Re: [OpenSIPS-Users] Question about AUTH in Deregister and Update

[Alex Balashov]

Raúl Alexis Betancor Santana wrote:

> You could check for the presecense of the expire header and then do no ask 
> for 
> AUTH ... but that will be a very, very, very, very, very, very, very, bad 
> idea.

Framed in this perspective, it seems that it is fully beyond human 
comprehension to entertain an idea more bad than this one!  :-)

-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [OpenSIPS-Users] Question about AUTH in Deregister and Update

[Raúl Alexis Betancor Santana]

On Monday 16 November 2009 03:08:50 jia li wrote:
> Hi,
>
> Is there any possibility to disable authentication in DEREGSITER and update
> the register flow in OpenSIPS server, and maintain authentication in
> REGISTER flow?
> Any configuration needed? Thanks!

Better if you try to explain what you need ... because a "DEREGISTER" doesn't 
exits as it, it's a REGISTER with contact info and expire time set to 0, so 
in fact it's a REGISTER request.

You could check for the presecense of the expire header and then do no ask for 
AUTH ... but that will be a very, very, very, very, very, very, very, bad 
idea.

-- 
Raúl Alexis Betancor Santana
Dimensión Virtual

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[OpenSIPS-Users] Question about AUTH in Deregister and Update

[jia li]

Hi,

Is there any possibility to disable authentication in DEREGSITER and update
the register flow in OpenSIPS server, and maintain authentication in
REGISTER flow?
Any configuration needed? Thanks!

BRs,
Jasper
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[OpenSIPS-Users] B2bua - possibly related question...

[Jeff Kronlage]

How is the destination route set with the B2BUA?  For this scenario, I
have only one gateway and do not require an LCR or drouting-style
system.

 

I understand that separate functions are required, which I am using.
However, setting $du or using t_relay("destination.com:5060"); doesn't
seem to work.

 

Thanks,

 

Jeff K 

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Re: [OpenSIPS-Users] INVITE not forwarded, call fails

[Bogdan-Andrei Iancu]

Or maybe the INVITE from the biascica to yhe asymmetric is not going 
through because of the NAT (maybe the client is doing a poor job in 
keeping the pinhole opened in the nat). So, as time as the call is 
originated from behind the nat, it will work, but when  from outside, it 
will not be able to traverse the nat and reach the client.

Regards,
Bogdan

Raúl Alexis Betancor Santana wrote:
> On Sunday 08 November 2009 15:47:38 lorenzo wrote:
>   
>> On 04/11/09 13:41, lorenzo wrote:
>> 
>>> On 04/11/09 13:32, Iñaki Baz Castillo wrote:
>>>   
 El Miércoles, 4 de Noviembre de 2009, lorenzo escribió:
 
> and this is the wireshark capture of the "conversation":
> http://pastie.org/683069
>   
 this shows a call which is cancelled (CANCEL) and later a call which
 receives 180 Ringing,

 It's difficukt to inspect  trace if you don't say what exactly occurs in
 that trace,
 
>>> hi Iñaki, thanks for your interest!
>>>   
>> guys, nobody got any advice on where to look for to solve this problem?
>> if anybody needs more info, just let me know!
>> 
>
> What I see from you trace, (better if next time you put a ngrep-sip trace 
> ..), 
> is:
>
> Apple DLink  GW
> INVITE  ->
>  <- 100 Trying
> CANCEL->
>  <- 200 Canceling
>  <- 486 Request Canceled
> ACK   ->
> -
> New INVITE ->
> <-  100 Trying
> [the next is supposed, not in trace]
>   INVITE  ->
>   <-  100 Trying
> [... end supposing]
><- INVITE
> <-  INVITE
> 180 Ringing ->
>
>
> Maybe you are rerouting the second invite back to your UAC, or you do some 
> aliasing ... or something similar.
>
>   


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Re: [OpenSIPS-Users] Modify SDP

[Iñaki Baz Castillo]

El Domingo, 15 de Noviembre de 2009, Bogdan-Andrei Iancu escribió:
> Hi Iñaki,
> 
> dropping in a bit later :).
> 
> What are talking about here is about a proxy "influencing" the codec
> negotiation and not about a proxy getting into a proxy negotiation  -
> the ops we have here are just for removing codecs and changing the
> priorities - these kinds of ops does not break the negation between the
> end parties, but just influence it. The actual decisions about codecs is
> still left to end parties.
> 
> There are no ops like adding new codecs in the SDP - this kind of ops
> will indeed break the negotiation and proxy should not do it.

Yes, it makes sense in order to force a light codec and so, sure :)


-- 
Iñaki Baz Castillo 

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Re: [OpenSIPS-Users] Modify SDP

[Bogdan-Andrei Iancu]

Hi Iñaki,

dropping in a bit later :).

What are talking about here is about a proxy "influencing" the codec 
negotiation and not about a proxy getting into a proxy negotiation  - 
the ops we have here are just for removing codecs and changing the 
priorities - these kinds of ops does not break the negation between the 
end parties, but just influence it. The actual decisions about codecs is 
still left to end parties.

There are no ops like adding new codecs in the SDP - this kind of ops 
will indeed break the negotiation and proxy should not do it.

