Re: [OpenSIPS-Users] corrupted header

[Jan D.]

Looks like Iñaki Baz Castillo is right, could be a buggy Sip ALG, the client
is using a Zyxel P660.

Still, I want to 'catch' the frame as a 'normal' error, also because it
floods my log files with a lot of error lines. I just want to log one line,
so I added a error_route at the end of my script:


# Error route

error_route
{
xlog("L_ERROR","--- error route class=$(err.class)
level=$(err.level) info=$(err.info) rcode=$(err.rcode)
rreason=$(err.rreason) ---\n");
xlog("L_ERROR","--- error from [$si:$sp]\n+\n$mb\n\n\n");
sl_send_reply("$err.rcode", "$err.rreason");
exit;
}

But the logfile is still full of errors, the error route does not kick in:

Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_via: bad port
Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_via: 
#015#012Max-Forwards:
70#015#012From: 31***285
;tag=25d7b2df4a#015#012To:
;tag=9D821F9C-C66#015#012Call-ID:
16b79f8c2c80a9cd#015#012CSeq: 18992 BYE#015#012Authorization: Digest
username="username0023",realm="my.sipserver.com",nonce="4a93d91f0001382fafda66c1***52",uri="sip:31***...@212.*.*.245:5060",response="da7fdedfec2df8798ca3559f0b4cfd9e"#015#012Reason:
Q.850; cause=16; text="Normal call clearing"#015#012Supported:
timer#015#012User-Agent: HiPath 3000 V7.0 M5T SIP
Stack/4.0.26.26#015#012Content-Length: 0#015#012#015#012>
Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_via: parsed
so far:
Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:get_hdr_field: bad
via
Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_msg:
message=#015#012Max-Forwards:
70#015#012From: 31***285
;tag=25d7b2df4a#015#012To:
;tag=9D821F9C-C66#015#012Call-ID:
16b79f8c2c80a9cd#015#012CSeq: 18992 BYE#015#012Authorization: Digest
username="username0023",realm="my.sipserver.com",nonce="4a93d91f0001382fafda66c1***52",uri="sip:31***...@212.*.*.245:5060",response="da7fdedfec2df8798ca3559f0b4cfd9e"#015#012Reason:
Q.850; cause=16; text="Normal call clearing"#015#012Supported:
timer#015#012User-Agent: HiPath 3000 V7.0 M5T SIP
Stack/4.0.26.26#015#012Content-Length: 0#015#012#015#012>
Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:receive_msg:
parse_msg failed

Is there a way I can just 'drop' the INVITE (or whatever corrupt header is
sent) and log a normal error? I hoped the error_route could do this.

Jan
-- 
View this message in context: 
http://n2.nabble.com/corrupted-header-tp3487025p3514297.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] how to uninstall

[Saúl Ibarra]

Have you removed everything OpenSIPS related under /usr/local/tec,
/usr/local/bin and /usr/local/lib ?



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Re: [OpenSIPS-Users] how to uninstall

[Tseveendorj Ochirlantuu]

I removed some folder and files named opensips. but It didn't work.

$sudo opensipsctl start
Aug 26 14:33:30 beastie opensips: ERROR:core:main: loading config
file(/usr/local/etc/opensips/opensips.cfg): No such file or directory

Where is it reading this path ?

Sincerely,
Tseveen.

On Wed, Aug 26, 2009 at 2:30 PM, Saúl Ibarra  wrote:

> There is no uninstall script (yet) so you'll have to remove files manually.
>
>
> 2009/8/26, Tseveendorj Ochirlantuu :
> > Dear all,
> >
> > I have installed OpenSIPS-1.5.2 from source with make prefix=/usr/local .
> > But I need to change prefix=/ or uninstall opensips completely.
> >
> > Sorry for my body language.
> >
> > Sincerely,
> > Tseveen.
> >
>
> --
> Enviado desde mi dispositivo móvil
>
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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Re: [OpenSIPS-Users] how to uninstall

[Saúl Ibarra]

There is no uninstall script (yet) so you'll have to remove files manually.


2009/8/26, Tseveendorj Ochirlantuu :
> Dear all,
>
> I have installed OpenSIPS-1.5.2 from source with make prefix=/usr/local .
> But I need to change prefix=/ or uninstall opensips completely.
>
> Sorry for my body language.
>
> Sincerely,
> Tseveen.
>

-- 
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Re: [OpenSIPS-Users] init.d file?

[Saúl Ibarra]

Look at /var/run to see if you still have an opensips pid file,
regardless it has been stopped.

2009/8/26, Alex G :
> not using the dpkg because i have all my opensips servers built already. for
> the sake of using monit, i require an init.d file to be there to start/stop
> opensips.
>
> i've changed the pointer for DAEMON and added the default file. I'm getting
> "Starting opensips: opensips already running." but its obviously not
> running. I can see I'm getting closer, but still missing something... any
> more tips? :)
>
>
>
> On Tue, Aug 25, 2009 at 5:14 PM, Saúl Ibarra  wrote:
>
>> You'll need to make it executable as well as copy the opensips.default
>> file to /etc/default. That should do the job :)
>>
>> Anyway, why don't you make the debian packages so that you install
>> opensips by just doing dpkg -i and everything is done automagically?
>>
>>
>>
>> --
>> /Saúl
>> http://www.saghul.net | http://www.sipdoc.net
>>
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>>
>

-- 
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/Saúl
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[OpenSIPS-Users] how to uninstall

[Tseveendorj Ochirlantuu]

Dear all,

I have installed OpenSIPS-1.5.2 from source with make prefix=/usr/local .
But I need to change prefix=/ or uninstall opensips completely.

Sorry for my body language.

Sincerely,
Tseveen.
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Re: [OpenSIPS-Users] Multiple Area Codes in Customer Area

[osiris123d]

Got it working!  Man OpenSIPS sure can do anything with SIP

So here is what I did for future searchers

So the users account is a 7 digit DID number XXX at blah.com

I set up an AVP called areacode for the whole domain blah.com (this assumes
that the whole domain blah.com is only in one areacode)

opensipsctl avp add -T usr_preferences 0 at blah.com areacode 1 201
opensipsctl avp add -T usr_preferences 0 at foo.com areacode 1 339


In the opensips.cfg file I do the following (it depends on your config as to
where you want to put this)
if (uri=~"^sip:[2-9][0-9]{6}@") {
avp_db_load("$ru/domain","$avp(s:areacode)");
subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
};

So when someone calls a 7 digit number the avp_db_load() loads the variable
for areacode and the subst_uri adds the areacode at the beginning of the
Request-URI.






