Re: [OpenSIPS-Users] Opensips 2.1.x siptrace not sending packet to homer

[Kneeoh]

I have the same issue in 1.11.5. Did you ever get a resolution to this?



--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-2-1-x-siptrace-not-sending-packet-to-homer-tp7595668p7602360.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] uac_registrant module

[Ovidiu Sas]

You didn't define the db_url parameter for uac_registrant 
I don't know how to put it more obvious in the logs.

-ovidiu

On Thu, Mar 24, 2016 at 12:45 PM, Francjos <35...@heb.be> wrote:
> These are the errors
>
> ERROR:uac_registrant:mod_init: DB URL is not defined
> ERROR:core:init_mod: fail to initialize module uac_registrant
> ERROR:core:main: error while initializing modules
>
> Thanks
>
>
>
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/uac-registrant-module-tp7602335p7602357.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



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

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.5

[Louis Rochon]

I recompiled with the 1.11.6, and did not resolve the issue. Traces show 
behaviour identical between 1.11.5 and 1.11.6.

Looking at the release notes for 1.11.6, there are some fixes that are closely 
related, but don't fix my specific issue. i.e. 95f5f79, 26a0a62. 

Any suggestions? Is there some yet-to-be documented B2BLogic module parameter 
that needs to be set? 

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Monday, March 21, 2016 2:22 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Thanks Liviu,

I will recompile and give it a try.

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Hi Louis,

There was a similar discussion back in August [1], concluding with a fix [2], 
made between the 1.11.5 and 1.11.6 releases. Updating to the latest OpenSIPS 
1.11 will most likely solve this problem.

[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250

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

On 21.03.2016 14:55, Louis Rochon wrote:
> Anybody?
>
> Louis
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
> Sent: Wednesday, March 16, 2016 10:02 AM
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
> 1.11.5
>
> SDP in ACK lost in OpenSIPS 1.11.5
>
> This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
> broken 1.11.5.
>
> Using the B2BUA facilities, I make the second leg of the call using 
> b2b_init_request. Then, if required, I move the call to another user agent 
> via a bash shell script, which issues a opensipsctl fifo b2b_bridge.
>
> For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
> is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.
>
> Anyway of reintroducing the SDP in the ACK in 1.11.5?
>
> Trace Explanation
> -
> Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
>   10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
> CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
> OpenSIPs 1.11.5.
>   
> Call Flow in Trace:
> 10.10.10.205 10.10.8.103
>--Invite-->Indialog reinvite, so SDP
> <--Trying--
> <--200 OK--With SDP
> --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
>   
> OpenSIPS 1.8.1:
> ---
> INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 INVITE
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: 
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Contact: 
> Content-Type: application/sdp
> Content-Length:   185
>
> v=0
> o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 ACK
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 186
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Allow: INVITE, ACK, CANCEL, BYE
> Priority: emergency
> Calluid: 1537A73F8B0C005C
> Geolocation: 
> Geolocation-Routing: yes
> Initial-CallID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Contact: 
> Content-Type: application/sdp
>
> v=0
> o=VOIPSIL_SIP 322321996 322321996 IN IP4 10.10.10.203 s=Sip Call c=IN
> IP4 10.10.10.7
> t=0 0
> m=audio 21336 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> BYE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK4037.cd227b95.0
> To:

Re: [OpenSIPS-Users] uac_registrant module

[Francjos]

These are the errors 

ERROR:uac_registrant:mod_init: DB URL is not defined
ERROR:core:init_mod: fail to initialize module uac_registrant
ERROR:core:main: error while initializing modules

Thanks



--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/uac-registrant-module-tp7602335p7602357.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [2.2/devel] Heads up, major changes

[Bogdan-Andrei Iancu]

My fault, it will be updated along with the rest of the docs for the 2.2 
release.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 24.03.2016 16:36, Ovidiu Sas wrote:

$log_level is not documented in the wiki.
Same for debug_mode.

