date:20180802

[OpenSIPS-Users] core dump on startup

2018-08-02 Thread Richard Revels

I'm using opensips version 2.3 from rpm package on fedora.  I've noticed
that if I use the wrong auth info for the mysql db and the connection fails
then opensips does a core dump in db_table_version in both the acc and
permissions modules.  Probably more, I didn't bother going further down the
list.

Since this keeps the proxy from starting there is limited damage it can do
(like dropping calls and all that) but it did have me stumped a lot longer
than a "db access denied" message or something like that would have.

Richard
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Re: [OpenSIPS-Users] mid_registrar work arround

2018-08-02 Thread Ben Newlin

M4 will process the file before OpenSIPS runs and will not be changeable at 
runtime. It sounds like that will not work for you if I am understanding 
properly.

My first thought would be to use the shared variables from CFGUTILS [1]. They 
can be accessed from inside the script, but can also be set via MI. So when 
your monitor detects a failure and switches the IPs, it could also use MI to 
change the value of the variable in OpenSIPS for future use.

As a side note to OpenSIPS devs, it appears this module was missed when the 
Module Index page [2] was redesigned as I can't find it there for 2.x, but the 
direct links to the 2.x versions still work.

[1] http://www.opensips.org/html/docs/modules/2.4.x/cfgutils.html
[2] http://www.opensips.org/Documentation/Modules-2-4

Ben Newlin 

On 8/2/18, 5:56 PM, "Users on behalf of volga...@networklab.ca" 
 wrote:

Hello Liviu,
Yes, that correct.  Right now each node in cluster have own vip from 
keepalived on LAN side.
The issue how to specify correct vip in case of failover, so if node 2 
fail and node 2 vip was relocated to node 1. All sessions should be 
process on node 1 and
append_hf("Path: \r\n"); should 
have correct ip's.  Same with rtpengine offer
rtpengine_offer("replace-origin replace-session-connection 
in-iface=external out-iface=internal media-address=$avp(media_addr) 
RTP/AVP ICE=remove") where need introduce media-address to specify 
correct vip ip.
Can you exapand on m4 ? I never used the templating.
Might db avp  to predefine  cluster nodes vip ip, not sure.

volga629

On Thu, Aug 2, 2018 at 10:50 AM, Liviu Chircu  
wrote:
> Hi Volga,
> 
> If I understood your problem correctly, one idea would be to use m4 
> over opensips.cfg and define a different MAIN_VIP variable for each 
> of your three servers:
> 
> append_hf("Path: \r\n");
>  Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 09.07.2018 07:34, volga...@networklab.ca wrote:
>> Hello Everyone,
>> 
>> I have work around on this issue
>> # Work arround for github issue #1109
>> append_hf("Path: \r\n"
>> 
>> 
>> Right now we use 3 node cluster with 3 vips in active/active mode
>> I need some idea how to determine path header listen ip address from 
>> 3 vips.
>> 
>> 
>> volga629
>> 
>> 
>> 
>> 
>> ___
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> 

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Re: [OpenSIPS-Users] json decode and access?

2018-08-02 Thread Richard Revels

I'm using a webserver response rather than cache_fetch but this may help.

#we got a valid response from the web server.
convert from string to json object
$json(resp-obj) := $avp(router-resp);
if( $avp(ok_log_this) == "true" )
xlog("L_INFO", "the whole object returned
is $json(resp-obj) and the json route first choice is
$json(resp-obj/routes[0]/ruri) \n");

...
if( $json(resp-obj/lang) )
{
$avp(jsonLangPref) :=
$json(resp-obj/lang);
$avp(jsonStatus) :=
$json(resp-obj/status);
$avp(jsonSID) :=
$json(resp-obj/sid);
$avp(jsonDashAddress) :=
$json(resp-obj/caller_id);
...
and so on.




On Thu, Aug 2, 2018 at 6:44 PM J E H A N Z A I B 
wrote:

> Hi team,
>
> I have another application storing the json encoded object in the redis
> cache.
>
> I read we can fetch using cache_fetch by passing the key. This is perfect
> but I cant find a function to decode the json I have since my other
> application is encoding the json object and putting into the redis. I need
> a function to decode it first.
>
> Second question is, after the decode the object is like
>
> Object
> (
> [id] => 161
> [user] => xxx
> [xx] => 
> [] => 
> )
>
> How to access user for example ?
>
> Thank you
>
>
> --
> Regards,
> Jehanzaib
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Re: [OpenSIPS-Users] Use RTPPROXY to bridge ipv4/ipv6

2018-08-02 Thread Maxim Sobolev

Daniel, you can find some v4-v6 examples here:

https://github.com/sippy/voiptests/blob/master/test_run.sh

${RTPPROXY} -p "${RTPP_PIDF}" -d dbug -f -s stdio: -s "${RTPP_SOCK_UDP}" \
  -s "${RTPP_SOCK_CUNIX}" -s "${RTPP_SOCK_UNIX}" -s "${RTPP_SOCK_UDP6}" -s
"${RTPP_SOCK_TCP}" \
  -s "${RTPP_SOCK_TCP6}" -m 12000 -M 15000 -6 '/::' -l '0.0.0.0'
${RTPP_NOTIFY_ARG}

In your case that would be (note "/" in front of IPv6 addr):

/bin/rtpproxy -F -l "200.200.200.200" -6 "/2607:3f00:2
"

-Max

On Thu, Aug 2, 2018 at 1:50 PM Daniel Zanutti 
wrote:

> Hi
>
> I'm trying to configure RTPPROXY to bridge ipv4 and ipv6 networks, but
> didn't find the proper way.
> Supposing IPs "200.200.200.200" and  "2607:3f00:2 " both on ETH0
> interface.
>
> Tried:
> /bin/rtpproxy -F -l 200.200.200.200/2607:3f00:2
>
> Got this error: Restarting rtpproxy: rtpproxy: host2bindaddr: Address
> family for hostname not supported
>
> Then used -6 option and got same error:
> /bin/rtpproxy -F -6 200.200.200.200/2607:3f00:2
>
> What is the right way?
>
> Thanks
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[OpenSIPS-Users] json decode and access?

