[OpenSIPS-Users] Help with "t_relay"

2012-11-22 Thread spady
Hi list, i need some hints by you experts to get working my idea.

What I am trying to do is parallel forking toward 2 different IP PBXs whose
have different carateristics. Let me explain:

IP PBX #1: accept UDP connections on port 5060
IP PBX #2: accept TCP connections on port 5068

What i tried to do is as follow:

if (src_ip == IP_MEDIANT || (method=="INVITE")) { 
rewritehostport("FQDN_IP_PBX_2:PORT_IP_PBX_2");
route(10);
exit;
} 

.

route[10] {
append_branch();

  t_relay("tcp:IP_PBX_2:PORT_IP_PBX_2");
  exit;
}

So now I am stuck because i have some dubts that i can't answer:

1- How Can I forward the original request and the new BRANCH created to
different "t_relays" ( which they have to have different features (UDP, TCp
etc.. ) ) ?

I tried with 

$(branch(uri)[0]) = "sip:???@IP_PBX_1:PORT_IP_PBX_1";

But seems not working. In this last snippet of code, Can I use pseudo
variables like $rU??
$(branch(uri)[0]) = "sip:$rU@IP_PBX_1:PORT_IP_PBX_1"; ( ? )

Hope someone can point me in a right way.

Regards

I did not found into documentation something about protocol for Branches
sections. In my case i also need to change it to UDP.





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Re: [OpenSIPS-Users] how to forward SIP messages?

2012-11-22 Thread Muhammad Shahzad
It means you are still doing forwarding. Find "forward" method and comment
it. :-)

Thank you.


On Thu, Nov 22, 2012 at 8:07 PM, Christian Cambier  wrote:

>  Hello.
>
> Direct registration (not passing via the proxy) with extension:
> 5006@10.0.2.16 works
> (see SIP trace 1)
>
>
> Registration with the same extension but via the proxy doesn't work
>
> (see SIP trace 2)
>
> I've added in the proxy as you suggested:
>  record_route();
>
>  rewritehostport("10.0.2.16:5060");
>
>  t_relay();
>
>  exit;
>
>
> Interesting to note though is that the message has been forwarded to the
> PBX (see Server-header in the response) but the PBX didn't accept the
> registration
>
>
>
> ==
> SIP trace 1  (OK)
>
> ** **
>
> REGISTER sip:10.0.2.16:5060 SIP/2.0 
>
> Via: SIP/2.0/UDP 
> 10.0.46.1:5082;rport;branch=z9hG4bKPj28b4d6575de442a5afb372fe9ddc47a9
> 
>
> Max-Forwards: 70 
>
> From: "cid5006" ;tag=43098562225648238bd899bc85994e05
> 
>
> To: "cid5006"  
>
> Call-ID: 78017ce971b646e7a6f41bd60601fd9c 
>
> CSeq: 38561 REGISTER 
>
> User-Agent: VoxtronSipPhone_01.00.010.000 
>
> Contact: "cid5006"  
>
> Expires: 300 
>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
> MESSAGE, OPTIONS 
>
> Content-Length: 0 
>
> ** **
>
> SIP/2.0 200 OK 
>
> Via: SIP/2.0/UDP 
> 10.0.46.1:5082;rport;branch=z9hG4bKPj7621a9f9aa544be7b50abc499f68b004
> 
>
> From: "cid5006" ;tag=43098562225648238bd899bc85994e05
> 
>
> To: "cid5006" ;tag=3609800608 
>
> Call-ID: 78017ce971b646e7a6f41bd60601fd9c 
>
> CSeq: 38562 REGISTER 
>
> Contact: ;expires=120 
>
> Allow:
> REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH
> 
>
> Content-Length: 0 
>
> Date: Thu, 22 Nov 2012 19:55:46 GMT 
>
> Expires: 120 
>
> Server: (innovaphone IP6000/8.00 hotfix25 [80500.65/8050065/107]) 
>
> Allow-Events: reg,dialog,message-summary,presence 
>
> Presence-State: register-action="added" 
>
> P-Alias: 0,4,5006 
>
> P-Alias: 2,6,id5006 
>
> ** **
>
>
> ==
> 
>
> SIP trace 2  (Not OK)
>
>
> REGISTER sip:10.0.2.16:5060 SIP/2.0 
>
> Via: SIP/2.0/TCP 
> 10.0.46.1:49370;rport;branch=z9hG4bKPjd606484bdde5459a96175285593932df
> 
>
> Route:  
>
> Max-Forwards: 70 
>
> From: "cid5006" ;tag=9bf43fc5cfb04bce8f6f42150223c301
> 
>
> To: "cid5006"  
>
> Call-ID: c6752588892f4ba499ddba6d5d9bc90e 
>
> CSeq: 26467 REGISTER 
>
> User-Agent: VoxtronSipPhone_01.00.010.000 
>
> Supported: outbound, path 
>
> Contact: "cid5006" 
> ;reg-id=1;+sip.instance=""
> 
>
> Expires: 300 
>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
> MESSAGE, OPTIONS 
>
> Content-Length: 0 
>
> ** **
>
> SIP/2.0 301 Moved Permanently 
>
> Via: SIP/2.0/TCP 
> 10.0.46.1:49370;received=10.0.46.1;rport=49370;branch=z9hG4bKPjd606484bdde5459a96175285593932df
> 
>
> From: "cid5006" ;tag=9bf43fc5cfb04bce8f6f42150223c301
> 
>
> To: "cid5006" ;tag=3609800590 
>
> Call-ID: c6752588892f4ba499ddba6d5d9bc90e 
>
> CSeq: 26467 REGISTER 
>
> Contact:  
>
> Content-Length: 0 
>
> Server: (innovaphone IP6000/8.00 hotfix25 [80500.65/8050065/107])
>
>
> How to solve this?
>
> thx
> Chris
>
> ** **
>
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[OpenSIPS-Users] how to forward SIP messages?

