[OpenSIPS-Users] Getting asterisk to reply to OPTIONS keepalive

[Alejandro Recarey]

Hi all,

I'm trying to use the keepalive / ping option in OpenSIPS Loadbalancer
module, but asterisk always responds with a 404 not found. Has anybody
managed to get this working with Asterisk 1.6.1.X + OpenSIPS 1.6.2 ?

Thanks!

Alex

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Re: [OpenSIPS-Users] [dialog] Can't fetch data in 200 reply route of BYE

[liuf]

Yes, you are right. They're the same.

Had this bug been fixed yet? It seems still open now.


Best Regards, 
Liu Fan
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Re: [OpenSIPS-Users] Opensips children process

[Bogdan-Andrei Iancu]

Hi Rajib,

each SIP message (request or reply) is read from the network by a child 
(any of them) and it will be processed (to the end) only by that child. 
A child processes only one message at a time..

Regards,
Bogdan

rajib deka wrote:
> Hi All,
>  
> Can anybody please explain me the use of opensips children processes? 
> Is it something like if children=4 and opensips gets 4 INVITES 
> tranjactions, then one children will work on one INVITE. Correct me if 
> I am wrong.
>  
> Thanks
> -- 
> Rajib Deka
> Software Engineer
> Servion Global Solution
> Chennai, India
>
> Mobile No: + 91 80157 09130
> 
>
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-- 
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www.voice-system.ro


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Re: [OpenSIPS-Users] db_berkeley append

[liuf]

Thanks. 

Base svn revision 6624, I check files in db_berkeley/opensips/ again. I
found the files which have no METADATA_KEY value as below:

  dialplan
  dr_gateways
  dr_groups
  dr_rules
  globalblacklist
  nh_sockets
  userblacklist

maybe you need fix them also. 
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[OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[erick]

> I have the following problem
>
> With install OpenSIPS  1.6 and  Control Panel 4.0.
>
> When try the access the Control Panel didn't accept the user o password
> (Invalid Username / Password)
>
> With the  commands opensips-c, monit-t said  that the configuration files
> are fine.
>
>  I can register phone.
>
> The question is how do I know the password to the web interface, try to
> file
> db.inc.php, boxes.global.inc.php, or  if I have to change something in the
> database
>
> The reference for the installation of the control panel are Book OpenSIPS
> 1.6 and the link  http://opensips-cp.sourceforge.net/ , but both have
> differences
>
> This is my first time with OpenSIPS
>
> I appreciate the help
>
> Thanks
>
> Erick Ch.
>



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[OpenSIPS-Users] pua/rls modules pidf-diff+xml support

[Patrice Dalle]

Hi,

Does pua/rls modules properly support pidf-diff+xml?

When subscribing to a rls service (that contains an embedded resource list)
I am able to get presence state notifications for buddies that publish their
state using pidf full (pidf+xml).
PUBLISH request is as below:

PUBLISH sip:5...@raftman SIP/2.0
...
Event: presence
Content-Type: application/pidf+xml
Content-Length: 253





open





But when buddies use pidf-diff such as below I cannot get presence
notification updates.

PUBLISH sip:5...@raftman SIP/2.0
...
Event: presence
Content-Type: application/pidf-diff+xml
Content-Length: 301





open





According to RFC5262 it is legal to have a pidf-full XML element within an
"application/pidf-diff+xml" payload.
Is it a bug or it is simply not supported?

Thanks
Patrice
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Re: [OpenSIPS-Users] lb_reload

[Bogdan-Andrei Iancu]

Hi Dimitri,

reloading the data should not delete the dialog profiles, unless you 
change the name of the resources (in the destination definition).
Is this your case?

Regards,
Bogdan

Dmitri G. wrote:
> Hi,
>
> Is there a way to keep the number of the dialogs after running the 
> lb_reload command through openipsctl?
> When I add a new gateway to the mysql able and execute lb_reload 
> command, the number of the dialogs are resets to zero in every 
> gateways (I have checked it with lb_list, load counters are zero)
>
> Thanks,
> Dimitri
> 
>
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www.voice-system.ro


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Re: [OpenSIPS-Users] global variables

[Josip Djuricic]

Sorry I missed this email,

I'll think about using db_virtual, but back to global vars issue, for a 
test I defined just a global counter variable, and created new function 
inside siptrace module, so every time a siptrace function is called I 
also call this function and do counter ++. since counter is defined 
outside of the function, it should be defined as global variable, so 
I've outputed with LM_INFO value of counter and it's always 1, sometime 
it gets pass that value, but on the next call it's 1 again. Perhaps I'm 
thinking wrong and siptrace module is not having it's own fork, but 
instead loads inside other forks? That would explain this behaviour.

If that is the case, what would you suggest? I would then need to use 
interprocess communication, perhaps I could reuse something from opensips?

Best regards,

Josip

Bogdan-Andrei Iancu wrote:
> Hi Josip,
>
> Josip Djuricic wrote:
>> Hi,
>>
>> one probably stupid question, I'm customizing siptrace module further,
>> so I am adding queue for db writing, in case we loose db connectivity
>> and threads for popping the queue and writing to db.
>>   
> why don't you use the new db_virtual module with a failover 
> configuration: define a new virtual db connection that has behind a 
> primary mysql conn and a secondary (failover only) flatstore/dbtext 
> conn. So, if DB fails, you will start writing to a local file.
>> Written almost everything, but just to be sure, if I use global variable
>> for linked list, will they keep the values until opensips ends, or will
>> they always be reinitialized, because for test purposes I defined
>> counter but it's always initialized to default value.
>>   
> To be honest I do not advices using thread in current opensips as you 
> have multiple processes already. And the combination of threads and 
> processes may be unpredictable.
> Anyhow, regarding your question - the global vars are init only at 
> startup time - but note that "global" is as visibility in the code and 
> not global "in all processes". Each process will have its own copy of 
> the var.
>
> Regards,
> Bogdan
>


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Re: [OpenSIPS-Users] "Explaining the default script" is the next webinar

[Noah Pugsley]

Is it possible to download the movie files themselves? Not everyone can 
or is allowed to use Flash.