2018-08-02 Thread J E H A N Z A I B

Hi team,

I have another application storing the json encoded object in the redis
cache.

I read we can fetch using cache_fetch by passing the key. This is perfect
but I cant find a function to decode the json I have since my other
application is encoding the json object and putting into the redis. I need
a function to decode it first.

Second question is, after the decode the object is like

Object
(
[id] => 161
[user] => xxx
[xx] => 
[] => 
)

How to access user for example ?

Thank you


-- 
Regards,
Jehanzaib
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Re: [OpenSIPS-Users] mid_registrar work arround

2018-08-02 Thread volga629


Hello Liviu,
Yes, that correct.  Right now each node in cluster have own vip from 
keepalived on LAN side.
The issue how to specify correct vip in case of failover, so if node 2 
fail and node 2 vip was relocated to node 1. All sessions should be 
process on node 1 and
append_hf("Path: \r\n"); should 
have correct ip's.  Same with rtpengine offer
rtpengine_offer("replace-origin replace-session-connection 
in-iface=external out-iface=internal media-address=$avp(media_addr) 
RTP/AVP ICE=remove") where need introduce media-address to specify 
correct vip ip.

Can you exapand on m4 ? I never used the templating.
Might db avp  to predefine  cluster nodes vip ip, not sure.


volga629



On Thu, Aug 2, 2018 at 10:50 AM, Liviu Chircu  
wrote:

Hi Volga,

If I understood your problem correctly, one idea would be to use m4 
over opensips.cfg and define a different MAIN_VIP variable for each 
of your three servers:


append_hf("Path: \r\n");
 Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 09.07.2018 07:34, volga...@networklab.ca wrote:

Hello Everyone,

I have work around on this issue
# Work arround for github issue #1109
append_hf("Path: \r\n"


Right now we use 3 node cluster with 3 vips in active/active mode
I need some idea how to determine path header listen ip address from 
3 vips.



volga629




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[OpenSIPS-Users] Use RTPPROXY to bridge ipv4/ipv6

2018-08-02 Thread Daniel Zanutti

Hi

I'm trying to configure RTPPROXY to bridge ipv4 and ipv6 networks, but
didn't find the proper way.
Supposing IPs "200.200.200.200" and  "2607:3f00:2 " both on ETH0
interface.

Tried:
/bin/rtpproxy -F -l 200.200.200.200/2607:3f00:2

Got this error: Restarting rtpproxy: rtpproxy: host2bindaddr: Address
family for hostname not supported

Then used -6 option and got same error:
/bin/rtpproxy -F -6 200.200.200.200/2607:3f00:2

What is the right way?

Thanks
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Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

2018-08-02 Thread Giovanni Maruzzelli

study, Fatma, study :)

btw, if you can't "find a softphone capable of being a presence user agent"
you may be in the wrong field of studies.

-giovanni

On 2 August 2018 at 15:52, Fatma Raissi  wrote:

> Good morning,
>
>
> Thanks again for your answer.
> But I can't find a softphone capable of being a presence user agent.
> Plus the presence information I need to publish is one variable which is
> "workload" of the machine.
>
> Here is the SIP message I am using and joined the configuration file.
> Maybe you can Identify the problem. Thanks
>
> test.xml
>
>
> 
> 
>
>
> 
>   
>   
>   
>   
>   
>   
> 
>   
> 
>   
>
>
> Reponse: 489 Bad event
>
>
>
> Aug  2 06:49:44 [40701] DBG:core:get_hdr_field: cseq : <1>
> 
> Aug  2 06:49:44 [40701] DBG:maxfwd:is_maxfwd_present: value = 70
> Aug  2 06:49:44 [40701] DBG:uri:has_totag: no totag
> Aug  2 06:49:44 [40701] DBG:core:parse_headers: flags=78
> Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: start searching:
> hash=22792, isACK=0
> Aug  2 06:49:44 [40701] DBG:tm:matching_3261: RFC3261 transaction matched,
> tid=nashds7
> Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request:
> REF_UNSAFE:[0x7f73f6c49708] after is 1
> Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: transaction found
> (T=0x7f73f6c49708)
> Aug  2 06:49:44 [40701] DBG:tm:t_retransmit_reply: buf=0x7f73f644f600:
> SIP/2.0 4..., shmem=0x7f73f6c4c678: SIP/2.0 4
> Aug  2 06:49:44 [40701] DBG:tm:t_check_trans: UNREF_UNSAFE:
> [0x7f73f6c49708] after is 0
> Aug  2 06:49:44 [40701] DBG:core:destroy_avp_list: destroying list (nil)
> Aug  2 06:49:44 [40701] DBG:core:receive_msg: cleaning up
> Aug  2 06:49:44 [40700] DBG:core:parse_msg: SIP Request:
> Aug  2 06:49:44 [40700] DBG:core:parse_msg:  method:  
> Aug  2 06:49:44 [40700] DBG:core:parse_msg:  uri: 
> Aug  2 06:49:44 [40700] DBG:core:parse_msg:  version: 
> Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=2
> Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 235,
>  = ; state=6
> Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 232,
>  = ; state=16
> Aug  2 06:49:44 [40700] DBG:core:parse_via: end of header reached, state=5
> Aug  2 06:49:44 [40700] DBG:core:parse_headers: via found, flags=2
> Aug  2 06:49:44 [40700] DBG:core:parse_headers: this is the first via
> Aug  2 06:49:44 [40700] DBG:core:receive_msg: After parse_msg...
> Aug  2 06:49:44 [40700] DBG:core:receive_msg: preparing to run routing
> scripts...
> Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=100
> Aug  2 06:49:44 [40700] DBG:core:_parse_to: end of header reached, state=9
> Aug  2 06:49:44 [40700] DBG:core:_parse_to: display={}, ruri={sip:
> 127.0.0.1:5060}
> Aug  2 06:49:44 [40700] DBG:core:get_hdr_field:  [20]; uri=[sip:
> 127.0.0.1:5060]
> Aug  2 06:49:44 [40700] DBG:core:get_hdr_field: to body [sip:
> 127.0.0.1:5060
>
>
>
>
>
> Fatma RAISSI  - ENIT Junior Entreprise
>
>
>
>
> *Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
> mandat 2016-2017* Tel: (+216) 53 411 311 | Email: raissifa...@gmail.com
> 
>
>
> 2018-08-02 13:48 GMT+02:00 Giovanni Maruzzelli :
>
>> Be ause they have working presence client embedded, and you seems not be
>> able to model it in sipp.
>>
>> Start with something known to work, softphones, trace the sip messages,
>> then (if needed) do the sipp xml modelization.
>>
>> -giovanni
>>
>> On Thu, Aug 2, 2018, 13:45 Fatma Raissi  wrote:
>>
>>> Good morning Sir,
>>>
>>>
>>> Thank you a lot for your answer.
>>> But could you explain why would I use softphones while I have nothing
>>> to do with voice or voice over IP.
>>>
>>> Cordially,
>>>
>>>
>>>
>>>
>>> Fatma RAISSI  - ENIT Junior Entreprise
>>>
>>>
>>>
>>>
>>> *Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
>>> mandat 2016-2017* Tel: (+216) 53 411 311 | Email: raissifa...@gmail.com
>>> 
>>>
>>>
>>> 2018-08-02 10:34 GMT+02:00 Giovanni Maruzzelli :
>>>
 Use softphones instead of sipp

