[OpenSIPS-Users] Opensips and FS scenario

[Miha]

Hi to all,

I am having on one FS server about 1300 users. As I would like to 
migrate for abour 10k users from other softswitch to FS I need a little 
help about implementation.



I would like to put Opensips before FS so that Opensips would do 
load_balacing, registration, etc (more FS servers).


My question. Now FS is only one but it is doing everything 
(registration, enum, CFWD, radius, xml_cdr (posting to web server which 
parse and put it in mysql)).


What whould you suggest to use Opensips for load_balacing and 
registration and to do FS enum, radius_accounting,cdr,... or you use FS 
for media handling and put enum, radius,.. etc on Opensips?


I am doing this for the first the with opensips so that is why I am 
asking to know which is the best way.
And one another thing: what about opensips performace for about 10k 
users (registration and load_balacing).


Thanks for help!

Miha

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Re: [OpenSIPS-Users] RTPProxy Problem

[Ignacio Gonzalez]

Hi, I red the RFC 4961 Symetric RTP, i set the flags of rtp_offer "a", and
then i stop receiving packets to the udp port i use to send RTP, but the
audio is not being sent to the port that i set on the SDP message. Do i
have to run rtpproxy with any special parameter to support asymmetric rtp?


2012/11/14 Ignacio Gonzalez 

> I found where the other port is taken from. I use Java, JMF to create the
> RTP part of the UA, I create two RTPManagers, one to send and one to
> receive, and I use two different ports, one is for sending and one for
> receiving, and rtpproxy send me packets to the sending port, I don't know
> if i have to use only one RTPManager to send and receive rtppackets using
> the port I send in the SDP message. This has been always a doubt for me.
>
>
> m=audio 11670 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=sendrecv
>
> this SDP message is the one i send in the INVITE
>
> so i listen to RTP packets in the 11760 udp port
>
> and create a "rtp sender" using another random port chosen by JMF, this
> random port is where packets are sent from rtpproxy.
>
> Thanks for your help.
>
>
>
> 2012/11/14 Ignacio Gonzalez 
>
>> Here is my complete scenario:
>>
>> OPENSIPS is in a private network inside of a virtual machine
>> 192.168.1.220 and is using port 5062 on udp and tcp
>> RTPProxy is in a private network inside of the same virtual machine
>> 192.168.1220 and this is the command line I'm using to run rtpproxy:
>>
>> ./rtpproxy -l 192.168.1.220 -s udp:192.168.1.220:12400 -n tcp:
>> 192.168.1.220:12333 -u user -m 35000 -M 35200
>>
>> The host machine is 192.168.1.200
>>
>> My router create the binding rules for  public_ip:35000 ->
>> 192.168.1.220:35000to public_ip:35200 -> 192.168.1.220:35200
>>
>> I'm making a call between to UAs outside of the LAN 192.168.1.0, both
>> clients are outside of the LAN.
>>
>>
>> I tried to set the flags "i" and "e" but still not working, I also tried
>> setting the autobridged parameter for the rtpproxy module.
>>
>> Thanks for helping.
>>
>> P.D. I want to see the debug of rtpproxy but I don't know how to do that,
>> can any one help me with this?
>>
>>
>>
>>
>>
>>
>>
>>
>> 2012/11/14 Răzvan Crainea 
>>
>>>  Hi, Ignacio!
>>>
>>> So you are trying to use RTPProxy in bridge mode (between a private and
>>> a public network). This means that for INVITE you should have a public IP,
>>> and for 200OK a private IP (or the other way around, depending on your
>>> scenario). You have three possible solutions to achieve this:
>>>
>>> 1. For INVITE, call rtpproxy_offer("flags", "PUBLIC_IP") and for 200OK,
>>> rtpproxy_answer("flags", "PRIVATE_IP") (or reverse the IPs in case of a
>>> different scenario). I am not sure this works, as RTPProxy will have no
>>> idea about where exactly is your UAC/UAS - therefore it might use wrong
>>> sockets to send data (for example, it might use a PRIVATE_IP socket to send
>>> RTP to a public ip).
>>> 2. Enable the autobridge mode, by setting the 'rtpproxy_autobrige'
>>> parameter [1]. I've never used this, so I can't confirm whether it works
>>> properly or not.
>>> 3. Call the rtpproxy_offer/answer functions with the 'E' or 'I'
>>> flags[2]. This is the most reliable method I've used I can confirm it works.
>>>
>>> [1] http://www.opensips.org/html/docs/modules/1.8.x/rtpproxy#id250154
>>> [2] http://www.opensips.org/html/docs/modules/1.8.x/rtpproxy#id292744
>>>
>>> Regards,
>>>
>>> Razvan Crainea
>>> OpenSIPS Core Developerhttp://www.opensips-solutions.com
>>>
>>> On 11/14/2012 05:14 PM, Ignacio Gonzalez wrote:
>>>
>>> Ok i will check the flags, I set the domain parameter of rtpproxy_offer
>>> to PUBLIC_IP, i create the rules in my router to bind the ports 35000 to
>>> 35200 of the public_ip to the ports 35000 to 35200 in the PRIVATE_IP of
>>> rtpproxy 192.168.1.220, I'm not sure if this is correct.
>>>
>>> If i don't put the domain parameter in the rtpproxy_offer the SDP
>>> message contains the private ip of the RTPPROXY (192.168.1.220) and my
>>> clients are outside this nat.
>>>
>>> Thanks
>>>
>>>
>>> 2012/11/14 Răzvan Crainea 
>>>
  Hi, Ignacio!