Thank you,
noah

Bogdan-Andrei Iancu wrote:
> The recording of this webinar is now available under :
>http://www.opensips.org/html/docs/video/webinar005/
> 
> Regards,
> Bogdan
> 
> 
> 
> Bogdan-Andrei Iancu wrote:
>> Next webinar is scheduled for 25th of February 2010.
>>
>> The topic is "Explaining the default script" - A detailed going though 
>> and explanation of the OpenSIPS default script - configuration params, 
>> the flow of messages in the script, what each part is good for, etc
>>
>> Free registration  at  http://www.opensips.org/Training/Webinars#toc6
>>
>>
>> Best regards,
>> Bogdan
>>
>>   
> 
> 


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[OpenSIPS-Users] Problem with OpenSIPS Control Panel 4.0

[Erick Chinchilla Berrocal]

I have the following problem 

With install OpenSIPS  1.6 and  Control Panel 4.0.

When try the access the Control Panel didn't accept the user o password
(Invalid Username / Password)

With the  commands opensips-c, monit-t said  that the configuration files
are fine.

 I can register phone.

The question is how do I know the password to the web interface, try to file
db.inc.php, boxes.global.inc.php, or  if I have to change something in the
database

The reference for the installation of the control panel are Book OpenSIPS
1.6 and the link  http://opensips-cp.sourceforge.net/ , but both have
differences 

This is my first time with OpenSIPS

I appreciate the help 

Thanks

Erick Ch.

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Re: [OpenSIPS-Users] [OpenSIPS-Devel] Soon you can Blink!

[Adrian Georgescu]

On Mar 10, 2010, at 7:49 PM, Thomas Gelf wrote:

> Adrian Georgescu schrieb:
>>> * How does it figure out it's reflective candidate if I didn't
>>> configure a STUN server?
>>
>> The stun server must be in DNS of the caller domain or you can add
>> them manually in Account -> Advanced  -> Nat Traversal section
>
> Ok, that explains it -> STUN server has been in the DNS :) But also
> adding it manually to "Account -> Adv..." AND enabling ICE somehow
> "blocked" Blink when trying to place a call, at least it didn't do
> anything - and it didn't send packets to my SIP proxy.

Can you enable Debug -> Notifications next time and paste them back?

> At least that's
> what I have been told, have not been sitting in front of that MAC  
> while
> we did the tests with those different settings...
>
>> Still it has to work when stun does not work. Any visible reason in /
>> var/log/system.log or Debug window ?
>
> No idea, sorry. And the MAC is no longer here :( I'll try it with more
> calm once I get one next time!

Again all you do with Blink can you can reproduce with the command  
line tools from SIP SIMPLE client SDK. Any luck installing those?

You can toggle Trace Notifications in sip-session or sip-audio-session  
too.

Adrian




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Re: [OpenSIPS-Users] lb_reload

[Dmitri G.]

Hi Bogdan,

I'm running OpenSIPS 1.6.1

version: opensips 1.6.1-notls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: 2:6599M
@(#) $Id: main.c 6169 2009-09-22 12:48:37Z bogdan_iancu $
main.c compiled on 01:18:50 Feb 15 2010 with gcc 4.3.4

BTW, what I'm doing frequently is to resize the existing resources, and
adding/removing new asterisk gws in the load_balancer table.

Thanks,
Dimitri



2010/3/10 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 12:52 (acum 5
ore)

> Hi Dimitri,
>
> Ok, so you do lb_reload just because you need to add some more
> destinations in the balancer. You do not change the properties of the
> already existing destination - this was what I was interested in.
>
> Let me run some checks to see what about. BTW, what opensips version are
> you using ?
>
> Regards,
> Bogdan
>
>
> Dmitri G. wrote:
> > I'm changing the resource frequently, btw these are asterisk boxes
> > with agents, an inbound call center.
> > Every morning they call me and tell "dimitri today we have 200 agent
> > for xx.xx.xx.xx, and we have 150 for yy.yy.yy.yy".
> > So I run a lb_resize to change the resource figure (the number of
> > available agents).
> > There is no problem with this and I never change the name of the
> > resource, just the figure. I can resize it without any problems.
> >
> > But in case if they call me to add a new asterisk ip with resources,
> > then I need to insert it to a DB, to the load_balancer table.
> > But how to enable to use the new gw other than using lb_reload?
> >
> > If I run the lb_reload, and there was concurrent calls to the asterisk
> > boxes (so agent=200, and agent=150), the load to each destination will
> > be zero (load=0). Buit there are still active calls to the
> > destinations so if I have 150 calls to one, and if there will be a new
> > call to the destination, after a lb_reload I see load=1, where in
> > reality there was 151 calls.
> >
> > See my problem?
> >
> > Thanks,
> >
> > Dimitri.
> >
> > 2010/3/9 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 16:26
> > (acum 27 minute)
> >
> > Ok, I got that :)
> >
> > my question was - what you change in DB (before doing the reload)? do
> > you change the definition of the already existing destinations (like
> > resources) ?
> >
> > Regards,
> > Bogdan
> >
> > Dmitri G. wrote:
> > > Hi Bogdan,
> > >
> > > I mean it's setting the load= figures to zero for all gws in the
> > > load_balancer table.
> > > So if I have 2 gws with load=136 and load=227, after running
> > > opensipsctl fifo lb_reload, then load will be zero (load=0).
> > >
> > > Dimitri
> > >
> > > 2010/3/9 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 15:57
> > > (acum 18 minute)
> > >
> > > Hi Dimitri,
> > >
> > > reloading the data should not delete the dialog profiles, unless
> you
> > > change the name of the resources (in the destination definition).
> > > Is this your case?
> > >
> > > Regards,
> > > Bogdan
> > >
> > > Dmitri G. wrote:
> > > > Hi,
> > > >
> > > > Is there a way to keep the number of the dialogs after running
> the
> > > > lb_reload command through openipsctl?
> > > > When I add a new gateway to the mysql able and execute lb_reload
> > > > command, the number of the dialogs are resets to zero in every
> > > > gateways (I have checked it with lb_list, load counters are zero)
> > > >
> > > > Thanks,
> > > > Dimitri
> > > >
> > >
> >
> 
> > > >
> > > > ___
> > > > Users mailing list
> > > > Users@lists.opensips.org 
> > >
> > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > > >
> > >
> > >
> > > --
> > > Bogdan-Andrei Iancu
> > > www.voice-system.ro 
> > 
> > >
> > >
> > > ___
> > > 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

Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk

[Web Support]

Bogdan,

I am using the fix_nated_contact() because if I don't, it tries to register
with the private IP of the device on my network... still not the public IP
of it.

