - how can I grab or compare users password? I know
>>>> nonce, which I understand is MD5 salt. Can I, for example, grab users
>>>> password from API, then grab the MD5 string and the nonce user sent me,
>>>> calculate MD5 on base of API password and nonce - and then compare MD5
>>>> strings sent by user and calculated?
>>>>
>>>> --
>>>> Alexandru Covalschi
>>>> ABRISS-Solutions
>>>> VoIP engineer and system administrator
>>>> phone: +37367398493
>>>> web: http://abs-telecom.com/
>>>>
>>>
>>>
>>>
>>> --
>>> Alexandru Covalschi
>>> ABRISS-Solutions
>>> VoIP engineer and system administrator
>>> phone: +37367398493
>>> web: http://abs-telecom.com/
>>>
>>> ___
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users@lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> ___
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>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
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[SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

Hello!
My problem is I need to do users authentication through API. So I need to
replace

if (!www_authenticate("$td", "subscriber")) {
www_challenge("$td", "1");
}

With

if (!my_auth_script()) {
www_challenge("$td", "1");
}

The main problem is - how can I grab or compare users password? I know
nonce, which I understand is MD5 salt. Can I, for example, grab users
password from API, then grab the MD5 string and the nonce user sent me,
calculate MD5 on base of API password and nonce - and then compare MD5
strings sent by user and calculated?

-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Grab users password from WWW-Auth header

[Alexandru Covalschi]

UPD: If upper method is possible - I assume I can check if message has Auth
header using

if (has_credentials("myrealm")) {
...
}
Can you please specify how to grab it?


2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:

> Hello!
> My problem is I need to do users authentication through API. So I need to
> replace
>
> if (!www_authenticate("$td", "subscriber")) {
>   www_challenge("$td", "1");
> }
>
> With
>
> if (!my_auth_script()) {
>   www_challenge("$td", "1");
> }
>
> The main problem is - how can I grab or compare users password? I know
> nonce, which I understand is MD5 salt. Can I, for example, grab users
> password from API, then grab the MD5 string and the nonce user sent me,
> calculate MD5 on base of API password and nonce - and then compare MD5
> strings sent by user and calculated?
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] Kamailio + Rtpengine < - > Freeswitch + ZRTP

[Alexandru Covalschi]

Hello! Was trying to perfrom subj interconnection via
http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbс
<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
via that guide.
Kamailio and FS are on same host on Amazon EC2. The issue is - when I have
proxy_media=true on FS, ZRTP is OK, but the leg_b user can't hear anything
(but his voice is transmitted OK). When I set proxy-media to false the
voice is OK, but something is with ZRTP so Jitsi can't establish ZRTP
handshake. When I set bypass-media=true on FS both ZRTP and voice
established, but quality is sometimes bad and the delay is too big.

Anyone has any ideas? Kamailio 4.3, FS 1.4
<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>

<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] authorize a client in bulk registration

[Alexandru Covalschi]

If it is OK to trust all clients from PBX - just add a check in route[AUTH]
if traffic comes from PBX and execute return; if TRUE
Else, you may create a list of allowed users in db/config and perform
checks based on $au

2015-08-31 20:49 GMT+03:00 Al S <ali...@outlook.com>:

> My PBX registered in bulk to Kamailio successfully.
>
> PBX --> Register --> kamailio
> PBX <-- 401 with nonce value <-- kamailio
> PBX --> Register with nonce and md5 response values --> kamailio
> BOX <-- 200ok <-- kamailio
>
> However, when one of the PBX client sends an invite out, kamailio AUTH
> module does not authorize this client:
>
> client (813-111-) --> Invite --> kamailio
>
> What would be the right way to authorize a 10-digit client to send a call
> out when a bulk registration and not 10-digit registration is performed.
>
> Your assistance in this matter is greatly appreciated
>
> Thanks,
> Al
>
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>
>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Using PSTN as fallback

[Alexandru Covalschi]

it depends on which PBX you use for media relay and which codes when no
user available does it return.
What I'd suggest is to check if call is coming not from PSTN (if it comes
from PSTN - it's for sure must be routed to PBX) and if TRUE, then first
send call to PBX and if answer is not 180/183 200 etc. (you can catch that
in a specific failure_route) route calls back to PSTN.

2015-08-30 12:04 GMT+03:00 Michael Nielsen mic.nie...@gmail.com:

 I have Kamailio running and connected to a PSTN gateway.

 My subscribers are named ex. +442071234567 - same as their real GSM
 number from my PSTN gateway.

 I'm using the standard kamailio.cfg which ships with version 4.3.

 When I'm trying to dial SIP client to SIP client I would like to have
 Kamailio route the call internally if a subscriber exists with ex.
 +442071234567.
 If no subscriber exists with ex. +442071234567 it should send it to my
 PSTN gateway.

 As it is now it seems as if it are trying to both call internally and
 via the PSTN gateway.

 How should one fix this issue the best way?

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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] unsupported route_type 64

[Alexandru Covalschi]

looks like you've defined wrong variables fro cgrates indlg route jumping.
can you share the cgrates part of your config?

2015-08-29 5:37 GMT+03:00 Admin smont...@twc.com:

 Hi,
  I am running Kamailio (4.4.0-dev2 (x86_64/linux)) with cgrates
 real-time billing application. At the end of a basic call (prepaid to
 postpaid), I receive the following error:

 ERROR:tmx [t_var.c:521]:pv_get_tm_reply_code():unsupported route_type 64
 8(6740) DEBUG:tmx [t_var.c:526]:pv_get_tm_reply_code():reply code is 0

 What does route_type 64 mean? I checked the archive posts (FS#456) and
 seemed to be fixed in branch 4.1 (part of 4.1.5).

 Thanks.


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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] Kamailio TLS configuration

[Alexandru Covalschi]

Hello!