2012-11-22 Thread Christian Cambier
Hello.

Direct registration (not passing via the proxy) with extension:
5006@10.0.2.16 works
(see SIP trace 1)


Registration with the same extension but via the proxy doesn't work

(see SIP trace 2)

I've added in the proxy as you suggested:
 record_route();

 rewritehostport("10.0.2.16:5060");

 t_relay();

 exit;


Interesting to note though is that the message has been forwarded to the
PBX (see Server-header in the response) but the PBX didn't accept the
registration



==
SIP trace 1  (OK)

 

REGISTER sip:10.0.2.16:5060 SIP/2.0 

Via: SIP/2.0/UDP
10.0.46.1:5082;rport;branch=z9hG4bKPj28b4d6575de442a5afb372fe9ddc47a9 

Max-Forwards: 70 

From: "cid5006"
;tag=43098562225648238bd899bc85994e05 

To: "cid5006"  

Call-ID: 78017ce971b646e7a6f41bd60601fd9c 

CSeq: 38561 REGISTER 

User-Agent: VoxtronSipPhone_01.00.010.000 

Contact: "cid5006"  

Expires: 300 

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS 

Content-Length: 0 

 

SIP/2.0 200 OK 

Via: SIP/2.0/UDP
10.0.46.1:5082;rport;branch=z9hG4bKPj7621a9f9aa544be7b50abc499f68b004 

From: "cid5006"
;tag=43098562225648238bd899bc85994e05 

To: "cid5006" ;tag=3609800608 

Call-ID: 78017ce971b646e7a6f41bd60601fd9c 

CSeq: 38562 REGISTER 

Contact: ;expires=120 

Allow:
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO
,UPDATE,PUBLISH 

Content-Length: 0 

Date: Thu, 22 Nov 2012 19:55:46 GMT 

Expires: 120 

Server: (innovaphone IP6000/8.00 hotfix25 [80500.65/8050065/107]) 

Allow-Events: reg,dialog,message-summary,presence 

Presence-State: register-action="added" 

P-Alias: 0,4,5006 

P-Alias: 2,6,id5006 

 


==

SIP trace 2  (Not OK)


