Re: [OpenSIPS-Users] opensips crash with location data

2014-06-12 Thread Pasan Meemaduma
Hi Bogdan,

Awesome :) . Thanks for all the help. I still can't believe I just missed that 
change. :(. I took care of lot of things when I migrate the script from 1.5 -> 
1.10.



On Thursday, 12 June 2014, 17:06, Bogdan-Andrei Iancu  
wrote:
 


Hi Pasan,

Glad the problem was fixed.

Most of the modules rely on the fact the DB schema is correct
and all columns have the correct type. So, they do not actually
check that - just a few are doing.
Probably it will be a good practice to have all of them doing
that kind of checks. I will add that on the TODO list :).

Regards,
 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com___
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Re: [OpenSIPS-Users] status on t_on_branch

2014-06-12 Thread Bogdan-Andrei Iancu

Hi,

"status" is related to a reply (to see the reply status). In branch 
route you are handling a request. So checking the "status" in branch 
route is a non-sense.


Regards,

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

On 12.06.2014 16:36, spady wrote:

Hi all,
I am using opensips 1.11 and using an old code, coming from 1.8, i noticed
that (status="200") does not work.
Is that normal?
Is changed way to use it?

Regards.



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[OpenSIPS-Users] status on t_on_branch

2014-06-12 Thread spady
Hi all,
I am using opensips 1.11 and using an old code, coming from 1.8, i noticed
that (status="200") does not work.
Is that normal?
Is changed way to use it?

Regards.



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Re: [OpenSIPS-Users] Real Time CDR

2014-06-12 Thread Dan Christian Bogos

Hey Bogdan,

Since we got your attention on this one, I was wondering what do u think 
about the other proposal within this thread - http posted CDRs (via db 
module)?


I understand that the event module could shortcut processing but in my 
approach I am trying to use a built in module to have the responsibility 
of dispatching the CDRs out of Opensips (eg: an external, custom build 
socket server can easier crash or loose socket connection, plus http 
libraries are much more universal than particular protocols for events 
processing (eg: it is faster to prototype a http server than a 
particular socket protocol one).


Did u have any bad experience with db -> http conversion/performance?

Ta,
DanB

On 12.06.2014 13:45, users-requ...@lists.opensips.org wrote:

Hi Ricky,

If you want to get realtime updates on the completed calls, I suggest
using the ACC module (to generate CDRs) and have the CDR delivered as
event (via rabbitMQ for example).
Check:
  - generating CDRs -
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295374
  - the CDR event -
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295744
  - using events -
http://www.opensips.org/Documentation/Interface-Events-1-11

Regards,

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



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Re: [OpenSIPS-Users] Problem about the audio conference

2014-06-12 Thread Muhammad Shahzad
I think you have posted the problem on wrong mailing list. OpenSIPS is NOT
an MCU for A/V conferences. Also whatever problem you have seems to be
related to linphone, NOT opensips.

Thank you.



On Tue, Jun 10, 2014 at 4:00 AM, CC  wrote:

> Hi, everyone.
>
> Now I try to merge some calls into a conference, but I have some problems.
>
> The description of the problems is as flows:
>
> First
>
> In the class which named “FriendAdapter.class”, when I click the item of
> the listview which contains the information of friends, I wrote a method to
> call the selected friend’s sip: ”* LinphoneManager*.*getInstance*
> ().callTo(sipid);”.Then in the class which named “LinphoneManager.class”,
> I add the “addToConference(LinphoneCall)” in the method “callTo(String
> address)” .
>
> Then, I try to test the function. Connection can be established, but both
> the callers cannot hear to each other. And I make a LOG message in the code
> to show the size of the conference and to indicate whether the local user
> is part of the conference. The message shows the size of the conference is
> “1” and the indication whether the local user is part of the conference is
> “false”. But in the doc of Linphone, I see this:
>
> void *addToConference*(LinphoneCall
> 
>  call)
>
> Merge a call into a conference. If this is the first call that enters the
> conference, the virtual conference will be created automatically. If the
> local user was actively part of the call (ie not in paused state), then the
> local user is automatically entered into the conference. If the call was in
> paused state, then it is automatically resumed when entering into the
> conference.
>
> But in my test result, the local user wasn’t automatically entered into
> the conference. I don’t know why.
>
> Hope you put forward guidance to help me solve the problems, Thank you.
>
>
>
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Re: [OpenSIPS-Users] opensips crash with location data