-ovidiu

On Mon, Feb 8, 2016 at 7:17 AM, Bogdan-Andrei Iancu  wrote:

Hi all,

This is a heads up regarding some latest changes over OpenSIPS 2.2 / devel
branch, changes that do impact your scripts :

1) "fork" option replaced with "debug_mode"
The fork/don't fork option made no sense anymore in the new OpenSIPS
architecture (with async reactor). Even more, the so called "fork" was
related only to SIP workers (even with fork=no, opensips was forking timer
process, per-module processes); not to mention the fork=no limitation to
only one UDP interface. Shortly, the "fork" option became outdated, causing
more problem (coding and experience), rather than something useful.
"debug_mode" option was introduce as better way to switch on and off a fast
way to debug your opensips. Upon enabling the "debug_mode", opensips will
automatically force:
  - staying in foreground (do not detach from console)
  - set logging level to 4 (debug)
  - set logging to standard error
  - enable core dumping
  - set UDP worker processes to 2
  - set TCP worker processes to 2

2) "debug" option replaced with "log_level"
The global parameter "debug" was renamed "log_level" for a more accurate
description of the setting. The values and the behavior were kept exactly as
before. Of course the MI function "debug" was renamed to "log_level".

3) "set_debug()" function replaced with "$log_level" variable
Th3e $log_level with behave upon writing exactly as old set_debug() - a new
logging level will be set (as number) or the log level will be reset (for a
NULL input). Even more, the variable will accept to be populated from other
variables/transformations. Upon reading, the current per process logging
level will be returned.
 Ex :
 $avp(custom_log_level) = 4;
 xlog("Current logging level is $log_level\n");
 $log_level = $avp(custom_log_level); /* switch to debug level */
 xlog("Running in debug level ($log_level) now\n");
 $log_level = NULL;
 xlog("Resetting back to logging level $log_level\n");


Any feedback or reports are welcome !

--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users







___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Query about avp in siptrace module

[Ramachandran, Agalya (Contractor)]

Hi Daniel,

It worked now after replacing the changes in the config file you sent.
Thank you so much for helping out me in order to understand the usage and how 
to use AVP variable.

Regards,
Agalya
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Daniel Zanutti
Sent: Wednesday, March 23, 2016 2:19 PM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Query about avp in siptrace module

Try this one, it should work.

Siptrace module will only record the packets, if the AVP variable is set.

On Wed, Mar 23, 2016 at 11:09 AM, Ramachandran, Agalya (Contractor) 
mailto:agalya_ramachand...@comcast.com>> wrote:
Hi Daniel,

Here is the snippet of the opensips log where I am getting AVP value is NULL. I 
have added this debug statement in order to ensure that AVP is NULL.
Also am attaching the opensips.cfg file.

Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:core:parse_headers: this is the first via
Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:core:receive_msg: After parse_msg...
Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:core:receive_msg: preparing to run routing scripts...
Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:siptrace:sip_trace: **AVP IS NULL**...
Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:siptrace:sip_trace: nothing to trace...
Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:core:parse_headers: flags=
Mar 22 13:59:20 ma-siplb-as-a-001 /usr/local/sbin/opensips[869]: 
DBG:core:parse_via_param: found param type 232,  = 
; state=16

What is the use of avp variable with respect to siptrace module? It would be 
great if you help me out on this.

Regards,
Agalya

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org]
 On Behalf Of Daniel Zanutti
Sent: Tuesday, March 22, 2016 12:22 PM

To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] Query about avp in siptrace module

Hi Agalya

Could you post you opensips.cfg file and some trace with the AVP = NULL error?

On Tue, Mar 22, 2016 at 12:07 PM, Ramachandran, Agalya (Contractor) 
mailto:agalya_ramachand...@comcast.com>> wrote:
Hi Daniel,

I have already set the onreply_avp_mode.
modparam("tm", "onreply_avp_mode", 1)

Added  the below line in route at the very beginning as suggested by you.
$avp(traced_user) = "1";

Am calling sip_trace() in ‘route’ and ‘onreply_route’.
Still am seeing AVP is NULL in my traces.  Can you please clarify me on what am 
missing out.

