[OpenSIPS-Users] Some other OpenSIPS newbie questions

[Gary Patton]

Hello again!  I'm trying to read the documentation and view the Kick Start 
video from 2012 before I install OpenSIPS.  A couple of questions came up: 
(1) I'd like to compile/install OpenSIPS 1.11.1 (on Debian 7.5) to use TLS.  
So, should I create / self-publish / install the certificate for TLS *before* 
(a) compiling and/or (b) installing OpenSIPS or can I do that after OpenSIPS is 
installed?
(2) Similar question about MySQL.  Should I create the "opensipsrw" admin 
account with the "opensips" password in MySQL before compiling/installing 
OpenSIPS?  Or does the opensipsdbctl config file automatically create those in 
MySQL when the "create" option is run?
Thanks.
Gary  ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-Devel] Fwd: RTPproxy project

[ag]

Guys,

All these softwares are mature with many years in service both for the media 
relays and the SIP part. They deal find with most of the expected failures, 
which is what the customers expect. For the un-expected failures, well the sky 
if the limit for optimising with infinite cost/benefit ratio. I personally did 
not hear my customers asking for any more resilience or scalability for the 
media relay component, so I stopped optimising long time ago.

A better question is where would OpenSIPS project go next, beyond 
optimisations, as the outside world does not stay still and the perception of 
some of my customers is that we are being left behind feature-wise.

Adrian

On 13 Jun 2014, at 14:18, Bogdan-Andrei Iancu  wrote:

> Hi Maxim,
> 
> It is good to know about the rtp_cluster, but aside simplifying things, it 
> does not bring any new functionality - the LB and failover between RTPproxy 
> nodes can be done now in OpenSIPS module .
> The most challenging thing we are looking at is the ability to move calls 
> between different instances of RTPP (for HA purposes)..or some restart 
> persistence for the sessions - without something like that it's very hard to 
> deal with SW/HW failures ; there are ways to go around for scheduled 
> stops/restarts (maintenance), but non for unexpected failures.
> 
> Thanks and Regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 13.06.2014 00:36, Maxim Sobolev wrote:
>> Brett, on the HA/carrier-grade side there is little-advertized middle-layer 
>> component called "rtp_cluster", which in essence is load-balancing, 
>> transparent dispatcher that can be inserted in between some call-controlling 
>> component like OpenSIPS or Sippy B2BUA and bunch of RTPP instances running 
>> on the same or multiple nodes. From the point of view of that OpenSIPS it's 
>> just another RTPP instance.
>> 
>> And it handles all logic necessary to load-balance incoming requests between 
>> online instances plus it can handle dynamic re-confiduration of the cluster 
>> and track individual nodes going up and down. The code is pretty usable, we 
>> have it deployed for several customers and it's being actively developed as 
>> well. We have it working reliably controlling up to 30-40 RTPP instances 
>> scattered over at least 5 nodes.
>> 
>> http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/
>> 
>> We have at least one pretty well known service provider whose name starts 
>> with capital V using it in combination with OpenSIPS to load balance RTP 
>> traffic via bunch of Amazon EC2 instances.
>> 
>> 
>> On Tue, May 27, 2014 at 6:52 AM, Brett Nemeroff  wrote:
>> Just wanted to add my 0.02 here.. 
>> 
>> I totally agree with Bogdan. For the applications where opensips + a RTP 
>> relay make sense, HA and persistence are much more important. 
>> 
>> WebRTC and ICE are kinda applications in of themselves. And although these 
>> applications are going to grow in popularity, the "legacy" needs for an RTP 
>> relay are still massively prevalent in the space. A general push towards 
>> "Carrier Grade", resiliency and redundancy I think is much better for the 
>> project as a whole. 
>> 
>> Not only that, consider that applications requiring ICE or WebRTC will 
>> greatly benefit from HA / persistence, but not so much the other way around 
>> :) 
>> 
>> YMMV
>> 
>> -Brett
>> 
>> 
>> 
>> On Sun, May 25, 2014 at 6:30 AM, Bogdan-Andrei Iancu  
>> wrote:
>> Hello,
>> 
>> As always, the truth is in the middle.
>> 
>> I agree RTPP is behind on certain things (and this is why we want to do 
>> them), but on the other hand it is a good platform with other good features 
>> (missing on the other relays). RTPP has better ability in individually 
>> controlling the stream (audio /video), ability to set timeouts and onhold 
>> with no conflicts, ability to generates events 
>> on timeout, more flexibility in handling symmetric / asymmetric NATs, 
>> ability to do media injection (playback), ability to do call recording
>> 
>> What neither  mediaproxy, nor rtpengine have is a mechanism for implementing 
>> RTP failover (for ongoing calls) or restart persistence . This is something 
>> we want to look into. I would love to have ICE and WebRTC on my media relay, 
>> for the HA and persistence are more important I would say.
>> 
>> Regards,
>>  Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>> On 24.05.2014 01:59, Muhammad Shahzad Shafi wrote:
>>> To be honest, i have stopped using rtpproxy for over 2 years now. It is not 
>>> evolving as fast as it should be, specially in the context of ICE and 
>>> WebRTC technologies.
>>> 
>>> I would like to suggest that opensips team should consider adding support 
>>> for rtpengine from SIPWise,
>>> 
>>> https://github.com/sipwise/rtpengine
>>> 
>>> For now mediaproxy from AG Projects is the only good choice for handling 
>>> media 