REGISTER sip:10.0.2.16:5060 SIP/2.0 

Via: SIP/2.0/TCP
10.0.46.1:49370;rport;branch=z9hG4bKPjd606484bdde5459a96175285593932df 

Route:  

Max-Forwards: 70 

From: "cid5006"
;tag=9bf43fc5cfb04bce8f6f42150223c301 

To: "cid5006"  

Call-ID: c6752588892f4ba499ddba6d5d9bc90e 

CSeq: 26467 REGISTER 

User-Agent: VoxtronSipPhone_01.00.010.000 

Supported: outbound, path 

Contact: "cid5006"
;reg-id=1;+sip.instance="" 

Expires: 300 

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS 

Content-Length: 0 

 

SIP/2.0 301 Moved Permanently 

Via: SIP/2.0/TCP
10.0.46.1:49370;received=10.0.46.1;rport=49370;branch=z9hG4bKPjd606484bd
de5459a96175285593932df 

From: "cid5006"
;tag=9bf43fc5cfb04bce8f6f42150223c301 

To: "cid5006" ;tag=3609800590 

Call-ID: c6752588892f4ba499ddba6d5d9bc90e 

CSeq: 26467 REGISTER 

Contact:  

Content-Length: 0 

Server: (innovaphone IP6000/8.00 hotfix25 [80500.65/8050065/107]) 


How to solve this?

thx
Chris

 

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Re: [OpenSIPS-Users] B2b custom_header not working!

2012-11-22 Thread Muhammad Shahzad
What headers you are trying to preserve, its possible that the headers your
are try to save are mandatory headers necessary for B2BUA function, which
will get duplicated if you try to preserve then from leg 1 to leg 2. See
custom_headers documentation for more info.

Thank you.


On Thu, Nov 22, 2012 at 5:25 PM, Jorge Henrique Pinho <
jorge-h-pi...@ext.ptinovacao.pt> wrote:

> Hi all,
> I have installed opensips-1.7.2-2.el5 and I am using b2b module.
> I need to preserve some headers that b2b module changes. I was using avp
> and textops modules in local_route to accomplish that. But this lead to
> unexpected behaviours by opensips.
> So i define the custom_headers variable in my configuration (
> http://www.opensips.org/html/docs/modules/1.7.x/b2b_logic.html#id250020),
> and the message that b2b passes to the other side of the dialog has the
> preserve headers repeated. One header with the original value and other
> with the new value generated by b2b module.
> Is this the correct behaviour? How do i solve this?
>
> Thanks in advance,
>
> Jorge Pinho
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Re: [OpenSIPS-Users] how to forward SIP messages?

2012-11-22 Thread Alex Ionescu

Hi,

You can use this, basically taken from the default OpenSIPS script  - at 
the end of the route check the "rewritehostport" :


### Routing Logic 

# main request routing logic

route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("INVITE")) {
# even if in most of the cases is 
useless, do RR for
# re-INVITEs alos, as some buggy 
clients do change route set

# during the dialog.
record_route();
}

# route it out to whatever destination was set 
by loose_route()

# in $du (destination URI).
t_relay();
exit;
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful 
ACK; must be an ACK after
# a 487 or e.g. 404 from 
upstream server

t_relay();
exit;
} else {
# ACK without matching 
transaction ->

# ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}

# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();


if (!is_method("INVITE")) {
sl_send_reply("500","Method not allowed");
exit;
}

record_route();
rewritehostport("10.0.2.16:5060");
t_relay();
exit;
}

On 11/22/2012 06:04 PM, Christian Cambier wrote:


Hi.