Bogdan-Andrei Iancu wrote:
> 
> Hi Duane,
> 
> You can correlate AVPs you a USER, a DOAMAIN, etc - it is up to you, 
> from the script, when loading the AVP - is a pure logical mapping.
> 
> Regards,
> Bogdan
> 
> osiris123d wrote:
>> I was reading Flavio's "Building Telephony Systems with OpenSER" chapter
>> about AVPOPs and he mentions that AVP's can be used for a whole domain. 
>> I
>> was thinking that I might be able to configure a area code for Company
>> A's
>> domain and then route calls that way.  If not that then I can set the AVP
>> on
>> the fly within the transaction by looking at the callers Request URI's
>> first
>> 3 digits and route it appropriately.
>>
>>
>> Bogdan-Andrei Iancu wrote:
>>   
>>> Hi,
>>>
>>> Requirements on the format of  CONTACT and TO headers are nonsense as 
>>> they are not used for routing at all. Only FROM (which provides info on 
>>> the caller) and RURI (request URI) (which provide info on callee).
>>>
>>> So, bottom line, only the normalization of the RURI should be required 
>>> on the system.
>>>
>>> Regards,
>>> Bogdan
>>>
>>> osiris123d wrote:
>>> 
 Thanks for the info.

 I will look into this and work up a config.

 I also got this direct email about my post from someone else who lives
 in
 the US.  I figured I would go ahead and post below what he sent just so
 its
 out there.


 Hello Duane --  

 You have hit on one of the more difficult areas in SIP and telephony in
 general -- especially here in the North American Numbering Plan.  Below
 I
 will address the problem in general, and not particularly related to
 the
 OpenSIPs question, because IMO you need a solution that will work in
 any
 architecture, not just OpenSIPs.

 In a nutshell, I recommend that for your USA users you:

 1.) Require From: and Contact: headers to be in NANPA National (10
 digit)
 format.  This is method is standard in the telephone industry, and will
 allow easy integration with North American ANI or Caller ID format,
 especially when a call may eventually be handed off to the PSTN.   

 2.)  Require incoming To: headers to be in e.164 International format,
 i.e. 
 NANPA-destination numbers all begin with the 1 digit, followed by the
 10
 digit National number.   Any incoming call to 612xxx goes to
 Sydney,
 Austrailia, and not Minneapolis, MN.  This requirement should be
 enforced
 at
 the perimeter of your network, where Customer Equipment can enforce the
 "local" digit normalization policy.  

 3.)  If you can't enforce #2 above, you will need to "Normalize"
 incoming
 calls to the e.164 International format prior to routing.  The
 unfortunate
 reality here in the USA is that the requirements for how many digits to
 dial
 for a given destination (the "dialing plan") depends on where the call
 comes
 from.   Here in the Chicago area, residents of the 847 area code must
 today
 dial all calls as 11 digits.  Calls within the 312 or 773 area code may
 today be dialed as 7 digits, however beginning on 07 November, all
 calls
 originating in 312 and 773 must be dialed as 1+10 digits, due to the
 new
 872
 overlay area code.For more information, see
 http://www.nanpa.com/reports/NPA_Dialing_Plans_05_09.pdf

 4.)  Finally, if you have any termination carriers who need special
 "prefixes,"  append them after you have made your route selection.  

 If you would like further information or discussion, please feel free
 to
 contact me.

 John S. Robi

 j...@communxxx.com



 Bogdan-Andrei Iancu wrote:
   
   
> Hi there,
>
> When you have to deal with local dialling you need consider the amount 
> of information yon need to keep in order to translate to national
> format 
> and the complexity of the processing you have to do.
>
> A compromise solution will be to keep in user profile some in

Re: [OpenSIPS-Users] init.d file?

[Alex G]

not using the dpkg because i have all my opensips servers built already. for
the sake of using monit, i require an init.d file to be there to start/stop
opensips.

i've changed the pointer for DAEMON and added the default file. I'm getting
"Starting opensips: opensips already running." but its obviously not
running. I can see I'm getting closer, but still missing something... any
more tips? :)



On Tue, Aug 25, 2009 at 5:14 PM, Saúl Ibarra  wrote:

> You'll need to make it executable as well as copy the opensips.default
> file to /etc/default. That should do the job :)
>
> Anyway, why don't you make the debian packages so that you install
> opensips by just doing dpkg -i and everything is done automagically?
>
>
>
> --
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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[OpenSIPS-Users] OPenSIPS with SQLite3 database

[Aryanto Rachmad]

Hello Everybody,

This is my first post to this list, so greeting to all of you.

I would like to know if anybody has successfully use OpenSIPS with 
SQLite3 database. I suppose we could use it through unixodbc, but is 
there any plan to support native interface like for MySQL?

Thanks a lot in advance for your response.

Kind regards,

Anto



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[OpenSIPS-Users] XCAP scalability: Integrated xcap server VS xcap_client mode

[Iñaki Baz Castillo]

Hi, I've read somewhere (but cannot find it right now) that integrated xcap 
mode (presence module getting XCAP documents vía SQL) is suitable for small 
environments.

I know that the other way, using xcap_client module, is not really suitable 
for now as the HTTP request is blocking (the opensips process gets blocked 
until the XCAP server replies). But let's imagine this issue is solved.

So we have two options:

1) Integrated server: OpenSIPS presence module gets the documents from the 
xcap table.

2) XCAP client mode: OpenSIPS acts as a xcap client to get the documents from 
the XCAP server.


In the option 1:
- OpenSIPS does a SQL query to get the document (faster than a HTTP request).
- So the XCAP server is not queried by the presence server (less work for the 
XCAP server).
- There could be various XCAP servers running at the same time (perhaps DNS 
random or a http proxy between clients and XCAP servers), all of them storing 
the documents in same DB. And the presence document uses directly that DB.


In the option 2:
- OpenSIPS must perform a HTTP request which takes more time than a SQL 
request (even if the DB is in other host), right?
- The XCAP server receives a XCAP request from the presence server, so the 
XCAP server must "work".
- The xcap client would contact just an unique XCAP server (it would learn the 
IP after the first DNS resolution, so ramdom DNS is not valid here).
  - Solution: Using a HTTP proxy between OpenSIPS and various XCAP servers.


Is it really the option 2 more suitable in order to favour scalability and big 
environments?



-- 
Iñaki Baz Castillo 

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Re: [OpenSIPS-Users] drouting module

[David Villasmil]

stop asking so much!

;)

D

El 25/08/2009, a las 18:05, "Sebastian Sastre"   
escribió:



Hi,



When applying the do_routing function without specifying the group  
id, what are the matching patterns that it will look for?




In other words If I want ALL the users behind domain test.com how  
would I do it?




Leaving the username field empty and putting test.com in the domain  
did not work.




Thanks





Sebastian









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Re: [OpenSIPS-Users] init.d file?

[Saúl Ibarra]

You'll need to make it executable as well as copy the opensips.default
file to /etc/default. That should do the job :)

Anyway, why don't you make the debian packages so that you install
opensips by just doing dpkg -i and everything is done automagically?



-- 
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Re: [OpenSIPS-Users] init.d file?