I'm having problems with Kamailio configuration with TLS. Or, maybe, that's
my misunderstanding about how it should work.
So, the issue - inbound TLS works just great, I can call everyone in my
domain. I have PositiveSSL certificate, so I have such files:
calist.crt  AddTrustExternalCARoot.crt + COMODORSAAddTrustCA.crt +
COMODORSADomainValidationSecureServerCA.crt divided by \n
server.key  - key
server.crt - cert
The configuration of tls.cfg

[server:default]
method = SSLv23
verify_certificate = no
require_certificate = no
private_key = /etc/ssl/sectel.io.ssl/sip/server.key
certificate = /etc/ssl/sectel.io.ssl/sip/server.crt
ca_list = /etc/ssl/sectel.io.ssl/sip/calist.crt
#crl = /etc/kamailio/crl.pem
(however with or without ca_list nothing changes)

[client:default]
verify_certificate = yes
require_certificate = yes


And with that configuration when I'm trying to call to ostel.co (public SIP
service supporting TLS) from my server I get such error:
ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL
routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed


Putting that in tls.cfg:
[client:default]
verify_certificate = no
require_certificate = no

Make everything work.
Cross-domain calling is essential and I'm just trying to figure out -
what's the problem? Is that my certificate, is that ostel.co certificate or
it is just the way it should be?

Thanks!

-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Kamailio TLS configuration

[Alexandru Covalschi]

Forgot to add
cat /etc/issue
Debian GNU/Linux 8 \n \l


kamailio -V
version: kamailio 4.3.1 (x86_64/linux)
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,
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: unknown
compiled with gcc 4.9.2

openssl version
OpenSSL 1.0.1k 8 Jan 2015


2015-08-28 20:01 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Hello!

 I'm having problems with Kamailio configuration with TLS. Or, maybe,
 that's my misunderstanding about how it should work.
 So, the issue - inbound TLS works just great, I can call everyone in my
 domain. I have PositiveSSL certificate, so I have such files:
 calist.crt  AddTrustExternalCARoot.crt + COMODORSAAddTrustCA.crt +
 COMODORSADomainValidationSecureServerCA.crt divided by \n
 server.key  - key
 server.crt - cert
 The configuration of tls.cfg

 [server:default]
 method = SSLv23
 verify_certificate = no
 require_certificate = no
 private_key = /etc/ssl/sectel.io.ssl/sip/server.key
 certificate = /etc/ssl/sectel.io.ssl/sip/server.crt
 ca_list = /etc/ssl/sectel.io.ssl/sip/calist.crt
 #crl = /etc/kamailio/crl.pem
 (however with or without ca_list nothing changes)

 [client:default]
 verify_certificate = yes
 require_certificate = yes


 And with that configuration when I'm trying to call to ostel.co (public
 SIP service supporting TLS) from my server I get such error:
 ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL
 routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed


 Putting that in tls.cfg:
 [client:default]
 verify_certificate = no
 require_certificate = no

 Make everything work.
 Cross-domain calling is essential and I'm just trying to figure out -
 what's the problem? Is that my certificate, is that ostel.co certificate
 or it is just the way it should be?

 Thanks!

 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Kamailio TLS configuration

[Alexandru Covalschi]

And server is under Amazon EC2, but that shouldn't really make any sense

2015-08-29 0:11 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Forgot to add
 cat /etc/issue
 Debian GNU/Linux 8 \n \l


 kamailio -V
 version: kamailio 4.3.1 (x86_64/linux)
 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,
 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: unknown
 compiled with gcc 4.9.2

 openssl version
 OpenSSL 1.0.1k 8 Jan 2015


 2015-08-28 20:01 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Hello!

 I'm having problems with Kamailio configuration with TLS. Or, maybe,
 that's my misunderstanding about how it should work.
 So, the issue - inbound TLS works just great, I can call everyone in my
 domain. I have PositiveSSL certificate, so I have such files:
 calist.crt  AddTrustExternalCARoot.crt + COMODORSAAddTrustCA.crt +
 COMODORSADomainValidationSecureServerCA.crt divided by \n
 server.key  - key
 server.crt - cert
 The configuration of tls.cfg

 [server:default]
 method = SSLv23
 verify_certificate = no
 require_certificate = no
 private_key = /etc/ssl/sectel.io.ssl/sip/server.key
 certificate = /etc/ssl/sectel.io.ssl/sip/server.crt
 ca_list = /etc/ssl/sectel.io.ssl/sip/calist.crt
 #crl = /etc/kamailio/crl.pem
 (however with or without ca_list nothing changes)

 [client:default]
 verify_certificate = yes
 require_certificate = yes


 And with that configuration when I'm trying to call to ostel.co (public
 SIP service supporting TLS) from my server I get such error:
 ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL
 routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed


 Putting that in tls.cfg:
 [client:default]
 verify_certificate = no
 require_certificate = no

 Make everything work.
 Cross-domain calling is essential and I'm just trying to figure out -
 what's the problem? Is that my certificate, is that ostel.co certificate
 or it is just the way it should be?

 Thanks!

 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Help with sip balancer

[Alexandru Covalschi]

First of all I'd suggest to use
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
guide in combination with
http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html
But, assuming your platform is behind NAT, you need:
1st. Use rtpengine instead of rtpproxy. You can read about how to advertise
your external public adress on rtpengine git page.
2nd. In Kamailio configuration when you define listen, you should use
listen - advertise construction (
http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen).
3d. Be sure to leave secret column empty on asterisk database, otherwise
all users registered on asterisks won't have OK status, what can cause
problems with queues etc.

2015-08-12 0:19 GMT+03:00 Bruno d4rks...@gmail.com:


 Hello,
 i'm on my first try with kamailio. I need to build a SIP balancer that
 should keep SIP
 registration from VoIP provider and route the calls to the asterisk boxes
 where an IVR
 will take care to answer.

 Here's my network topology:

   +--- [asterisk1]
 [public_ip]   |10.50.10.131
  [router]  ---NAT--- [kamailio] ---+
 10.50.10.110.50.10.120|
   +--- [asterisk2]
10.50.10.132

 In my setup i planned to use UAC and DISPATCHER modules. I started from
 the
 kamailio-basic.cfg and added some extra lines to handle UAC and
 DISPATCHER.

 All is working fine when i do a test call from a softphone inside network
 10.50.10.0/24.

 When a call is coming from the sip carrier, troubles occurs because
 asterisk boxes
 are sending their internal ip in SDP.