Perhaps I'm attacking this the wrong way, maybe you can clarify for me what
I actually want to do. The end I'm trying to reach is to secure my dialtone
via TLS, as Asterisk (1.4 due to PBXware) doesn't support it. 

After reading forums, mailing lists, googling non-stop... everything is
starting to blur. Maybe you can outline for me the steps I need to put into
place to make this work. As mentioned before, ideally I would like Asterisk
to acknowledge the UACs as their public IPs, not the OpenSIPS server, as I'm
putting together a multi-tenanted environment. My understanding is that I
would:

1. Register UAC through OpenSIPS via TLS.
2. Registration is relayed to Asterisk, Asterisk would see the public IP of
the UAC.
3. INVITES made from a registered UAC would in turn relay through OpenSIPS
to Asterisk, which would send it to the destination directly, bypassing
OpenSIPS. 

Is this logic flawed?

-Tyler

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, March 10, 2010 11:56
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk

Tyler,

The script logic (for REGISTER) is a bit broken - you authenticate the 
REGISTER, you save it locally, send a reply to UAC and then relay the 
REGISTER to asterisk; which will send you back a reply, so you will end 
up with 2 replies back to UAC :)

Regarding the contact part, I see you are using the fix_nated_contact()  
function - this is the only function that seams to modify the CT hdr.  
To validate this assumption, try commenting the function and run again.

Regards,
Bogdan

Web Support wrote:
> Bogdan, or anyone,
>
> How can I relay the public IP of my UAD off to Asterisk instead of the IP
of
> the OpenSIPS server?
>
> Everything I've tried has not changed anything. I am using
> fix_nated_contact(), as before using it, it was trying to register the
> private IP of the UAD. 
>
> -Tyler
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> Sent: Tuesday, March 09, 2010 13:34
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>
> forgot the script :)
>
> but look for fix_nated_contact() if present
>
> Regards,
> Bogdan
>
> Web Support wrote:
>   
>> I've attached the .cfg, I'm sure it's a bit of a mess in your eyes... I'm
>> still getting the grasp of OpenSIPS scripting and I'm having a hard time
>> figuring out what is returned from certain function calls.
>>
>> I am not knowingly changing the Contact hdr, I think I want to though, so
>> that it reflects the public IP of the UAD, no?
>>
>> Thanks for your help so far Bogdan.
>>
>> -Tyler
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
>> Sent: Tuesday, March 09, 2010 13:25
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>>
>> Ok, in this case I assume opensips is change it, right ?
>>
>> so the question is - does your opensips changes the Contact hdr in 
>> REGISTERs ?
>>
>> Regards,
>> Bogdan
>>
>> Web Support wrote:
>>   
>> 
>>> Hello Bogdan,
>>>
>>> The IP is of my OpenSIPS server.
>>>
>>> -Tyler
>>>
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org
>>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
>>>   
> Iancu
>   
>>> Sent: Tuesday, March 09, 2010 12:34
>>> To: OpenSIPS users mailling list
>>> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to
Asterisk
>>>
>>> Hi Tyler,
>>>
>>> can you check at network level (use tcpdump, ngrep), in the REGISTER 
>>> from opensips to asterisk, what is the Contact header? does this Contact

>>> hdr contain an IP of opensips or an IP of UAD ?
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Web Support wrote:
>>>   
>>> 
>>>   
 Currently I have OpenSIPS installed on one machine, and have my 
 Asterisk (PBXware) on another machine. When trying to register my UAD 
 through OpenSIPS, the relay is happening correctly and the device is 
 showing up in Asterisk, but it's showing up with the IP of OpenSIPS.

 Ideally, I would like it to be the public IP of the device trying to 
 register. As it stands right now, my UAD gets authenticated fine, 
 registered in Asterisk. I can make extension to extension calls, 
 extension to PSTN calls, but can NOT receive calls.

 How can I change the IP when relaying the REGISTER to Asterisk so that 
 it's the public IP of the deivce, instead of my OpenSIPS server?

 Note: I am a LAMP guy,

Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk

[Bogdan-Andrei Iancu]

Tyler,

The script logic (for REGISTER) is a bit broken - you authenticate the 
REGISTER, you save it locally, send a reply to UAC and then relay the 
REGISTER to asterisk; which will send you back a reply, so you will end 
up with 2 replies back to UAC :)

Regarding the contact part, I see you are using the fix_nated_contact()  
function - this is the only function that seams to modify the CT hdr.  
To validate this assumption, try commenting the function and run again.