 The ports you have listed in the SDP snippet belong to a single rtp
 stream - Callee-RTPProxy-Caller. You should also check the ports in the
 200OK.
 The "nortpproxy_str" parameter you are specifying is used by RTPProxy
 to determine if the SDP has to be changed, or somebody else already changed
 in the path.
 I noticed that you call rtpproxy_offer() function with two parameters:
 1. "ro" - if the SDP has a private IP, you shouldn't use the "r" flag.
 This flag is used to specify that RTPProxy should accept packages only from
 that IP, whereas in your case he will receive the packets from a public IP.
 2. "domain" - you shouldn't use this parameter unless you really know
 what you are doing. This overwrites the domain specified by RTPProxy, which
 is not always such a good idea.

 I hope these remarks will help you fix your pr

Re: [OpenSIPS-Users] RTPProxy on virtual machine

[Adrian Serafini]

Can rtpproxy run on a virtual machine?

It works on XEN.  It's a good test environment.

Adrian

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[OpenSIPS-Users] RTPProxy on virtual machine

[Ignacio Gonzalez]

Can rtpproxy run on a virtual machine?
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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Christian Cambier]

Hello Alex.



Thank you Alexe and Ali and Samy for your feedback.



> - if you register your phones on OpenSIPS there is no need to register


>them also on the PBX

Ok, I forget about the PBX for my registration. but trying to registrate
on openSIPS doesn't work.
It would be really if this problem could be solved!!!

So, I don't manage to register my sipphone.
Here's what I do.

Domain used on opensips-server (ip=10.0.4.34) is added in the mysql
table
mysql> select * from domain;

++---+-+

| id | domain| last_modified   |

++---+-+

|  1 | 10.0.4.34 | 2012-11-13 15:24:23 |

++---+-+

I have 1 user registered in OpenSIPS (in the subscribe-table, is this
the correct table to add users?)
mysql> select * from subscriber;

++--+---+--+---+
--+--+--+

| id | username | domain| password | email_address | ha1
| ha1b | rpid |

++--+---+--+---+
--+--+--+

|  1 | 5006 | 10.0.4.34 |  |   |
9d576f5a62c6bc6e46b20d1b6d9e2ad7 | c33ca435f9c3c8b87a11b013c32df2c8 |
NULL |

++--+---+--+---+
--+--+--+


Here's what I send from UAC
username/extension: 5006
no password
registrar: 10.0.4.34

REGISTER sip:10.0.4.34:5060 SIP/2.0 
Via: SIP/2.0/TCP
10.0.46.1:50666;rport;branch=z9hG4bKPj5be69cb0d02144bdb9d2777431d88264 
Route:  
Max-Forwards: 70 
From: "cid5006"
;tag=2992022966fa4dca862118d12678d4a1 
To: "cid5006"  
Call-ID: 56966ef0ba474bca919ef8a67db0659b 
CSeq: 27976 REGISTER 
User-Agent: VoxtronSipPhone_01.00.009.002 
Supported: outbound, path 
Contact: "cid5006"
;reg-id=1;+sip.instance="" 
Expires: 300 
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS 
Content-Length: 0 

here's the response

SIP/2.0 403 Preload Route denied 
Via: SIP/2.0/TCP
10.0.46.1:50666;received=10.0.46.1;rport=50666;branch=z9hG4bKPj5be69cb0d
02144bdb9d2777431d88264 
From: "cid5006"
;tag=2992022966fa4dca862118d12678d4a1 
To: "cid5006"
;tag=cb89901742d7fa34dfcce4fffa472cde.5c91 
Call-ID: 56966ef0ba474bca919ef8a67db0659b 
CSeq: 27976 REGISTER 
Server: OpenSIPS (1.8.0-notls (x86_64/linux)) 
Content-Length: 0 


The response comes from following code that is executred in my config
file 

   # preloaded route checking

if (loose_route()) {

xlog("L_ERR",

"Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci] );

if (!is_method("ACK"))

sl_send_reply("403","Preload Route denied");

exit;

}


How to solve this?

regards
CHris

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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Christian Cambier]

Hello.

>... you can simply include the ip addresses in your opensips.cfg

>file. The call will be forwarded to those ip addresses.

I know I'll have to read more about it (and I will) but in order to save
some time now

could you show some sample code about how to include the ip addresses in
a opensips.cfg?