Re: [OpenSIPS-Users] [OpenSIPS-Devel] Fwd: RTPproxy project

[Bogdan-Andrei Iancu]

Hi Maxim,

It is good to know about the rtp_cluster, but aside simplifying things, 
it does not bring any new functionality - the LB and failover between 
RTPproxy nodes can be done now in OpenSIPS module .
The most challenging thing we are looking at is the ability to move 
calls between different instances of RTPP (for HA purposes)..or some 
restart persistence for the sessions - without something like that it's 
very hard to deal with SW/HW failures ; there are ways to go around for 
scheduled stops/restarts (maintenance), but non for unexpected failures.


Thanks and Regards,

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

On 13.06.2014 00:36, Maxim Sobolev wrote:
Brett, on the HA/carrier-grade side there is little-advertized 
middle-layer component called "rtp_cluster", which in essence is 
load-balancing, transparent dispatcher that can be inserted in between 
some call-controlling component like OpenSIPS or Sippy B2BUA and bunch 
of RTPP instances running on the same or multiple nodes. From the 
point of view of that OpenSIPS it's just another RTPP instance.


And it handles all logic necessary to load-balance incoming requests 
between online instances plus it can handle dynamic re-confiduration 
of the cluster and track individual nodes going up and down. The code 
is pretty usable, we have it deployed for several customers and it's 
being actively developed as well. We have it working reliably 
controlling up to 30-40 RTPP instances scattered over at least 5 nodes.


http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/

We have at least one pretty well known service provider whose name 
starts with capital V using it in combination with OpenSIPS to load 
balance RTP traffic via bunch of Amazon EC2 instances.



On Tue, May 27, 2014 at 6:52 AM, Brett Nemeroff > wrote:


Just wanted to add my 0.02 here..

I totally agree with Bogdan. For the applications where opensips +
a RTP relay make sense, HA and persistence are much more important.

WebRTC and ICE are kinda applications in of themselves. And
although these applications are going to grow in popularity, the
"legacy" needs for an RTP relay are still massively prevalent in
the space. A general push towards "Carrier Grade", resiliency and
redundancy I think is much better for the project as a whole.

Not only that, consider that applications requiring ICE or WebRTC
will greatly benefit from HA / persistence, but not so much the
other way around :)

YMMV

-Brett



On Sun, May 25, 2014 at 6:30 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hello,

As always, the truth is in the middle.

I agree RTPP is behind on certain things (and this is why we
want to do them), but on the other hand it is a good platform
with other good features (missing on the other relays). RTPP
has better ability in individually controlling the stream
(audio /video), ability to set timeouts and onhold with no
conflicts, ability to generates events on timeout, more
flexibility in handling symmetric / asymmetric NATs, ability
to do media injection (playback), ability to do call recording

What neither  mediaproxy, nor rtpengine have is a mechanism
for implementing RTP failover (for ongoing calls) or restart
persistence . This is something we want to look into. I would
love to have ICE and WebRTC on my media relay, for the HA and
persistence are more important I would say.

Regards,

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

On 24.05.2014 01 :59, Muhammad Shahzad
Shafi wrote:


To be honest, i have stopped using rtpproxy for over 2 years
now. It is not evolving as fast as it should be, specially in
the context of ICE and WebRTC technologies.

I would like to suggest that opensips team should consider
adding support for rtpengine from SIPWise,

https://github.com/sipwise/rtpengine

For now mediaproxy from AG Projects is the only good choice
for handling media in opensips with ICE support (though it
still lacks WebRTC features).

Thank you.

On 2014-05-23 14:55, Bogdan-Andrei Iancu wrote:


Going for a public exposure on this question to Maxim, maybe
we will get an answer here.


 Original Message 
Subject:RTPproxy project
Date:   Mon, 14 Apr 2014 15:03:31 +0300
From:   Bogdan-Andrei Iancu
To: Maxim Sobolev
CC: Razvan Crainea



Hello Maxim,

Long time, no talks, but I hope everything is fine on your side.

I'm reaching you in order to ask about your future plans in regards to
the rtpproxy project? We see no mu

Re: [OpenSIPS-Users] rtpproxy_offer() adds IP and Port twice in SDP

[Bogdan-Andrei Iancu]

If you what the RTP to be sent to the public IP, why are you using the 
"r" flag ? have you read the docs to understand that flag ?

http://www.opensips.org/html/docs/modules/1.11.x/rtpproxy.html#id293915

Regards,

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

On 13.06.2014 14:52, kaushik parmar wrote:

Hi,

Thank you so much. I solved this problem. Now i have another problem. 
i am using rtpproxy_offer("r","xx.xx.xx.xx") in onreply_route[] 
function. Now i can initiate  call but problem is that rtpproxy is 
sending rtp packet on device's private address instead of public address.