 On Wed, Aug 1, 2018, 12:01 Fatma Raissi  wrote:

> Good morning Everyone,
>
>
> I am using OpenSIPS as *presence server*. I need it just to
> accomplish very basic and simple presence server functions.
>
> Here is the purpose of my work:
>
> I have 3 machines P, A, B and C.
>
> 1)* P* is the machine in which I have installed *OpenSIPS*, thus the
> presence server
> 2) I want *A and B to be the presentities* and thus publishing its
> own presence information (that I precise) into the presence server.
> 3) I want OpenSIPS to update* the watcher C* each time there is a
> change in A or B presence information.
>
> What I have done so far is that I am using the presence server config
> file that I found here :
>
> https://www.opensips.org/Documentation/Tutorials-Presence-Si
> mplePresConfig
>
> and added all the parameters I found here
>
> http://www.opensips.org/html/docs/modules/2.1.x/presence.html
>
> I am

Re: [OpenSIPS-Users] AG Project - Media Proxy

2018-08-02 Thread Dan Pascu

The log suggests that the mediaproxy dispatcher is not replying in time, yet it 
is not dead/frozen.
The timeout is 500ms and can be set through the mediaproxy_timeout module 
parameter (see the mediaproxy module README for more info).

I suggest you increase the timeout until you don't see the problem again, but 
keep in mind that whatever value you use, that is how long the opensips worker 
process can be blocked waiting for a reply, during which time it cannot do 
anything else (communication with the dispatcher is not asynchronous).

Better would be to identify what causes the dispatcher to lag like that (maybe 
over time you got an increasing number of call setups which resulted in a 
higher load that makes the dispatcher take more than 500ms to reply sometimes). 
In that case it would be better to consider some horizontal scaling and use 
some clustering solution to distribute the load.

P.S. subsequent calls work, because opensips reconnects automatically to the 
dispatcher as needed.

On 24 Jul 2018, at 14:56, Alex Tatham wrote:

> Hello everyone,
>  
> We have been using AG Projects Media Proxy for several years.  Recently we’ve 
> been experiencing intermittent problems with calls not being initiated and 
> we’ve traced this to the media proxy component.
>  
> The frequency of the issue is around 1 in 2000 calls.
>  
> The following error is logged by the SIP proxy “ERROR: mediaproxy 
> [mediaproxy.c:1528]: send_command(): did timeout waiting for an answer”.
>  
> The following error corresponds in the dispatcher component of the Media 
> Proxy software “Connection to OpenSIPS lost: Connection was closed cleanly.”.
>  
> The problem is temporary and subsequent calls are successful without any 
> intervention.
>  
> I can’t find any information on the WWW about either of these messages.
>  
> The mediaproxy components are running the latest debian packages as is the 
> corresponding module.  I do have some older servers running a much older 
> version of mediaproxy that don’t have this fault but I’ve got to update 
> because of a bug with the relay component on that build.
>  
> Does anyone have any suggestions.
>  
> Thanks, Alex
> Alex Tatham
> Technical Director
> 
> T:01233 220 943
> E:alex.tat...@dmcplc.co.uk
> W:www.dmctechnologies.co.uk
>  
> DMC Technologies   formerly HP Technology
> Unit 5, Invicta Business Centre
>  | 
> Orbital Park
>  | 
> Ashford
>  | 
> Kent   | 
> TN24 0HB
> 
> DMC Technologies is a trading name of Hobbs Parker 2000 Limited 
> Registered Office: Romney House, Orbital Park, Monument Way, Ashford TN24 0HB
> Registered in England No.04027755
> 
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Re: [OpenSIPS-Users] RTPengine provisioning by db

2018-08-02 Thread Bogdan-Andrei Iancu


Hi all,

Just pushed a fix to include rtpengine table into the default create 
tables, so `opensipsctldb create` will now create the rtpengine table too.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 07/12/2018 03:30 PM, Răzvan Crainea wrote:

Hi, Alain!