I'd like to use openSIPS proxy (10.0.4.34) for tracing but leave all 
SIP-handling to a PBX that is on the same network (10.0.2.16)


I tried just forwarding the sip-messages on the proxy using
forward("10.0.2.16:5060 ");
but the UAC receives a "Moved permanently"

How can this be achieved?

thx
Chris



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Re: [OpenSIPS-Users] installing OpenSIPS on Centos 5.8

2012-11-22 Thread Laszlo
you need to install the ncurses-devel package via yum

-Laszlo

2012/11/22 Carlos Cruz :
>
>
> I really don't know why it needs to be so difficult to install a piece of
> software?
>
>
>
> I'm trying to install OpenSIPS on CentOS 5.8 below are the errors I get:
>
>
>
> I've done quite a bit research, I've tried the suggestions described in the
> mailing list under the subject "Re: libmysqlclient-dev for CentOS?" but I
> still can't get off the ground. I would appreciate any suggestions or tips!!
>
>
>
> I am willing to pay someone to assist me with this and to also help with
> getting me started with basic integration with Asterisk. If you're
> interested I can pay you via PayPal. I can give you temporary access to a
> VM. If you're interested or know someone who could be, let me know.
>
>
>
>
>
>
>
> [root@localhost opensips-1.8.2-tls]# make menuconfig
> bison -d -b cfg cfg.y
> cfg.y: conflicts: 1 shift/reduce
> flex cfg.lex
> cd menuconfig; make ; cd ..
> make[1]: Entering directory
> `/usr/src/opensips/opensips-1.8.2-tls/menuconfig'
> gcc -g -Wall -DMENUCONFIG_CFG_PATH=\"menuconfig/configs/\"
> -DMENUCONFIG_GEN_PATH=\"etc/\" -DMENUCONFIG_HAVE_SOURCES=1   -c -o cfg.o
> cfg.c
> In file included from main.h:33,
>  from cfg.c:30:
> curses.h:31:19: error: curses.h: No such file or directory
> In file included from cfg.c:30:
> main.h:35: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before
> ‘*’ token
> main.h:40: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before
> ‘*’ token
> make[1]: *** [cfg.o] Error 1
> make[1]: Leaving directory `/usr/src/opensips/opensips-1.8.2-tls/menuconfig'
> ./menuconfig/configure --local
> make: ./menuconfig/configure: Command not found
> make: *** [menuconfig] Error 127
>
>
>
>
>
>
>
>
>
>
>
>
> ___
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[OpenSIPS-Users] installing OpenSIPS on Centos 5.8

2012-11-22 Thread Carlos Cruz
 

I really don't know why it needs to be so difficult to install a piece of
software? 

 

I'm trying to install OpenSIPS on CentOS 5.8 below are the errors I get:

 

I've done quite a bit research, I've tried the suggestions described in the
mailing list under the subject "Re: libmysqlclient-dev for CentOS?" but I
still can't get off the ground. I would appreciate any suggestions or tips!!

 

I am willing to pay someone to assist me with this and to also help with
getting me started with basic integration with Asterisk. If you're
interested I can pay you via PayPal. I can give you temporary access to a
VM. If you're interested or know someone who could be, let me know.

 

 

 

[root@localhost opensips-1.8.2-tls]# make menuconfig
bison -d -b cfg cfg.y
cfg.y: conflicts: 1 shift/reduce
flex cfg.lex
cd menuconfig; make ; cd ..
make[1]: Entering directory
`/usr/src/opensips/opensips-1.8.2-tls/menuconfig'
gcc -g -Wall -DMENUCONFIG_CFG_PATH=\"menuconfig/configs/\"
-DMENUCONFIG_GEN_PATH=\"etc/\" -DMENUCONFIG_HAVE_SOURCES=1   -c -o cfg.o
cfg.c
In file included from main.h:33,
 from cfg.c:30:
curses.h:31:19: error: curses.h: No such file or directory
In file included from cfg.c:30:
main.h:35: error: expected '=', ',', ';', 'asm' or '__attribute__' before
'*' token
main.h:40: error: expected '=', ',', ';', 'asm' or '__attribute__' before
'*' token
make[1]: *** [cfg.o] Error 1
make[1]: Leaving directory `/usr/src/opensips/opensips-1.8.2-tls/menuconfig'
./menuconfig/configure --local
make: ./menuconfig/configure: Command not found
make: *** [menuconfig] Error 127

 

 

 

 

 

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[OpenSIPS-Users] B2b custom_header not working!