 

regards
Chris

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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Alex Ionescu]

Hi Christian,

It would be interesting if you could describe a little bit more the 
scenario that you are trying to setup. From what you've told us until 
now this is what I understand:

- you have a PBX where you used to register your phones
- you want to use OpenSIPS as a proxy ... here things are getting a bit 
blurry


So, a few main ideas:
- if you register your phones on OpenSIPS there is no need to register 
them also on the PBX
- if you use this PBX in order to terminate calls then it is very simple 
to make it work with Opensips, this is what you must do:

a) register your phones on OpenSIPS
b) configure a routing rule in OpenSIPS to instruct OpenSIPS to 
send all calls from your phones to your PBX(s)

c) configure your PBX to accept traffic from OpenSIPS

In order to configure a routing rule use OpenSIPS Control Panel >System 
-> Dynamic Routing and do the following:

a) Use the Gateways tool to define your PBX as a gateway for OpenSIPS
b) Use the Rules tool to define rule that will send all the calls 
to your newly defined carrier

c) Apply your changes and your are ready to go - is that simple

Now, let me know if this is what you want to do and if we can assist in 
doing this.


Regards,
Alex Ionescu

On 11/15/2012 04:46 PM, Christian Cambier wrote:


Hello.

>"Your sip phones only register to the opensips servers. Your pbx dosen't

>need to sip registrations."
But what do you do then with the account-settings that were created on the PBX?

Account-settings like username and password, do you need to redefine them on 
the proxy then and not anymore on the PBX?



>Your sip phones only talk to the opensips servers, opensips then will
>forward the calls to the pbx.

But how does the opensips know to which PBX to forward the call to.
Is has to be defined somewhere? But where and how please?


regards



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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Ali Pey]

Yes, all the account info would be in opensips. Your pbx's configuration
will be quite simpler.
There are many ways that opensips can distribute the calls to PBXs. You can
use the load balancer, drouting or dispatcher modules. If you don't have
many pbxs, you can simply include the ip addresses in your opensips.cfg
file. The call will be forwarded to those ip addresses.

Have a look at registrar, authentication, load balancer, drouting,
dispatcher and dial plan modules. Also there are three sample opensips.cfg
files created by opensips installation that you can use as a reference.

Regards,
Ali Pey


On Thu, Nov 15, 2012 at 9:46 AM, Christian Cambier  wrote:

>  Hello.
>
> >"Your sip phones only register to the opensips servers. Your pbx dosen't*
> ***
>
> >need to sip registrations."
>
> But what do you do then with the account-settings that were created on the 
> PBX?
>
> Account-settings like username and password, do you need to redefine them on 
> the proxy then and not anymore on the PBX?
>
> >Your sip phones only talk to the opensips servers, opensips then will
>
> >forward the calls to the pbx.
>
> But how does the opensips know to which PBX to forward the call to.
> Is has to be defined somewhere? But where and how please?
>
> 
>
>
> regards
>
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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[SamyGo]

I really suggest you to first go through the opensips.org pages. Read some
basic tutorials from there. If possible read the opensips book too. There
areplenty of blog posts our there as well on exactly what you are saying
and want.

Just follow a few and you'll start understanding what you need to do in
order to tell opensips what to do on which conditions and where to route
the calls based on those decisions.

BR
Sammy




On Thu, Nov 15, 2012 at 7:46 PM, Christian Cambier  wrote:

>  Hello.
>
> >"Your sip phones only register to the opensips servers. Your pbx dosen't*
> ***
>
> >need to sip registrations."
>
> But what do you do then with the account-settings that were created on the 
> PBX?
>
> Account-settings like username and password, do you need to redefine them on 
> the proxy then and not anymore on the PBX?
>
> >Your sip phones only talk to the opensips servers, opensips then will
>
> >forward the calls to the pbx.
>
> But how does the opensips know to which PBX to forward the call to.
> Is has to be defined somewhere? But where and how please?
>
> 
>
>
> regards
>
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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Christian Cambier]

Hello.

>"Your sip phones only register to the opensips servers. Your pbx
dosen't

>need to sip registrations."
But what do you do then with the account-settings that were created on
the PBX?

Account-settings like username and password, do you need to redefine
them on the proxy then and not anymore on the PBX? 



>Your sip phones only talk to the opensips servers, opensips then will
>forward the calls to the pbx.

But how does the opensips know to which PBX to forward the call to.
Is has to be defined somewhere? But where and how please?




regards

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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Ali Pey]

Your sip phones only register to the opensips servers. Your pbx dosen't
need to sip registrations.
Your sip phones only talk to the opensips servers, opensips then will
forward the calls to the pbx.
The pbx only talks to the opensips servers, then opensips will send the
call to the sip phones based on the registration server.
Basically opensips would be your proxy server. It keeps and digests all the
registration info and will handle the routing between your sip
clients/trunks and your pbxs.
You need some reading and some help. This is the general idea.

Regards,
Ali Pey



On Thu, Nov 15, 2012 at 7:59 AM, Christian Cambier  wrote:

>  Hi.
>
> Here's what i try to do: I have a SipPhone with extension 5006
> Without proxy i register at a PBX with address (10.1.2.3). Ok, no problem
>
> What I'm struggling with is, how can i have this phone register at that
> PBX when every SIP request has to pass via a OpenSIPS proxy first.
> Somehow the proxy must know how to forward REGISTER messages to the PBX
> no? I'd say, It must know the existence of the PBX.
> Or don't I need the PBX for REGISTER?
> But surely, INVITE must pass the PBX no?
>
> Anyway, how do I have a cooporate OpenSIPS proxy with a PBX?
> Where to configure what?
>
> Please help!!!
>
> thx
> Chris
>
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Re: [OpenSIPS-Users] How to best check a request's method?