No, rtpengine is not created when you run `opensipsctldb create`. 
You'll have to create it yourself using something like:


mysql -uUSER -pPASS -dDB -hHOST < scripts/mysql/rtpengine-create.sql

Best regards,
Răzvan

On 07/12/2018 03:30 PM, Alain Bieuzent wrote:

Hi Razvan,



are you sure this new support of provisioning via DB is supported by 
tools "opensipctldb create", because new table "rtpengine" does not 
exist when I create a new database from scratch.




thanks



https://github.com/OpenSIPS/opensips/commit/094c850dd91286ff75bf3463dafbf83a398aaf8f 











Le 12/07/2018 11:58, « Users au nom de Alain Bieuzent » 
 a 
écrit :




 Hi all,




 I just created a new database and version of table 
rtpproxy_sockets is 0, so i think there is a mismatch somewhere 
because opensips 2.4.1 expected 1.





 Is there somewhere the mysql schema of 2.4 version?




 thanks




 Le 11/07/2018 18:02, « Users au nom de Răzvan Crainea » 
 a 
écrit :





 No problem, thanks for sharing the initial cause!




 Best regards,


 Răzvan




 On 07/11/2018 06:59 PM, Alain Bieuzent wrote:


 > never forgot to check the logs.


 >


 > yes, you are right opensips didn't start with this error:


 >


 >


 >


 > Jul 11 17:52:21 opensips-test opensips[20625]: 
ERROR:core:db_check_table_version: invalid version 0 for table 
rtpproxy_sockets found, expected 1



 >


 > Jul 11 17:52:21 opensips-test opensips[20625]: 
ERROR:rtpengine:mod_init: error during table version check



 >


 > Jul 11 17:52:21 opensips-test opensips[20625]: 
ERROR:core:init_mod: failed to initialize module rtpengine



 >


 > Jul 11 17:52:21 opensips-test opensips[20625]: 
ERROR:core:main: error while initializing modules



 >


 >


 >


 > My fault sorry for the noise.


 >


 >


 >


 > Thanks, Razvan.


 >


 >


 >


 > Le 11/07/2018 17:48, « Users au nom de Răzvan Crainea » 
 a 
écrit :



 >


 >


 >


 >  Hello!


 >


 >


 >


 >  After you comment the first line, can you make sure 
opensips actually



 >


 >  starts (ps aux | grep opensips). After that, can you 
check that the



 >


 >  /tmp/opensips_fifo file exists?


 >


 >


 >


 >  Best regards,


 >


 >  Răzvan


 >


 >


 >


 >  On 07/11/2018 06:37 PM, Alain Bieuzent wrote:


 >


 >  > Hi, Rasvan!


 >


 >  >


 >


 >  >


 >


 >  >


 >


 >  > I double check and the typo with the "i" is not the 
cause.



 >


 >  >


 >


 >  >


 >


 >  >


 >


 >  > I used these parameters:


 >


 >  >


 >


 >  > 
##



 >


 >  >


 >


 >  >  RTPENGINE module


 >


 >  >


 >


 >  > 
##



 >


 >  >


 >


 >  > modparam("rtpengine", "rtpengine_sock", 
"udp:10.207.201.19:2223 udp:10.207.201.20:2223")



 >


 >  >


 >


 >  >


 >


 >  >


 >


 >  > modparam("rtpengine", "db_url", 
"mysql://opensips:@ZZZ/opensips_lb_did_2_4")



 >


 >  >


 >


 >  > modparam("rtpengine", "db_table", "rtpproxy_sockets")


 >


 >  >


 >


 >  > modparam("rtpengine", "socket_column", 
"rtpproxy_sock")



 >


 >  >


 >


 >  > modparam("rtpengine", "set_column", "set_id")


 >


 >  >


 >


 >  >


 >


 >  >


 >


 >  > If i comment the 4 last line to do not use db, 
opensips start i can use opensipsctl fifo ds_list (so no issue with 
fifo file)



 >


 >  >


 >


 >  > If i comment only the first line to use db, 
opensipsctl produce the

Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

2018-08-02 Thread Fatma Raissi

Good morning,


Thanks again for your answer.
But I can't find a softphone capable of being a presence user agent.
Plus the presence information I need to publish is one variable which is
"workload" of the machine.

Here is the SIP message I am using and joined the configuration file. Maybe
you can Identify the problem. Thanks

test.xml







  
  
  
  
  
  

  

  