 I understand that i need to rewrite SDP in that case, but i actually don't
 know how/where.

 I've attached kamailio configuration and a sip trace taken with sngrep
 where the problem
 is visible.

 For security reasons, i would like to force the RTP through RTPProxy.

 I'm missing something, and need your help me to understand my errors.

 Best Regards,
 Bruno



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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Help with sip balancer

[Alexandru Covalschi]

Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/

2015-08-12 0:41 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 First of all I'd suggest to use
 http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
 guide in combination with
 http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html
 But, assuming your platform is behind NAT, you need:
 1st. Use rtpengine instead of rtpproxy. You can read about how to
 advertise your external public adress on rtpengine git page.
 2nd. In Kamailio configuration when you define listen, you should use
 listen - advertise construction (
 http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen).
 3d. Be sure to leave secret column empty on asterisk database, otherwise
 all users registered on asterisks won't have OK status, what can cause
 problems with queues etc.

 2015-08-12 0:19 GMT+03:00 Bruno d4rks...@gmail.com:


 Hello,
 i'm on my first try with kamailio. I need to build a SIP balancer that
 should keep SIP
 registration from VoIP provider and route the calls to the asterisk boxes
 where an IVR
 will take care to answer.

 Here's my network topology:

   +--- [asterisk1]
 [public_ip]   |10.50.10.131
  [router]  ---NAT--- [kamailio] ---+
 10.50.10.110.50.10.120|
   +--- [asterisk2]
10.50.10.132

 In my setup i planned to use UAC and DISPATCHER modules. I started from
 the
 kamailio-basic.cfg and added some extra lines to handle UAC and
 DISPATCHER.

 All is working fine when i do a test call from a softphone inside network
 10.50.10.0/24.

 When a call is coming from the sip carrier, troubles occurs because
 asterisk boxes
 are sending their internal ip in SDP.

 I understand that i need to rewrite SDP in that case, but i actually
 don't know how/where.

 I've attached kamailio configuration and a sip trace taken with sngrep
 where the problem
 is visible.

 For security reasons, i would like to force the RTP through RTPProxy.

 I'm missing something, and need your help me to understand my errors.

 Best Regards,
 Bruno



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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Debian/jessie and 4.3.1 default config doesn't work

[Alexandru Covalschi]

Hmf... I saw the advice to put them on /tmp/ somewhere on mailing lists and
had same thoughts. Thanks, will fix that on my servers!

2015-08-10 14:16 GMT+03:00 Daniel Tryba d.tr...@pocos.nl:

 On Monday 10 August 2015 13:12:12 Alexandru Covalschi wrote:
  Shouldn't they be /tmp/kamailio_fifo and /tmp/kamailio_ctl in 4.3.x?

 Putting these files in /tmp makes it possible for mortal users to prevent a
 service from running by simply creating a file/dir (unless startup scripts
 forcefully remove them first and then there still is a possible race
 condition
 in actually creating the file/dir).


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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Debian/jessie and 4.3.1 default config doesn't work

[Alexandru Covalschi]

Shouldn't they be /tmp/kamailio_fifo and /tmp/kamailio_ctl in 4.3.x?

2015-08-10 13:10 GMT+03:00 Daniel Tryba d.tr...@pocos.nl:

 It looks like the compile defaults of the ctl and mi_fifo module and the
 default kamailio.conf/kamctlrc conflict.

 kamailio tries to open:

 /tmp/buildd/kamailio-4.3.1+jessie/debian/kamailio/var/run/kamailio//kamailio_ctl

 /tmp/buildd/kamailio-4.3.1+jessie/debian/kamailio/var/run/kamailio/kamailio_fifo

 and will fail to run, unless explicitly set to conf paths are set to
 /var/run/kamailio/kamailio_(ctl|fifo)

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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] LCR Routing - Cost based routing

[Alexandru Covalschi]

DanB, well, I never used it - can you please describe how does it work? :)
I mean, the logic in short

2015-08-10 13:15 GMT+03:00 DanB danb.li...@gmail.com:

 Guys,

 Since I saw the thread growing, I am around if you got questions on
 CGRateS related LCR, fully compatible with Kamailio via evapi module.

 DanB


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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] LCR Routing - Cost based routing

[Alexandru Covalschi]

try using CGRateS

2015-08-09 12:06 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 Hi

  is there any way to achive following usecase using carrier route /
 LCR / droude module

 i have 10 different termination gateways, each gateways will have more
 than 1 destinations and each gateways will have different rate, for
 example

 91 Destination ( there will be 1 entries like 91 )
 gw1gw2   gw3   gw4gw5
 0.001 0.007 0.003 0.002 0.003

 now need to logic to load the these destination with cost and sort
 gateways in order of longest match of dialled number,cost, priority for the
 dialled destination, which will not use too much CPU and memory

 Regards
 Arun

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




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] LCR Routing - Cost based routing

[Alexandru Covalschi]

Or, well, see that guide
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have
priority and weight on LCR module

2015-08-09 12:37 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 try using CGRateS

 2015-08-09 12:06 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 Hi

  is there any way to achive following usecase using carrier route /
 LCR / droude module

 i have 10 different termination gateways, each gateways will have more
 than 1 destinations and each gateways will have different rate, for
 example

 91 Destination ( there will be 1 entries like 91 )
 gw1gw2   gw3   gw4gw5
 0.001 0.007 0.003 0.002 0.003

 now need to logic to load the these destination with cost and sort
 gateways in order of longest match of dialled number,cost, priority for the
 dialled destination, which will not use too much CPU and memory

 Regards
 Arun

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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] LCR Routing - Cost based routing

[Alexandru Covalschi]

however you can try building LCR based on prefix and weight, why not?