Regards,
Bogdan

Web Support wrote:
> Bogdan, or anyone,
>
> How can I relay the public IP of my UAD off to Asterisk instead of the IP of
> the OpenSIPS server?
>
> Everything I've tried has not changed anything. I am using
> fix_nated_contact(), as before using it, it was trying to register the
> private IP of the UAD. 
>
> -Tyler
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> Sent: Tuesday, March 09, 2010 13:34
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>
> forgot the script :)
>
> but look for fix_nated_contact() if present
>
> Regards,
> Bogdan
>
> Web Support wrote:
>   
>> I've attached the .cfg, I'm sure it's a bit of a mess in your eyes... I'm
>> still getting the grasp of OpenSIPS scripting and I'm having a hard time
>> figuring out what is returned from certain function calls.
>>
>> I am not knowingly changing the Contact hdr, I think I want to though, so
>> that it reflects the public IP of the UAD, no?
>>
>> Thanks for your help so far Bogdan.
>>
>> -Tyler
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
>> Sent: Tuesday, March 09, 2010 13:25
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>>
>> Ok, in this case I assume opensips is change it, right ?
>>
>> so the question is - does your opensips changes the Contact hdr in 
>> REGISTERs ?
>>
>> Regards,
>> Bogdan
>>
>> Web Support wrote:
>>   
>> 
>>> Hello Bogdan,
>>>
>>> The IP is of my OpenSIPS server.
>>>
>>> -Tyler
>>>
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org
>>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
>>>   
> Iancu
>   
>>> Sent: Tuesday, March 09, 2010 12:34
>>> To: OpenSIPS users mailling list
>>> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>>>
>>> Hi Tyler,
>>>
>>> can you check at network level (use tcpdump, ngrep), in the REGISTER 
>>> from opensips to asterisk, what is the Contact header? does this Contact 
>>> hdr contain an IP of opensips or an IP of UAD ?
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Web Support wrote:
>>>   
>>> 
>>>   
 Currently I have OpenSIPS installed on one machine, and have my 
 Asterisk (PBXware) on another machine. When trying to register my UAD 
 through OpenSIPS, the relay is happening correctly and the device is 
 showing up in Asterisk, but it's showing up with the IP of OpenSIPS.

 Ideally, I would like it to be the public IP of the device trying to 
 register. As it stands right now, my UAD gets authenticated fine, 
 registered in Asterisk. I can make extension to extension calls, 
 extension to PSTN calls, but can NOT receive calls.

 How can I change the IP when relaying the REGISTER to Asterisk so that 
 it's the public IP of the deivce, instead of my OpenSIPS server?

 Note: I am a LAMP guy, programming is not new to me, however. OpenSIPS 
 scripting is still sort of foggy.

 Thanks in advance.

 -Tyler

 

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>>>   
>>> 
>>>   
>>   
>> 
>
>
>   


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Re: [OpenSIPS-Users] [dialog] Can't fetch data in 200 reply route of BYE

[Bogdan-Andrei Iancu]

Hi,

I guess what you are describing is the same as this bug report:

https://sourceforge.net/tracker/?func=detail&aid=2941492&group_id=232389&atid=1086410

Am I right ?

Regards,
Bogdan

liuf wrote:
> I'm sorry. My real question is about where to set dialog pointer to the new
> transaction of sequential request, so that opensips can retrieve the correct
> dialog pointer in the correspond reply.
>
> My actions as below, please help me check:
>
> 1. I used fetch_dlg_value function in onreply_route, when opensips received
> 100/200 reply of INVITE, I can get dialog stored value, but in 200 reply of
> BYE, I can't get any data.
>
> 2. First, I found opensips had not get dialog pointer in w_fetch_dlg_value
> function.
>
> 3. In get_current_dialog function, if 200 reply, ONREPLY_ROUTE,
> trans->dialog_ctx will be return.
>
> 4. I found dialog_ctx seem only can be set in function dlg_onreq when
> opensips received BYE.
>
> 5. According my addtion debug info, I found when opensips received BYE,
> dlg_onreq will be exec.
>
> 6. I guess when opensips received sequential request (BYE), it will create
> new transaction, then set current dialog pointer to this new transaction in
> dlg_onreq function. Then when reply of these requests (200) be received,
> current dialog pointer can be get via get_current_dialog function.
>
> 7. So I consider in function dlg_onreq, move t->dialog_ctx =
> (void*)current_dlg_pointer; to the position that before "if (
> current_dlg_pointer->flags & DLG_FLAG_ISINIT )".
>
> ===
> void dlg_onreq(struct cell* t, int type, struct tmcb_params *param)
> {
> /* is the dialog already created? */
> if (current_dlg_pointer!=NULL) {
>
> t->dialog_ctx = (void*)current_dlg_pointer;
>
> /* new, un-initialized dialog ? */
> if ( current_dlg_pointer->flags & DLG_FLAG_ISINIT )
> return;
>
> /* dialog was previously created by create_dialog()
>-> just do the last settings */
> run_create_callbacks( current_dlg_pointer, param->req);
> ..
> === 
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: [OpenSIPS-Users] "Explaining the default script" is the next webinar

[Bogdan-Andrei Iancu]

The recording of this webinar is now available under :
   http://www.opensips.org/html/docs/video/webinar005/

Regards,
Bogdan



Bogdan-Andrei Iancu wrote:
> Next webinar is scheduled for 25th of February 2010.
>
> The topic is "Explaining the default script" - A detailed going though 
> and explanation of the OpenSIPS default script - configuration params, 
> the flow of messages in the script, what each part is good for, etc
>
> Free registration  at  http://www.opensips.org/Training/Webinars#toc6
>
>
> Best regards,
> Bogdan
>
>   


-- 
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] Soon you can Blink!

[Adrian Georgescu]

On Mar 10, 2010, at 2:49 PM, Thomas Gelf wrote:

> As I work on Linux only

If you get the latest SIP SIMPLE command line clients (they work on  
Debian at least) they all have the ICE feature and display the outcome  
of ICE negotiation when RTP stream is active.

http://sipsimpleclient.com/wiki/SipTesting#TestingGuide

sip-audio-sessions
sip-session

Regards,
Adrian


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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Iñaki Baz Castillo]

2010/3/10 Pascal Maugeri :
> Hi Anca
> We just tested the patch you sent us but we don't see any difference.

Could you paste a SIP flow for the SUBSCRIBE and NOTIFY?

-- 
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] Soon you can Blink!

[Adrian Georgescu]

> * How does it figure out it's reflective candidate if I didn't  
> configure
> a STUN server?