Reponse: 489 Bad event



Aug  2 06:49:44 [40701] DBG:core:get_hdr_field: cseq : <1> 
Aug  2 06:49:44 [40701] DBG:maxfwd:is_maxfwd_present: value = 70
Aug  2 06:49:44 [40701] DBG:uri:has_totag: no totag
Aug  2 06:49:44 [40701] DBG:core:parse_headers: flags=78
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: start searching:
hash=22792, isACK=0
Aug  2 06:49:44 [40701] DBG:tm:matching_3261: RFC3261 transaction matched,
tid=nashds7
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request:
REF_UNSAFE:[0x7f73f6c49708] after is 1
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: transaction found
(T=0x7f73f6c49708)
Aug  2 06:49:44 [40701] DBG:tm:t_retransmit_reply: buf=0x7f73f644f600:
SIP/2.0 4..., shmem=0x7f73f6c4c678: SIP/2.0 4
Aug  2 06:49:44 [40701] DBG:tm:t_check_trans: UNREF_UNSAFE:
[0x7f73f6c49708] after is 0
Aug  2 06:49:44 [40701] DBG:core:destroy_avp_list: destroying list (nil)
Aug  2 06:49:44 [40701] DBG:core:receive_msg: cleaning up
Aug  2 06:49:44 [40700] DBG:core:parse_msg: SIP Request:
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  method:  
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  uri: 
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  version: 
Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=2
Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 235,
 = ; state=6
Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 232,
 = ; state=16
Aug  2 06:49:44 [40700] DBG:core:parse_via: end of header reached, state=5
Aug  2 06:49:44 [40700] DBG:core:parse_headers: via found, flags=2
Aug  2 06:49:44 [40700] DBG:core:parse_headers: this is the first via
Aug  2 06:49:44 [40700] DBG:core:receive_msg: After parse_msg...
Aug  2 06:49:44 [40700] DBG:core:receive_msg: preparing to run routing
scripts...
Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=100
Aug  2 06:49:44 [40700] DBG:core:_parse_to: end of header reached, state=9
Aug  2 06:49:44 [40700] DBG:core:_parse_to: display={}, ruri={sip:
127.0.0.1:5060}
Aug  2 06:49:44 [40700] DBG:core:get_hdr_field:  [20]; uri=[sip:
127.0.0.1:5060]
Aug  2 06:49:44 [40700] DBG:core:get_hdr_field: to body [sip:127.0.0.1:5060





Fatma RAISSI  - ENIT Junior Entreprise




*Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
mandat 2016-2017* Tel: (+216) 53 411 311 | Email: raissifa...@gmail.com



2018-08-02 13:48 GMT+02:00 Giovanni Maruzzelli :

> Be ause they have working presence client embedded, and you seems not be
> able to model it in sipp.
>
> Start with something known to work, softphones, trace the sip messages,
> then (if needed) do the sipp xml modelization.
>
> -giovanni
>
> On Thu, Aug 2, 2018, 13:45 Fatma Raissi  wrote:
>
>> Good morning Sir,
>>
>>
>> Thank you a lot for your answer.
>> But could you explain why would I use softphones while I have nothing to
>> do with voice or voice over IP.
>>
>> Cordially,
>>
>>
>>
>>
>> Fatma RAISSI  - ENIT Junior Entreprise
>>
>>
>>
>>
>> *Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
>> mandat 2016-2017* Tel: (+216) 53 411 311 | Email: raissifa...@gmail.com
>> 
>>
>>
>> 2018-08-02 10:34 GMT+02:00 Giovanni Maruzzelli :
>>
>>> Use softphones instead of sipp
>>>
>>> On Wed, Aug 1, 2018, 12:01 Fatma Raissi  wrote:
>>>
 Good morning Everyone,


 I am using OpenSIPS as *presence server*. I need it just to accomplish
 very basic and simple presence server functions.

 Here is the purpose of my work:

 I have 3 machines P, A, B and C.

 1)* P* is the machine in which I have installed *OpenSIPS*, thus the
 presence server
 2) I want *A and B to be the presentities* and thus publishing its own
 presence information (that I precise) into the presence server.
 3) I want OpenSIPS to update* the watcher C* each time there is a
 change in A or B presence information.

 What I have done so far is that I am using the presence server config
 file that I found here :

 https://www.opensips.org/Documentation/Tutorials-
 Presence-SimplePresConfig

 and added all the parameters I found here

 http://www.opensips.org/html/docs/modules/2.1.x/presence.html

 I am using SIPp in machine A and B  and C to send publish and subscribe
 messages but I keep getting no answer at all from OpenSIPS although I am
 sure he is receiving the messages.
 I think I am not using the right syntax of SIP messages.

 Can you please help me by sending to me an example of SIP subscribe and
 SIP publish message that I can send from SIPp to OpenSIPS.

 - *SIP Subscribe message* : in

Re: [OpenSIPS-Users] mid_registrar work arround

2018-08-02 Thread Liviu Chircu


Hi Volga,

If I understood your problem correctly, one idea would be to use m4 over 
opensips.cfg and define a different MAIN_VIP variable for each of your 
three servers:


append_hf("Path: \r\n");

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 09.07.2018 07:34, volga...@networklab.ca wrote:

Hello Everyone,

I have work around on this issue
# Work arround for github issue #1109
append_hf("Path: \r\n"


Right now we use 3 node cluster with 3 vips in active/active mode
I need some idea how to determine path header listen ip address from 3 
vips.



volga629




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Re: [OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite

2018-08-02 Thread Sebastian Sastre

Hey guys, sorry to keep insisting, im just very curious why its not finding
that TCP connection.

Nothing anywhere tells or yields an error.

What else can I look at?