Regards,
Bogdan

Iñaki Baz Castillo wrote:
> El Miércoles, 4 de Noviembre de 2009, Victor Pascual Avila escribió:
>   
>> On Wed, Nov 4, 2009 at 3:43 PM, Iñaki Baz Castillo  wrote:
>> 
>>> El Miércoles, 4 de Noviembre de 2009, Jeff Pyle escribió:
>>>   
 Hello,

 Saul is absolutely correct.  There are some amazing functions in the
  textops module that will permit you to completely destroy the SDP.  No
  better way to learn.
 
>>> When a SIP proxy left being a SIP proxy?
>>>   
>> At the very right moment that fundamentalism was replaced by
>> pragmatism and operational requirements.
>> 
>
> Thanks for your reply. I need a voicemail system for my opensips, could you 
> please code such a module? Also a IVR module would be great.
>
> XD
>
>   


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Re: [OpenSIPS-Users] dialog module errors

[Mouncif Benniane]

Bogdan,

I figured out the issue, you are right the errors that were generated
because Mysql client I have installed was not compiled properly with
opensips (version 5.1).
thnanks

On Sun, Nov 15, 2009 at 1:50 PM, Bogdan-Andrei Iancu  wrote:

> Hi Mouncif,
>
> Do you get db errors only for the DIALOG module? do you have other
> modules using DB perfectly working?
>
> Have you tried to re-create the dialog table ?
>
> The error is generated directly by the mysql driver (from opensips)
> because it does not recognize the types of the columns from the dialog
> table - are you sure you are using the structure as provided by the
> opensips db installer ?
>
> Regards,
> Bogdan
>
> Mouncif Benniane wrote:
> > I get this errors when starting opensips 1.6 with dialog module
> > enabled ( centos 5.2)
> >
> > WARNING:db_mysql:db_mysql_get_columns: unhandled data type column
> > (hash_id) type id (154246320), use DB_STRING as default
> > WARNING:db_mysql:db_mysql_get_columns: unhandled data type column
> > (dialog) type id (154246392), use DB_STRING as default
> > WARNING:db_mysql:db_mysql_get_columns: unhandled data type column
> > (dialog) type id (154246432), use DB_STRING as default
> > WARNING:db_mysql:db_mysql_get_columns: unhandled data type column
> > (opensips) type id (154246512), use DB_STRING as default
> > WARNING:db_mysql:db_mysql_get_columns: unhandled data type column
> > (def) type id (154246544), use DB_STRING as default
> >
> >
> > my config looks like this:
> >
> > #dialog
> > modparam("dialog", "profiles_with_value", "inbound")
> > modparam("dialog", "dlg_flag", 4)
> > modparam("dialog", "db_url", "mysql://root:passx...@localhost/opensips")
> > modparam("dialog", "db_mode", 1)
> > modparam("dialog", "db_update_period", 60)
> > modparam("dialog", "table_name", "dialog")
> >
> >
> > opensips never starts and exit. any ideas?? BTW same config and
> > version worked on centos 4.6.
> >
> >
> > Thanks
> >
> > 
> >
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> >
>
>
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Re: [OpenSIPS-Users] Next OpenSIPS Webinar Schedule?

[Bogdan-Andrei Iancu]

Hi,

I will work out the topic and the date for the next webinar  with Flavio 
- I would suggest as topic "Script Variables in OpenSIPS" for the next week?

Regards,
Bogdan

bay2x1 wrote:
> I have noticed that for the month of October and November there was no
> schedule for a webinar.  I know that you are all busy, but this is only my
> way of getting first hand information regarding OpenSIPS.  I am hoping that
> the webinars continue.
>
> -
> http://opensips.blogspot.com http://opensips.blogspot.com 
>   


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Re: [OpenSIPS-Users] Question: presence:destroy: destroy module ..

[Bogdan-Andrei Iancu]

Hi,

Those are not errors :)) the text says very clear "INFO" or "NOTICE" 
- those are normal logs during shutdown...nothing to solve :)

Regards,
Bogdan

Ȩΰ wrote:
> I installed opensips with module presence.when I start opensips,there 
> seems is some errors about "NOTICE:presence:destroy: destroy module ..."
> Error report as follows:
> Nov 6 17:33:00 debian /usr/local/sbin/opensips[3300]: 
> NOTICE:core:main: version: opensips 1.6.0-notls (i386/linux)
> Nov 6 17:33:00 debian /usr/local/sbin/opensips[3300]: INFO:core:main: 
> using 32 Mb shared memory
> Nov 6 17:33:00 debian /usr/local/sbin/opensips[3300]: INFO:core:main: 
> using 1 Mb private memory per process
> Nov 6 17:33:00 debian /usr/local/sbin/opensips[3300]: 
> NOTICE:signaling:mod_init: initializing module ...
> Nov 6 17:33:00 debian /usr/local/sbin/opensips[3300]: 
> INFO:sl:mod_init: Initializing StateLess engine
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:tm:mod_init: TM - initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:maxfwd:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:usrloc:ul_init_locks: locks array size 512
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:registrar:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:textops:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:xlog:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:acc:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:auth:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:auth_db:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:alias_db:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> NOTICE:presence:mod_init: initializing module ...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:presence_mwi:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> INFO:presence_xcapdiff:mod_init: initializing...
> Nov 6 17:33:01 debian /usr/local/sbin/opensips[3300]: 
> NOTICE:presence:destroy: destroy module ...
> how can I resolve this problem?
> 
>
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Re: [OpenSIPS-Users] Load balancer ORed resource types

[Bogdan-Andrei Iancu]

Hi Taner,

I would suggest a different approach - as from LB point of view you do 
not care too much if it VM or CONF - what is important is that each of 
your box has only on channel, so I would define a resource the number of 
channels each box can handle .