2012-11-22 Thread Jorge Henrique Pinho
Hi all,
I have installed opensips-1.7.2-2.el5 and I am using b2b module.
I need to preserve some headers that b2b module changes. I was using avp and 
textops modules in local_route to accomplish that. But this lead to unexpected 
behaviours by opensips.
So i define the custom_headers variable in my configuration 
(http://www.opensips.org/html/docs/modules/1.7.x/b2b_logic.html#id250020), and 
the message that b2b passes to the other side of the dialog has the preserve 
headers repeated. One header with the original value and other with the new 
value generated by b2b module.
Is this the correct behaviour? How do i solve this?

Thanks in advance,

Jorge Pinho
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[OpenSIPS-Users] how to forward SIP messages?

2012-11-22 Thread Christian Cambier
Hi.

I'd like to use openSIPS proxy (10.0.4.34) for tracing but leave all
SIP-handling to a PBX that is on the same network (10.0.2.16)

I tried just forwarding the sip-messages on the proxy using 
forward("10.0.2.16:5060  ");
but the UAC receives a "Moved permanently"

How can this be achieved?

thx
Chris

 

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Re: [OpenSIPS-Users] [RFC] New Release Policy for OpenSIPS project

2012-11-22 Thread Bogdan-Andrei Iancu

Hi Saul,

On 11/22/2012 12:46 PM, Saúl Ibarra Corretgé wrote:

Release cycles
===
 - instead of a feature driven release cycle, I would prefer a time driven 
release cycle - because it is more predictable and being feature driven may 
actually escalate the time to the next release (the snowball effect) - see the 
timing for 1.7, 1.8 versions
 - have a 5-7 months release cycle (depending on the required volume of 
work)
 - smaller steps in releases will be more friendly to users as there are no 
big gaps between releases, easier and more appealing to upgrade ; also shorter 
release cycles will make new features available in stable versions much faster.


While a time-based release cycle sounds good to me for minor releases (1.8.1, 
1.8.2, ...) I'm not sure if it can also be applied to major releases. What if 
feature X takes longer than expected to develop? Things may be inadvertently 
rushed and that's not good. For major releases I'd go with the Debian-ish 
policy: it's ready when it's ready :-)

[bogdan]
The problem I see with the features-based release cycle is that they are 
unpredictable as time - some features may not be properly (or 
impossible) time evaluated -> it may stretch the interval between 
releases ; IMHO, for a project to reliable it is a must to be 
predictable. The best examples are what is happening now with OpenSIPS 
(the interval between releases is keep growing) and Debian (lack of 
predictability and huge intervals between release ended up in the Ubuntu 
alternative).
Being able to predict the releases (as time) without huge differences 
between versions (to make an upgrade something easy you are not scared 
like hell to do it) should be some key-feature of the project.


The time-based releases should not be affected by how long a feature 
takes to be implemented - 6 months of development for a feature is 
really more than enough, IMHO.



PS: let me ask you: how many OpenSIPS installations do you still have 
running old versions because upgrade is really painful ? ;)



Next Release TODO
==
 - on a new cycle, we should start with a brainstorming on what the next 
release should contain (or focus on). This will open up the development and 
roadmap of the project to the entire community.
 - maintain a web page with the TODO features that will be updated (this 
process is to be continuous); also the items that where address to be 
documented and listed as new available features (see 
http://www.opensips.org/Main/Ver190)
 - as the release is time driven, the next release will contain only the 
features (from TODO list, based on priorities) that can be done in that time 
frame; the remaining list will be inherited by the next release.

Steps inside a Cycle

 - brainstorming on TODO list
 - estimating the release time (T) based on the volume of work (between 5-7 
months)
 - actual work on implementing the items on TODO list ; it is critical 
important to have a
 better description / documentation / examples on the newly added feature 
->  it will help
 people to understand and use them from day 0 (an undocumented super 
feature is an
 inexistent feature)
 - SVN freeze (no more new stuff) at T - 1 months ; at this point the SVN 
trunk code
 is moved in a new separate SVN branch (dedicated to that release)->  
Release Candidate
 (or beta release) ; this will make the trunk free and available for 
new work in the
 mean while (we overlap the testing on release N with the start of 
release N+1)
 - testing, debugging - 1 month ->  at T we have the GA release (full 
stable release)

Version Management
==
 - at any moment, officially we will support only the last 2 stable release 
(by support I mean
 troubleshooting, fixing bugs, backporting, etc)
 - whatever is older than 2 stable release (like older than 1.7 now) is 
unsupported (no fixing,
 no packing, no new tarballs)
 - when a new release gets to a full stable state, the window of 2 
supported versions is shifted
 (like when 1.9 will become stable, 1.7 will become obsolete and 
unsupported).