2014-06-12 Thread Pasan Meemaduma
Hi Bogdan,

appologies for my late reply. I got problem with accessing my email account 
from home. :(

Thanks for the help in spotting the mistake I did :'(. I'll fix the location 
table and see how it goes. :)




On Friday, 6 June 2014, 16:58, Bogdan-Andrei Iancu  wrote:
 


no, it is not related to that :) - see my previous email.

Regards,
 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer http://www.opensips-solutions.com
On 06.06.2014 06:35, Pasan Meemaduma wrote:

Hi Bogdan,
>
>
>I just found that I have got the key constraint wrong. Will that be the issue ?
>
>
>| location | CREATE TABLE `location` (
>  `id` int(10) unsigned NOT NULL AUTO_INCREMENT,
>  `username` char(64) NOT NULL DEFAULT '',
>  `domain` char(64) DEFAULT NULL,
>  `contact` char(255) NOT NULL DEFAULT '',
>  `received` char(128) DEFAULT NULL,
>  `path` char(128) DEFAULT NULL,
>  `expires` datetime NOT NULL DEFAULT '2020-05-28 21:32:15',
>  `q` float(10,2) NOT NULL DEFAULT '1.00',
>  `callid` char(255) NOT NULL DEFAULT 'Default-Call-ID',
>  `cseq` int(11) NOT NULL DEFAULT '13',
>  `last_modified` datetime NOT NULL DEFAULT '1900-01-01
00:00:01',
>  `flags` int(11) NOT NULL DEFAULT '0',
>  `cflags` int(11) NOT NULL DEFAULT '0',
>  `user_agent` char(255) NOT NULL DEFAULT '',
>  `socket` char(64) DEFAULT NULL,
>  `methods` int(11) DEFAULT NULL,
>  `sip_instance` char(255) DEFAULT NULL,
>  `attr` char(255) DEFAULT NULL,
>  PRIMARY KEY (`id`),
>  KEY `account_contact_idx` (`username`,`domain`,`contact`)
>) ENGINE=MyISAM AUTO_INCREMENT=25375081 DEFAULT
CHARSET=latin1 |
>
>
>
>account_contact_idx is missing callid column. Will this has any relation to 
>crash ?
>
>
>
>
>On Friday, 6 June 2014, 8:48, Pasan Meemaduma  wrote:
> 
>
>
>Hi Bogdan,
>
>
>Thanks for the reply. I created the table using normal schema it looks fine to 
>me. I was refering to the old url you gave me. There in that table attr was 
>missing :)
>
>
>this is the url missing attr in location table
>http://www.opensips.org/Documentation/Install-DBSchema-1-10#AEN7254  
>
>
>below is my table schema,
>
>mysql> desc location;
>+---+--+--+-+-++
>| Field | Type | Null |
  Key | Default | Extra  |
>+---+--+--+-+-++
>| id    | int(10) unsigned | NO   |
  PRI | NULL    | auto_increment |
>| username  | char(64) | NO   |
  MUL | |    |
>| domain    | char(64) | YES 
  | | NULL    |    |
>| contact   | char(255)    | NO  
  | | |    |
>| received  | char(128)    | YES 
  | | NULL    |    |
>| path  | char(128)    | YES 
  | | NULL    |    |
>| expires   | datetime | NO  
  | | 2020-05-28 21:32:15 |    |
>| q | float(10,2)  | NO  
  | | 1.00    |    |
>| callid    | char(255)    | NO  
  | | Default-Call-ID |    |
>| cseq  | int(11)  | NO  
  | | 13  |    |
>| last_modified | datetime | NO  
  | | 1900-01-01 00:00:01 |    |
>| flags | int(11)  | NO  
  | | 0   |    |
>| cflags    | int(11)  | NO  
  | | 0   |    |
>| user_agent    | char(255)    | NO  
  | | |    |
>| socket    | char(64) | YES 
  | | NULL    |    |
>| methods   | int(11)  | YES 
  | | NULL    |    |
>| sip_instance  | char(255)    | YES 
  | | NULL    |    |
>| attr  | char(255)    | YES 
  | | NULL    |    |
>+---+--+--+-+-++
>
>
>attr holds NULL values for all entries.
>
>do I have to run below in gdb after running bt
  full right ?
>
>this is what i got
>
>(gdb) f 1
>#1  0x7f64823a60a2 in dbrow2info
  (contact=0x7fffe0246180, vals=0x7f648287a260)
  at udomain.c:306
>306    in udomain.c
>(gdb) p (vals+6)->val.string_val
>$2 = 0x0
>(gdb) 
>
>
>-