++--++-+
| id | group_id | dst_uri| resources   |
++--++-+

|  1 |  555 |  sip:serv...@host1.net  | 
channel=1   |
|  2 |  555 |  sip:serv...@host2.net  | 
channel=1   |

++--++-+


and do (when routing a VM or conference call) :
'load_balance(.., "channel");'

Regards,
Bogdan


Taner Sener wrote:
> Hi,
>
> I'm using load balancer module on 1.6.0-notls (i386/linux) to balance 
> incoming sip calls to sip clients. The case is; my sip clients are 
> capable of serving multiple types of calls. But not in the same time 
> like in the definition "vm=1;conf=1". Only one of them is available at 
> that moment. They can process either a vm call, either a conf call; 
> not one vm call and one conf call. Is there a way to define it?
>
> ++--++-+
> | id | group_id | dst_uri| resources   |
> ++--++-+
>
> |  1 |  555 |  sip:serv...@host1.net  | 
> vm=1; conf=1|
> |  2 |  555 |  sip:serv...@host2.net  | 
> vm=1; conf=1|
>
> ++--++-+
>   
> Current 'resources' field of load_balancer table ANDs resource types 
> given with semi-colon and I when I invoke
>
> 'load_balance(.., "vm")'
>
> and
>
> 'load_balance(.., "conf")'
>
> second load balanced call receives busy and balancing fails there 
> because that client is serving to vm call. How can I configure 
> load_balanced module to select it right?
>
> Thanks,
> Taner
> 
>
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Re: [OpenSIPS-Users] TLS about Openxcap

[Bogdan-Andrei Iancu]

Hi,

Can you check at network level if the https support is really enabled? 
Use "netstats -talnp" to see that.

Regards,
Bogdan

Ȩΰ wrote:
> The system log has show TLS Started~ But where is the error from?
> Nov 9 17:08:37 debian openxcap[2556]: Main loop terminated.
> Nov 9 17:08:38 debian openxcap[3266]: Log opened.
> Nov 9 17:08:38 debian openxcap[3266]: Starting OpenXCAP 1.1.2
> Nov 9 17:08:38 debian openxcap[3266]: Trusted peers: 
> 192.168.190.129/192.168.190.2/127.0.0.1/192.168.1.213 
> 
> Nov 9 17:08:38 debian openxcap[3266]: xcap.server.HTTPFactory starting 
> on 80
> *Nov 9 17:08:38 debian openxcap[3266]: TLS started*
> Nov 9 17:08:38 debian openxcap[3266]: 13 xcap documents in the database
>
> 2009/11/9 权伟 mailto:quanwe...@gmail.com>>
>
>
>   
>
>
>   
> I have installed openxcap and opensips successfully in debian.
> Everything seems run well except one little problem...
> that is :
> xcap root = https://192.168.190.129/xcap-root/
> 
> 
> 
> debian:~# xcapclient -i resource-lists.xml put
> put
> 
> https://192.168.190.129/xcap-root/resource-lists/users/sip:1...@localhost/index
> FATAL: URLError: 
> 
> But when I modify XCAP URI is http://192.168.190.129/xcap-root/,
> that is ok~
> 
> debian:~# xcapclient -i resource-lists.xml put
> put
> 
> http://192.168.190.129/xcap-root/resource-lists/users/sip:1...@localhost/index
> 200 OK
> etag: "3533027a0fa9bb6b18851e8ebc3e9e8c"
> content-type: application/resource-lists+xml
>   
>
>   
>
> -
>
> what's wrong with it?
>
> It seems some thing wrong with TLS, what can I doo
>
>   
>
>
>
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Re: [OpenSIPS-Users] ratelimit per method per subscriber

[Bogdan-Andrei Iancu]

Hi Jeff,

Maybe you can use the memcache support (with variable names) to keep the 
information you need for counting and limiting traffic based on whatever 
you want.

So, using variables in cache, you can implement (directly in script) 
some custom limiting alg.