What about security fixes? I can understand that when 1.9 is released 1.7 goes 
to EOL (sort of), but what if there is a bug in the parser (for example) which 
can cause a crash just by using a stupid script? IMHO there should be a 
security-fixes-only period, since migrating to a new OpenSIPS version is not  a 
task to be taken lightly.

[bogdan]
That is true problem that may have as solutions:
1) simply upgrade (most common way to go in open source world) , 
considering that upgrades should become easier.
2) try to define what is really critical (based on what??) and 
still do backporting - but at the end of the day we need to encourage 
people to use the new versions - keep patching and supporting really old 
versions (consider 1.6 at this point) is

Re: [OpenSIPS-Users] Fix empty lines in SIP headers generated by buggy hardware

2012-11-22 Thread Muhammad Shahzad
Use TextOPS module,

http://www.opensips.org/html/docs/modules/1.8.x/textops.html#id250208

You probably need to use it something like this,

replace_all("^\s+$", "");

You have to do this BEFORE involving B2BUA stuff.

Thank you.


On Thu, Nov 22, 2012 at 2:03 PM, Adam Raszynski wrote:

> Hi
>
> I have problem with some buggy hardware ATAs and routers, some of them
> mess SIP requests by adding empty lines (\n or \r\n) between headers.
>
> For example:
>
> Via: SIP/2.0/UDP 80.1.1.1:5060;branch=z9hG4bKe89.28ac75a.0
> To: sip:80.1.1.1
> From:  >;tag=13e3d64e25956fb4c1e2442af82fd0e3-c2c5
> CSeq: 14 INVITE
>
> Call-ID: 1af0eccc5994a91b-19144@80.1.1.1
> Max-Forwards: 70
> Content-Length: 0
>
>
> Note empty line between CSeq and Call-Id. That's just example, empty lines
> appear in random order.
>
> I route calls to sippy B2BUA wich does not accept malformed requests and
> throw exception in that condition.
>
> Is it possible to detect such requests in OpenSIPS and remove unnecessary
> empty lines from between headers?
>
> Regards
>
>
>
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Fix empty lines in SIP headers generated by buggy hardware

2012-11-22 Thread Adam Raszynski
Hi

I have problem with some buggy hardware ATAs and routers, some of them mess
SIP requests by adding empty lines (\n or \r\n) between headers.

For example:

Via: SIP/2.0/UDP 80.1.1.1:5060;branch=z9hG4bKe89.28ac75a.0
To: sip:80.1.1.1
From: ;tag=13e3d64e25956fb4c1e2442af82fd0e3-c2c5
CSeq: 14 INVITE

Call-ID: 1af0eccc5994a91b-19144@80.1.1.1
Max-Forwards: 70
Content-Length: 0


Note empty line between CSeq and Call-Id. That's just example, empty lines
appear in random order.

I route calls to sippy B2BUA wich does not accept malformed requests and
throw exception in that condition.

Is it possible to detect such requests in OpenSIPS and remove unnecessary
empty lines from between headers?

Regards
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Re: [OpenSIPS-Users] [RFC] New Release Policy for OpenSIPS project

2012-11-22 Thread Saúl Ibarra Corretgé
Hi Bogdan,

It's great to see this. See some comments inline.