[Michael Renzmann]

Hi Vlad.

> You assumptions are right :)
> [...]

Ok, great :) Thanks for your confirmation and clarification.

Bye, Mike

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Re: [OpenSIPS-Users] how to Register with OpenSIPS?: help needed!!

[Christian Cambier]

Hi.

Here's what i try to do: I have a SipPhone with extension 5006
Without proxy i register at a PBX with address (10.1.2.3). Ok, no
problem

What I'm struggling with is, how can i have this phone register at that
PBX when every SIP request has to pass via a OpenSIPS proxy first.
Somehow the proxy must know how to forward REGISTER messages to the PBX
no? I'd say, It must know the existence of the PBX.
Or don't I need the PBX for REGISTER? 
But surely, INVITE must pass the PBX no?

Anyway, how do I have a cooporate OpenSIPS proxy with a PBX?
Where to configure what?

Please help!!!

thx
Chris

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Re: [OpenSIPS-Users] Dialog timeout_avp and onreply_route

[Mariana Arduini]

Hello Vlad,

Thanks for explaining, I will try to use sst module then!

Regards,
Mariana.


On Thu, Nov 15, 2012 at 8:28 AM, Vlad Paiu  wrote:

> **
> Hello,
>
> When using the SST module, refreshing in-dialog replies should work, since
> the internal callbacks are called before the dialog module updates it's
> timeout values.
> Still, is you set the AVP in the onreply_route, that is known not to work
> since the updating of timeout values is done before the onreply_route is
> triggered.
>
> indeed  this is some sort of bug, that we intend to fix in the future 1.9
> release, by triggering event routes for different events that occur for the
> dialog module.
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
>
> On 11/14/2012 07:59 PM, Muhammad Shahzad wrote:
>
> Then its a bug and need to be reported at,
>
>  http://sourceforge.net/tracker/?group_id=232389&atid=1086410
>
>  Thank you.
>
>
> On Wed, Nov 14, 2012 at 6:43 PM, Mariana Arduini 
> wrote:
>
>>
>>  Hi Muhammad,
>>
>>  "Well, i think if you have enabled onreply_avp_mode in TM module then
>> you can access and change AVP you set in route block in on_reply block. I
>> don't see any reason why you need to modify opensips source code for this."
>>  Yes, I have that enabled in tm module, and in fact I can change avp
>> values in onreply_route, from what I see in other avps we use.
>>
>>  The problem is that it seems the dialog module cannot apply the new avp
>> value to the dialog timeout. Check dialog module docs:
>>
>>
>> http://www.opensips.org/html/docs/modules/devel/dialog.html#timeout-avp-id
>>
>>  ** It may be used only in a request (initial or sequential) context**
>>
>>  Using opensipsctl fifo dlg_list, I can see that the timeout value is
>> set to 43200 seconds, which is the dafault value, even after setting the
>> timeout avp in onreply_route:
>>
>>   # opensipsctl fifo dlg_list
>> dialog::  hash=1571:444747590
>>  state:: 4
>>  user_flags:: 0
>>  timestart:: 1352912021
>>  timeout:: 1352955222
>>  callid:: 1-8696@91.190.218.180
>>  from_uri:: sip:daitan.en...@connect-lync.connect.com
>>  to_uri:: sip:lync_u...@lync.com
>>  caller_tag:: uac191.190.218.180
>>  caller_contact:: sip:10.22.136.172;transport=tcp;did=326.53fb7ac1
>>  callee_cseq:: 0
>>  caller_route_set::
>>  caller_bind_addr:: tcp:10.22.137.64:5060
>>  callee_tag:: uas291.190.218.170
>>  callee_contact:: sip:10.22.136.170;transport=tcp;did=326.7efc7653
>>  caller_cseq:: 1
>>  callee_route_set::
>>  callee_bind_addr:: tcp:10.22.137.25:5060
>>
>>  I understand the dialog module won´t apply any change in the avp from
>> onreply_route, doesn´t matter if done manually or through SST module... =/
>>
>>  Thanks again!
>> Mariana.
>>
>>
>>  On Wed, Nov 14, 2012 at 3:17 PM, Muhammad Shahzad <
>> shaherya...@googlemail.com> wrote:
>>
>>> Well, i think if you have enabled onreply_avp_mode in TM module then you
>>> can access and change AVP you set in route block in on_reply block. I don't
>>> see any reason why you need to modify opensips source code for this.
>>>
>>>  http://www.opensips.org/html/docs/modules/1.8.x/tm.html#id292825
>>>
>>>  Just change value of SST AVP as i mentioned in previous email when you
>>> get final reply from upstream. However make sure onreply_avp_mode is set as
>>> mentioned in about link.
>>>
>>>  Thank you.
>>>
>>>
>>>
>>> On Wed, Nov 14, 2012 at 5:37 PM, Mariana Arduini <
>>> marianardu...@gmail.com> wrote:
>>>
 Hi Muhammad!

  Thanks for replying.