Re: [OpenSIPS-Users] Real Time CDR

2014-06-12 Thread Bogdan-Andrei Iancu

Hi Ricky,

If you want to get realtime updates on the completed calls, I suggest 
using the ACC module (to generate CDRs) and have the CDR delivered as 
event (via rabbitMQ for example).

Check:
- generating CDRs - 
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295374
- the CDR event - 
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295744
- using events - 
http://www.opensips.org/Documentation/Interface-Events-1-11


Regards,

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

On 11.06.2014 22:26, Ricky Keele wrote:
Do you have to refresh or does it just automatically display after the 
end of each call?


Thanks
On 6/11/2014 2:01 PM, Tito Cumpen wrote:

Ricky Keele,

You can use a variety of things depending on requirements. I use 
db_http conjunction with the acc module. This means you can post cdr 
 to an http source.



Thanks,
Tito


On Wed, Jun 11, 2014 at 1:47 PM, Ricky Keele 
mailto:rjke...@digitalipvoice.com>> wrote:


No I need to display in real time as soon as the call is completed.

Thanks,



On 6/10/2014 3:14 PM, Schneur Rosenberg wrote:


Do u mean live calls while the call is still active? If that's
what you need, I did it by querying the dialog table.

On Jun 10, 2014 10:56 PM, "Ricky Keele"
mailto:rjke...@digitalipvoice.com>>
wrote:

Hello,

Could someone tell me what is the best way to generate a
real time CDR for display to screen. Need to display 1 line
CDR showing both legs of call.

Thanks,

Ricky

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Re: [OpenSIPS-Users] opensips crash with location data

2014-06-12 Thread Bogdan-Andrei Iancu

Hi Pasan,

Glad the problem was fixed.

Most of the modules rely on the fact the DB schema is correct and all 
columns have the correct type. So, they do not actually check that - 
just a few are doing.
Probably it will be a good practice to have all of them doing that kind 
of checks. I will add that on the TODO list :).


Regards,

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

On 12.06.2014 06:53, Pasan Meemaduma wrote:


Hi Bogdan.

I can confirm the opensips crash stopped after I used the correct 
schema. Is it possible to add better erorr handling to such situation 
? or is it a overhead ?


On Wednesday, 11 June 2014, 7:46, Pasan Meemaduma  
wrote:






Hi Bogdan,

appologies for my late reply. I got problem with accessing my email 
account from home. :(


Thanks for the help in spotting the mistake I did :'(. I'll fix the 
location table and see how it goes. :)




On Friday, 6 June 2014, 16:58, Bogdan-Andrei Iancu 
 wrote:



no, it is not related to that :) - see my previous email.

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


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Re: [OpenSIPS-Users] RTPproxy errors

2014-06-12 Thread Răzvan Crainea

Hi, Rick!

I was asking for RTPProxy debugs, not OpenSIPS. Can you provide it?

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 06/06/2014 03:42 PM, Rik Broers wrote:

I've sent you some Debug logs on a separate mail.

Kind Regards,
Met vriendelijke groet,

Rik Broers



-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 
Namens Razvan Crainea
Verzonden: vrijdag 6 juni 2014 13:37
Aan: users@lists.opensips.org
Onderwerp: Re: [OpenSIPS-Users] RTPproxy errors

Hi, Rik!

The R flag indicates that the IP advertised in SDP should be trusted.
Can you increase the debugging level of rtpproxy to DEBUG? This way we can see 
exactly what commands are sent by OpenSIPS and check what's wrong with them.