On Nov 22, 2012, at 11:34 AM, Bogdan-Andrei Iancu wrote:

> Hi all,
> 
> I want to bring to public discussion the changing of the release policy of 
> the project. Why, because I had an interesting feedback from the community 
> after the email on shaping the 1.9 release and I felt the need of straighting 
> some things up.
> 
> First of all, what this change should target? It should make the release 
> process :
> - more open - anyone from community (and not only developers) should be 
> able to 
> contribute to roadmap of the next release (on what should be done)
> - more predictable - everyone should know when and how the next release 
> will be 
> available, so they can rely and sync their own private schedules (for 
> using opensips)
> with the project scheduling. You will know when the next release will 
> be available 
> as RC, as GA, etc, you will know what features will contain, you will 
> know when to 
> get involved for bringing in discussion some new features for the 
> next release.
> - more transparent - the entire releasing process to be generally known 
> in details, so
> we can achieve a better collaboration and interfacing between 
> community and developers
> (we should avoid a separation between these two entities and rather 
> put them together 
> to work)
> 
> 
> Now, I'm listing here what I see as a starting point and I'm eager to hear 
> your comments, suggestions, improvements or any other ideas related to this 
> topic.
> 
> Release cycles
> ===
> - instead of a feature driven release cycle, I would prefer a time driven 
> release cycle - because it is more predictable and being feature driven may 
> actually escalate the time to the next release (the snowball effect) - see 
> the timing for 1.7, 1.8 versions
> - have a 5-7 months release cycle (depending on the required volume of 
> work)
> - smaller steps in releases will be more friendly to users as there are 
> no big gaps between releases, easier and more appealing to upgrade ; also 
> shorter release cycles will make new features available in stable versions 
> much faster.
> 

While a time-based release cycle sounds good to me for minor releases (1.8.1, 
1.8.2, ...) I'm not sure if it can also be applied to major releases. What if 
feature X takes longer than expected to develop? Things may be inadvertently 
rushed and that's not good. For major releases I'd go with the Debian-ish 
policy: it's ready when it's ready :-)

> Next Release TODO
> ==
> - on a new cycle, we should start with a brainstorming on what the next 
> release should contain (or focus on). This will open up the development and 
> roadmap of the project to the entire community.
> - maintain a web page with the TODO features that will be updated (this 
> process is to be continuous); also the items that where address to be 
> documented and listed as new available features (see 
> http://www.opensips.org/Main/Ver190)
> - as the release is time driven, the next release will contain only the 
> features (from TODO list, based on priorities) that can be done in that time 
> frame; the remaining list will be inherited by the next release.
> 
> Steps inside a Cycle
> 
> - brainstorming on TODO list
> - estimating the release time (T) based on the volume of work (between 
> 5-7 months)
> - actual work on implementing the items on TODO list ; it is critical 
> important to have a 
> better description / documentation / examples on the newly added 
> feature -> it will help
> people to understand and use them from day 0 (an undocumented super 
> feature is an 
> inexistent feature)
> - SVN freeze (no more new stuff) at T - 1 months ; at this point the SVN 
> trunk code 
> is moved in a new separate SVN branch (dedicated to that release)-> 
> Release Candidate 
> (or beta release) ; this will make the trunk free and available for 
> new work in the 
> mean while (we overlap the testing on release N with the start of 
> release N+1)
> - testing, debugging - 1 month -> at T we have the GA release (full 
> stable release)
> 
> Version Management
> ==
> - at any moment, officially we will support only the last 2 stable 
> release (by support I mean 
> troubleshooting, fixing bugs, backporting, etc)
> - whatever is older than 2 stable release (like older than 1.7 now) is 
> unsupported (no fixing,
> no packing, no new tarballs)
> - when a new release gets to a full stable state, the window of 2 
> supported versions is shifted
> (like when 1.9 will become stable, 1.7 will become obsolete and 
> unsupported).
> 

What about security fixes? I can understand that when 1.9 is released 1.7 goes 
to EOL (sort of), but what if there is a bug in the par

Re: [OpenSIPS-Users] OpenSIPS and Sip-Provider like gateway authenticate.

2012-11-22 Thread Saúl Ibarra Corretgé
Checkout the uac_registrant module.