On Mon, Jul 23, 2018 at 11:58 AM, Sebastian Sastre <
sastre.sebast...@gmail.com> wrote:

> I put a full debug on this paste https://pastebin.com/BEJ6fAR8
>
> Should be
>
> Jul 13 11:42:39 gcwregistrar151 /sbin/opensips[3647]: ERROR:tm:msg_send:
> send() to 192.0.2.246:443 for proto wss/6 failed
>
> Thanks
>
>
> On Mon, Jul 23, 2018 at 11:47 AM, Răzvan Crainea 
> wrote:
>
>> Hi, Sebastien!
>>
>> It looks like the contact is not changing. Can you indicate what
>> connection he is trying to find, i.e. what IP, port and proto? You should
>> see them in the error line.
>>
>> Best regards,
>> Răzvan
>>
>> On 07/23/2018 06:43 PM, Sebastian Sastre wrote:
>>
>>> Hey Razvan,
>>>
>>> I’ve been playing around a lot with this but I can’t seem to make it
>>> work. Whatever I do without the fix route doesn’t find a suitable tcp
>>> connection.
>>>
>>> Do you see something on the dlg_list before that would indicate the
>>> problem Or is there any other debug I can use ?
>>>
>>> Thanks again !
>>>
>>>
>>> On Thu, Jul 19, 2018 at 8:03 PM, Sebastian Sastre <
>>> sastre.sebast...@gmail.com > wrote:
>>>
>>> Razvan,
>>> Thanks ! I tried what you indicated but I don’t see the contact
>>> changing. Im taking care of the fix contacts where it needs to be
>>> bet but still on the bye it can’t find it.
>>>
>>>
>>> root@gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call
>>> Connected)*
>>> dialog::  ID=5820137817639
>>>  state:: 4
>>>  user_flags:: 0
>>>  timestart:: 1532043804
>>>  datestart:: 2018-07-19 19:43:24
>>>  timeout:: 1532044163
>>>  dateout:: 2018-07-19 19:49:23
>>>  callid:: fp436dll6pcmdqk78gn6
>>>  from_uri:: sip:u...@domain.com >> sip%3au...@domain.com>
>>>  to_uri:: sip:18889990...@domain.com
>>> 
>>>  caller_tag:: 1i4vfmjico
>>>  caller_contact:: sip:lccpphv2@192.168.202.3
>>> :51292;transport=wss;ob
>>>  callee_cseq:: 0
>>>  caller_route_set::
>>>  caller_bind_addr:: wss:10.101.10.151:443
>>> 
>>>  caller_sdp::
>>>  CALLEES::
>>>  callee::
>>>  callee_tag::
>>> d651df12-c9c2-4db1-99ad-b15d6240ffee
>>>  callee_contact:: sip:10.101.10.161:5060
>>> 
>>>  caller_cseq:: 1094
>>>  callee_route_set::
>>>  callee_bind_addr:: udp:10.101.10.151:5060
>>> 
>>>  callee_sdp::
>>>
>>> root@gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )*
>>> dialog::  ID=5820137817639
>>>  state:: 4
>>>  user_flags:: 0
>>>  timestart:: 1532043804
>>>  datestart:: 2018-07-19 19:43:24
>>>  timeout:: 1532044163
>>>  dateout:: 2018-07-19 19:49:23
>>>  callid:: fp436dll6pcmdqk78gn6
>>>  from_uri:: sip:u...@domain.com >> sip%3au...@domain.com>
>>>  to_uri:: sip:18889990...@domain.com
>>> 
>>>  caller_tag:: 1i4vfmjico
>>>  caller_contact:: sip:lccpphv2@192.168.202.3
>>> :51292;transport=wss;ob
>>>  callee_cseq:: 0
>>>  caller_route_set::
>>>  caller_bind_addr:: wss:10.101.10.151:443
>>> 
>>>  caller_sdp::
>>>  CALLEES::
>>>  callee::
>>>  callee_tag::
>>> d651df12-c9c2-4db1-99ad-b15d6240ffee
>>>  callee_contact:: sip:10.101.10.161:5060
>>> 
>>>  caller_cseq:: 1095
>>>  callee_route_set::
>>>  callee_bind_addr:: udp:10.101.10.151:5060
>>> 
>>>  callee_sdp::
>>>
>>>
>>> On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea >> > wrote:
>>>
>>> Hi, Sebastian!
>>>
>>> The re-invite probably generates a remote contact update. And if
>>> you don't "fix" the contact on re-invites and their 200 OK, you
>>> might end up with broken contacts in the dialog, thus sequential
>>> signaling will not work.
>>> I suggest you do two things to debug this:
>>> 1. remove the fix_route_dialog() call - the call should still be
>>> routed

Re: [OpenSIPS-Users] Opensips TCP children deadlock

2018-08-02 Thread Bogdan-Andrei Iancu


Hi all,

For the sake of completion, here is the commit fixing the issue:
https://github.com/OpenSIPS/opensips/commit/058cc22cb55dce9b890308b9f83a42a88691f2c8

Thank you Yuval for the report and for investigating this!

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 07/12/2018 04:07 PM, Yuval Dinari via Users wrote:

Hi,
I have a state in which opensips gets into an unrecoverable bad state, 
in which some of the tcp children process are stuck waiting to acquire 
a lock which they never get.

The issue occurs in the following load test scenario:

 1. About 25K clients register in TCP (but also happens with less)
 2. All the TCP connections become unresponsive (by blocking outgoing
traffic on the test clients machine)
 3. INVITEs are sent for each of those clients, putting their
connection in retransmit mode
 4. After a few minutes opensips gets into a bad state - some tcp
children run at 90-100% cpu, no traffic is being sent from the
machine (including OPTIONS pings)
 5. After all the tcp connections die due to timeouts, opensips does
not recover, the mentioned symptoms stay
 6. After all the registered users are removed from internal table
there's still no change

When attaching debugger to the problematic processes (with high cpu 
usage) we see that they're all stuck trying to get a lock which they 
never seem to get. Stack traces:


#0  0x7fd6b72d1bb7 in sched_yield () at 
../sysdeps/unix/syscall-template.S:81
#1  0x00549e65 in get_lock (lock=) at 
net/proto_tcp/../../net/../fastlock.h:221
#2  _tcp_write_on_socket (len=, buf=, 
fd=, c=) at net/proto_tcp/proto_tcp.c:724
#3  proto_tcp_send (send_sock=0x7ffd8e12c140, buf=0x0, len=399, 
to=0x7fd5c7ccdcc0, id=1) at net/proto_tcp/proto_tcp.c:922
#4  0x7fd5a5cb7b30 in msg_send (msg=, 
len=, buf=, id=, 
to=, proto=,

send_sock=0x7fd6a7208168) at ../../forward.h:123
#5  send_pr_buffer (rb=0x7fd5c7ccdca0, buf=0x7fd6a76b4a50, len=0, 
ctx=0x) at t_funcs.c:66


And:

#0  0x7fd6b72d1bb7 in sched_yield () at 
../sysdeps/unix/syscall-template.S:81
#1  0x005349b8 in get_lock (lock=) at 
net/../fastlock.h:221
#2  handle_io (event_type=, idx=, 
fm=) at net/net_tcp_proc.c:210
#3  io_wait_loop_epoll (repeat=287, t=, h=out>) at net/../io_wait_loop.h:280


This traces look the same every time we attach.
The machine opensips runs on has 4 cpus.
Thanks





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Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

2018-08-02 Thread Giovanni Maruzzelli

Be ause they have working presence client embedded, and you seems not be
able to model it in sipp.

Start with something known to work, softphones, trace the sip messages,
then (if needed) do the sipp xml modelization.

-giovanni

On Thu, Aug 2, 2018, 13:45 Fatma Raissi  wrote:

> Good morning Sir,
>
>
> Thank you a lot for your answer.
> But could you explain why would I use softphones while I have nothing to
> do with voice or voice over IP.
>
> Cordially,
>
>
>
>
> Fatma RAISSI  - ENIT Junior Entreprise
>
>
>
>
> *Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
> mandat 2016-2017* Tel: (+216) 53 411 311 | Email: raissifa...@gmail.com
> 
>
>
> 2018-08-02 10:34 GMT+02:00 Giovanni Maruzzelli :
>
>> Use softphones instead of sipp
>>
>> On Wed, Aug 1, 2018, 12:01 Fatma Raissi  wrote:
>>
>>> Good morning Everyone,
>>>
>>>
>>> I am using OpenSIPS as *presence server*. I need it just to accomplish
>>> very basic and simple presence server functions.
>>>
>>> Here is the purpose of my work:
>>>
>>> I have 3 machines P, A, B and C.
>>>
>>> 1)* P* is the machine in which I have installed *OpenSIPS*, thus the
>>> presence server
>>> 2) I want *A and B to be the presentities* and thus publishing its own
>>> presence information (that I precise) into the presence server.
>>> 3) I want OpenSIPS to update* the watcher C* each time there is a
>>> change in A or B presence information.
>>>
>>> What I have done so far is that I am using the presence server config
>>> file that I found here :
>>>
>>>
>>> https://www.opensips.org/Documentation/Tutorials-Presence-SimplePresConfig
>>>
>>> and added all the parameters I found here
>>>
>>> http://www.opensips.org/html/docs/modules/2.1.x/presence.html
>>>
>>> I am using SIPp in machine A and B  and C to send publish and subscribe
>>> messages but I keep getting no answer at all from OpenSIPS although I am
>>> sure he is receiving the messages.
>>> I think I am not using the right syntax of SIP messages.
>>>
>>> Can you please help me by sending to me an example of SIP subscribe and
>>> SIP publish message that I can send from SIPp to OpenSIPS.
>>>
>>> - *SIP Subscribe message* : in which I can precise the list of IP
>>> addresses that the machine C needs to know their updates
>>> - *SIP Publish message*: in which I can precise the presence
>>> information I want A and B to publish.
>>>
>>> Tell me if you need my config file or the SIP message I am using ( I am
>>> pretty sure it is wrong though...)
>>>
>>> An other question please: Is it normal that I couldn't find pua.so and
>>> rls.so in my modules file?
>>>
>>> I really appreciate the help. I couldn't find any example else where.
>>>
>>> Cordially,
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
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>>
>>
>
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Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

2018-08-02 Thread Fatma Raissi

Good morning Sir,


Thank you a lot for your answer.
But could you explain why would I use softphones while I have nothing to do
with voice or voice over IP.

Cordially,




Fatma RAISSI  - ENIT Junior Entreprise




*Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
mandat 2016-2017* Tel: (+216) 53 411 311 | Email: raissifa...@gmail.com



2018-08-02 10:34 GMT+02:00 Giovanni Maruzzelli :

> Use softphones instead of sipp
>
> On Wed, Aug 1, 2018, 12:01 Fatma Raissi  wrote:
>
>> Good morning Everyone,
>>
>>
>> I am using OpenSIPS as *presence server*. I need it just to accomplish
>> very basic and simple presence server functions.
>>
>> Here is the purpose of my work:
>>
>> I have 3 machines P, A, B and C.
>>
>> 1)* P* is the machine in which I have installed *OpenSIPS*, thus the
>> presence server
>> 2) I want *A and B to be the presentities* and thus publishing its own
>> presence information (that I precise) into the presence server.
>> 3) I want OpenSIPS to update* the watcher C* each time there is a change
>> in A or B presence information.
>>
>> What I have done so far is that I am using the presence server config
>> file that I found here :
>>
>> https://www.opensips.org/Documentation/Tutorials-
>> Presence-SimplePresConfig
>>
>> and added all the parameters I found here
>>
>> http://www.opensips.org/html/docs/modules/2.1.x/presence.html
>>
>> I am using SIPp in machine A and B  and C to send publish and subscribe
>> messages but I keep getting no answer at all from OpenSIPS although I am
>> sure he is receiving the messages.
>> I think I am not using the right syntax of SIP messages.
>>
>> Can you please help me by sending to me an example of SIP subscribe and
>> SIP publish message that I can send from SIPp to OpenSIPS.
>>
>> - *SIP Subscribe message* : in which I can precise the list of IP
>> addresses that the machine C needs to know their updates
>> - *SIP Publish message*: in which I can precise the presence information
>> I want A and B to publish.
>>
>> Tell me if you need my config file or the SIP message I am using ( I am
>> pretty sure it is wrong though...)
>>
>> An other question please: Is it normal that I couldn't find pua.so and
>> rls.so in my modules file?
>>
>> I really appreciate the help. I couldn't find any example else where.
>>
>> Cordially,
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
> ___
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>
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Re: [OpenSIPS-Users] change_reply_status() do not work with 2xx replies