2015-08-09 18:52 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I know CGRateS allow cost-based LCR for Kamailio. Maybe there are some
 internal Kamailio modules, but I don't know about them

 2015-08-09 17:10 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 but above guide is only prefix priority and weight based   , but we
 should involve rate as well.
  ,

 On Sun, Aug 9, 2015 at 3:09 PM, Alexandru Covalschi 568...@gmail.com
 wrote:

 Or, well, see that guide
 http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have
 priority and weight on LCR module

 2015-08-09 12:37 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 try using CGRateS

 2015-08-09 12:06 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 Hi

  is there any way to achive following usecase using carrier route
 / LCR / droude module

 i have 10 different termination gateways, each gateways will have more
 than 1 destinations and each gateways will have different rate, for
 example

 91 Destination ( there will be 1 entries like 91 )
 gw1gw2   gw3   gw4gw5
 0.001 0.007 0.003 0.002 0.003

 now need to logic to load the these destination with cost and sort
 gateways in order of longest match of dialled number,cost, priority for 
 the
 dialled destination, which will not use too much CPU and memory

 Regards
 Arun

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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/

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



 ___
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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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Re: [SR-Users] LCR Routing - Cost based routing

[Alexandru Covalschi]

I know CGRateS allow cost-based LCR for Kamailio. Maybe there are some
internal Kamailio modules, but I don't know about them

2015-08-09 17:10 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 but above guide is only prefix priority and weight based   , but we should
 involve rate as well.
  ,

 On Sun, Aug 9, 2015 at 3:09 PM, Alexandru Covalschi 568...@gmail.com
 wrote:

 Or, well, see that guide
 http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have
 priority and weight on LCR module

 2015-08-09 12:37 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 try using CGRateS

 2015-08-09 12:06 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 Hi

  is there any way to achive following usecase using carrier route /
 LCR / droude module

 i have 10 different termination gateways, each gateways will have more
 than 1 destinations and each gateways will have different rate, for
 example

 91 Destination ( there will be 1 entries like 91 )
 gw1gw2   gw3   gw4gw5
 0.001 0.007 0.003 0.002 0.003

 now need to logic to load the these destination with cost and sort
 gateways in order of longest match of dialled number,cost, priority for the
 dialled destination, which will not use too much CPU and memory

 Regards
 Arun

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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/

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



 ___
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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
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Re: [SR-Users] LCR Routing - Cost based routing

[Alexandru Covalschi]

you already have cost, just write a script to transform it into weight, I
don't see a hard task here. It can be easily automated

2015-08-09 21:15 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 hi Alexandru

so we have manually  set prefix , weight and priority
 depanding upon rates  ?  is there any avaliable way to automate or we have
 to rewrite/modify  the lcr/drouting module for rate selection  ,



 On Sun, Aug 9, 2015 at 9:22 PM, Alexandru Covalschi 568...@gmail.com
 wrote:

 however you can try building LCR based on prefix and weight, why not?

 2015-08-09 18:52 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I know CGRateS allow cost-based LCR for Kamailio. Maybe there are some
 internal Kamailio modules, but I don't know about them

 2015-08-09 17:10 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 but above guide is only prefix priority and weight based   , but we
 should involve rate as well.
  ,

 On Sun, Aug 9, 2015 at 3:09 PM, Alexandru Covalschi 568...@gmail.com
 wrote:

 Or, well, see that guide
 http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we
 have priority and weight on LCR module

 2015-08-09 12:37 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 try using CGRateS

 2015-08-09 12:06 GMT+03:00 Arun Kumar mi2a...@gmail.com:

 Hi

  is there any way to achive following usecase using carrier
 route / LCR / droude module

 i have 10 different termination gateways, each gateways will have
 more than 1 destinations and each gateways will have different rate,
 for example

 91 Destination ( there will be 1 entries like 91 )
 gw1gw2   gw3   gw4gw5
 0.001 0.007 0.003 0.002 0.003

 now need to logic to load the these destination with cost and sort
 gateways in order of longest match of dialled number,cost, priority for 
 the
 dialled destination, which will not use too much CPU and memory

 Regards
 Arun

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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/

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



 ___
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 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/

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



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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Rtpengine/rtpproxy and zrtp

[Alexandru Covalschi]

thanks!

2015-08-06 22:29 GMT+03:00 Frank Carmickle fr...@carmickle.com:

 Zrtp passes through rtpengine just fine.

 --FC
 Sent from my 6 plus

 On Aug 6, 2015, at 14:12, Alexandru Covalschi 568...@gmail.com wrote:

 Sorry if writing to wrong mailing list, I am very limited to traffic now
 amd don't know if there is any for rtpproxy/rtpengine.
 My question is - can they support ZRTP at least in pass-through mode? Will
 rtpengine fail on trying to recognize unknown SDP fields?

 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
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[SR-Users] Rtpengine/rtpproxy and zrtp

[Alexandru Covalschi]

Sorry if writing to wrong mailing list, I am very limited to traffic now
amd don't know if there is any for rtpproxy/rtpengine.
My question is - can they support ZRTP at least in pass-through mode? Will
rtpengine fail on trying to recognize unknown SDP fields?
___
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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

I got bridging working well on internal interfaces in case of simple SIP
calls on a bit other configuration. But editing this config to support
WebRTC causes same problems. I need internal interfaces on asterisk to
completely close external ones (Security etc.).
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Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Also, an interesting thing - if you can see in Kamailio log, a check of the
proto of user 300 is being made. But 300 is $tU, and $tU proto is being
checked only if source IP is asterisks IP.

Here's the part of config where rtpengine is engaged (in NATmanage route)

if((src_ip==10.0.0.87))
{
xlog(L_NOTICE,== select proto from sipusers where
name=$tU);
sql_xquery(ca_asterisk, select proto from sipusers where
name=$tU, ra);
 xlog(L_NOTICE,= $tU has proto $xavp(ra=proto));
if ($xavp(ra=proto)==ws)
{
 xlog(L_NOTICE,= $tU has WEBSOCKETS);

rtpengine_manage(trust-address replace-origin
replace-session-connection ICE=force RTP/SAVPF);
}
else
{
xlog(L_NOTICE,= $tU has NO fucken WEBSOCKETS);
rtpengine_manage(trust-address replace-origin
replace-session-connection);
}
} else {
xlog(L_NOTICE,== select proto from sipusers where
name=$fU);
   sql_xquery(ca_asterisk, select proto from sipusers where
name=$fU, ra);
  if ($xavp(ra=proto)==ws)
{

xlog(L_NOTICE,= $fU has WEBSOCKETS);
rtpengine_manage(trust-address replace-origin
replace-session-connection ICE=force RTP/AVP);
}
else
{
xlog(L_NOTICE,= $fU has NO WEBSOCKETS);
rtpengine_manage(replace-origin
replace-session-connection RTP/AVP);
}

}


2015-06-24 16:14 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Heh...
 Well, I still have troubles with my configuration. And in SDP media adress
 is Amazon public interface - but rtpengine has replace-origin
 replace-session-connection session, so it must be local address.
 Any ideas?
 Asterisk log http://pastebin.com/MFt9V9qK
 Kamailio log http://pastebin.com/jZceP2Rn
 Javascript log http://pastebin.com/4ZLePyKz


 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Well.. Guys, sorry, it was totally my fault. I just used VPN.