[Brett Nemeroff]

Did you make it executable? You may also need to adjust the script to point
to the proper binary location. Open up the script, it's not too tricky.
I'm pretty sure that this line:
test -f $DAEMON || exit 0

Says, "If the binary isn't there, just quietly die, without giving the user
a useful error message"



-Brett



On Tue, Aug 25, 2009 at 3:57 PM, Alex G  wrote:

> so i changed the name of the file to opensips and moved it into init.d, but
> it does not respond to any commands. any hints you could give me?
>
>
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[OpenSIPS-Users] drouting module

[Sebastian Sastre]

Hi, 

 

When applying the do_routing function without specifying the group id, what
are the matching patterns that it will look for?

 

In other words If I want ALL the users behind domain test.com how would I do
it? 

 

Leaving the username field empty and putting test.com in the domain did not
work. 

 

Thanks 

 

 

Sebastian

 

 

 

 

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Re: [OpenSIPS-Users] init.d file?

[Alex G]

so i changed the name of the file to opensips and moved it into init.d, but
it does not respond to any commands. any hints you could give me?

On Tue, Aug 25, 2009 at 3:38 PM, Saúl Ibarra  wrote:

> Look at packaging/debian directory, file is opensips.init.
>
>
> --
> /Saúl
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>
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Re: [OpenSIPS-Users] init.d file?

[Saúl Ibarra]

Look at packaging/debian directory, file is opensips.init.


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[OpenSIPS-Users] init.d file?

[Alex G]

can anyone point me to a current init.d script for starting/stopping
opensips? am trying use in conjunction  with monit and have an older init.d
from openser days


thanks in advance!
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Re: [OpenSIPS-Users] Next OpenSIPS releases

[Bogdan-Andrei Iancu]

Hi Julien,


Julien Chavanton wrote:
>
> Hi Bogdan, speaking about the futur, what do you think about the 
> B2B_ENTITIES and B2B_LOGIC module, to be realistic/strategic, you will 
> delay this to a futur version maybe even 2.x, will this require 
> modification to the core ?
>

Not sure how you see this, but the B2BUA related modules are part of the 
1.6 release. I admit they are in alpha-stage and probably it will take 
some more time (in the next releases) to (1) become stable and (2) 
functionality consistent.
>
> Personnaly, I want to take some time to test them in a lab, I am 
> curious to see how it is implemented.
>
> I beleive that a lot of users could benefit from this, in term of :
>  - security : hide interconnection information
>  - interoperability : handle/fix "possible" compatibility problem 
> between other SIP UA
>
and also there is a third case, quite large, when comes to implementing 
complex SIP scenarios (not possible by a simple proxy) which does not 
need any media manipulation (we still have a signalling b2bua) - like 
the examples we have in the tutorial, with  inserting announcements  in 
the call, tele-marketing, etc...More or less better integration of calls 
with media services.
>
> Opensips could then become a one peice solution for SIP carrier, as 
> now they may require an SBC as well for such concern
>
Indeed.
>
> Thank you for your continuous effort and accomplishement.
>

Thank you :)

Regards,
Bogdan
> 
> *From:* users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu
> *Sent:* Tue 25/08/2009 9:59 AM
> *To:* users@lists.opensips.org; de...@lists.opensips.org
> *Subject:* [OpenSIPS-Users] Next OpenSIPS releases
>
> Hi,
>
> Here are the plans for the next OpenSIPS releases (minor and major).
> This is an initial draft (content and dates), so please comment and
> contribute (if necessary):
>
>
> 1) Minor release 1.5.3
> ---
>
> Why: This is needed as more than 50 fixes were done on the 1.5 branch
> since 1.5.2.
>
> Date: during this week
>
> Pending: personally I'm hunting an memory leak in SNMP module (mainly
> design issues). If someone is aware of any other issues that requires
> fixing in 1.5 branch, please speak up.
>
>
> 2) Major release 1.6.0
> ---
>
> Code freeze:  estimated for mid September (depending of how fast the
> pending work is completed)
>
> Release date: estimated for October
>
> What we have so far: http://www.opensips.org/Main/Ver160
>
> Pending work:
> - adding context for PVs (like reply, request)
> - route types - init, onreply per branch, timer based
> - dialog - early dialog support to be finished; new functions to
> check dialog consistency (cseq numbers, route set, contacts); dialog
> direction function
> - pike enhancement for catching more events (replies, non-SIP
> traffic attacks)
> - json support
> If there is something missing or if somebody is working some (new) code
> and needs time and support, please let me know.
>
>
> Regards,
> Bogdan
>
>
>
>
>
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Re: [OpenSIPS-Users] B2BUA module question

[Saúl Ibarra]

> The headers that are now taken from the initial message and inserted in
> the message sent on the other side are: Supported, Require,
> Proxy-Require, Accept and Content-Type.
> We can extend the rules action part to include this one of requesting a
> certain header to be added since it can indeed be useful.

I'd be really happy to see this, as I stated on the dev list. It would
be nice if we had a configuration parameter so that all custom headers
are passed to the other leg, so OpenSIPS could act as a transparent
b2bua :)


-- 
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Re: [OpenSIPS-Users] B2BUA module question

[Brett Nemeroff]

Anca,What I was imaging was something like the dialplan module to perform
the rewrite, and the B2BUA module to track was it was originally so the
reverse direction doesn't include the translation. That kind of thing.

inside -> b2bua + translation ---> outside see's translated TO URI

outside with translated TOURI ->b2bua -> inside see's restored TO URI
(original TO URI)

Forgive me for not entirely understanding the B2BUA scenarios and rules
quite yet. ;)

-Brett

On Tue, Aug 25, 2009 at 10:28 AM, Anca Vamanu  wrote:

> Hi Brett,
>
>
> Brett Nemeroff wrote:
> > All,
> > Question about the direction of the B2BUA module. I know one of the
> > key feature is topology hiding. Does this also occur in the SDP? I
> > would expect that it would need to still be paired with something like
> > mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is
> > this correct? Do you expect the B2BUA module will ever integrate into
> > any of the media proxying solutions?
> >
> As you correctly assumed, the B2BUA implementation in OpenSIPS is only a
> signaling B2BUA and it does not deal with sdp. The media will still go
> end to end and you need to use something like rtpproxy for a full b2b.
> > Also, what's the possibility of doing things like changing headers,
> > removing headers and such. For example, internally, I may have an
> > "X-Account-Number:" field that is used between servers and I never
> > want an request from the outside to ever come in with one of those and
> > likewise I don't ever want a request to go out with one of those. I
> > know a lot of that can be done in the script already, but I'm
> > wondering if the B2BUA portions have any special handling for that
> > kind of thing (ie: remove all non-standard headers).  Also, there are
> > a lot of non-rfc-ish things that I have to do on a regular basis that
> > a B2BUA always performs better. For example, I have partners that
> > insist on specific formatting of the From or To headers (like adding
> > or removing prefixes to from/to headers.. yes.. I know..).
> >
> The headers that are now taken from the initial message and inserted in
> the message sent on the other side are: Supported, Require,
> Proxy-Require, Accept and Content-Type.
> We can extend the rules action part to include this one of requesting a
> certain header to be added since it can indeed be useful.
> But the one with formatting the to or from header in a certain way is
> quite hard to express as a rule..
>
> regards,
> Anca
> > Thanks!
> > -Brett
> >
> > 
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
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Re: [OpenSIPS-Users] B2BUA module question