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 06/06/2014 12:37 PM, Rik Broers wrote:

These errors suddenly appear after an update from 1.8 to 1.10.1

Script is unchanged.

Audio/RTP seems to work in almost all the cases that this error comes.

ERROR:rtpproxy:force_rtp_proxy: can't extract reply destination from
transaction/reply_via2

RTPProxy logs the following:

rtpproxy[8051]: ERR:handle_command: unknown command modifier `R'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `1'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `9'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `2'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `.'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `1'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `6'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `.'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `1'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `.'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `2'

rtpproxy[8051]: ERR:handle_command: unknown command modifier `7'

rtpproxy[8051]: ERR:handle_command: getaddrinfo: Address family for
hostname not supported

What strikes me is that one character is missing everytime this comes
up, in the second octet in all cases this is logged the last number is
missing.

In the Traces the SDP is complete.

What does the R mean? I’m not using recording from my script in any way.
I use RIEW/REIW combinations on rtpproxy_offer and rtpporxy_answer
which have always worked before.

Why does it throw these errors now? And what can I do to fix them.

Kind regards,

Vriendelijke groet,

*Rik Broers*
Voice Engineer



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Re: [OpenSIPS-Users] OpenSIPScrashing after specific interval of time

2014-06-12 Thread Bogdan-Andrei Iancu

Hi Juned,

See http://www.opensips.org/Documentation/TroubleShooting-Crash - try 
all the suggestions, maybe you will get a core file at the end


Regards,

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

On 12.06.2014 08:21, Juned Khan wrote:

double checked the all possible logs but didn't get anything suspected.
also tried to disable some scripts but still having same problem,

It crashes every four hour almost.


On Tue, Jun 10, 2014 at 9:01 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


checked all possible log file ? syslog, messages, etc ?

Regards,

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

On 09.06.2014 08:42, Juned Khan wrote:

nope I don't see such messages in Log, also I confirmed about
scripts,
but i don’t have any scripts which causing this issue.


On Thu, Jun 5, 2014 at 9:35 PM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

btw, do you see message from kernel like "opensips[xxx]:
segfault at X ip ." ?

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

On 05.06.2014 15:03, Bogdan-Andrei Iancu wrote:

Those are just memory reports (on usage) and it is done at
shutdown - nothing unusual.

Are you sure your opensips is not getting terminated
(stopped) by mistake by other scripts on your server ? I see
no indication of a crash so far.

Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05.06.2014 14:59, Juned Khan wrote:

Hi Bogdan

No I don't see any other signal than 15.
I was little bit curious about this logs

Jun 5 06:39:57 proxy02 opensips[24206]: used= 374656,
used+overhead=453616, free=1722496
Jun 5 06:39:57 proxy02 opensips[24206]:  max used
(+overhead)= 453616
Jun 5 06:39:57 proxy02 opensips[24206]: dumping free list:
Jun 5 06:39:57 proxy02 opensips[24200]:  max used
(+overhead)= 453616
Jun 5 06:39:57 proxy02 opensips[24180]: used= 334936,
used+overhead=417856, free=1762216
Jun 5 06:39:57 proxy02 opensips[24200]: dumping free list:
Jun 5 06:39:57 proxy02 opensips[24180]:  max used
(+overhead)= 421648
Jun 5 06:39:57 proxy02 opensips[24199]: TOTAL: 4 free
fragments = 1781264 free bytes
Jun 5 06:39:57 proxy02 opensips[24179]:  max used
(+overhead)= 405112
Jun 5 06:39:57 proxy02 opensips[24197]:  max used
(+overhead)= 395824
This is the only log when OpenSIPS crashes, which quite different then 
normal logs.
And it looks like memory problem to me but I am not sure, you better 
know about this.


On Thu, Jun 5, 2014 at 5:12 PM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Juned,

Your log (as per IRC discussion ) shows only signal 15
(which is SIG TERM) when doing normal shutdown - do you
have any other "signal" reported in the logs ?

Regards,

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

On 05.06.2014 09:39, Juned Khan wrote:

Hi All.

Since last few months I am facing very strange issue
of OpenSIPS, its crashing after specific interval of time.

I have enabled core dump parameters but dump is not
generating. it seems some memory related problem. I
hope someone expert can give me idea through this. how
to solve this issue.