  I looked at SST module and, at least from docs, there is no mention to
 using it for the 200 OK, only for INVITES...

  Even so, I think the problem is that the dialog module does not
 update the timeout value for 200 OK. For example, if you want to refresh
 the timeout value when you get an UPDATE message, you have to set the avp
 configured in timeout_avp modparam and then call
 match_dialog()/loose_route(), because either of these functions will
 actually update the dialog, and setting the avp without calling them will
 take no effect. I guess the same thing is happening when I set the avp
 after receiving the 200 OK.

  Any help on what should be changed in OpenSIPS code so that setting
 the avp in onreply_route will take effect on the dialog timeout?

  Thanks!
  Mariana


 On Wed, Nov 14, 2012 at 1:38 PM, Muhammad Shahzad <
 shaherya...@googlemail.com> wrote:

> Use SST module,
>
>  http://www.opensips.org/html/docs/modules/1.8.x/sst.html#id250008
>
>  Thank you.
>
>
>  On Tue, Nov 13, 2012 at 8:25 PM, Mariana Arduini <
> marianardu...@gmail.com> wrote:
>
>>  Hello all,
>>
>>  I know from the docs that dialog module param timeout_avp " ... may
>> be used only in a request (initial or sequential) context", but we
>> have a scenario where the UAS should be the session timer refresher, then
>> we can only set the 

Re: [OpenSIPS-Users] Errors in OpenSIPS

[Jorge Ortea]

Hello,

can someone said me if really modify dialog table and allow callee_contract 
field to be NULL on DB could break anything??

Thanks.
Regards.

Date: Tue, 13 Nov 2012 14:48:51 +0100
From: shaherya...@googlemail.com
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Errors in OpenSIPS

Its db schema problem. The field dialog.callee_contact is set as "NOT NULL", 
while you are getting NULL contact from upstream (most likely in BYE from 
callee side, as some old PSTN gateways running at many termination providers do 
not add "Contact" header in BYE request). Please check SIP trace and confirm or 
attach that here for our review.

For a workaround you can modify dialog table and allow callee_contract field to 
be NULL. However i am not sure if this won't break anything else.
Thank you.


On Mon, Nov 12, 2012 at 1:52 PM, Jorge Ortea  wrote:




Hello,

I keep getting these errors.

My opensips version is 1.6.4-2-tls and my mysql version is 5.1.34

Would it be better to change the version of opensips?

What can mean  "Column 'callee_contact' cannot be null" ??


Thanks.
Regards.

From: dar...@hotmail.com

To: users@lists.opensips.org
Date: Fri, 9 Nov 2012 08:30:03 +0100
Subject: Re: [OpenSIPS-Users] Errors in OpenSIPS




Hello Vlad,

You are right, I had searched for ERROR, but before each Error message I got 
this:

Nov  5 10:24:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12562]: 
CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error (1048): 
Column 'callee_contact' cannot be null 

Nov  5 10:24:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12562]: 
ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db 

Why could this be happening?


Thanks.
Regards.




Date: Thu, 8 Nov 2012 19:21:45 +0200
From: vladp...@opensips.org
To: users@lists.opensips.org

Subject: Re: [OpenSIPS-Users] Errors in OpenSIPS



  

  
  
Hello Jorge,



Do you see any other errors from the DB level in the OpenSIPS logs ?
What DB backend are you using ?

It seems that for some reason, the dialog DB inserts are failing,
but more ERROR messages should appear in such a case from the
db_mysql module..



Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com 


On 11/08/2012 07:10 PM, Ali Pey wrote:
I think it's telling you it can't add another dialog
  to db. I am almost certain that's what's happening.
  

  
  Regards,
  Ali Pey
  

  
  




  On Thu, Nov 8, 2012 at 4:22 AM, Jorge Ortea  
wrote:

  

  Hi all,



I am getting the
following errors:





Nov  7 11:47:19 hgt-tero45
/usr/local/opensips/sbin/opensips[12582]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 11:47:19 hgt-tero45
/usr/local/opensips/sbin/opensips[12548]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 11:47:20 hgt-tero45
/usr/local/opensips/sbin/opensips[12558]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 11:57:32 hgt-tero45
/usr/local/opensips/sbin/opensips[12554]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 11:57:32 hgt-tero45
/usr/local/opensips/sbin/opensips[12578]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 11:59:07 hgt-tero45
/usr/local/opensips/sbin/opensips[12556]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 11:59:07 hgt-tero45
/usr/local/opensips/sbin/opensips[12574]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 12:23:23 hgt-tero45
/usr/local/opensips/sbin/opensips[12580]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 12:23:24 hgt-tero45
/usr/local/opensips/sbin/opensips[12544]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 12:23:25 hgt-tero45
/usr/local/opensips/sbin/opensips[12572]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db 

Nov  7 12:24:47 hgt-tero45
/usr/local/opensips/sbin/opensips[12558]:
ERROR:dialog:update_dialog_dbinfo: could not add another
di

Re: [OpenSIPS-Users] dialplan table and Control Panel tool.