Regards,
Bogdan

Jeff Pyle wrote:
> Ovidiu,
>
> I'd like to be able to impose a limit on each of my subscribers.  I could
> have thousands of subscribers.  Apparently this module isn't appropriate for
> that kind of application.
>
>
> - Jeff
>
>
> On 11/9/09 12:28 PM, "Ovidiu Sas"  wrote:
>
>   
>> Hello Jeff,
>>
>> Since you want to limit only the INVITEs, just go with the forced
>> limiting by forcing a known pipe:
>> http://www.opensips.org/html/docs/modules/devel/ratelimit.html#id228413
>> Define a pipe for each IP and the map the IP to a pipe before ratelimiting.
>> The only thing that you need to be aware are the maximum number of pipes:
>> http://www.opensips.org/html/docs/modules/devel/ratelimit.html#id271306
>> If you need more then 16 pipes, you will need to recompile.
>>
>> Hope this helps.
>> Regards,
>> Ovidiu Sas
>>
>> On Mon, Nov 9, 2009 at 11:31 AM, Jeff Pyle  wrote:
>> 
>>> Hello,
>>>
>>> It appears our options today for limiting traffic quantity into Opensips are
>>> the pike and ratelimit modules.  As my oversimplified minds understand it,
>>> pike works by limiting all traffic based on source IP.  Ratelimit works by
>>> limiting all traffic based on method.
>>>
>>> Is there a way with ratelimit to limit traffic based on method per source
>>> IP?  For example, can I specify a usr_preference that indicates calls
>>> (INVITEs) per second per subscriber?
>>>
>>> I've got a functioning ratelimit configuration to limit the amount of
>>> INVITEs into the proxy based on the example configurations but I don't see
>>> how to do it based on subscriber/source IP like pike does.
>>>
>>>
>>> Regards,
>>> Jeff
>>>
>>>
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>
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Re: [OpenSIPS-Users] Centralized authentication service

[Bogdan-Andrei Iancu]

Hi Jonathan,

If you have a SIP proxy between the client and the registrar, that proxy 
should add a PATH header (if you want to keep that proxy in the path for 
the calls).

Also, when you say that the calling is not working, could you post the 
call flow you expect and what exactly is going on (how the INVITE is 
routed and where it fails) ?

Regards,
Bogdan

Jonathan González wrote:
> Hi there,
>
> First of all I would like to say that I solved the problem I had with 
> RTP traffic... it was not routed directly between clients due to the 
> xlite version I was using (4.0 beta). Once I changed the client 
> version everything was working properly.
>
> I would like to configure a new scenario: 2 OpenSIPS acting as SIP 
> Proxy and one acting as SIP registrar. The idea is for example 
> configure a centralized authentication service for users from 
> different offices.
>
> If a user from one office where the OpenSIPS is configured as SIP 
> proxy launch his client, that proxy should redirect that REGISTER to 
> the registrar server. I would like to know what is the appropriate way 
> to configure an OpenSIPS like this. I have been able to authenticate a 
> user in this scenario using applying the method 
> set_host_port(ipRegistrar:5060) to all REGISTER requests that the 
> proxy receives, but then I am unable to call any other user 
> registered... So I don't know if the registration is right and I have 
> to modify something for the INVITE method or that's not the way to do it.
>
> Any suggestion is appreciated.
>
> Thanks in advance,
> Jonathan
>
> -- 
> Personal webpage - www.jonbaraq.eu
> 
>
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Re: [OpenSIPS-Users] BYE does not work / Opensips stops working

[Bogdan-Andrei Iancu]

Hi Alberto,

First, please do not post scripts as nobody will do script reviewing or 
analysis.

Now, going back to your issues:

 - when opensips receives the BYE, disregarding the warning messages 
from dialog module, is the BYE request routed out opensips to the 
correct destination (the other party) ? Can you check this by taking a 
network capture with ngrep or wireshark.

- regarding the "blocking" during INVITEs.maybe enabling full 
logging (debug=4) and posting the logs starting with the first BYE will 
help in seeing what opensips is internally doing.

Regards,
Bogdan

Alberto Listas wrote:
> Hello,
>
> I am installing the new 1.6 and getting a strange behavior.
> When the call ended on one side the other side does not get disconnected,
> and the log shows:
>
> Nov 10 18:40:38 en4 /sbin/opensips[20610]: WARNING:dialog:dlg_onroute: 
> unable to find dialog for BYE with route param 'a94.cd4c751'
> Nov 10 18:40:39 en4 /sbin/opensips[20609]: WARNING:dialog:dlg_onroute: 
> unable to find dialog for BYE with route param 'a94.cd4c751'
> Nov 10 18:40:41 en4 /sbin/opensips[20611]: WARNING:dialog:dlg_onroute: 
> unable to find dialog for BYE with route param 'a94.cd4c751'
> Nov 10 18:40:45 en4 /sbin/opensips[20609]: WARNING:dialog:dlg_onroute: 
> unable to find dialog for BYE with route param 'a94.cd4c751'
>
>
> I did a ngrep and the SIP Route: shows the correct param.
> Eventually the Opensips starts not to accept INVITEs or creates a very long 
> setup time for INVITEs (20-40s).
> I think I am doing something wrong, could someone please check my config 
> below.
>
> Thanks,
> Alberto
>
>   


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Re: [OpenSIPS-Users] Problem with prefix() during call forwarding

[Bogdan-Andrei Iancu]

Hi Andrew,

Noticed you fixed the problem, but here are some ideas/questions:

1) what version on opensips do you use?