2018-08-02 Thread Bogdan-Andrei Iancu


Hi,

What you are trying to do it not logically correct ; even more, it is 
not SIP wise correct (a proxy must not mess up with the final replies as 
the end points will get de-syncronized .


This a well know race in SIP - caller canceling versus callee answering 
the call. The RFC says that the proxy must pass the 200 OK INVITE back 
to caller and let the caller device to decide what to do. If the device 
really wants to terminate, it will generate a BYE on the spot. If not, 
it can keep the call and ignore the fact it sent the cancel.
And you accounting must follow the caller's decision, by simply 
accounting the call as normal.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
  http://opensips.org/training/OpenSIPS_Bootcamp_2018/

On 08/01/2018 05:20 PM, kerto...@yandex.ru wrote:

Hello everyone! Need your help.
Scheme: A <---> opensips <> B, simple scheme of call:
1) INVITEA ---> opensips > B
2) 100   A <--- opensips < B
3) 183   A <--- opensips < B
4) CANCELA ---> opensips > B
5) 200 CancellingA <--- opensips < B
6) 200 okA <--- opensips < B
When customer B received CANCEL request, he answers "200 Cancelling" (Cseq: CANCEL) and 
"200 ok" (Cseq: INVITE) at the same time. This behavior generates is wrong CDR - i use 
accounting. For resolve this situation i trying to use this code in route BRANCH_REPLY, in snippet 
below you may see routes initializing chain:
```
route {
...
 t_on_failure("GETDATA");
...
}

failure_route[GETDATA] {
...
  route("TRYING");
...
}
   
route[TRYING] {

...
 t_on_branch("BRANCH_CARRIER");
...
}

branch_route[BRANCH_CARRIER] {
...
 t_on_reply("BRANCH_REPLY");
...
}

onreply_route[BRANCH_REPLY] {
...
 xlog("L_INFO", "DEBUG OUT rs=$rs rr=$rr si=$si DLG_status=$DLG_status\n");
 if ((t_check_status("18[0-9]|200")) && (t_was_cancelled())) { # Don't 
allow continue cancelled call
 xlog("L_INFO", "DEBUG IN rs=$rs rr=$rr si=$si 
DLG_status=$DLG_status\n");
 $DLG_timeout = 1;
 change_reply_status("487", "Request Terminated");
 $var(local_sip_code) = "487";
 exit;
 }
...
}
```

Reply "200 ok" falls into the route "BRANCH_REPLY" and block "Don't allow continue 
cancelled call", but change_reply_status() do not change rr and rs.
One more question about accounting, any chance use function drop_accounting() 
in this scheme? I want drop accounting if opensips receives 200ok AFTER 
processing CANCEL.
Thanks!

opensips version:
```
version: opensips 2.3.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, HP_MALLOC, 
FAST_LOCK-FUTEX-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
main.c compiled on 16:48:49 Jul  5 2018 with gcc 5.4.0
```
The same ticket in git: https://github.com/OpenSIPS/opensips/issues/1435

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Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

2018-08-02 Thread Giovanni Maruzzelli

Use softphones instead of sipp

On Wed, Aug 1, 2018, 12:01 Fatma Raissi  wrote:

> Good morning Everyone,
>
>
> I am using OpenSIPS as *presence server*. I need it just to accomplish
> very basic and simple presence server functions.
>
> Here is the purpose of my work:
>
> I have 3 machines P, A, B and C.
>
> 1)* P* is the machine in which I have installed *OpenSIPS*, thus the
> presence server
> 2) I want *A and B to be the presentities* and thus publishing its own
> presence information (that I precise) into the presence server.
> 3) I want OpenSIPS to update* the watcher C* each time there is a change
> in A or B presence information.
>
> What I have done so far is that I am using the presence server config file
> that I found here :
>
> https://www.opensips.org/Documentation/Tutorials-Presence-SimplePresConfig
>
> and added all the parameters I found here
>
> http://www.opensips.org/html/docs/modules/2.1.x/presence.html
>
> I am using SIPp in machine A and B  and C to send publish and subscribe
> messages but I keep getting no answer at all from OpenSIPS although I am
> sure he is receiving the messages.
> I think I am not using the right syntax of SIP messages.
>
> Can you please help me by sending to me an example of SIP subscribe and
> SIP publish message that I can send from SIPp to OpenSIPS.
>
> - *SIP Subscribe message* : in which I can precise the list of IP
> addresses that the machine C needs to know their updates
> - *SIP Publish message*: in which I can precise the presence information
> I want A and B to publish.
>
> Tell me if you need my config file or the SIP message I am using ( I am
> pretty sure it is wrong though...)
>
> An other question please: Is it normal that I couldn't find pua.so and
> rls.so in my modules file?
>
> I really appreciate the help. I couldn't find any example else where.
>
> Cordially,
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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