Regards,
Agalya


From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org]
 On Behalf Of Daniel Zanutti
Sent: Monday, March 21, 2016 2:24 PM
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] Query about avp in siptrace module

Please take a look at AVP page:
http://www.opensips.org/html/docs/modules/1.9.x/avpops.html#id248034

'AVPs are persistent per SIP transaction, being available in "route", 
"branch_route" and "failure_route". To make them available in "onreply_route" 
armed via TM module, set "onreply_avp_mode" parameter of TM module (note that 
in the default "onreply_route", the AVPs of the transaction are not available)'

The AVP is persistent at the SIP transaction. Even if not, you are setting it 
again on every message received.

Don't forget to call sip_trace again on reply_route

Regards


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [2.2/devel] Heads up, major changes

[Ovidiu Sas]

$log_level is not documented in the wiki.
Same for debug_mode.

-ovidiu

On Mon, Feb 8, 2016 at 7:17 AM, Bogdan-Andrei Iancu  wrote:
> Hi all,
>
> This is a heads up regarding some latest changes over OpenSIPS 2.2 / devel
> branch, changes that do impact your scripts :
>
> 1) "fork" option replaced with "debug_mode"
> The fork/don't fork option made no sense anymore in the new OpenSIPS
> architecture (with async reactor). Even more, the so called "fork" was
> related only to SIP workers (even with fork=no, opensips was forking timer
> process, per-module processes); not to mention the fork=no limitation to
> only one UDP interface. Shortly, the "fork" option became outdated, causing
> more problem (coding and experience), rather than something useful.
> "debug_mode" option was introduce as better way to switch on and off a fast
> way to debug your opensips. Upon enabling the "debug_mode", opensips will
> automatically force:
>  - staying in foreground (do not detach from console)
>  - set logging level to 4 (debug)
>  - set logging to standard error
>  - enable core dumping
>  - set UDP worker processes to 2
>  - set TCP worker processes to 2
>
> 2) "debug" option replaced with "log_level"
> The global parameter "debug" was renamed "log_level" for a more accurate
> description of the setting. The values and the behavior were kept exactly as
> before. Of course the MI function "debug" was renamed to "log_level".
>
> 3) "set_debug()" function replaced with "$log_level" variable
> Th3e $log_level with behave upon writing exactly as old set_debug() - a new
> logging level will be set (as number) or the log level will be reset (for a
> NULL input). Even more, the variable will accept to be populated from other
> variables/transformations. Upon reading, the current per process logging
> level will be returned.
> Ex :
> $avp(custom_log_level) = 4;
> xlog("Current logging level is $log_level\n");
> $log_level = $avp(custom_log_level); /* switch to debug level */
> xlog("Running in debug level ($log_level) now\n");
> $log_level = NULL;
> xlog("Resetting back to logging level $log_level\n");
>
>
> Any feedback or reports are welcome !
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



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

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] t_relay and forward accepts only literals

[Daniel Moreira Yokoyama]

Thanks Liviu.

To be precise it only worked when I changed both $ru and $du. I'm not sure
if this is a good use.

Also, because of that, now, when it comes from the other endpoint I have to
force it to do the opposite (force to use UDP), what wasn't necessary
before.

Does it make any sense to you? Am I doing something that should not be
necessary to achieve this behavior?

I still have to make the stress test to see how much impact it does
comparing to the regular scenario. But I finally could make it work thanks
to you. Now I'll be able to give my customer some numbers.

Thank you very much.


Atenciosamente,

Daniel Moreira Yokoyama.
@dmyoko
http://twitter.com/dmyoko

TrafficTalks
Um podcast sobre cinema feito a partir de conversas de trânsito.
http://traffictalks.com.br


2016-03-23 13:03 GMT-03:00 Liviu Chircu :