2) keep in mind that all the changes you do before creating the 
transaction (which is typically done by the first t_relay()) are 
inherited by all the following branched (you create via failure route). 
If you want to do changes to affect only a specific branch, you should 
use the onbranch route (see 
http://www.opensips.org/Resources/DocsCoreRoutes#toc2)

Regards,
Bogdan

Andrew Pogrebennyk wrote:
> Hi,
> I'm trying to put together some configuration for unconditional call 
> forwarding. The carrier requires me to send the call with prefix "400" 
> in R-URI. Here are the relevant routes:
>
> route[6]
> {
>  if(avp_db_load("$ru/username","$avp(s:callfwd)")) {
>  avp_pushto("$ru", "$avp(s:callfwd)");
>  xlog("L_INFO", "forwarded to: $avp(s:callfwd)");
>  avp_delete("$avp(s:callfwd)");
>  $avp(i:25) = 20;
>  }
>  route(7);
> }
>
> route[7]
> {
>   prefix("400");
>   rewritehost("XX.YY.ZZ.WW");
>   t_on_failure("2");
>   xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru 
> F=$fu T=$tu IP=$si ID=$ci\n");
>   t_relay("XX.YY.ZZ.WW:5060");
>   exit;
> }
>
> failure_route[2]
> {
>  # forward on busy
>  if(t_check_status("(486)|(408)") && avp_pushto("$ru", 
> "$avp(s:callfwd)")) {
>  append_branch();
>  xlog("forwarded on $T_reply_code to: $avp(s:callfwd)");
>  avp_pushto("$du", "$avp(s:callfwd)");
>  avp_delete("$avp(s:callfwd)");
>  $avp(i:25) = 20;
>  route(7);
>  }
> }
>
> The problem is that while the call is sent to the first call forward 
> destination correctly (with prefix 400 in R-URI), it goes to the next 
> destination (triggered from failure_route) without the prefix in R-URI! 
> There are the following messages in the log file:
>
> Nov 12 21:12:00 sip2 /usr/local/sbin/openser[52971]: forwarded on 408
> to: sip:89151793...@xx.yy.zz.ww
> Nov 12 21:12:00 sip2 /usr/local/sbin/openser[52971]:  1 avps were removed
> Nov 12 21:12:00 sip2 /usr/local/sbin/openser[52971]:
> DBG:core:pv_get_dsturi: no destination URI
> Nov 12 21:12:00 sip2 /usr/local/sbin/openser[52971]: Request leaving
> server, D-URI='' - M=INVITE RURI=sip:40089151793...@xx.yy.zz.ww
> F=sip:84957978...@xx.yy.zz.ww T=sip:4953801...@aa.bb.cc.dd
> IP=XX.YY.ZZ.WW id=470bed43-cece11de-b158f4a9-da974...@xx.yy.zz.ww
>
>
> Despite R-URI appears with 400, it is sent without the prefix as I've 
> confirmed by the trace.
>
> That "no destination URI" line looked suspicious to me, in fact I would 
> expect that prefix() handles destination URI as well. I thought that 
> could be the case so I've added explicit "$du = $ru;" after prefix and 
> rewritehost. D-URI looks fine now, but it is still sent on the wire 
> without 400:
>
> ov 12 21:59:29 sip2 /usr/local/sbin/openser[53183]: forwarded on 408 to: 
> sip:89165438...@xx.yy.zz.ww
> Nov 12 21:59:29 sip2 /usr/local/sbin/openser[53183]: 
> DBG:avpops:ops_pushto_avp: 1 avps were processed
> Nov 12 21:59:29 sip2 /usr/local/sbin/openser[53183]:  1 avps were removed
> Nov 12 21:59:29 sip2 /usr/local/sbin/openser[53183]: Request leaving 
> server, D-URI='sip:40089165438...@xx.yy.zz.ww' - M=INVITE 
> RURI=sip:40089165438...@xx.yy.zz.ww F=sip:84957978...@xx.yy.zz.ww 
> T=sip:4953801...@aa.bb.cc.dd IP=XX.YY.ZZ.WW 
> id=e9c3fa8b-ced411de-b59ef4a9-da974...@xx.yy.zz.ww
>
> What is the problem?
>
>   


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Re: [OpenSIPS-Users] Cleanup acc & mediaproxy when dialog bye_on_timeout triggers

[Bogdan-Andrei Iancu]

Hi John,

John Quick wrote:
> I am using the bye_on_timeout_flag within the Dialog module to limit the 
> maximum duration for a call.
> When it is triggered I am having problems cleaning up - i.e. accounting the 
> end of call correctly and
> ending the mediaproxy session. I am not using engage_media_proxy() function, 
> instead preferring to
> have more control with use_media_proxy() and end_media_session().
>
> >From a search of the archived questions on this forum I found the trick is 
> >to create a local_route
> section and use a check based on is_method("BYE") to determine when to 
> execute acc_aaa_request().
>
> This works ok, but it is triggered twice at the end of the call - once for 
> the BYE sent upstream and
> again for the BYE sent downstream. The only solution I could find (I've spent 
> many hours testing ideas
> for this) is the use of a global flag that is toggled each time the 
> local_route block gets a BYE
> method. Then the code only calls acc_aaa_request() when the global flag is 
> set - i.e. only one in
> every two times.
>   if (is_gflag("1")) {
>   acc_aaa_request("Internal BYE");
>   }
>
>   # Use Global flag 1 to avoid double reporting/accounting of timeout BYE's
>   if (is_gflag("1")) {
>   reset_gflag("1");
>   } else {
>   set_gflag("1");
>   }
>
> This is not a very elegant or satisfactory solution but it should just about 
> work and hopefully not
> many timeouts will occur anyway. Any suggestions for a better solution would 
> be welcome. I have tried
> Dialog values and transaction flags - they don't work.
>   
This solution is not correct (using the gflags) - triggering the acc 
from local_route is the right thing to do, but not using the gflags - 
these flags are global and shared by entire opensips which can run (in 
the same time) multiple local_routes for different calls -> so you may 
have a mixture between different calls.