On Nov 22, 2012, at 11:28 AM, Miguel J. López Valverde wrote:

> Hello users@list.opensips
> 
> I've a query for you. I want to connect my OpenSIP Server with a SIP 
> provider that requires validate the user account and password instead of by 
> IP. The question is... ¿It's posible to resgistar OpenSIPS like an UAC 
> against a SIP-Provider like gateway authenticate by user/pass for send and 
> recive calls throught it?
> 
> Regards.
> 
> Miguel J. López.
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Saúl Ibarra Corretgé
AG Projects




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[OpenSIPS-Users] [RFC] New Release Policy for OpenSIPS project

2012-11-22 Thread Bogdan-Andrei Iancu

Hi all,

I want to bring to public discussion the changing of the release policy 
of the project. Why, because I had an interesting feedback from the 
community after the email on shaping the 1.9 release and I felt the need 
of straighting some things up.


First of all, what this change should target? It should make the release 
process :
- *more open* - anyone from community (and not only developers) 
should be able to

contribute to roadmap of the next release (on what should be done)
- *more predictable* - everyone should know when and how the next 
release will be
available, so they can rely and sync their own private 
schedules (for using opensips)
with the project scheduling. You will know when the next 
release will be available
as RC, as GA, etc, you will know what features will contain, 
you will know when to
get involved for bringing in discussion some new features for 
the next release.
- *more transparent* - the entire releasing process to be generally 
known in details, so
we can achieve a better collaboration and interfacing between 
community and developers
(we should avoid a separation between these two entities and 
rather put them together

to work)


Now, I'm listing here what I see as a starting point and I'm eager to 
hear your comments, suggestions, improvements or any other ideas related 
to this topic.


Release cycles
===
- instead of a feature driven release cycle, I would prefer a time 
driven release cycle - because it is more predictable and being feature 
driven may actually escalate the time to the next release (the snowball 
effect) - see the timing for 1.7, 1.8 versions
- have a 5-7 months release cycle (depending on the required volume 
of work)
- smaller steps in releases will be more friendly to users as there 
are no big gaps between releases, easier and more appealing to upgrade ; 
also shorter release cycles will make new features available in stable 
versions much faster.


Next Release TODO
==
- on a new cycle, we should start with a brainstorming on what the 
next release should contain (or focus on). This will open up the 
development and roadmap of the project to the entire community.
- maintain a web page with the TODO features that will be updated 
(this process is to be continuous); also the items that where address to 
be documented and listed as new available features (see 
http://www.opensips.org/Main/Ver190)
- as the release is time driven, the next release will contain only 
the features (from TODO list, based on priorities) that can be done in 
that time frame; the remaining list will be inherited by the next release.


Steps inside a Cycle

- brainstorming on TODO list
- estimating the release time (T) based on the volume of work 
(between 5-7 months)
- actual work on implementing the items on TODO list ; it is 
critical important to have a
better description / documentation / examples on the newly 
added feature -> it will help
people to understand and use them from day 0 (an undocumented 
super feature is an

inexistent feature)
- SVN freeze (no more new stuff) at T - 1 months ; at this point 
the SVN trunk code
is moved in a new separate SVN branch (dedicated to that 
release)-> Release Candidate
(or beta release) ; this will make the trunk free and available 
for new work in the
mean while (we overlap the testing on release N with the start 
of release N+1)
- testing, debugging - 1 month -> at T we have the GA release (full 
stable release)


Version Management
==
- at any moment, officially we will support only the last 2 stable 
release (by support I mean

troubleshooting, fixing bugs, backporting, etc)
- whatever is older than 2 stable release (like older than 1.7 now) 
is unsupported (no fixing,

no packing, no new tarballs)
- when a new release gets to a full stable state, the window of 2 
supported versions is shifted
(like when 1.9 will become stable, 1.7 will become obsolete and 
unsupported).




Regards,

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

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[OpenSIPS-Users] OpenSIPS and Sip-Provider like gateway authenticate.

2012-11-22 Thread Miguel J.
Hello users@list.opensips

I've a query for you. I want to connect my OpenSIP Server with a SIP
provider that requires validate the user account and password instead of
by IP. The question is... ¿It's posible to resgistar OpenSIPS like an
UAC against a SIP-Provider like gateway authenticate by user/pass for
send and recive calls throught it?

Regards.

Miguel J. López.
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