[Alex Ionescu]

Hi Miguel,

Can you please do a SVN checkout :
|svn co 
https://opensips-cp.svn.sourceforge.net/svnroot/opensips-cp/trunk 
/path/to/your/opensips-cp|


Then just copy your boxes and db config files from your old installation 
to this new one and you should be ready to go.


Regards,
Alex

On 11/15/2012 01:08 PM, Miguel J. López Valverde wrote:

Hi Alex,

Thankyou for your answer.

I installed it downloading the tarball file, (opensips-cp_5.0.tgz).

Rergards.



-Mensaje original-
*De*: Alex Ionescu >

*Reply-to*: OpenSIPS users mailling list 
*Para*: OpenSIPS users mailling list >

*Asunto*: Re: [OpenSIPS-Users] dialplan table and Control Panel tool.
*Fecha*: Thu, 15 Nov 2012 12:45:49 +0200

Hi,

No, don't change anything. In 5 minutes a fix will be available on SVN 
trunk. There is  a small bug in the code.

Miguel, how did you install OpenSIPS Control Panel ? SVN or tarball ?

Regards,
Alex Ionescu

On 11/15/2012 12:40 PM, Max Mühlbronner wrote:


Hi,


maybe you need to adjust memory_limit in php.inf and restart apache. 
At least this helped me a few times when i had the same problem with 
dynamic routing/dr_rules table.



Max M.


On 11/15/2012 11:23 AM, Miguel J. López Valverde wrote:


Dear Opensips lists:

I've a trouble with the OpenSips Control Panel tool when the 
dialplan table has more than 4 rows.


If this table contains up to 40,000 rows, the content is 
provided in the web tool and everything works fine but if this table 
contains a record more, the tool leaves the panel empty.


Can I configure this tool to avoid this limitation?

Thank you very much and best regards

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Re: [OpenSIPS-Users] dialplan table and Control Panel tool.

[Miguel J.]

Hi Alex,

Thankyou for your answer.

I installed it downloading the tarball file, (opensips-cp_5.0.tgz).

Rergards.



-Mensaje original-
De: Alex Ionescu 
Reply-to: OpenSIPS users mailling list 
Para: OpenSIPS users mailling list 
Asunto: Re: [OpenSIPS-Users] dialplan table and Control Panel tool.
Fecha: Thu, 15 Nov 2012 12:45:49 +0200

Hi,

No, don't change anything. In 5 minutes a fix will be available on SVN
trunk. There is  a small bug in the code.
Miguel, how did you install OpenSIPS Control Panel ? SVN or tarball ?

Regards,
Alex Ionescu

On 11/15/2012 12:40 PM, Max Mühlbronner wrote:

> Hi,
> 
> 
> maybe you need to adjust memory_limit in php.inf and restart apache.
> At least this helped me a few times when i had the same problem with
> dynamic routing/dr_rules table.
> 
> 
> Max M.
> 
> 
> On 11/15/2012 11:23 AM, Miguel J. López Valverde wrote:
> 
> > 
> > Dear Opensips lists:
> > 
> > I've a trouble with the OpenSips Control Panel tool when the
> > dialplan table has more than 4 rows.
> > 
> > If this table contains up to 40,000 rows, the content is
> > provided in the web tool and everything works fine but if this table
> > contains a record more, the tool leaves the panel empty.
> > 
> > Can I configure this tool to avoid this limitation?
> > 
> > Thank you very much and best 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



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Re: [OpenSIPS-Users] how to Register with OpenSIPS?

[Ali Pey]

What do you mean by 'Registrar Changes'?

You can basically have your sip phones register with the opensips
server(s). Your pbx would only need to know the opensips ip address. You
can have redundant opensips servers with a virtual ip address and
heartbeat. When one goes down, the other one would take over. All the
registration info would be in a database such as mysql.

Regards,
Ali Pey


On Thu, Nov 15, 2012 at 3:13 AM, Christian Cambier  wrote:

>  Hi.
>
> ** **
>
> I'm new to using SIP proxies (OpenSIPS) so maybe it is a basic question
> but anyway...
>
> ** **
>
> I have a few SIP phones that register at some PBX (say 10.2.3.4)
>
> Registration without a proxy works fine
>
> ** **
>
> What I'd like to do now is to use OpenSIPS as proxy and define the
> registrar for all my clients there.
>
> So that when the registrar changes i only have to change it in the proxy
> instead of in all the clients
>
> ** **
>
> Is that possible? if so, how do you implement it?
>
> By specifying a route in some config file? 
>
> ** **
>
> thx
>
> Chris
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] dialplan table and Control Panel tool.

[Alex Ionescu]

Hi,

No, don't change anything. In 5 minutes a fix will be available on SVN 
trunk. There is  a small bug in the code.

Miguel, how did you install OpenSIPS Control Panel ? SVN or tarball ?

Regards,
Alex Ionescu

On 11/15/2012 12:40 PM, Max Mühlbronner wrote:

Hi,


maybe you need to adjust memory_limit in php.inf and restart apache. 
At least this helped me a few times when i had the same problem with 
dynamic routing/dr_rules table.



Max M.


On 11/15/2012 11:23 AM, Miguel J. López Valverde wrote:

Dear Opensips lists:

I've a trouble with the OpenSips Control Panel tool when the 
dialplan table has more than 4 rows.