> That's because I just copy-pasted your example and adapted it without
> thinking. It should be:
>
> force_send_socket(sctp:10.12.8.108:5060)
> $ru = "sip:" + $od + ":" + $op + ";transport=sctp";
> forward();
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 23.03.2016 15:36, Daniel Moreira Yokoyama wrote:
>
> Hi Liviu.
>
> I did the change and Opensips started up ok. But when I run my tests, no
> message get to the UAS.
>
> When I check the dump there's a lot of Destination Unreachable error for
> messages in UDP. It didn't change the transport protocol.
>
> Thanks anyway.
>
>
> Atenciosamente,
>
> Daniel Moreira Yokoyama.
> @dmyoko
> http://twitter.com/dmyoko
>
> TrafficTalks
> Um podcast sobre cinema feito a partir de conversas de trânsito.
> http://traffictalks.com.br
>
>
> 2016-03-23 6:57 GMT-03:00 Liviu Chircu :
>
>> Hi Daniel,
>>
>> It's not a limitation, but rather a performance optimization which kills
>> some of the user experience :)
>>
>> You can still do stateless forwarding to various destinations by setting
>> the Request-URI ahead:
>>
>> force_send_socket(sctp:10.12.8.108:5060)
>> $ru = "sctp:" + $od + ":" + $op;
>> forward();
>>
>> Liviu Chircu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>> On 22.03.2016 22:23, Daniel Moreira Yokoyama wrote:
>>
>> Hi everyone.
>>
>> I'm still trying to undestand why I can't use t_relay(dest) or
>> forward(dest) passing a variable instead of a literal.
>>
>> In my scenario, all I have to do is to receive a UDP message from a
>> client and relay it on SCTP to the remote endpoint.
>>
>> The way I'm trying to achieve this is by:
>>
>>  force_send_socket(sctp:10.12.8.108:5060);
>>  forward("sctp:$od:$op");
>>
>>
>> But Opensips doesn't even start up, complaining that the domain and the
>> port are invalid.
>>
>> I only works in my tests when I put the literal value  (e.g: "sctp:
>> 10.0.8.104:5060").
>>
>> I don't get it. Why would it restrict something like this?
>>
>> Anyone can give me any guidance?
>>
>> Thanks a lot.
>>
>>
>> Atenciosamente,
>>
>> Daniel Moreira Yokoyama.
>> @dmyoko
>> http://twitter.com/dmyoko
>>
>> TrafficTalks
>> Um podcast sobre cinema feito a partir de conversas de trânsito.
>> http://traffictalks.com.br
>>
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] "WARNING:core:timer_ticker: timer task already schedualed issue." Some hint ?

[Bogdan-Andrei Iancu]

Hi Rodrigo,

How many processes you have configured with opensips (the "children" 
parameter) -I assume you do run in "fork=no" mode.


Also, how many records do you have in location table ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.03.2016 23:27, Rodrigo Pimenta Carvalho wrote:


Hi.


Finally, after 2 weeks analyzing the case, I found the solution. So, I 
would like to share it here:



modparam("usrloc", "timer_interval", 300)


After configuring this timer interval equals to 300, the issue 
disappeared.



Does it make sense?


In my case, my system will never has more than 2 simultaneous calls. 
It is a project requirement. So, i guess this new timer interval will 
not affect the system.



Any comment will be very helpful!


Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
 em nome de Rodrigo Pimenta Carvalho 


*Enviado:* quinta-feira, 17 de março de 2016 14:25
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] "WARNING:core:timer_ticker: timer task 
 already schedualed issue." Some hint ?



Hi.


I was analyzing more details about this issue.

I guess the problem rises when OpenSIPS receives a SIP REGISTER 
message in some moment. When OpenSIPS receives a SIP REGISTER message, 
it has to update the location table, according to my opensips.cfg file.


See:


if (!save("location")){
sl_reply_error();
}
avp_db_query("UPDATE location SET callerName='$fn' 
WHERE id = last_insert_rowid()");



So, immediately after saving data in location table, there is a db 
query to update the same table. Maybe it is occurring some kind of 
concurrence here. So I asK:



1 - Does " " means something related to user location?


2 - Is the save function a kind of asynchronous routine?


3 - If it is asynchronous, could it be a good idea to put a 'sleep' 
between such function and the db query?



Any hint will be very helpful!


Thanks a a lot.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
 em nome de Rodrigo Pimenta Carvalho 


*Enviado:* quinta-feira, 17 de março de 2016 09:01
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] "WARNING:core:timer_ticker: timer task 
 already schedualed issue." Some hint ?


Hi.


What is the module that has Asyn radius? Do you mean module AAA-Radius?

In my script opensips.cfg I don't have any module for RADIUS. That is 
why I think my case might be little different from those seen on old 
discussions.



Have you seen any another function from OpenSIPS 2.2 that could cause 
such issue? I'm using OpenSIPS 2.2 too.



Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
 em nome de Dragomir Haralambiev 


*Enviado:* quinta-feira, 17 de março de 2016 03:35
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] "WARNING:core:timer_ticker: timer task 
 already schedualed issue." Some hint ?

Hi,

I check my log. I have same messages too.

This is appear when use Asyn radius at Opensips 2.2.

Regards,
Dragomir
*
*


2016-03-16 20:03 GMT+02:00 Rodrigo Pimenta Carvalho >:


Hi.

Since some days ago I'm analyzing an issue related to
"WARNING:core:timer_ticker: timer task  already scheduled".


Some times my OpenSIPS starts printing 'like in a loop' too many
messages about "WARNING:core:timer_ticker: timer task 
already schedualed". See below a part of my log. I have read some
discussions about it, but the information didn't help me, because
my case might be little different from those seen on old discussions.


So, I have some basic questions:


1 - What does mean "WARNING:core:timer_ticker: timer task
 already schedualed" in terms of OpenSIPS task scheduler?


2 - What should I try to avoid this issue? Some special
configuration on opensips.cfg? Or maybe give up accessing the
database sqlite?


3 - If I do nothing about it, after some time receiving such
warning, what will be the logic consequence?


Any hint will be very helpful!


Thanks alot.



-


Mar 16 17:36:27 colibri-imx6 opensips[12369]: new branch at
sip:6001@192.168.100.156:64917;transport=TCP;ob
Mar 16 17:36:27 colibri-imx6 opensips[12369]: new branch at
sip:6001@177.79.5.103:1880;transport=TCP;rin

Re: [OpenSIPS-Users] uac_registrant module

[Ovidiu Sas]

Do you see any errors in the logs?

-ovidiu
On Mar 24, 2016 04:38, "Francjos" <35...@heb.be> wrote:

> Hello,
> I've successed to laod and use uac_registrant module on one of my two
> opensips sip servers.I can register one of the two opensips sip server.
> But now, the problem i have on the other opensips, when i load
> uac_registrant module, opensips does not want to start.
> Is there something you can suggest?
> Thank you.
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/uac-registrant-module-tp7602335.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS installation: permission denied

[Bogdan-Andrei Iancu]

Hello Marty,

I see that is an OpenSIPS 1.9 - is that correct ?? Keep it mind that is 
a very, very old version which is not supported any more.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 23.03.2016 17:02, Marty van de Veerdonk wrote:

Hi,
during installation of OpenSIPS on a virtual Debian server I get the 
errors below after adding some services in the menuconfig menu.


After Exit & Save I'm not able anymore to state the menuconfig.

Anybody any idea/tips?

Regards, Marty

##

debian@vm-debian:/usr/local/src/opensips_1_9$ make menuconfig
/bin/sh: 2: cannot create cfg.tab.d: Permission denied
In file included from evi/../locking.h:68:0,
 from evi/event_interface.h:31,
 from evi/evi_modules.h:30,
 from ut.h:42,
 from socket_info.h:39,
 from cfg.y:99:
evi/../lock_alloc.h:60:2: error: #error "locking requires shared 
memory support"

 #error "locking requires shared memory support"
  ^
/bin/sh: 2: cannot create cachedb/cachedb_id.d: Permission denied
In file included from cachedb/../evi/../locking.h:68:0,
 from cachedb/../evi/event_interface.h:31,
 from cachedb/../evi/evi_modules.h:30,
 from cachedb/../ut.h:42,
 from cachedb/cachedb_id.c:30:
cachedb/../evi/../lock_alloc.h:60:2: error: #error "locking requires 
shared memory support"

 #error "locking requires shared memory support"

etc etc

cd menuconfig; make ; cd ..
make[1]: Map '/usr/local/src/opensips_1_9/menuconfig' wordt binnengegaan
gcc -o configure -g -Wall 
-DMENUCONFIG_CFG_PATH=\"menuconfig/configs/\" 
-DMENUCONFIG_GEN_PATH=\"etc/\" -DMENUCONFIG_HAVE_SOURCES=1  cfg.o 
curses.o items.o commands.o menus.o parser.o main.o -lncurses