[Anca Vamanu]

Hi Brett,


Brett Nemeroff wrote:
> All,
> Question about the direction of the B2BUA module. I know one of the 
> key feature is topology hiding. Does this also occur in the SDP? I 
> would expect that it would need to still be paired with something like 
> mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is 
> this correct? Do you expect the B2BUA module will ever integrate into 
> any of the media proxying solutions?
>
As you correctly assumed, the B2BUA implementation in OpenSIPS is only a 
signaling B2BUA and it does not deal with sdp. The media will still go 
end to end and you need to use something like rtpproxy for a full b2b.
> Also, what's the possibility of doing things like changing headers, 
> removing headers and such. For example, internally, I may have an 
> "X-Account-Number:" field that is used between servers and I never 
> want an request from the outside to ever come in with one of those and 
> likewise I don't ever want a request to go out with one of those. I 
> know a lot of that can be done in the script already, but I'm 
> wondering if the B2BUA portions have any special handling for that 
> kind of thing (ie: remove all non-standard headers).  Also, there are 
> a lot of non-rfc-ish things that I have to do on a regular basis that 
> a B2BUA always performs better. For example, I have partners that 
> insist on specific formatting of the From or To headers (like adding 
> or removing prefixes to from/to headers.. yes.. I know..). 
>
The headers that are now taken from the initial message and inserted in 
the message sent on the other side are: Supported, Require, 
Proxy-Require, Accept and Content-Type.
We can extend the rules action part to include this one of requesting a 
certain header to be added since it can indeed be useful.
But the one with formatting the to or from header in a certain way is 
quite hard to express as a rule..

regards,
Anca
> Thanks!
> -Brett
>
> 
>
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>   


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Re: [OpenSIPS-Users] B2BUA module question

[Alex Balashov]

Brett Nemeroff wrote:

> Question about the direction of the B2BUA module. I know one of the key 
> feature is topology hiding. Does this also occur in the SDP? I would 
> expect that it would need to still be paired with something like 
> mediaproxy or rtpproxy to achieve topology hiding with SDP as well, is 
> this correct? 

Yes.

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

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[OpenSIPS-Users] B2BUA module question

[Brett Nemeroff]

All,Question about the direction of the B2BUA module. I know one of the key
feature is topology hiding. Does this also occur in the SDP? I would expect
that it would need to still be paired with something like mediaproxy or
rtpproxy to achieve topology hiding with SDP as well, is this correct? Do
you expect the B2BUA module will ever integrate into any of the media
proxying solutions?

Also, what's the possibility of doing things like changing headers, removing
headers and such. For example, internally, I may have an "X-Account-Number:"
field that is used between servers and I never want an request from the
outside to ever come in with one of those and likewise I don't ever want a
request to go out with one of those. I know a lot of that can be done in the
script already, but I'm wondering if the B2BUA portions have any special
handling for that kind of thing (ie: remove all non-standard headers).
 Also, there are a lot of non-rfc-ish things that I have to do on a regular
basis that a B2BUA always performs better. For example, I have partners that
insist on specific formatting of the From or To headers (like adding or
removing prefixes to from/to headers.. yes.. I know..).

Thanks!
-Brett
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Re: [OpenSIPS-Users] Next OpenSIPS releases

[Julien Chavanton]

Hi Bogdan, speaking about the futur, what do you think about the B2B_ENTITIES 
and B2B_LOGIC module, to be realistic/strategic, you will delay this to a futur 
version maybe even 2.x, will this require modification to the core ?

Personnaly, I want to take some time to test them in a lab, I am curious to see 
how it is implemented.

I beleive that a lot of users could benefit from this, in term of :
 - security : hide interconnection information 
 - interoperability : handle/fix "possible" compatibility problem between other 
SIP UA

Opensips could then become a one peice solution for SIP carrier, as now they 
may require an SBC as well for such concern

Thank you for your continuous effort and accomplishement.




From: users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu
Sent: Tue 25/08/2009 9:59 AM
To: users@lists.opensips.org; de...@lists.opensips.org
Subject: [OpenSIPS-Users] Next OpenSIPS releases



Hi,

Here are the plans for the next OpenSIPS releases (minor and major).
This is an initial draft (content and dates), so please comment and
contribute (if necessary):


1) Minor release 1.5.3
---

Why: This is needed as more than 50 fixes were done on the 1.5 branch
since 1.5.2.

Date: during this week

Pending: personally I'm hunting an memory leak in SNMP module (mainly
design issues). If someone is aware of any other issues that requires
fixing in 1.5 branch, please speak up.


2) Major release 1.6.0
---

Code freeze:  estimated for mid September (depending of how fast the
pending work is completed)

Release date: estimated for October

What we have so far: http://www.opensips.org/Main/Ver160

Pending work:
- adding context for PVs (like reply, request)
- route types - init, onreply per branch, timer based
- dialog - early dialog support to be finished; new functions to
check dialog consistency (cseq numbers, route set, contacts); dialog
direction function
- pike enhancement for catching more events (replies, non-SIP
traffic attacks)
- json support
If there is something missing or if somebody is working some (new) code
and needs time and support, please let me know.


Regards,
Bogdan





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Re: [OpenSIPS-Users] 1.5.2 dispatcher module behaviour

[Taner Sener]

It's working now as expected.

Thanks

On Thu, Aug 13, 2009 at 12:59 PM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:

> Hi Taner,
>
> Taner Sener wrote:
>
>> Hi Bogdan,
>>
>> On Wed, Aug 12, 2009 at 3:56 PM, Bogdan-Andrei Iancu <
>> bog...@voice-system.ro > wrote:
>>
>>Hi Taner,
>>
>>Taner Sener wrote:
>>
>>Hi,
>>
>>I'm using Opensips 1.5.2 to distribute incoming calls to my
>>clients using dispatcher module. I'm keeping my gateway list
>>in db_mysql and use ds_select_dst("1", "4"); to select a
>>gateway using round-robin algorithm. I have a few issues about
>>the module behaviour.
>>
>>- The first one is about pinging. I've configured dispatcher
>>to send ping requests every 20 seconds. But if destination is
>>not available, ping requests are repeated every 4 seconds. I
>>guess there is another module which repeats the unresponded
>>sip messages. How can I prevent this and change the repeat
>>timeout about this?
>>
>>there should be no second module to do the pinging, and there is
>>no way the module can dynamically change the pinging interval.
>>
>>try enabling full debug (debug=6) and look for the log messages like:
>>  "probing set #n, URI   "
>>
>>  I looked inside logs and found "DBG:dispatcher:ds_check_timer: probing
>> set #1, URI sip:" lines there. So i guess it means that timer has expired
>> and dispatcher is sending SIP OPTIONS at that time. But later found that TM
>> module was enabled in my configuration and it was TM retransmitting SIP
>> OPTIONS to dead destinations (with T2_timer which is 4 seconds). I can
>> increase T2_timer but it will effect other messages, so I will leave it as
>> is.
>>
> AhaThe dispatcher module uses TM support for sending the pings in a
> statefull manner - so, if there is no reply at all, the TM will do
> retransmission of the original request it send. It was not clear from your
> original email if new OPTIONS are fire (at each 2 secs) or what you are
> simply retransmissions (copies) of the pings that were already sent out.
>
> You not control the retransmission interval via T1 and T2 params in TM (see
> http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id228598), but
> note that this will have a global impact.
>
> Also you can configure how long the retransmission will be done via the
> fr_timer (see
> http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id271112).
>
>
>
>>
>>
>>- The second issue is about selecting gateways. When I receive
>>busy from one of the destinations I'm calling ds_next_dst()
>>and this returns me a destination which is not alive and does
>>not respond to ping requests. I'm expecting to have only
>>destinations which are alive, and don't understand why it is
>>returned. Another issue here is: I'm sending INVITE request to
>>this dead destination and dead host is not responding as
>>expected. After that, every 4 seconds INVITE request is
>>repeated for this dead destination.
>>
>>you should call the ds_mark_dst() function from failure route,
>>when you detect a destination as failed (and before the
>>ds_next_dst() ). See:
>>
>>
>> http://www.opensips.org/html/docs/modules/1.5.x/dispatcher.html#id271344
>>
>>
>> I thought that if a destination is not alive and not responding to PING
>> requests (in my case Destination Unreachable ICMP messages are received), it
>> is marked as failure route automatically, but it looks like I must mark it
>> by myself. At this point I want to ask if I can listen for results of PING
>> resuls. So if I receive REPLY I will mark it as healthy and if PING timeout
>> occurs I can mark it as dead. BTW are Destination Unreachable ICMP messages
>> identified by opensips?
>>
>
> There are two ways to mark (as failed) a destination:
>
>   1) from script, via ds_mark_dst() function, based on the negative replies
> you get when routing traffic to your destination.
>
>   2) automatically, based on 408 replies received. You should see in logs
> debug like:
>OPTIONS-Request was finished with code XX (to xx, group
> )
>Setting the probing state failed (xx, group XX)
>
>
>
> Regards,
> Bogdan
>
>
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Re: [OpenSIPS-Users] Regrarding is_user_in problem in opensips-1.5

[ASHWINI NAIDU]

Hi Bogdan,

Thank You for reply. It was network mapping issue and not opensips
issue. I solved it.



On Mon, Aug 24, 2009 at 4:09 PM, Bogdan-Andrei Iancu  wrote:

> Hi Ashwini,
>
> As your script shows, you do either IP auth (allow trusted) or digest
> auth , but "credentials" are present only after digest auth.
>
> So, if it is a trusted peer, there will be no digest auth, no
> credentials and is_user_in() will fail.
>
> My advice is to replace is_user_in("credentials"); with
> is_user_in("from");   - anyhow you required both FROM USERNAME and AUTH
> USERNAME to be the same when doing check_from().
>
> Regards,
> Bogdan
>
> ASHWINI NAIDU wrote:
> >
> > Hi Bogdan,
> >
> > Authen tication is done
> >
> > *# - auth_db params -
> > /* uncomment the following lines if you want to enable the DB based
> >authentication */
> > modparam("auth_db", "calculate_ha1", yes)
> > modparam("auth_db", "password_column", "password")
> > modparam("auth_db", "db_url",
> > "mysql://opensips:opensip...@localhost/opensips")
> > modparam("auth_db", "load_credentials", "")
> > *
> >
> > *if (is_from_local()){
> > # From an internal domain -> check the credentials and the FROM
> > if (method=="MESSAGE") {
> > log(1,"\n--> ROUTE
> > 3 MESSAGE Looop---\n");
> > route(17);
> > };
> >if(!allow_trusted()){
> > if (!proxy_authorize("","subscriber")) {
> > proxy_challenge("","0");
> > exit;
> > } else if(!check_from()) {
> >   sl_send_reply("403", "Forbidden, use From=ID");
> >   exit;
> > };
> > };
> > if (client_nat_test("3")) {
> > append_hf("P-hint: setflag7|forcerport|fix_contact\r\n");
> > setbflag(7);
> > force_rport();
> > fix_contact();
> > };
> > #unconditional call forward
> > if(avp_db_load("$ru/username","$avp(s:callfwd)")) {
> > avp_pushto("$ru", "$avp(s:callfwd)");
> > route(1);
> > exit;
> > }
> >
> > *   *consume_credentials();*
> >
> >   * if (uri=~"^sip:00[0-9]{6,20}@") {
> > if (is_user_in("Credentials","local")) {
> > route(6);
> > log(1,"\n*** I AM GOING TO ENTER ROUTE
> > 4");
> > route(4);
> > exit;
> > } else {
> > sl_send_reply("403", "No permissions for local calls");
> > exit;
> > };
> > };*
> >
> >
> > Can you tell me where i may be going wrong
> >
> >
> > This is the piece of script
> > On Fri, Aug 21, 2009 at 5:58 PM, Bogdan-Andrei Iancu
> > mailto:bog...@voice-system.ro>> wrote:
> >
> > HI Ashwini,
> >
> > If you wan to used the Credentials, then you need to be sure you did
> > authentication before (in script).
> >
> > Regards,
> > Bogdan
> >
> > ASHWINI NAIDU wrote:
> > > Hi all,
> > >
> > >  I have installed opensips-1.5. I have applied the required
> > patch
> > > for group. When i use
> > >
> > >* is_user_in("Credentials", "local") { *
> > >
> > >   I get the following error
> > >
> > >  *ERROR:auth:consume_credentials: no authorized credentials found
> > > (error in scripts)
> > > Aug 21 17:09:30 debian /sbin/opensips[18916]:
> > > ERROR:group:get_username_domain: no authorized credentials found
> > > (error in scripts)
> > > Aug 21 17:09:30 debian /sbin/opensips[18916]:
> > ERROR:group:is_user_in:
> > > failed to get usern...@domain*
> > >
> > > Can anyone say what may be the problem.
> > > --
> > > Thanking You,
> > > Ashwini BR Naidu
> > >
> >
> 
> > >
> > > ___
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> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
> >
> > ___
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> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> >
> > --
> > Thanking You,
> > Ashwini BR Naidu
> > 
> >
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> >
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-- 
Thanking You,
Ashwini BR Naidu
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[OpenSIPS-Users] Dialog information tracing in opensips Issue

[urmi lakkad]

Hello Bogdan,

Thank you very much.
Here I have attached my OpenSIPs Log. so please find the attachment.

-Urmi



Looks good - can you post the opensips logs (in debug=6) for this single
call ? just to verify.