If this table contains up to 40,000 rows, the content is provided 
in the web tool and everything works fine but if this table contains 
a record more, the tool leaves the panel empty.


Can I configure this tool to avoid this limitation?

Thank you very much and best regards


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




___
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Re: [OpenSIPS-Users] dialplan table and Control Panel tool.

[Max Mühlbronner]

Hi,


maybe you need to adjust memory_limit in php.inf and restart apache. At 
least this helped me a few times when i had the same problem with 
dynamic routing/dr_rules table.



Max M.


On 11/15/2012 11:23 AM, Miguel J. López Valverde wrote:

Dear Opensips lists:

I've a trouble with the OpenSips Control Panel tool when the 
dialplan table has more than 4 rows.


If this table contains up to 40,000 rows, the content is provided 
in the web tool and everything works fine but if this table contains a 
record more, the tool leaves the panel empty.


Can I configure this tool to avoid this limitation?

Thank you very much and best regards


___
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Re: [OpenSIPS-Users] Increasing time to call new module´s method

[Vlad Paiu]

Hello,

Do you initialise the libcurl session in the mod_init or child_init ?
If you want to have each process have it's own curl session, initialise 
it in child_init, as that function is ran after forking.


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


On 11/12/2012 08:23 PM, Aldo Jose Spanghero Romao wrote:


Hi,

I´ve created a module that uses libcurl to send requests to a HTTP server.

The problem is that the time to the function 'curl_easy_perform' be 
called is increasing each time the method of the module is called.


I´ve made some profiling, and looks like 'all' opensips' processes are 
using the same instance of the module, calling the same function, and 
not each process having its own copy of the module.


This forces the next process to wait until the current one is finished 
the call to the 'curl_easy_perform' function.


Is there any way to force that each process uses its own copy of the 
module, so the calls can be performed in parallel?


Thank a lot for any tip!

Regards

Aldo


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Re: [OpenSIPS-Users] Dialog timeout_avp and onreply_route

[Vlad Paiu]

Hello,

When using the SST module, refreshing in-dialog replies should work, 
since the internal callbacks are called before the dialog module updates 
it's timeout values.
Still, is you set the AVP in the onreply_route, that is known not to 
work since the updating of timeout values is done before the 
onreply_route is triggered.


indeed  this is some sort of bug, that we intend to fix in the future 
1.9 release, by triggering event routes for different events that occur 
for the dialog module.


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


On 11/14/2012 07:59 PM, Muhammad Shahzad wrote:

Then its a bug and need to be reported at,

http://sourceforge.net/tracker/?group_id=232389&atid=1086410 



Thank you.


On Wed, Nov 14, 2012 at 6:43 PM, Mariana Arduini 
mailto:marianardu...@gmail.com>> wrote:



Hi Muhammad,

"Well, i think if you have enabled onreply_avp_mode in TM module
then you can access and change AVP you set in route block in
on_reply block. I don't see any reason why you need to modify
opensips source code for this."
Yes, I have that enabled in tm module, and in fact I can change
avp values in onreply_route, from what I see in other avps we use.

The problem is that it seems the dialog module cannot apply the
new avp value to the dialog timeout. Check dialog module docs:

http://www.opensips.org/html/docs/modules/devel/dialog.html#timeout-avp-id

** It may be used only in a request (initial or sequential) context**

Using opensipsctl fifo dlg_list, I can see that the timeout value
is set to 43200 seconds, which is the dafault value, even after
setting the timeout avp in onreply_route:

# opensipsctl fifo dlg_list
dialog::  hash=1571:444747590
state:: 4
user_flags:: 0
timestart:: 1352912021
timeout:: 1352955222
callid:: 1-8696@91.190.218.180 
from_uri:: sip:daitan.en...@connect-lync.connect.com

to_uri:: sip:lync_u...@lync.com 
caller_tag:: uac191.190.218.180
caller_contact:: sip:10.22.136.172;transport=tcp;did=326.53fb7ac1
callee_cseq:: 0
caller_route_set::
caller_bind_addr:: tcp:10.22.137.64:5060 
callee_tag:: uas291.190.218.170
callee_contact:: sip:10.22.136.170;transport=tcp;did=326.7efc7653
caller_cseq:: 1
callee_route_set::
callee_bind_addr:: tcp:10.22.137.25:5060 

I understand the dialog module won´t apply any change in the avp
from onreply_route, doesn´t matter if done manually or through SST
module... =/

Thanks again!
Mariana.


On Wed, Nov 14, 2012 at 3:17 PM, Muhammad Shahzad
mailto:shaherya...@googlemail.com>>
wrote:

Well, i think if you have enabled onreply_avp_mode in TM
module then you can access and change AVP you set in route
block in on_reply block. I don't see any reason why you need
to modify opensips source code for this.

http://www.opensips.org/html/docs/modules/1.8.x/tm.html#id292825

Just change value of SST AVP as i mentioned in previous email
when you get final reply from upstream. However make sure
onreply_avp_mode is set as mentioned in about link.

Thank you.



On Wed, Nov 14, 2012 at 5:37 PM, Mariana Arduini
mailto:marianardu...@gmail.com>> wrote:

Hi Muhammad!