I am using

OpenSIPS 1.7.1
Debian squeeze 6.0.9
RAM : 16GB


Here is the opensips log

http://pastebin.com/5YnvF6ub

-- 
Thanks,

Juned Khan



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Juned Khan




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-- 
Thanks,

Juned Khan
iNextrix Technologies Pvt Ltd.
www.inextrix.com 





--
Thanks,
Juned Khan
iNextrix Technologies Pvt Ltd.
www.inextrix.com 


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Re: [OpenSIPS-Users] Route RTP packets through

2014-06-12 Thread Răzvan Crainea

Hi, Kaushik!

Indeed, as Salman pointed out, you should not use the "r" flag unless 
you really want to use the private IP in the SDP for RTP.


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 06/12/2014 08:44 AM, Salman Zafar wrote:

Kaushik,

  'r' means to trust the IP in SDP, so if it is private IP it will take
private IP for media. Try changing connection string with 'c' flag also
owner 'o' as per your requirement based on source public IP.

http://www.opensips.org/html/docs/modules/devel/rtpproxy.html

/r/ - flags that IP address in SDP should be trusted. Without this flag,
rtpproxy ignores address in the SDP and uses source address of the SIP
message as media address which is passed to the RTP proxy.


On Thu, Jun 12, 2014 at 10:32 AM, kaushik parmar mailto:androidj...@gmail.com>> wrote:

Hello Razvan


Yes i am using "r" as rtpproxy_offer("r","xx.xx.xx.xx");


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M. Salman Zafar

VoIP Professional



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Re: [OpenSIPS-Users] RabbitMQ Timer Route Not Working

2014-06-12 Thread Răzvan Crainea

Yes, subscribing with expiration 0 unsubscribes the rabbitmq node.

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 06/11/2014 08:08 PM, Kneeoh wrote:

Is there an unsubscribe option? I'm going to try removing haproxy and just let 
the vip be shared among the rabbit cluster nodes and see if that tricks it.



On Tuesday, June 10, 2014 9:01 AM, Răzvan Crainea  wrote:
Hi, Kneeoh!

I think that the only solution that would work properly was your first
approach. However, since this is not yet implemented, all I can think of
is an external process that periodically test if the node is up. If it
is not, unsubscribe it and re-subscribe the second node.

PS: I haven't really used haproxy so I have no idea how it works.

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com




On 06/06/2014 06:15 PM, Kneeoh wrote:

Hi Razvan, thank you for the reply. I don't necessarily need expiration of 
subscriptions to rabbit or the timer route per se. I'm just trying to figure 
out (with the existing capabilities) how to make opensips fail to another 
member in the rabbit cluster in the event that the current node dies. My first 
thought was that you could simply stack entry points like:

subscribe_event("E_ACC_CDR", 
"rabbitmq:rabbitmq:rabbit@192.168.2.30;rabbitmq:rabbit@192.168.2.31;rabbitmq:rabbit@192.168.2.33/cdr1")

However, it sounds like that's not in the present implementation of the rabbit 
module.

So my second thought was to trick opensips and put HAProxy between it and 
Rabbit, which works, but if I fail an HAProxy via corosync to the other HAproxy 
something with the subscription breaks. Since it looked like the two options 
were either put the subscribes in the startup route (only happens once so 
probably won't failover) OR use the timer route to subscribe (which is what i'm 
doing) I figured that in the event of an HAProxy failure, I might miss a few 
messages but on the next timer fire opensips would resubscribe to haproxy which 
would relay that to the appropriate rabbit server (I haven't failed over any 
rabbit servers in this scenario so haproxy2 is talking to the same rabbit 
server as haproxy1. All i'm doing is killing haproxy1 right now and letting the 
VIP go to haproxy2). However it doesn't look like this is working and I can't 
tell if its because the subscription isn't happening, OR it is happening but 
opensips sees it already exists in the
subscribers list and does nothing (I think this is the case). If this IS 
the case perhaps a solution would be to kill the subscriber entry on new 
subscribe. If I'm way off, let me know, I'd really like to figure this out. Am 
I going about this all wrong? How would you handle a rabbit node failure?

Regards




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