 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I used https://github.com/caruizdiaz/kamailio-ws configuration that
 100% works on other then Amazon EC2 environment and I still get this error.
 Maybe it is somehow related to NAT traversal?

 Kamailio log: http://pastebin.com/jZceP2Rn
 javascript log: http://pastebin.com/9Y4Pv43W


 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com
 :

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio 
 can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists.
 What kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla 
 mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA 
 tries to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not 
 create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Asterisk localip=10.0.0.87, sorry

2015-06-24 16:24 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Ok, so my scheme.
 Kamailio and Asterisk are in Amazon EC2
 Kamailio externip=54.197.230.121 localip=10.145.45.103
 Asterisk localip=10.145.45.103, externip doesn't matter

 Call should flow like that:
 webrtc -- kamailio-externip -- kamailio-localip -- asterisk-localip
 but now it's webrtc -- kamailio-externip -- kamailio--localip --
 asterisk-localip -- kamailio-externip -- peer

 I have the voice, but it's wrong scheme, and Asterisk drops call because
 of retransmissions failure


 2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Can you specify exactly which side received what IP and what you would
 expect there? It is not easy to digests lots of logs and also guess what
 would you expect to happen...

 Cheers,
 Daniel


 On 24/06/15 15:14, Alexandru Covalschi wrote:

  Heh...
  Well, I still have troubles with my configuration. And in SDP media
 adress is Amazon public interface - but rtpengine has replace-origin
 replace-session-connection session, so it must be local address.
  Any ideas?
  Asterisk log http://pastebin.com/MFt9V9qK
  Kamailio log http://pastebin.com/jZceP2Rn
  Javascript log http://pastebin.com/4ZLePyKz


 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Well.. Guys, sorry, it was totally my fault. I just used VPN.

 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

  I used https://github.com/caruizdiaz/kamailio-ws configuration that
 100% works on other then Amazon EC2 environment and I still get this error.
 Maybe it is somehow related to NAT traversal?

  Kamailio log: http://pastebin.com/jZceP2Rn
  javascript log: http://pastebin.com/9Y4Pv43W


 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com
 :

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio 
 can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists.
 What kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla 
 mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA 
 tries to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in
 the signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING:
 core [msg_translator.c:2778]: via_builder(): TCP/TLS connection 
 (id: 0)
 for WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not 
 create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to
 Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


  ___
 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

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Ok, so my scheme.
Kamailio and Asterisk are in Amazon EC2
Kamailio externip=54.197.230.121 localip=10.145.45.103
Asterisk localip=10.145.45.103, externip doesn't matter

Call should flow like that:
webrtc -- kamailio-externip -- kamailio-localip -- asterisk-localip
but now it's webrtc -- kamailio-externip -- kamailio--localip --
asterisk-localip -- kamailio-externip -- peer

I have the voice, but it's wrong scheme, and Asterisk drops call because of
retransmissions failure


2015-06-24 16:18 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Can you specify exactly which side received what IP and what you would
 expect there? It is not easy to digests lots of logs and also guess what
 would you expect to happen...

 Cheers,
 Daniel


 On 24/06/15 15:14, Alexandru Covalschi wrote:

  Heh...
  Well, I still have troubles with my configuration. And in SDP media
 adress is Amazon public interface - but rtpengine has replace-origin
 replace-session-connection session, so it must be local address.
  Any ideas?
  Asterisk log http://pastebin.com/MFt9V9qK
  Kamailio log http://pastebin.com/jZceP2Rn
  Javascript log http://pastebin.com/4ZLePyKz


 2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Well.. Guys, sorry, it was totally my fault. I just used VPN.

 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

  I used https://github.com/caruizdiaz/kamailio-ws configuration that
 100% works on other then Amazon EC2 environment and I still get this error.
 Maybe it is somehow related to NAT traversal?

  Kamailio log: http://pastebin.com/jZceP2Rn
  javascript log: http://pastebin.com/9Y4Pv43W


 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com
 :

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio 
 can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists.
 What kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla 
 mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA 
 tries to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not 
 create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to
 Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


  ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Heh...
Well, I still have troubles with my configuration. And in SDP media adress
is Amazon public interface - but rtpengine has replace-origin
replace-session-connection session, so it must be local address.
Any ideas?
Asterisk log http://pastebin.com/MFt9V9qK
Kamailio log http://pastebin.com/jZceP2Rn
Javascript log http://pastebin.com/4ZLePyKz


2015-06-24 1:27 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Well.. Guys, sorry, it was totally my fault. I just used VPN.

 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
 works on other then Amazon EC2 environment and I still get this error.
 Maybe it is somehow related to NAT traversal?

 Kamailio log: http://pastebin.com/jZceP2Rn
 javascript log: http://pastebin.com/9Y4Pv43W


 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists. What
 kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla 
 mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA tries 
 to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create 
 Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to
 Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


  ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

without fix_nated_contact error behaviour is the same
maybe I should upgrade to 4.3 ?

2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists. What
 kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive audio
 from asterisk, but I can't transmit audio there - my SIP UA tries to send
 data to Kamailio-s local EC2 IP.


 you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Here is it
http://pastebin.com/JkkM4M5m

2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists. What
 kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive audio
 from asterisk, but I can't transmit audio there - my SIP UA tries to send
 data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to
 Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


  ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
works on other then Amazon EC2 environment and I still get this error.
Maybe it is somehow related to NAT traversal?