Thanks for replying.

I looked at SST module and, at least from docs, there is
no mention to using it for the 200 OK, only for INVITES...

Even so, I think the problem is that the dialog module
does not update the timeout value for 200 OK. For example,
if you want to refresh the timeout value when you get an
UPDATE message, you have to set the avp configured in
timeout_avp modparam and then call
match_dialog()/loose_route(), because either of these
functions will actually update the dialog, and setting the
avp without calling them will take no effect. I guess the
same thing is happening when I set the avp after receiving
the 200 OK.

Any help on what should be changed in OpenSIPS code so
that setting the avp in onreply_route will take effect on
the dialog timeout?

Thanks!
Mariana


On Wed, Nov 14, 2012 at 1:38 PM, Muhammad Shahzad
mailto:shaherya...@googlemail.com>> wrote:

Use SST module,


http://www.opensips.org/html/docs/modules/1.8.x/sst.html#id250008

Thank you.


On Tue, Nov 13, 2012 at 8:25 PM, Mariana Arduini
mailto:marianardu...@gmail.com>> wrote:

  

Re: [OpenSIPS-Users] Can I modify register time

[Alex Ionescu]

Hi,

If you want to do this globally then, check this link (default_expires, 
min_expires, max_expires), should be really easy :

http://www.opensips.org/html/docs/modules/1.8.x/registrar.html#id249191

Regards,
Alex

On 11/15/2012 12:13 PM, Nick Chang wrote:


Hello

Can I modify register timeout??

Thanks

Nick



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[OpenSIPS-Users] dialplan table and Control Panel tool.

[Miguel J.]

Dear Opensips lists:

I've a trouble with the OpenSips Control Panel tool when the
dialplan table has more than 4 rows.

If this table contains up to 40,000 rows, the content is provided in
the web tool and everything works fine but if this table contains a
record more, the tool leaves the panel empty.

Can I configure this tool to avoid this limitation?

Thank you very much and best regards
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[OpenSIPS-Users] Can I modify register time

[Nick Chang]

Hello

 

Can I modify register timeout??

 

Thanks

Nick

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Re: [OpenSIPS-Users] How to best check a request's method?

[Vlad Paiu]

Hello,

You assumptions are right :)
From coding point of view, is_method is more flexible.
From performance point of view, is_method is faster, since at startup 
it goes through the list and if it contains only well known header 
names, at runtime it will only do binary ops for checking the method, 
instead of checking the actual method string.


Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


On 11/15/2012 10:17 AM, Michael Renzmann wrote:

Hi.

For the records and just in case anyone stumbles across my post in the
archives (feel free to ignore it otherwise):


Let's assume that I want to check whether the current request uses a given
method. The following statements would work for that:

  1. if (method == INVITE) { ... }
  2. if (method == "INVITE") { ... }
  3. if (is_method("INVITE")) { ... }

No one answered so far, so it appears to me this is more or less a
personal decision thing rather than a real performance issue. :-)

I for me decided to go with is_method(). It's widely used in the demo
configuration.

Plus: it offers more flexibility when it comes to expanding an existing
test. Say, if you wanted to expand the statements given as example above
to also test for SUBSCRIBE requests, for 1 or 2 you would end up with
something like:

   if ((method == INVITE) || (method == SUBSCRIBE)) { ... }

while with 3 it would be:

   if (is_method("INVITE|SUBSCRIBE")) { ... }

The latter appears to be more intuitive and easier to read, at least for me.

Bye, Mike

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Re: [OpenSIPS-Users] How to best check a request's method?

[Michael Renzmann]

Hi.

For the records and just in case anyone stumbles across my post in the
archives (feel free to ignore it otherwise):

> Let's assume that I want to check whether the current request uses a given
> method. The following statements would work for that:
>
>  1. if (method == INVITE) { ... }
>  2. if (method == "INVITE") { ... }
>  3. if (is_method("INVITE")) { ... }

No one answered so far, so it appears to me this is more or less a
personal decision thing rather than a real performance issue. :-)

I for me decided to go with is_method(). It's widely used in the demo
configuration.

Plus: it offers more flexibility when it comes to expanding an existing
test. Say, if you wanted to expand the statements given as example above
to also test for SUBSCRIBE requests, for 1 or 2 you would end up with
something like:

  if ((method == INVITE) || (method == SUBSCRIBE)) { ... }

while with 3 it would be:

  if (is_method("INVITE|SUBSCRIBE")) { ... }

The latter appears to be more intuitive and easier to read, at least for me.

Bye, Mike

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[OpenSIPS-Users] how to Register with OpenSIPS?

[Christian Cambier]

Hi.

 

I'm new to using SIP proxies (OpenSIPS) so maybe it is a basic question
but anyway...

 

I have a few SIP phones that register at some PBX (say 10.2.3.4)

Registration without a proxy works fine

 

What I'd like to do now is to use OpenSIPS as proxy and define the
registrar for all my clients there.

So that when the registrar changes i only have to change it in the proxy
instead of in all the clients

 

Is that possible? if so, how do you implement it?

By specifying a route in some config file? 

 

thx

Chris

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