/usr/bin/ld: cannot open output file configure: Toegang geweigerd
collect2: error: ld returned 1 exit status
Makefile:11: recept voor doel 'all' is mislukt
make[1]: *** [all] Fout 1
make[1]: Map '/usr/local/src/opensips_1_9/menuconfig' wordt verlaten
./menuconfig/configure --local
Error opening output file
Makefile:889: recept voor doel 'menuconfig' is mislukt
make: *** [menuconfig] Fout 255


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Hold call timedout on RTPproxy

[Bogdan-Andrei Iancu]

Hi Hamid,

The rtpproxy should be able to disable the rtptimeout when the call is 
on hold. So, are you sure you pass to the same rtpproxy the re-INVITE 
putting the call on hold ?


Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 22.03.2016 12:40, Hamid Hashmi wrote:
Trying to implement call hold feature and facing an issue of media 
timeout after 45 sec on RTPproxy (-T 45). If I increase the media 
timeout time on RTPporxy ( -T 3600) then it fulfill our call hold 
requirement but disturbs media timeout settings for other calls.


Is there any other solution to do this. Can I set per call media 
timeout on RTPproxy ? OR how can I initiate another media session on 
RE-INVITE of same dialog while the first session was timedout ?  OR 
any other good possible solution ?


I have also tried using two RTPproxies one with -T 45 and second with 
-T 3600. For Hold RE-INVITE i tried to send call to seconds RTPproxy 
but It didnt work :(



*/Hamid R. Hashmi/*
Software Engineer - VoIP
Vopium A/S


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] How can the global variable "listen" to listen WLAN ?

[Bogdan-Andrei Iancu]

Hi Rodrigo,

The listen param may accept as input the name of any network interface 
you may have on your system - eth0, eth1, lo, ma1, wlan1, 
etcwhatever is listed by "ifconfig" as configured interface may be used.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 22.03.2016 13:54, Rodrigo Pimenta Carvalho wrote:

Hi Alex.

I have tried something similar in the past and I have problems to execute 
opensips.
I'm not sure now if I had done exactly this way, but I will try it as your 
suggestion.

By the way, the opensips documentation doesn't say about wlan for 'listen' 
variable.
Let me try it and post the result here with more details.

Regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


De: users-boun...@lists.opensips.org  em nome de 
Alex Balashov 
Enviado: segunda-feira, 21 de março de 2016 21:03
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How can the global variable "listen" to listen 
WLAN ?

On 03/21/2016 04:36 PM, Rodrigo Pimenta Carvalho wrote:


According to the documentation "...It can be an IP address, hostname or
network interface id".

So, can I do the following configuration?


listen=tcp:wlan0:5060

Why can't you just do exactly that?

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

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

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [RELEASES] OpenSIPS 2.2 beta is scheduled for 31 of March

[Bogdan-Andrei Iancu]

Hi Dragomir,

For 1) and 2) John is working on them. The fix should be available soon.

In regards to 3), I will try to push a more detailed log to understand 
what is the problem you experience.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 22.03.2016 21:35, Dragomir Haralambiev wrote:

Hello developers,

Now I do report for all problems with Opensips 2.2.
Please fix this problems before release l 2.2.

1. Async - not make CDR (issue #799)
2. TM Request In Callback breaking async flow (issue #810)
3. When using Async I receive follow message in log:
WARNING:core:handle_timer_job: utimer job  has a 6 us 
delay in execution


Best regards,
Dragomir

2016-03-21 11:35 GMT+02:00 Bogdan-Andrei Iancu >:


Hi all,

The release of OpenSIPS 2.2 beta is scheduled for 31 of March - we
still have a tremendous amount of work to do in terms of finishing
pending new work, doing fixes, preparing documentation and many other.

So heads up , time is ticketing - if anyone has new input or
fixes/reports, now is the best time to do it.

Best regards,

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
http://www.opensips-solutions.com


___
Users mailing list
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] uac_registrant module

[Francjos]

Hello, 
I've successed to laod and use uac_registrant module on one of my two
opensips sip servers.I can register one of the two opensips sip server.
But now, the problem i have on the other opensips, when i load
uac_registrant module, opensips does not want to start.
Is there something you can suggest?
Thank you.



--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/uac-registrant-module-tp7602335.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users