The stun server must be in DNS of the caller domain or you can add  
them manually in Account -> Advanced  -> Nat Traversal section


> * Configuring a STUN server AND enabling ICE seems to cause  
> problems, it
> didn't place calls any more. I must confess that the person testing it
> was sitting in another room, I was an the phone with him and  
> sniffing at
> proxy side. Therefore it could also be that he misstyped the STUN  
> server
> domain...

Still it has to work when stun does not work. Any visible reason in / 
var/log/system.log or Debug window ?

> * It has problems with special characters (example: german umlaut) if
> configured as display name. Blink immediately says "...cannot encode
> charaters..." in the "Add New Account" dialog

We will check this.

Adrian


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Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk

[Web Support]

Bogdan, or anyone,

How can I relay the public IP of my UAD off to Asterisk instead of the IP of
the OpenSIPS server?

Everything I've tried has not changed anything. I am using
fix_nated_contact(), as before using it, it was trying to register the
private IP of the UAD. 

-Tyler

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, March 09, 2010 13:34
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk

forgot the script :)

but look for fix_nated_contact() if present

Regards,
Bogdan

Web Support wrote:
> I've attached the .cfg, I'm sure it's a bit of a mess in your eyes... I'm
> still getting the grasp of OpenSIPS scripting and I'm having a hard time
> figuring out what is returned from certain function calls.
>
> I am not knowingly changing the Contact hdr, I think I want to though, so
> that it reflects the public IP of the UAD, no?
>
> Thanks for your help so far Bogdan.
>
> -Tyler
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> Sent: Tuesday, March 09, 2010 13:25
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>
> Ok, in this case I assume opensips is change it, right ?
>
> so the question is - does your opensips changes the Contact hdr in 
> REGISTERs ?
>
> Regards,
> Bogdan
>
> Web Support wrote:
>   
>> Hello Bogdan,
>>
>> The IP is of my OpenSIPS server.
>>
>> -Tyler
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
>> Sent: Tuesday, March 09, 2010 12:34
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Wrong IP when relaying REGISTER to Asterisk
>>
>> Hi Tyler,
>>
>> can you check at network level (use tcpdump, ngrep), in the REGISTER 
>> from opensips to asterisk, what is the Contact header? does this Contact 
>> hdr contain an IP of opensips or an IP of UAD ?
>>
>> Regards,
>> Bogdan
>>
>> Web Support wrote:
>>   
>> 
>>> Currently I have OpenSIPS installed on one machine, and have my 
>>> Asterisk (PBXware) on another machine. When trying to register my UAD 
>>> through OpenSIPS, the relay is happening correctly and the device is 
>>> showing up in Asterisk, but it's showing up with the IP of OpenSIPS.
>>>
>>> Ideally, I would like it to be the public IP of the device trying to 
>>> register. As it stands right now, my UAD gets authenticated fine, 
>>> registered in Asterisk. I can make extension to extension calls, 
>>> extension to PSTN calls, but can NOT receive calls.
>>>
>>> How can I change the IP when relaying the REGISTER to Asterisk so that 
>>> it's the public IP of the deivce, instead of my OpenSIPS server?
>>>
>>> Note: I am a LAMP guy, programming is not new to me, however. OpenSIPS 
>>> scripting is still sort of foggy.
>>>
>>> Thanks in advance.
>>>
>>> -Tyler
>>>
>>> 
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>   
>>> 
>>>   
>>   
>> 
>
>
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Anca Vamanu]

Hi Pascal,

Please try this revised patch. It is also against svn, take a new copy 
of the tm module and apply it.


Thanks and regards,

--
Anca Vamanu
www.voice-system.ro



Pascal Maugeri wrote:

Hi Anca

We just tested the patch you sent us but we don't see any difference.

Just in case, we recompiled the whole server (first we updated to rev 
6700) and applied the patch you kindly sent us:


svn update
...
patch -p0 < tm_proto_fix.patch
make all
make install
service opensips restart   



The NOTIFY message is still sent over UDP :-(

What information could we send you in order to help you ?

A big thank you for your help and fast answer!

Cheers
Pascal

On Tue, Mar 9, 2010 at 4:55 PM, Anca Vamanu > wrote:


Hi Pascal,

We found the problem - the socket on which the Subscribe was
received was always used when sending the Notify.
Find attached a patch for tm module that should fix this problem.
Please test and report the results.

Regards,

-- 
Anca Vamanu

www.voice-system.ro 



Pascal Maugeri wrote:

Hi

Using opensips 1.6 we need to have the presence module to send
NOTIFY over TCP rather than UDP when the size of the message
is greater than a given MTU.

We tried to force it sending a SUBSCRIBE with transport=tcp in
Contact header:

   SUBSCRIBE sip:echo...@presence.net

   > SIP/2.0

   [...]
   Contact: 
   Event: presence


but the NOTIFY messages are sent over UDP (observe the
"transport=tcp" in the request line):

   NOTIFY sip:albe...@10.1.1.24:5233;transport=tcp SIP/2.0
   [...]
   Contact: http://sip:s...@213.151.105.170:6667>
   >

   Subscription-State: active;expires=120
   Content-Type: application/pidf+xml
   Content-Length: 1254


Could you please let me know what we should do to enable the
sending over TCP ? I don't understand what we're doing wrong.