Try using the "$DLG_dir" script variable 
(http://www.opensips.org/html/docs/modules/devel/dialog.html#id273200) 
to do the acc only for only one direction will be ideally, but reviewing 
the code I think that variable is not populated in local_route - if you 
could give it a try and let me know, it will be great.

Other idea will be to use the dialog flag or values - you said they do 
not work, even if they should - what seams to be the problem here ?

> The other problem is how to end the media proxy session. The transaction 
> flags set for the Invite are
> not visible in the local_route block when it is handling the internally 
> generated BYE's so I cannot
> try the normal checks that would be used in the main route block for BYE. 
> Perhaps I should just call
> end_media_session(), but will it even work from local_route? If I change my 
> script to start using
> engage_media_proxy() will I lose the flexibility of being able to check which 
> calls need media proxy
> and which don't? I don't want them all using it - only those with far-end NAT.
>   
Shouldn't the media proxy automatically stop when the dialog is 
destroyed ? if you use the enagage_mediaproxy(), I think you do not have 
to explicitly terminate the relay session as the this op will be done 
based on the "dialog destroyed" event.

If it does not work like this, you can use a dialog flag to remember 
(from INVITE time) if the NAT was detected or not. And in local_route 
you can do an explicit end_mediaproxy().

Regards,
Bogdan

> John Quick
> Smartvox Limited
>
>
>
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Re: [OpenSIPS-Users] sip-chatserver sends 500 internal server error - for im-conf INVITE

[Bogdan-Andrei Iancu]

Hi Mani,

The 500 reply you received indicates that opensips was not able to 
forward the request - you have to check the opensips logs to see the 
cause of the failure.

Post the error messages (from the logs) to help you with the error.

Regards,
Bogdan

mani sivaraman wrote:
> Hi Adrian
> I changed by SDP to support TLS and now I'm getting "500 Internal 
> Server Error" from sip-chatserver. Please see the trace below. The 
> only change from your working trace and mine is that, my client tries 
> to send TLS connect to sip-chatserver and not listen on port 2855. 
> Where as your client seems like it acts as a TLS server that listen on 
> port 2855. Is this the problem ?
>
> here is the trace.
>
>  
>  
>
> INVITE sip:imconfere...@mydomain.com 
>  SIP/2.0
>  Via: SIP/2.0/UDP 172.16.1.160:7927;rport;branch=z9hG4bK227591048-91588957
>  Max-Forwards: 70
>  Allow: 
> INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,PUBLISH,MESSAGE,REFER,REGISTER,UPDATE
>  Supported: timer,replaces,norefersub,100rel
>  Accept-Contact: *;+g.oma.sip-im
>  User-Agent: SMSI IMS Client R02.0.00
>  P-Preferred-Identity: sip:msivara...@mydomain.com 
> 
>  From: mani >;tag=smsiUA_227591048-91588958
>  To:  >
>  Call-ID: 227591048-91588956
>  CSeq: 1 INVITE
>  Contact: 
> msivaraman;+g.oma.sip-im
>  Content-Type: application/sdp
>  Content-Length: 328
>  
>  v=0
>  o=SMSI 256 2 IN IP4 172.16.1.160
>  s=SipSession with SMSIUA
>  i=SIP Session in SMSIUA
>  u=http://www.mydomain.com
>  c=IN IP4 172.16.1.160
>  t=0 0
>  m=message 9602 TCP/TLS/MSRP *
>  c=IN IP4 172.16.1.160
>  a=accept-types:message/cpim text/*
>  a=path:msrps://172.16.1.160:9602/z3mfbweby6vb;tcp 
> 
>  a=accept-wrapped-types:*
>  a=sendrecv
>  --
>  
>  
>  Got a message on UDP
>  
>  
>  
>  Trace generated at :
>  Thu Nov 12 11:41:28 2009
>  
>  
>  SIP/2.0 100 Giving a try
>  Via: SIP/2.0/UDP 
> 172.16.1.160:7927;rport=25279;branch=z9hG4bK227591048-91588957;received=63.148.166.3
>  From: mani >;tag=smsiUA_227591048-91588958
>  To:  >
>  Call-ID: 227591048-91588956
>  CSeq: 1 INVITE
>  Server: OpenSIPS (1.5.3-tls (i386/linux))
>  Content-Length: 0
>  
>  --
>  
>  
>  Got a message on UDP
>  
>  
>  
>  Trace generated at :
>  Thu Nov 12 11:41:28 2009
>  
>  
>  SIP/2.0 500 Internal Server Error
>  Via: SIP/2.0/UDP 
> 172.16.1.160:7927;rport=25279;received=63.148.166.3;branch=z9hG4bK227591048-91588957
>  Call-ID: 227591048-91588956
>  From: "mani"  >;tag=smsiUA_227591048-91588958
>  To:  >;tag=id9HOCrk1cfHWEoKDHmKpZaHX60esLl1
>  CSeq: 1 INVITE
>  Server: sip-chatserver-0.9.2
>  Content-Length:  0
>  
>  --
>
> On Fri, Nov 6, 2009 at 3:50 AM, Adrian Georgescu  > wrote:
>
> You are not using TLS:
>
> m=message 8876 TCP/MSRP *
>
> --
> Adrian
>
>
>
>
>
>
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Re: [OpenSIPS-Users] lcr- ideas?