Regards,
Bogdan

urmi lakkad wrote:
> Hello Bogdan,
>
> Thank you for ur response.
>
> Here with this mail I have _attached my SIP call capture_ using ngrep.
> So, please find the attachment. and do needful.
>
>
> -Thanks
> Urmi
>
> On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
> Post the SIP capture of the call you are testing with. Use "ngrep
> -d any
> . port 5060" to get the capture - this will solve the mystery.
>
> Regards,
> Bogdan
>
> urmi lakkad wrote:
> > Hello,
> >
> > Can u please suggest me some solution of my problem of DIALLOG
> module ?
> >
> > Thank you for your attention.
> >
> > -Urmi
> >
> > 2009/8/21 urmi lakkad  
> > >>
> >
> > Hello Brett,
> >
> > Thank you very much for quick response.
> >
> > My calls are working fine. I have checked through SIPp and also
> > with Grandstream Phones. The call is working fine with out
> > failure. At the time of call, I have started the wireshark to
> > capture the packets, but there also I m not getting any negative
> > reply like 400 or 300.
> >
> > See, my call is working fine, call dialog created successfully,
> > but after that it destroyed, again new dialog is created n that
> > too destroy. For a single call it creates 2 dialogs. But that
> > dialog entry is not going to DB. Please suggest me the right thing
> > to do.
> >
> > Thanks a lot for your attention.
> >
> >
> > -Urmi
> >
> >
> >
> > On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> > mailto:br...@nemeroff.com>
> >> wrote:
> >
> > Urmi,
> > You log shows the call having failed. I'm not sure why you
> > think it runs for the proper duration. But as far as OpenSIPs
> > is concerned, the call failed. It's likely a problem in your
> > sipp scenario. It's very possible that sipp thinks the call is
> > up, but the proxy does not.
> >
> > In any case, OpenSIPs is behaving as expected, the call fails,
> > the dialog is destroyed.
> > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > 0x2d55af90 changed from state 1 to state 5, due event 1
> > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > 0x2d55af90
> > failed (negative reply)
> >
> > BTW, a negative reply is >=400 (or may also include >= 300,
> > can't remember). Check your traces, see where that comes from.
> > -Brett
> >
> > On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> > mailto:urmi.lak...@gmail.com>
> >> wrote:
> >
> > Hello Stanisław Pitucha,
> >
> > Thank you for support.
> >
> > No, My call is established perfectly and is running for
> > the specified duration without fail.
> > I m firing the call using SIPp.
> >
> > Also, the dialog state gives me 1.
> >
> > Thanks for ur attention.
> >
> > -Urmi
> >
> > 2009/8/20 Stanisław Pitucha  
> > >>
> >
> > 2009/8/20 urmi lakkad  
> > >>:
> > > Am I doing right or not ? If not, please tell me the
> > correct way.
> > > One more thing, Is my configuration is correct or not ??
> >
> > It looks like your call doesn't even get accepted:
> > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > 0x2d55af90 changed from state 1 to state 5, due
> > event 1
> > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > 0x2d55af90
> > failed (negative reply)
> >
> > Maybe you require authentication, or something else?
> > Just take care of
> > the call not failing first. So far it's rejected
> > before an OK answer
> > (state 1 is "after sending an INVITE", state 5 is
> > "deleted" - more or
> > less).
> > Capture the traffic and see what's going on.
> >
> > --
> > Kind regards,
> >
> > Stanisław Pitucha, Gradwell Voip Engineer
> >
> > T: 01225 800 831 | F: 01225 800 801 | E:
> > s...@gradwell.net 
> > |
> > www.gradwell.com  
> >
> > Gradwell – Internet for Business People
> > Phone Services | Business Broadband | Email & W

[OpenSIPS-Users] Dialog information tracing in opensips Issue

[urmi lakkad]

Hello Bogdan,

Thank you very much for your quick response.
Here I have attached OpenSIPs call log.

-Urmi


On Tue, Aug 25, 2009 at 3:54 PM, Bogdan-Andrei Iancu  wrote:

> Looks good - can you post the opensips logs (in debug=6) for this single
> call ? just to verify.
>
> Regards,
> Bogdan
>
> urmi lakkad wrote:
> > Hello Bogdan,
> >
> > Thank you for ur response.
> >
> > Here with this mail I have _attached my SIP call capture_ using ngrep.
> > So, please find the attachment. and do needful.
> >
> >
> > -Thanks
> > Urmi
> >
> > On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu
> > mailto:bog...@voice-system.ro>> wrote:
> >
> > Post the SIP capture of the call you are testing with. Use "ngrep
> > -d any
> > . port 5060" to get the capture - this will solve the mystery.
> >
> > Regards,
> > Bogdan
> >
> > urmi lakkad wrote:
> > > Hello,
> > >
> > > Can u please suggest me some solution of my problem of DIALLOG
> > module ?
> > >
> > > Thank you for your attention.
> > >
> > > -Urmi
> > >
> > > 2009/8/21 urmi lakkad  > 
> > > >>
> > >
> > > Hello Brett,
> > >
> > > Thank you very much for quick response.
> > >
> > > My calls are working fine. I have checked through SIPp and also
> > > with Grandstream Phones. The call is working fine with out
> > > failure. At the time of call, I have started the wireshark to
> > > capture the packets, but there also I m not getting any negative
> > > reply like 400 or 300.
> > >
> > > See, my call is working fine, call dialog created successfully,
> > > but after that it destroyed, again new dialog is created n that
> > > too destroy. For a single call it creates 2 dialogs. But that
> > > dialog entry is not going to DB. Please suggest me the right thing
> > > to do.
> > >
> > > Thanks a lot for your attention.
> > >
> > >
> > > -Urmi
> > >
> > >
> > >
> > > On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> > > mailto:br...@nemeroff.com>
> > >> wrote:
> > >
> > > Urmi,
> > > You log shows the call having failed. I'm not sure why you
> > > think it runs for the proper duration. But as far as OpenSIPs
> > > is concerned, the call failed. It's likely a problem in your
> > > sipp scenario. It's very possible that sipp thinks the call is
> > > up, but the proxy does not.
> > >
> > > In any case, OpenSIPs is behaving as expected, the call fails,
> > > the dialog is destroyed.
> > > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > > 0x2d55af90 changed from state 1 to state 5, due event 1
> > > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > > 0x2d55af90
> > > failed (negative reply)
> > >
> > > BTW, a negative reply is >=400 (or may also include >= 300,
> > > can't remember). Check your traces, see where that comes from.
> > > -Brett
> > >
> > > On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> > > mailto:urmi.lak...@gmail.com>
> > >>
> wrote:
> > >
> > > Hello Stanisław Pitucha,
> > >
> > > Thank you for support.
> > >
> > > No, My call is established perfectly and is running for
> > > the specified duration without fail.
> > > I m firing the call using SIPp.
> > >
> > > Also, the dialog state gives me 1.
> > >
> > > Thanks for ur attention.
> > >
> > > -Urmi
> > >
> > > 2009/8/20 Stanisław Pitucha  > 
> > > >>
> > >
> > > 2009/8/20 urmi lakkad  > 
> > > >>:
> > > > Am I doing right or not ? If not, please tell me the
> > > correct way.
> > > > One more thing, Is my configuration is correct or not ??
> > >
> > > It looks like your call doesn't even get accepted:
> > > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > > 0x2d55af90 changed from state 1 to state 5, due
> > > event 1
> > > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > > 0x2d55af90
> > > failed (negative reply)
> > >
> > > Maybe you require authentication, or something else?
> > > Just take care of
> > > the call not failing first. So far it's rejected
> > > before an OK answer
> > > (state 1 is "after sending an INVITE", state 5 is
> > > "deleted" - more or
> > > less).
> > > Capture the traffic and see what's going on.
>
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Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue

[Bogdan-Andrei Iancu]

Looks good - can you post the opensips logs (in debug=6) for this single 
call ? just to verify.