Cheers
Pascal




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Index: modules/tm/uac.c
===
--- modules/tm/uac.c	(revision 6694)
+++ modules/tm/uac.c	(working copy)
@@ -207,6 +207,7 @@
 	int ret, flags, sflag_bk;
 	int backup_route_type;
 	unsigned int hi;
+	struct socket_info* send_sock;
 
 	ret=-1;
 	
@@ -218,21 +219,26 @@
 
 	LM_DBG("next_hop=<%.*s>\n",dialog->hooks.next_hop->len,
 			dialog->hooks.next_hop->s);
-	/* it's a new message, so we will take the default socket */
-	if (dialog->send_sock) {
-		if (uri2su( dialog->hooks.next_hop, &to_su,
-		dialog->send_sock->proto)==-1) {
-			goto error2;
-		}
-	} else {
-		dialog->send_sock = uri2sock(0, dialog->hooks.next_hop, &to_su,
+
+	/* calculate the socket corresponding to next hop */
+	send_sock = uri2sock(0, dialog->hooks.next_hop, &to_su,
 			PROTO_NONE);
-		if (dialog->send_sock==0) {
-			ret=ser_error;
-			LM_ERR("no socket found\n");
-			goto error2;
+	if (send_sock==0) {
+		ret=ser_error;
+		LM_ERR("no socket found\n");
+		goto error2;
+	}
+	/* if a send socket defined verify if the same protocol */
+	if(dialog->send_sock) {
+		if(send_sock->proto != dialog->send_sock->proto)
+		{
+			dialog->send_sock = send_sock;
 		}
 	}
+	else
+	{
+		dialog->send_sock = send_sock;
+	}
 
 	new_cell = build_cell(0);
 	if (!new_cell) {
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Re: [OpenSIPS-Users] [Presence] How to receive presence NOTIFY over TCP ?

[Pascal Maugeri]

Hi Anca

We just tested the patch you sent us but we don't see any difference.

Just in case, we recompiled the whole server (first we updated to rev 6700)
and applied the patch you kindly sent us:

svn update
...
patch -p0 < tm_proto_fix.patch
make all
make install
service opensips restart


The NOTIFY message is still sent over UDP :-(

What information could we send you in order to help you ?

A big thank you for your help and fast answer!

Cheers
Pascal

On Tue, Mar 9, 2010 at 4:55 PM, Anca Vamanu  wrote:

> Hi Pascal,
>
> We found the problem - the socket on which the Subscribe was received was
> always used when sending the Notify.
> Find attached a patch for tm module that should fix this problem. Please
> test and report the results.
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
> Pascal Maugeri wrote:
>
>> Hi
>>
>> Using opensips 1.6 we need to have the presence module to send NOTIFY over
>> TCP rather than UDP when the size of the message is greater than a given
>> MTU.
>>
>> We tried to force it sending a SUBSCRIBE with transport=tcp in Contact
>> header:
>>
>>SUBSCRIBE sip:echo...@presence.net 
>>
>> SIP/2.0
>>
>>[...]
>>Contact: 
>>Event: presence
>>
>>
>> but the NOTIFY messages are sent over UDP (observe the "transport=tcp" in
>> the request line):
>>
>>NOTIFY sip:albe...@10.1.1.24:5233;transport=tcp SIP/2.0
>>[...]
>>Contact: >>
>>
>>Subscription-State: active;expires=120
>>Content-Type: application/pidf+xml
>>Content-Length: 1254
>>
>>
>> Could you please let me know what we should do to enable the sending over
>> TCP ? I don't understand what we're doing wrong.
>>
>> Cheers
>> Pascal
>> 
>>
>>
>> ___
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>>
>>
>
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Re: [OpenSIPS-Users] Opensips children process

[rajib deka]

Thank you Bogdan. I am clear now.

Regards
Rajib
On Wed, Mar 10, 2010 at 5:20 PM, Bogdan-Andrei Iancu  wrote:

> Hi Rajib,
>
> I meant "to the end" for that SIP message -  a SIP message is a request
> or a reply.
>
> For example, an INVITE request may be processed (entirely) by proc P1,
> while its 200 OK reply may be handled by proc P2.
>
> Regards,
> Bogdan
>
> rajib deka wrote:
> > Thank you Bogdan. When you say 'to the end' does it mean that INVITE
> > to BYE (may be sometime REGISTER or CANCEL) the whole transaction is
> > taken care of by the same process that reads the initial INVITE. So
> > for 200 CPS how many children we should have ?
> >
> > Regards
> > Rajib
> >
> > On Tue, Mar 9, 2010 at 8:29 PM, Bogdan-Andrei Iancu
>  > mailto:bog...@voice-system.ro>> wrote:
> >
> > Hi Rajib,
> >
> > each SIP message (request or reply) is read from the network by a
> > child
> > (any of them) and it will be processed (to the end) only by that
> > child.
> > A child processes only one message at a time..
> >
> > Regards,
> > Bogdan
> >
> > rajib deka wrote:
> > > Hi All,
> > >
> > > Can anybody please explain me the use of opensips children
> > processes?
> > > Is it something like if children=4 and opensips gets 4 INVITES
> > > tranjactions, then one children will work on one INVITE. Correct
> > me if
> > > I am wrong.
> > >
> > > Thanks
> > > --
> > > Rajib Deka
> > > Software Engineer
> > > Servion Global Solution
> > > Chennai, India
> > >
> > > Mobile No: + 91 80157 09130
> > >
> >
> 
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org 
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
> >
> > --
> > Bogdan-Andrei Iancu
> > www.voice-system.ro 
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
>  > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> >
> > --
> > Rajib Deka
> > Software Engineer
> > Servion Global Solution
> > Chennai, India
> >
> > Mobile No: + 91 80157 09130
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
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>



-- 
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Software Engineer
Servion Global Solution
Chennai, India

Mobile No: + 91 80157 09130
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[OpenSIPS-Users] OpenSIPS + FreeRadius Accounting and Authentication

[Ahmed Munir]

Hi,

I've configured OpenSIPS + FreeRadius Accounting and Authentication setup,
which was implemented success full. Using Authentication via Freeradius can
anybody tell me how can I populate data on  radius database tables? Mean
what sort of values do I required  for its tables so I can authenticate and
register my softphone? Like in tables radreply, radgroupcheck, radgroureply,
realms, etc.

Kindly put the light on it and assist me with some sample data.