[Bogdan-Andrei Iancu]

Hi Dmitri,

First of all, I would suggest to use the Dynamic Routing module for 
lcr-like purposes. The LCR module is old and does not scale as size and 
performance - anyhow, it will become obsolete in the next versions.
See http://www.opensips.org/html/docs/modules/devel/drouting.html

How you can implement your system? - use in DR module the routing groups 
- you can group the routing rules (the prefixes) into different 
individual sets. When you invoke the routing function (from the script), 
you can specify the routing group you want to be use (see the parameter 
of the do_routing() function); see:
http://www.opensips.org/html/docs/modules/devel/drouting.html#id272408
http://www.opensips.org/html/docs/modules/devel/drouting.html#id227281

 From the script, based on whatever information (dialled number, caller 
profile, domain, etc) you can choose the routing group to be used . So 
you can have a single system with multiple sets of routing rules.

Regards,
Bogdan


Dmitri G. wrote:
> Hi List,
>
> I am using lcr with opensips (version 1.5.2), I have 3 voip products 
> what I offer for my users.
> Let's name it as "bronze", "silver" and "gold".
>
> Each products have it's own lcr (with nearly the same prefixes, but 
> with different gateways).
> Basically the lcr will be different in each products, voip products 
> are calculated based on asr and pdd, so "gold" will be the highest 
> quality lcr.
>
> Each lcr contains various rows, from 5k to 130k rows (based on 
> prefixes). 50% of the prefixes can be found in all 3 products.Every 
> prefix in lcr table have multiple gateways with multiple priorities.
> The customers are dial with prefixes, so 101+number will identify the 
> call as "bronze", 102+number will be identified as "silver".
>
> What will be the best way to route calls based on the dialed number? I 
> am using a cfg file based on Ovidiu's example, so I have exactly the 
> same like this(ok few things are changed to fit opensips 1.5.2)
> http://www.voipembedded.com/resources/openser_dbtext_lcr.cfg
>
> I mean when the customer dial 101+number, then the call will be routed 
> based on the "bronze" lcr, when somebody else dial 102+number, then 
> that will be routed based on the "silver" lcr.
>
> I am thinking about to run multiple opensips instances, each instance 
> with it's own cfg file, so 1st instance ill run with using 
> opensips.cfg, second one will run with opensips2.cfg and so on, so 
> basically I'll run 3 instances with different cfg/pid files and with 3 
> different DBs + listen ports, first instance with "bronze" lcr, second 
> one with "silver" lcr, third one wwith "gold" lcr.
> With this I can have 3 separate databases for the 3 instances, and I 
> can have 3 different lcr tables, but I don't think it's the best way 
> to do this.
>
> I hope somebody can suggest me a solution for this.
>
> Any help would be appreciated.
>
>
> Thanks,
>
> Dmitri
>  
>
>
>
>
> 
>
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Re: [OpenSIPS-Users] dialog module errors

[Bogdan-Andrei Iancu]

Hi Mouncif,

Do you get db errors only for the DIALOG module? do you have other 
modules using DB perfectly working?

Have you tried to re-create the dialog table ?

The error is generated directly by the mysql driver (from opensips) 
because it does not recognize the types of the columns from the dialog 
table - are you sure you are using the structure as provided by the 
opensips db installer ?

Regards,
Bogdan

Mouncif Benniane wrote:
> I get this errors when starting opensips 1.6 with dialog module 
> enabled ( centos 5.2)
>
> WARNING:db_mysql:db_mysql_get_columns: unhandled data type column 
> (hash_id) type id (154246320), use DB_STRING as default
> WARNING:db_mysql:db_mysql_get_columns: unhandled data type column 
> (dialog) type id (154246392), use DB_STRING as default
> WARNING:db_mysql:db_mysql_get_columns: unhandled data type column 
> (dialog) type id (154246432), use DB_STRING as default
> WARNING:db_mysql:db_mysql_get_columns: unhandled data type column 
> (opensips) type id (154246512), use DB_STRING as default
> WARNING:db_mysql:db_mysql_get_columns: unhandled data type column 
> (def) type id (154246544), use DB_STRING as default
>
>
> my config looks like this:
>
> #dialog
> modparam("dialog", "profiles_with_value", "inbound")
> modparam("dialog", "dlg_flag", 4)
> modparam("dialog", "db_url", "mysql://root:passx...@localhost/opensips")
> modparam("dialog", "db_mode", 1)
> modparam("dialog", "db_update_period", 60)
> modparam("dialog", "table_name", "dialog")
>
>
> opensips never starts and exit. any ideas?? BTW same config and 
> version worked on centos 4.6.
>
>
> Thanks
>
> 
>
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Re: [OpenSIPS-Users] Presence server behind load-balancer