Kamailio log: http://pastebin.com/jZceP2Rn
javascript log: http://pastebin.com/9Y4Pv43W


2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists. What
 kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com
 :

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive audio
 from asterisk, but I can't transmit audio there - my SIP UA tries to send
 data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create 
 Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to
 Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


  ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

I solved the SIP voice trouble, but WebRTC problem still exists. What kind
of trace I must do to make my post more informative?

2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive audio
 from asterisk, but I can't transmit audio there - my SIP UA tries to send
 data to Kamailio-s local EC2 IP.


 you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Here's the trace on port which I use for ws server. Don't look at
fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
establish a ws connection properly. Client is SIPML5 demo phone
http://pastebin.com/LvAk2HkP

2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists. What kind
 of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive audio
 from asterisk, but I can't transmit audio there - my SIP UA tries to send
 data to Kamailio-s local EC2 IP.


 you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Well.. Guys, sorry, it was totally my fault. I just used VPN.

2015-06-24 0:59 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
 works on other then Amazon EC2 environment and I still get this error.
 Maybe it is somehow related to NAT traversal?

 Kamailio log: http://pastebin.com/jZceP2Rn
 javascript log: http://pastebin.com/9Y4Pv43W


 2015-06-23 20:40 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here is it
 http://pastebin.com/JkkM4M5m

 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:

  There are no major changes in 4.3 comparing with 4.2 in regards to
 websocket -- the implementation is quite mature for a long time.

 Looks like websocket connection is not available. Can you look at
 javascript debug console in the browser to see what is printing?

 Daniel


 On 23/06/15 17:23, Alexandru Covalschi wrote:

  without fix_nated_contact error behaviour is the same
  maybe I should upgrade to 4.3 ?

 2015-06-23 14:08 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Here's the trace on port which I use for ws server. Don't look at
 fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
 establish a ws connection properly. Client is SIPML5 demo phone
 http://pastebin.com/LvAk2HkP

 2015-06-23 14:03 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 I solved the SIP voice trouble, but WebRTC problem still exists. What
 kind of trace I must do to make my post more informative?

 2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com
 :

  Hello,

 On 23/06/15 04:10, Alexandru Covalschi wrote:

  Hello. I'm trying to set up this (v 4.2 stable):
  peer -- ec2 --kamailio+rtpengine-- asterisk
  scheme

  I use advertised adress for SIP and WS connections.
  The problem is that on SIP I get one way audio - I can receive
 audio from asterisk, but I can't transmit audio there - my SIP UA tries 
 to
 send data to Kamailio-s local EC2 IP.


  you should grab a ngrep trace on server to see what happens in the
 signaling in order to be able to provide some hints on solving it.

 Cheers,
 Daniel

In case of WebRTC I get lot's of erros:

 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
 [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
 WebSocket could not be found
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create 
 Via
 header
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
 [forward.c:584]: forward_request(): building failed
 Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
 [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
 terribly sorry, server error occurred (1/SL)

  The call reaches Asterisk, but not vice-versa. No media is being
 transferred.

  Rtpengine flags I use:
  For SIP:  rtpengine_manage(trust-adress replace-origin
 replace-session-connection RTP/AVP);
  For WS:  rtpengine_manage(trust-address replace-origin
 replace-session-connection ICE=force RTP/AVP);

  Do you have any ideas how ti fix that? I also make REGFWD's to
 Asterisk
  --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


  ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




 --
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


 ___
 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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




 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer

[SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

[Alexandru Covalschi]

Hello. I'm trying to set up this (v 4.2 stable):
peer -- ec2 --kamailio+rtpengine-- asterisk
scheme

I use advertised adress for SIP and WS connections.
The problem is that on SIP I get one way audio - I can receive audio from
asterisk, but I can't transmit audio there - my SIP UA tries to send data
to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros:

Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: core
[msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for
WebSocket could not be found
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
[msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via
header
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: core
[forward.c:584]: forward_request(): building failed
Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl
[sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm
terribly sorry, server error occurred (1/SL)

The call reaches Asterisk, but not vice-versa. No media is being
transferred.

Rtpengine flags I use:
For SIP:  rtpengine_manage(trust-adress replace-origin
replace-session-connection RTP/AVP);
For WS:  rtpengine_manage(trust-address replace-origin
replace-session-connection ICE=force RTP/AVP);

Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk
-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] SIP-over-Websocket Load Balancing

[Alexandru Covalschi]

thanks, will try that

2015-06-15 14:07 GMT+03:00 Juha Heinanen j...@tutpro.com:

 Alexandru Covalschi writes:

   sorry, i thought you use registrar/usrloc modules
  Well, I do use them - so if you could explain in which table does
 Kamailio
  write the user's proto and which flags I can use - I'll make a test to
 see
  which scheme is preferable :)

 before calling save() function, set one of the branch flags if
 registering ua uses ws protocol.  save() then causes branch flags to be
 stored in location table cflags field.  when you do lookup(), branch
 flags are then restored from that field.

 -- juha

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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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[SR-Users] Kamailio license usage

[Alexandru Covalschi]

Maybe it may be an offtopic, but I'm not really into legal issues - so I'm
sorry if this message is not fully related to this mailing list.

Can I use Kamailio to provide VoIP backend for kind of CRM system in case
of SaaS?

---
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] Kamailio license usage

[Alexandru Covalschi]

Thank you for clear responses guys!
15 июня 2015 г. 20:42 пользователь Daniel-Constantin Mierla 
mico...@gmail.com написал:

  Yes, you can use kamailio to provide VoIP backend -- all free of charge
 and without constraints.

 As Fred pointed in another email, kamailio is GPLv2 and the main
 restriction of that license is to distribute the sources to anyone that
 gets the binaries from you. If you install kamailio on your servers, then
 you do not distribute it and it is no need to give away the source code.
 That means, even if you have a module that you develop and you use it in
 your servers, then you don't need to give away its sources. If you simply
 use stock Kamailio, then the sources are already available for the public.

 That is rather basic description, if you are not sure about what you are
 doing with Kamailio, then as Fred suggested, it is better to consult a
 lawyer.