-- 
Regards,

Ahmed Munir
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Re: [OpenSIPS-Users] Request for clarification on tm timers

[Bogdan-Andrei Iancu]

Hi Mike,

RT_Txx are the retransmission timers (as defined per RFC3261). and you 
can change them via:
   http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id228598

It seams that the delay comes from RADIUS level, when doing RADIUS acc:
   

Mar 10 00:09:25 OpenSIPS /sbin/opensips[13343]: ACC: call missed: 
timestamp=1268197765;method=INVITE;from_tag=as027c059e;to_tag=3477186414-725808;call_id=1ef525222aa5126677bfec8276a37...@myipaddress;code=503;reason=Service
 
Unavailable
Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]: rc_send_server: no reply from 
RADIUS server OpenSIPS:1813

Regards,
Bogdan

Flatbush Telecom wrote:
> Thank you for the prompt response, Bogdan.
>
> I enabled "restart_fr_on_each_reply" and found that I would enter failure 
> route a lot faster (immediately in most cases) however, there are occasions 
> when I have to wait 30 seconds for the failure route. Is there a way to set 
> that 30 seconds to something smaller? Is it the RT_T2 timer? How do I 
> specify a shorter period for RT_T2?
>
> I have attached an example of the 30 second delay from when I receive a 
> negative INVITE in the reply route and when the failure route kicks in as it 
> appears in my syslog
>
> Thanks for your help,
>
> Mike Montgomery
>
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]: enter route [13] 
> RequestURI=sip:50375990...@myipaddress RequestUser=50375990162 
> Source=myIPaddress Method=INVITE TO sip:50375990...@myipaddress 
> FromURI=sip:5672512...@myipaddress FromUser=5672512350
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]: PRE Strip requested URI = 
> sip:50375990...@myipaddress
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]: INFO:pdt:prefix2domain: no 
> prefix found in [50375990162]
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]: Strip prefix  requested 
> URI = sip:50375990...@myipaddress
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]:  Canonical URI = 
> sip:50375990...@myipaddress
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]: valid CALLERID < 
>  sip:5672512...@myipaddress
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]:  load gateway... flags = 
> 
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13338]:  1. next gateway... flags = 
> 0
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13339]: enter onreply_route [2] 
> RequestURI= RequestUser= Source=nexthopIPaddress Method=INVITE 
> TO sip:50375990...@myipaddress FromURI=sip:5672512...@myipaddress 
> FromUser=5672512350
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13339]:  - Reply Route 100 
> Trying = 100, Trying
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13343]: enter onreply_route [2] 
> RequestURI= RequestUser= Source=nexthopIPaddress Method=INVITE 
> TO sip:50375990...@myipaddress FromURI=sip:5672512...@myipaddress 
> FromUser=5672512350
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13343]:  - Reply Route  FAILURE 
> status = 503, Service Unavailable
> Mar 10 00:09:25 OpenSIPS /sbin/opensips[13343]: ACC: call missed: 
> timestamp=1268197765;method=INVITE;from_tag=as027c059e;to_tag=3477186414-725808;call_id=1ef525222aa5126677bfec8276a37...@myipaddress;code=503;reason=Service
>  
> Unavailable
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]: rc_send_server: no reply 
> from RADIUS server OpenSIPS:1813
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]: ERROR:acc:acc_rad_request: 
> radius-ing failed
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]: enter failure_route [2] 
> Source=myIPaddress Method=INVITE TO sip:50375990...@myipaddress FROM 
> sip:5672512...@myipaddress
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]:  -
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]:  - FAILURE status = 
> 
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]:  -
> Mar 10 00:09:55 OpenSIPS /sbin/opensips[13343]: ACC: call missed: 
> timestamp=1268197795;method=INVITE;from_tag=as027c059e;to_tag=;call_id=1ef525222aa5126677bfec8276a37...@myipaddress;code=503;reason=Service
>  
> Unavailable
> - Original Message - 
> From: 
> To: 
> Sent: 03/09/2010 4:00 AM
> Subject: Users Digest, Vol 20, Issue 26
>
>
>   
>> Send Users mailing list submissions to
>> users@lists.opensips.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> or, via email, send a message with subject or body 'help' to
>> users-requ...@lists.opensips.org
>>
>> You can reach the person managing the list at
>> users-ow...@lists.opensips.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of Users digest..."
>>
>>
>> Today's Topics:
>>
>>   1. [NEW] OpenSIPS Control Panel 4.0 is released (with user
>>  provisioning) (Bogdan-Andrei Iancu)
>>   2. Re: Load Balance for OpenSIPS Proxies (osiris123d)
>>   3. basic over view of flags (Nigel Daniels)
>>   4. Re: Load Balance for OpenSIPS Proxies (Stanis?aw Pitucha)
>>   5. [dialog] Can't fetch data in 200 reply route of BYE (liuf)
>>   6. Re: Load Balance for OpenSIPS Proxies (

Re: [OpenSIPS-Users] lb_reload

[Bogdan-Andrei Iancu]

Hi Dimitri,

Ok, so you do lb_reload just because you need to add some more 
destinations in the balancer. You do not change the properties of the 
already existing destination - this was what I was interested in.

Let me run some checks to see what about. BTW, what opensips version are 
you using ?