Regards,
Bogdan

urmi lakkad wrote:
> Hello Bogdan,
>
> Thank you for ur response.
>
> Here with this mail I have _attached my SIP call capture_ using ngrep.
> So, please find the attachment. and do needful.
>
>
> -Thanks
> Urmi
>
> On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Post the SIP capture of the call you are testing with. Use "ngrep
> -d any
> . port 5060" to get the capture - this will solve the mystery.
>
> Regards,
> Bogdan
>
> urmi lakkad wrote:
> > Hello,
> >
> > Can u please suggest me some solution of my problem of DIALLOG
> module ?
> >
> > Thank you for your attention.
> >
> > -Urmi
> >
> > 2009/8/21 urmi lakkad  
> > >>
> >
> > Hello Brett,
> >
> > Thank you very much for quick response.
> >
> > My calls are working fine. I have checked through SIPp and also
> > with Grandstream Phones. The call is working fine with out
> > failure. At the time of call, I have started the wireshark to
> > capture the packets, but there also I m not getting any negative
> > reply like 400 or 300.
> >
> > See, my call is working fine, call dialog created successfully,
> > but after that it destroyed, again new dialog is created n that
> > too destroy. For a single call it creates 2 dialogs. But that
> > dialog entry is not going to DB. Please suggest me the right thing
> > to do.
> >
> > Thanks a lot for your attention.
> >
> >
> > -Urmi
> >
> >
> >
> > On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> > mailto:br...@nemeroff.com>
> >> wrote:
> >
> > Urmi,
> > You log shows the call having failed. I'm not sure why you
> > think it runs for the proper duration. But as far as OpenSIPs
> > is concerned, the call failed. It's likely a problem in your
> > sipp scenario. It's very possible that sipp thinks the call is
> > up, but the proxy does not.
> >
> > In any case, OpenSIPs is behaving as expected, the call fails,
> > the dialog is destroyed.
> > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > 0x2d55af90 changed from state 1 to state 5, due event 1
> > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > 0x2d55af90
> > failed (negative reply)
> >
> > BTW, a negative reply is >=400 (or may also include >= 300,
> > can't remember). Check your traces, see where that comes from.
> > -Brett
> >
> > On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> > mailto:urmi.lak...@gmail.com>
> >> wrote:
> >
> > Hello Stanisław Pitucha,
> >
> > Thank you for support.
> >
> > No, My call is established perfectly and is running for
> > the specified duration without fail.
> > I m firing the call using SIPp.
> >
> > Also, the dialog state gives me 1.
> >
> > Thanks for ur attention.
> >
> > -Urmi
> >
> > 2009/8/20 Stanisław Pitucha  
> > >>
> >
> > 2009/8/20 urmi lakkad  
> > >>:
> > > Am I doing right or not ? If not, please tell me the
> > correct way.
> > > One more thing, Is my configuration is correct or not ??
> >
> > It looks like your call doesn't even get accepted:
> > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > 0x2d55af90 changed from state 1 to state 5, due
> > event 1
> > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > 0x2d55af90
> > failed (negative reply)
> >
> > Maybe you require authentication, or something else?
> > Just take care of
> > the call not failing first. So far it's rejected
> > before an OK answer
> > (state 1 is "after sending an INVITE", state 5 is
> > "deleted" - more or
> > less).
> > Capture the traffic and see what's going on.
> >
> > --
> > Kind regards,
> >
> > Stanisław Pitucha, Gradwell Voip Engineer
> >
> > T: 01225 800 831 | F: 01225 800 801 | E:
> > s...@gradwell.net 
> > |
> > www.gradwell.com  
> >
> > Gradwell – Internet for Business People
> > Phone Services | Business Broadband | Email & Website
> > Hosting
> >
> > Can switching to VoIP today put some change in your
> > pocket?
> 

Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue

[urmi lakkad]

Hello Bogdan,

Thank you for ur response.

Here with this mail I have *attached my SIP call capture* using ngrep.
So, please find the attachment. and do needful.


-Thanks
Urmi

On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:

> Post the SIP capture of the call you are testing with. Use "ngrep -d any
> . port 5060" to get the capture - this will solve the mystery.
>
> Regards,
> Bogdan
>
> urmi lakkad wrote:
> > Hello,
> >
> > Can u please suggest me some solution of my problem of DIALLOG module ?
> >
> > Thank you for your attention.
> >
> > -Urmi
> >
> > 2009/8/21 urmi lakkad  > >
> >
> > Hello Brett,
> >
> > Thank you very much for quick response.
> >
> > My calls are working fine. I have checked through SIPp and also
> > with Grandstream Phones. The call is working fine with out
> > failure. At the time of call, I have started the wireshark to
> > capture the packets, but there also I m not getting any negative
> > reply like 400 or 300.
> >
> > See, my call is working fine, call dialog created successfully,
> > but after that it destroyed, again new dialog is created n that
> > too destroy. For a single call it creates 2 dialogs. But that
> > dialog entry is not going to DB. Please suggest me the right thing
> > to do.
> >
> > Thanks a lot for your attention.
> >
> >
> > -Urmi
> >
> >
> >
> > On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> > mailto:br...@nemeroff.com>> wrote:
> >
> > Urmi,
> > You log shows the call having failed. I'm not sure why you
> > think it runs for the proper duration. But as far as OpenSIPs
> > is concerned, the call failed. It's likely a problem in your
> > sipp scenario. It's very possible that sipp thinks the call is
> > up, but the proxy does not.
> >
> > In any case, OpenSIPs is behaving as expected, the call fails,
> > the dialog is destroyed.
> > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > 0x2d55af90 changed from state 1 to state 5, due event 1
> > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > 0x2d55af90
> > failed (negative reply)
> >
> > BTW, a negative reply is >=400 (or may also include >= 300,
> > can't remember). Check your traces, see where that comes from.
> > -Brett
> >
> > On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> > mailto:urmi.lak...@gmail.com>> wrote:
> >
> > Hello Stanisław Pitucha,
> >
> > Thank you for support.
> >
> > No, My call is established perfectly and is running for
> > the specified duration without fail.
> > I m firing the call using SIPp.
> >
> > Also, the dialog state gives me 1.
> >
> > Thanks for ur attention.
> >
> > -Urmi
> >
> > 2009/8/20 Stanisław Pitucha  > >
> >
> > 2009/8/20 urmi lakkad  > >:
> > > Am I doing right or not ? If not, please tell me the
> > correct way.
> > > One more thing, Is my configuration is correct or not
> ??
> >
> > It looks like your call doesn't even get accepted:
> > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > 0x2d55af90 changed from state 1 to state 5, due
> > event 1
> > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > 0x2d55af90
> > failed (negative reply)
> >
> > Maybe you require authentication, or something else?
> > Just take care of
> > the call not failing first. So far it's rejected
> > before an OK answer
> > (state 1 is "after sending an INVITE", state 5 is
> > "deleted" - more or
> > less).
> > Capture the traffic and see what's going on.
> >
> > --
> > Kind regards,
> >
> > Stanisław Pitucha, Gradwell Voip Engineer
> >
> > T: 01225 800 831 | F: 01225 800 801 | E:
> > s...@gradwell.net  |
> > www.gradwell.com 
> >
> > Gradwell – Internet for Business People
> > Phone Services | Business Broadband | Email & Website
> > Hosting
> >
> > Can switching to VoIP today put some change in your
> > pocket?
> > Registered Address: 26 Cheltenham Street, Bath, BA2
> > 3EX, UK. Company
> > Number: 3673235
> >
> > ___
> > Users ma