 Cheers,
 Daniel

 On 15/06/15 18:09, Alexandru Covalschi wrote:

  Maybe it may be an offtopic, but I'm not really into legal issues - so
 I'm sorry if this message is not fully related to this mailing list.

  Can I use Kamailio to provide VoIP backend for kind of CRM system in case
 of SaaS?

 ---
  Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/


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 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
 Book: SIP Routing With Kamailio - http://www.asipto.com


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Re: [SR-Users] SIP-over-Websocket Load Balancing

[Alexandru Covalschi]

 sorry, i thought you use registrar/usrloc modules
Well, I do use them - so if you could explain in which table does Kamailio
write the user's proto and which flags I can use - I'll make a test to see
which scheme is preferable :)

So, about script:

1.) Write to redis
Please read http://kamailio.org/docs/modules/4.3.x/modules/ndb_redis.html
this guide to understand how to connect redis to Kamailio
It route[AUTH] you shall add write to redis command:

if (is_method(REGISTER) || from_uri==myself)
{
# authenticate requests

redis_cmd(protobase, SET $fU $proto bar, r); #
Here is the redis

 if (!auth_check($fd, subscriber, 1)) {
auth_challenge($fd, 0);
exit;
}
# user authenticated - remove auth header
if(!is_method(REGISTER|PUBLISH))
consume_credentials();
}

You can find information about pseudo-variables on this
http://www.kamailio.org/wiki/cookbooks/4.0.x/pseudovariables page

2. Rtpengine algorithm
First of all, look through https://github.com/sipwise/rtpengine and
http://kamailio.org/docs/modules/4.3.x/modules/rtpengine.html to understand
what's the difference between rtpengine and rtpproxy
In your NATMANAGE route change rtpproxy_manage(); or rtpengine_manage();
string to this:

if(ds_is_from_list())
{
xlog(L_NOTICE,== selecting $tU proto\n);
redis_cmd(protobase, GET $tU, uproto);
 xlog(L_NOTICE,= $tU has proto $redis(uproto=value)\n);
if ($redis(uproto=value)==ws)
{
 xlog(L_NOTICE,= $tU is a websocket user\n);
rtpengine_manage(direction=internal direction=external force
trust-address replace-origin replace-session-connection ICE=force
RTP/SAVPF);
}
else
{
xlog(L_NOTICE,= $tU is classy user\n);
rtpengine_manage(direction=internal direction=external force
trust-address replace-origin replace-session-connection);
}
} else {
xlog(L_NOTICE,== $fU proto is $proto );
  if ($proto==ws)
{
xlog(L_NOTICE,= $fU is websocket user\n);
rtpengine_manage(direction=external direction=internal force
trust-address replace-origin replace-session-connection ICE=force RTP/AVP);
}
else
{
xlog(L_NOTICE,= $fU is a classy user);
rtpengine_manage(direction=external direction=internal
replace-origin replace-session-connection force trust-address RTP/AVP);
}

}


2015-06-14 22:24 GMT+03:00 Juha Heinanen j...@tutpro.com:

 Alexandru Covalschi writes:

   you don't need a database for that.  you can use location table flags
  Can you please describe how to do that? I chosen redis because I need to
  figure out the proto of the leg_b (called) user pretty fast - mysql is
 much
  slower.

 sorry, i thought you use registrar/usrloc modules.

 -- juha

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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] SIP-over-Websocket Load Balancing

[Alexandru Covalschi]

Hi, sorry, my previus answer wasn't clear enough - was writing it in a very
sleepy mood :)

No, kamailio acts as a full proxy server for websocket and SIP. P2P is for
caruzdias's configuration from github.
You can try following this
http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket guide
for editing your current configuration file to support WebRTC. But as I
said, you can face some problems with NAT-traversal, so you may to create
different routes for ws and simple SIP.
Also, if you use Asterisk - make sure your version doesn't have some
problems with understanding SRTP handshake (RTP/SAVPF) - be sure that you
have last stable version of your branch (my colleague spent 3 days to
figure out that there was a bug in his version). However, even after update
we couldn't perform a transparent proxy for SRTP, so I used rtpengine with
such scheme:

1. On each registration user's proto is stored in redis database
2. When rtpengine is being called, Kamailio checks user's proto
a) If user is WS and is incoming call, dispatch him to media relay with
RTP/AVP flag
b) If user is WS and is outgoing call (from media relay) send it to the
endpoint with RTP/SAVPF flag
c) If user is SIP and is incoming call, dispatch it to media-relay with
RTP/AVP flag (some SIP clients also have SRTP turned on by default)
d) If user is SIP and is outgoing call, send it to endpoint without any
RTP flags (most sipphones ca recognize which traffic is incomig)

This configurations works well both with Asterisk and Freeswitch, but
Freeswitch in my practice can provide more concurent calls for lesser cost.

2015-06-13 22:24 GMT+03:00 Murugan Pandian manpower13@gmail.com:

 HI Alexandru,

 i try to connect like this


 !--Freeswitch(IVR,Callcenter,dialplan,sip auth)

  Browser(chrome,firefox,opera)--(WS)---Kamailio---!


 !--Freeswitch(IVR,Callcenter,dialplan,sip auth)

 i understand Kamailio only handling signalling(using websocket) but
 stream goes to peer-to-peer ,But i need to play ivr and handle callcenter
 (freeswitch)

so here i try to kamailiio act proxy server

   Any idea how i can achieve thid





 On Sun, Jun 14, 2015 at 12:24 AM, Alexandru Covalschi 568...@gmail.com
 wrote:

 Well, I performed that by creating a media relay consisting of 2
 freeswitches and using rtpengine.

 You just need to handle WebRTC by kamailio using kamailio websocket
 module:
 http://kamailio.org/docs/modules/4.3.x/modules/websocket.html
 caruzdias-es configuration helped me a lot in understanding how
 websockets work on Kamailio:
 https://github.com/caruizdiaz/kamailio-ws
 But be aware, this configuration is for peer2peer connections, not for
 dispatching!

 Kamailio will send simple SIP packets to the media relay then.