Regards,
Bogdan


Dmitri G. wrote:
> I'm changing the resource frequently, btw these are asterisk boxes 
> with agents, an inbound call center.
> Every morning they call me and tell "dimitri today we have 200 agent 
> for xx.xx.xx.xx, and we have 150 for yy.yy.yy.yy".
> So I run a lb_resize to change the resource figure (the number of 
> available agents).
> There is no problem with this and I never change the name of the 
> resource, just the figure. I can resize it without any problems.
>
> But in case if they call me to add a new asterisk ip with resources, 
> then I need to insert it to a DB, to the load_balancer table.
> But how to enable to use the new gw other than using lb_reload?
>
> If I run the lb_reload, and there was concurrent calls to the asterisk 
> boxes (so agent=200, and agent=150), the load to each destination will 
> be zero (load=0). Buit there are still active calls to the 
> destinations so if I have 150 calls to one, and if there will be a new 
> call to the destination, after a lb_reload I see load=1, where in 
> reality there was 151 calls.
>
> See my problem?
>
> Thanks,
>
> Dimitri.
>
> 2010/3/9 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 16:26 
> (acum 27 minute)
>
> Ok, I got that :)
>
> my question was - what you change in DB (before doing the reload)? do
> you change the definition of the already existing destinations (like
> resources) ?
>
> Regards,
> Bogdan
>
> Dmitri G. wrote:
> > Hi Bogdan,
> >
> > I mean it's setting the load= figures to zero for all gws in the
> > load_balancer table.
> > So if I have 2 gws with load=136 and load=227, after running
> > opensipsctl fifo lb_reload, then load will be zero (load=0).
> >
> > Dimitri
> >
> > 2010/3/9 Bogdan-Andrei Iancu către OpenSIPS arată detaliile 15:57
> > (acum 18 minute)
> >
> > Hi Dimitri,
> >
> > reloading the data should not delete the dialog profiles, unless you
> > change the name of the resources (in the destination definition).
> > Is this your case?
> >
> > Regards,
> > Bogdan
> >
> > Dmitri G. wrote:
> > > Hi,
> > >
> > > Is there a way to keep the number of the dialogs after running the
> > > lb_reload command through openipsctl?
> > > When I add a new gateway to the mysql able and execute lb_reload
> > > command, the number of the dialogs are resets to zero in every
> > > gateways (I have checked it with lb_list, load counters are zero)
> > >
> > > Thanks,
> > > Dimitri
> > >
> >
> 
> > >
> > > ___
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> > > Users@lists.opensips.org 
> >
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
> >
> > --
> > Bogdan-Andrei Iancu
> > www.voice-system.ro 
> 
> >
> >
> > ___
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> > Users@lists.opensips.org 
> >
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> 
> >
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> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro 
>
>
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>   


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www.voice-system.ro


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Re: [OpenSIPS-Users] Opensips children process

[Bogdan-Andrei Iancu]

Hi Rajib,

I meant "to the end" for that SIP message -  a SIP message is a request 
or a reply.

For example, an INVITE request may be processed (entirely) by proc P1, 
while its 200 OK reply may be handled by proc P2.

Regards,
Bogdan

rajib deka wrote:
> Thank you Bogdan. When you say 'to the end' does it mean that INVITE 
> to BYE (may be sometime REGISTER or CANCEL) the whole transaction is 
> taken care of by the same process that reads the initial INVITE. So 
> for 200 CPS how many children we should have ?
>
> Regards
> Rajib
>
> On Tue, Mar 9, 2010 at 8:29 PM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Rajib,
>
> each SIP message (request or reply) is read from the network by a
> child
> (any of them) and it will be processed (to the end) only by that
> child.
> A child processes only one message at a time..
>
> Regards,
> Bogdan
>
> rajib deka wrote:
> > Hi All,
> >
> > Can anybody please explain me the use of opensips children
> processes?
> > Is it something like if children=4 and opensips gets 4 INVITES
> > tranjactions, then one children will work on one INVITE. Correct
> me if
> > I am wrong.
> >
> > Thanks
> > --
> > Rajib Deka
> > Software Engineer
> > Servion Global Solution
> > Chennai, India
> >
> > Mobile No: + 91 80157 09130
> >
> 
> >
> > ___
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> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro 
>
>
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>
>
>
> -- 
> Rajib Deka
> Software Engineer
> Servion Global Solution
> Chennai, India
>
> Mobile No: + 91 80157 09130
> 
>
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>   


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Re: [OpenSIPS-Users] db_berkeley append

[Bogdan-Andrei Iancu]

Hi,

you are right - I made a fix to solve this issue. SVN is update on 
trunk, 1.6 and 1.5 branches.

This fix will be part of the new 1.6.2 release from tomorrow.

Thanks and regards,
Bogdan

liuf wrote:
> I know this source directory, I have read all file in this directory before.
>
> Can I put another question?
>
> I found in SOURCE/scripts/db_berkeley/opensips, only load_balancer file has
> no METADATA_KEY info, is it right? 
>
> Because refer to document
> http://www.opensips.org/html/docs/modules/1.6.x/db_berkeley.html,
> METADATA_KEY is required.
>
>
> load_balancer
> ==
> METADATA_COLUMNS
> id(int) group_id(int) dst_uri(str) resources(str) probe_mode(int)
> description(str)
> METADATA_KEY
>
> METADATA_READONLY
> 0
> METADATA_LOGFLAGS
> 0
> METADATA_DEFAULTS
> NIL|0|NIL|NIL|0|''
> ==
>   


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Re: [OpenSIPS-Users] [OpenSER-Users] Adding "Reason: SIP ; cause=200" header when CANCEL in forking scenarios

[Bogdan-Andrei Iancu]

Hi Iñaki,

Iñaki Baz Castillo wrote:
> El Martes, 9 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
>   
>> Iñaki Baz Castillo wrote:
>> 
>>> El Martes, 9 de Febrero de 2010, Bogdan-Andrei Iancu escribió:
>>>   
 Hi Iñaki,

 uh...you are totally right :D...
 propagating a received Reason hdr is still missing

 I will take care of it - thanks for pointing it out .
 
>>> It would be a really interesting feature (i.e. Asterisk does include such
>>> Reason header when cancels a call answered elsewhere).
>>>
>>> However there should be possible to allow or deny that received "Reason"
>>> header before propagating it in the CANCEL generated by OpenSIPS (IMHO).
>>>   
>> And do you see this as a global or per CANCEL setting ??
>> 
>
> I could imagine accepting such header just for CANCEL coming from a known 
> source (i.e. my application server or PBX).
>
> Just as suggestion, perhaps a flag in t_relay() would be used for this, a 
> flag 
> that only makes sense for CANCEL rather than INVITE, so:
>
>   if (is_method("CANCEL")) {
> if ($si == MY_APPLICATION_SERVER_IP)
>   # Allow propagating "Reason" header.
>   t_relay(0x12);
> else
>   t_relay();
>   }
>   
I just added flag "0x08" for Reason hdr propagation - both in trunk and 
1.6 branch. It will part of the 1.6.2 release.

Thanks and regards,
Bogdan

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