[Bogdan-Andrei Iancu]

Hi Takeshi,

Routing the presence traffic does not require USRLOC/REGISTRATION at 
all.  The clients do send PUBLISH to the presence agent and they also 
send the SUBSCRIBE request to it. The PA (presence agent) is sending 
back the NOTIFY requests within the dialog created by the SUBSCRIBE 
request. So, the NOTIFY will be route back based on the Record-Route you 
did for SUBSCRIBE.

Regards,
Bogdan

mayamatakeshi wrote:
> Hello,
> I am testing opensips as registrar in the following scenario:
>
> UA ---REGISTER---> opensips (load-balancer) REGISTER with header 
> Path> opensips (registrar).
>
> This works fine, and I have the INVITEs that reach the registrar going 
> thru both proxies as expected:
>
> UA <-INVITE- opensips(load-balancer) 
> <---INVITE---opensips(registrar) <-- GW
>
> But now I want to make the registrar to work as a presence server and 
> I want the NOTIFY requests to go thru the load-balancer too (to permit 
> reach UAs behind NAT).
> Is this possible? I mean, when presence is processed by opensips, it 
> seems to me we cannot route NOTIFY requests using the usrloc data from 
> the registrar, because the NOTIFYs generated by this are not processed 
> by the cfg file.
> So, is there any way to do this?
>
> br,
> takeshi
>
> 
>
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[OpenSIPS-Users] Presence server behind load-balancer

[mayamatakeshi]

Hello,
I am testing opensips as registrar in the following scenario:

UA ---REGISTER---> opensips (load-balancer) REGISTER with header
Path> opensips (registrar).

This works fine, and I have the INVITEs that reach the registrar going thru
both proxies as expected:

UA <-INVITE- opensips(load-balancer)
<---INVITE---opensips(registrar) <-- GW

But now I want to make the registrar to work as a presence server and I want
the NOTIFY requests to go thru the load-balancer too (to permit reach UAs
behind NAT).
Is this possible? I mean, when presence is processed by opensips, it seems
to me we cannot route NOTIFY requests using the usrloc data from the
registrar, because the NOTIFYs generated by this are not processed by the
cfg file.
So, is there any way to do this?

br,
takeshi
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[OpenSIPS-Users] B2BUA and placing a call on hold. Bug?

[Jeff Kronlage]

Anca,

 

I've nearly got my B2BUA ready for deployment, however, in a
front-end/back-end solution, with my standard proxy configuration in the
front-end (passing calls to the B2BUA back-end), I'm having trouble
putting a call on hold.  Please note that all other functionality is
working great - REFERs, media-related reINVITEs, etc.

 

I noticed this because if I put the call on hold for too long while
attempting a REFER, I start having problems.

 

Basically, the TM module on the B2BUA keeps re-transmitting the final
200 OK.  The UAC responds to all of these with an ACK, which the B2B
passes on to the far SIP endpoint.  Eventually the UAC, getting hammered
by OKs, assumes there's a problem and sends a BYE after about 30
seconds.  Note that if I transfer during this time period, the call
stays up indefinitely.  It's only the reINVITE for "hold" that breaks
it.

 

My config on the B2B box is super-simplified (less than 10 lines) so I
don't think it's a configuration problem.  Happy to post B2B
confgs/debug/SIP trace/etc.  Please let me know what (if anything) is
necessary.

 

Please advise.  Thanks,

 

Jeff K

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Re: [OpenSIPS-Users] opensips and vserver

[Jeff Pyle]

Erik,

For performance reasons it¹s not a good idea to run a Mediaproxy Relay in a
vserver.  Testing is one thing, but production scaling is another entirely.
Running Mediaproxy Dispatcher attached to an Opensips instance in a vserver
works great, however.  Mediaproxy¹s architecture allows you to separate the
relays from the dispatchers.  In my case, I have two dedicated relay boxes
connecting to three dispatchers, where the dispatchers are running with
Opensips all in Xen VMs.  It works quite well for me.


- Jeff



On 11/14/09 7:31 PM, "erik pepermans"  wrote:

> Hi,
>  
> I setup opensips+radius+asterisk on a CentOS machine -
>  
> Currently there are 3 asterisk servers each running in its own vserver. Radius
> and the accounting part is running in another vserver.
> Opensips is again running in another vserver and communicating with the 3
> asterisks.
>  
> My question: If I want to add mediaproxy, is it better to run
> opensips+mediaproxy outside any vserver or can I join it to my current
> opensips vserver ?
>  
> Brgds
> Erik


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