 Also I used different NAT-traversal mechanism for sip and ws traffic
 (different routes based on client's transport protocol).
 Also you'll maybe need to have different rtpengine flags for sip and ws -
 remember that WebRTC MUST have SRTP, but I had some issues in transfering
 the SRTP handshake in sipphone--kamailio--freeswitch scheme, so on
 webrtc connection my incoming rtpengine had RTP/AVP flag, and on outgoing
 webrtc it MUST have RTP/SAVP
 For usual SIP calls I also conveted everything to RTP/AVP.

 So you'll need to know to which type of user - ws or tcp/udp you're
 calling to understand which type of RTP to send them.

 2015-06-13 19:07 GMT+03:00 Murugan Pandian manpower13@gmail.com:

 it's posible dispatching websocket request?

 I am try to connect browser(WebRTC) to sip-phone and vice versa,How i
 can achieve more concurrent call(more then 1000 call)

 On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov 
 abalas...@evaristesys.com wrote:

 That question is difficult to answer without some elaboration on your
 part as to what you want to achieve.

  --
 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/

 Sent from my BlackBerry.
   *From: *Murugan Pandian
 *Sent: *Saturday, June 13, 2015 09:47
 *To: *sr-users@lists.sip-router.org
 *Reply To: *Kamailio (SER) - Users Mailing List
 *Subject: *[SR-Users] SIP-over-Websocket Load Balancing

 HI,

   how to handle sip-over-websocket load balancing (WebRTC)


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 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com

Re: [SR-Users] Websocket Load Balancing

[Alexandru Covalschi]

Please take a look at this
http://lists.sip-router.org/pipermail/sr-users/2015-June/088669.html thread

2015-06-13 13:32 GMT+03:00 W5RTC murugan.pand...@w5rtc.com:

 HI,

how to handle  SIP-OVER-Websocket loab balancing (WebRTC)

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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] SIP-over-Websocket Load Balancing

[Alexandru Covalschi]

Well, I performed that by creating a media relay consisting of 2
freeswitches and using rtpengine.

You just need to handle WebRTC by kamailio using kamailio websocket module:
http://kamailio.org/docs/modules/4.3.x/modules/websocket.html
caruzdias-es configuration helped me a lot in understanding how websockets
work on Kamailio:
https://github.com/caruizdiaz/kamailio-ws
But be aware, this configuration is for peer2peer connections, not for
dispatching!

Kamailio will send simple SIP packets to the media relay then.

Also I used different NAT-traversal mechanism for sip and ws traffic
(different routes based on client's transport protocol).
Also you'll maybe need to have different rtpengine flags for sip and ws -
remember that WebRTC MUST have SRTP, but I had some issues in transfering
the SRTP handshake in sipphone--kamailio--freeswitch scheme, so on
webrtc connection my incoming rtpengine had RTP/AVP flag, and on outgoing
webrtc it MUST have RTP/SAVP
For usual SIP calls I also conveted everything to RTP/AVP.

So you'll need to know to which type of user - ws or tcp/udp you're calling
to understand which type of RTP to send them.

2015-06-13 19:07 GMT+03:00 Murugan Pandian manpower13@gmail.com:

 it's posible dispatching websocket request?

 I am try to connect browser(WebRTC) to sip-phone and vice versa,How i can
 achieve more concurrent call(more then 1000 call)

 On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov abalas...@evaristesys.com
 wrote:

 That question is difficult to answer without some elaboration on your
 part as to what you want to achieve.

  --
 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/

 Sent from my BlackBerry.
   *From: *Murugan Pandian
 *Sent: *Saturday, June 13, 2015 09:47
 *To: *sr-users@lists.sip-router.org
 *Reply To: *Kamailio (SER) - Users Mailing List
 *Subject: *[SR-Users] SIP-over-Websocket Load Balancing

 HI,

   how to handle sip-over-websocket load balancing (WebRTC)


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-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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Re: [SR-Users] SIP-over-Websocket Load Balancing

[Alexandru Covalschi]

Sorry, a mistake: on outgoing webrtc it MUST have RTP/SAVP or RTP/SAVPF


2015-06-13 21:54 GMT+03:00 Alexandru Covalschi 568...@gmail.com:

 Well, I performed that by creating a media relay consisting of 2
 freeswitches and using rtpengine.

 You just need to handle WebRTC by kamailio using kamailio websocket module:
 http://kamailio.org/docs/modules/4.3.x/modules/websocket.html
 caruzdias-es configuration helped me a lot in understanding how websockets
 work on Kamailio:
 https://github.com/caruizdiaz/kamailio-ws
 But be aware, this configuration is for peer2peer connections, not for
 dispatching!

 Kamailio will send simple SIP packets to the media relay then.

 Also I used different NAT-traversal mechanism for sip and ws traffic
 (different routes based on client's transport protocol).
 Also you'll maybe need to have different rtpengine flags for sip and ws -
 remember that WebRTC MUST have SRTP, but I had some issues in transfering
 the SRTP handshake in sipphone--kamailio--freeswitch scheme, so on
 webrtc connection my incoming rtpengine had RTP/AVP flag, and on outgoing
 webrtc it MUST have RTP/SAVP
 For usual SIP calls I also conveted everything to RTP/AVP.

 So you'll need to know to which type of user - ws or tcp/udp you're
 calling to understand which type of RTP to send them.

 2015-06-13 19:07 GMT+03:00 Murugan Pandian manpower13@gmail.com:

 it's posible dispatching websocket request?

 I am try to connect browser(WebRTC) to sip-phone and vice versa,How i can
 achieve more concurrent call(more then 1000 call)

 On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov abalas...@evaristesys.com
  wrote:

 That question is difficult to answer without some elaboration on your
 part as to what you want to achieve.

  --
 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/

 Sent from my BlackBerry.
   *From: *Murugan Pandian
 *Sent: *Saturday, June 13, 2015 09:47
 *To: *sr-users@lists.sip-router.org
 *Reply To: *Kamailio (SER) - Users Mailing List
 *Subject: *[SR-Users] SIP-over-Websocket Load Balancing

 HI,

   how to handle sip-over-websocket load balancing (WebRTC)


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 --
 Alexandru Covalschi
 ABRISS-Solutions
 VoIP engineer and system administrator
 phone: +37367398493
 web: http://abs-telecom.com/




-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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