[OpenSIPS-Users] Next OpenSIPS releases

[Bogdan-Andrei Iancu]

Hi,

Here are the plans for the next OpenSIPS releases (minor and major). 
This is an initial draft (content and dates), so please comment and 
contribute (if necessary):


1) Minor release 1.5.3
---

Why: This is needed as more than 50 fixes were done on the 1.5 branch 
since 1.5.2.

Date: during this week

Pending: personally I'm hunting an memory leak in SNMP module (mainly 
design issues). If someone is aware of any other issues that requires 
fixing in 1.5 branch, please speak up.


2) Major release 1.6.0
---

Code freeze:  estimated for mid September (depending of how fast the 
pending work is completed)

Release date: estimated for October

What we have so far: http://www.opensips.org/Main/Ver160

Pending work:
- adding context for PVs (like reply, request)
- route types - init, onreply per branch, timer based
- dialog - early dialog support to be finished; new functions to 
check dialog consistency (cseq numbers, route set, contacts); dialog 
direction function
- pike enhancement for catching more events (replies, non-SIP 
traffic attacks)
- json support
If there is something missing or if somebody is working some (new) code 
and needs time and support, please let me know.


Regards,
Bogdan





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Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue

[Bogdan-Andrei Iancu]

Post the SIP capture of the call you are testing with. Use "ngrep -d any 
. port 5060" to get the capture - this will solve the mystery.

Regards,
Bogdan

urmi lakkad wrote:
> Hello,
>
> Can u please suggest me some solution of my problem of DIALLOG module ?
>
> Thank you for your attention.
>
> -Urmi
>
> 2009/8/21 urmi lakkad  >
>
> Hello Brett,
>
> Thank you very much for quick response.
>
> My calls are working fine. I have checked through SIPp and also
> with Grandstream Phones. The call is working fine with out
> failure. At the time of call, I have started the wireshark to
> capture the packets, but there also I m not getting any negative
> reply like 400 or 300.
>
> See, my call is working fine, call dialog created successfully,
> but after that it destroyed, again new dialog is created n that
> too destroy. For a single call it creates 2 dialogs. But that
> dialog entry is not going to DB. Please suggest me the right thing
> to do.
>
> Thanks a lot for your attention.
>
>
> -Urmi
>
>
>
> On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> mailto:br...@nemeroff.com>> wrote:
>
> Urmi,
> You log shows the call having failed. I'm not sure why you
> think it runs for the proper duration. But as far as OpenSIPs
> is concerned, the call failed. It's likely a problem in your
> sipp scenario. It's very possible that sipp thinks the call is
> up, but the proxy does not.
>
> In any case, OpenSIPs is behaving as expected, the call fails,
> the dialog is destroyed.
> Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> 0x2d55af90 changed from state 1 to state 5, due event 1
> Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> 0x2d55af90
> failed (negative reply)
>
> BTW, a negative reply is >=400 (or may also include >= 300,
> can't remember). Check your traces, see where that comes from.
> -Brett
>
> On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> mailto:urmi.lak...@gmail.com>> wrote:
>
> Hello Stanisław Pitucha,
>
> Thank you for support.
>
> No, My call is established perfectly and is running for
> the specified duration without fail.
> I m firing the call using SIPp.
>
> Also, the dialog state gives me 1.
>
> Thanks for ur attention.
>
> -Urmi
>
> 2009/8/20 Stanisław Pitucha  >
>
> 2009/8/20 urmi lakkad  >:
> > Am I doing right or not ? If not, please tell me the
> correct way.
> > One more thing, Is my configuration is correct or not ??
>
> It looks like your call doesn't even get accepted:
> Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> 0x2d55af90 changed from state 1 to state 5, due
> event 1
> Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> 0x2d55af90
> failed (negative reply)
>
> Maybe you require authentication, or something else?
> Just take care of
> the call not failing first. So far it's rejected
> before an OK answer
> (state 1 is "after sending an INVITE", state 5 is
> "deleted" - more or
> less).
> Capture the traffic and see what's going on.
>
> --
> Kind regards,
>
> Stanisław Pitucha, Gradwell Voip Engineer
>
> T: 01225 800 831 | F: 01225 800 801 | E:
> s...@gradwell.net  |
> www.gradwell.com 
>
> Gradwell – Internet for Business People
> Phone Services | Business Broadband | Email & Website
> Hosting
>
> Can switching to VoIP today put some change in your
> pocket?
> Registered Address: 26 Cheltenham Street, Bath, BA2
> 3EX, UK. Company
> Number: 3673235
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@li

Re: [OpenSIPS-Users] SRV failover results in double call

[Bogdan-Andrei Iancu]

Brent Thomson wrote:
> Stanisław Pitucha wrote:
>   
>> 2009/8/24 Brent Thomson :
>> 
>>> Bogdan-Andrei Iancu wrote:
>>>   
 Hi Brent,

 This problem was reported last week by another person and fixed on SVN
 (including in 1.6 branch).

 What you have to do is to upgrade from SVN and hopefully the problem
 will be solved.

 
>>> Cool. Thanks.
>>>   
>> Or if you're ok with applying custom patches, just pull the trunk
>> change - rev 6007. It applies to opensips 1.5.2 just fine.
>> 
>
> Thanks for the tip. Patching the release version is definitely
> preferred. I ran (all on one line):
>
> svn diff -r 6006:6007
> https://opensips.svn.sourceforge.net/svnroot/opensips/trunk > opensips.diff
>
> and got about 8 lines of changes in modules/tm/tm_reply.c. Does this
> seem about right?
>   
Yes, that is correct.

Regards,
Bogdan


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