E.g.

Peer1 (Public IP) >>  rtpproxy+opensips
rtpproxy  -->> voip switch -->> peer2
rtpproxy --->> peer1 (Private IP)   .. (Here problem occurs).

How can i tell rtpproxy to send rtp packets on public ip instead of 
private. I am using "r" flag with rtpproxy_offer. See below wireshark 
trace image. Here 1) 1st line is for peer1 Public Ip 2) rtpproxy 3) 
asterisk voip server 4) peer1 private IP. Here rtpproxy is sending 
packets to private IP so no voice both the side.





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

Re: [OpenSIPS-Users] Real Time CDR

[Dan Christian Bogos]

Hey Bogdan,

Thanks for your valuable insides.
I agree that performance wise would be faster via events. I was more 
like into fast prototyping, but well, I will follow your advice and will 
more likely use the event via datagram sockets.


Have a good one!

DanB

On 13.06.2014 13:41, Bogdan-Andrei Iancu wrote:

Hi Dan,

To be honest, I'm not such a big fan of HTTP based stuff, mainly 
because of the performance penalty.  When handling CDRs, you need to 
be sure you can handle large amounts of data and very fast. The HTTP 
overhead doesn't fit in this picture, IMHO :).
RabbitMQ is light as transport and also flexible as connecting to the 
consumers.
If you want to stay with an SQL DB, db_http is at the end of the list 
as performance.


Regards,

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

On 12.06.2014 16:32, Dan Christian Bogos wrote:

Hey Bogdan,

Since we got your attention on this one, I was wondering what do u 
think about the other proposal within this thread - http posted CDRs 
(via db module)?


I understand that the event module could shortcut processing but in 
my approach I am trying to use a built in module to have the 
responsibility of dispatching the CDRs out of Opensips (eg: an 
external, custom build socket server can easier crash or loose socket 
connection, plus http libraries are much more universal than 
particular protocols for events processing (eg: it is faster to 
prototype a http server than a particular socket protocol one).


Did u have any bad experience with db -> http conversion/performance?

Ta,
DanB

On 12.06.2014 13:45, users-requ...@lists.opensips.org wrote:

Hi Ricky,

If you want to get realtime updates on the completed calls, I suggest
using the ACC module (to generate CDRs) and have the CDR delivered as
event (via rabbitMQ for example).
Check:
  - generating CDRs -
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295374
  - the CDR event -
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295744
  - using events -
http://www.opensips.org/Documentation/Interface-Events-1-11

Regards,

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



___
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

Re: [OpenSIPS-Users] Real Time CDR

[Bogdan-Andrei Iancu]

Hi Dan,

To be honest, I'm not such a big fan of HTTP based stuff, mainly because 
of the performance penalty.  When handling CDRs, you need to be sure you 
can handle large amounts of data and very fast. The HTTP overhead 
doesn't fit in this picture, IMHO :).
RabbitMQ is light as transport and also flexible as connecting to the 
consumers.
If you want to stay with an SQL DB, db_http is at the end of the list as 
performance.


Regards,

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

On 12.06.2014 16:32, Dan Christian Bogos wrote:

Hey Bogdan,

Since we got your attention on this one, I was wondering what do u 
think about the other proposal within this thread - http posted CDRs 
(via db module)?


I understand that the event module could shortcut processing but in my 
approach I am trying to use a built in module to have the 
responsibility of dispatching the CDRs out of Opensips (eg: an 
external, custom build socket server can easier crash or loose socket 
connection, plus http libraries are much more universal than 
particular protocols for events processing (eg: it is faster to 
prototype a http server than a particular socket protocol one).


Did u have any bad experience with db -> http conversion/performance?

Ta,
DanB

On 12.06.2014 13:45, users-requ...@lists.opensips.org wrote:

Hi Ricky,

If you want to get realtime updates on the completed calls, I suggest
using the ACC module (to generate CDRs) and have the CDR delivered as
event (via rabbitMQ for example).
Check:
  - generating CDRs -
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295374
  - the CDR event -
http://www.opensips.org/html/docs/modules/1.11.x/acc.html#id295744
  - using events -
http://www.opensips.org/Documentation/Interface-Events-1-11

Regards,

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



___
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

Re: [OpenSIPS-Users] sip forwarding

[Bogdan-Andrei Iancu]

Hi,

You can forward a SIP request by using one of the functions:
- forward() , core function, stateless relay
- t_relay() , TM module, stateful relay.

Regards,

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

On 13.06.2014 11:53, Mike Claudi Pedersen wrote:
when exactly does the package get forwarded, just from the default 
config ?




___
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

Re: [OpenSIPS-Users] OpenSIPS with LDAP directory

[Bogdan-Andrei Iancu]

Hi Cyrill,

Once you have the password into a variable, simply use auth module to do 
the authentications, via the pv_authorize() functions:

http://www.opensips.org/html/docs/modules/1.11.x/auth.html#id294212

As an example of using that functions, see:
http://www.opensips.org/Documentation/Tutorials-KeyValueInterface#toc4

Regards,

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

On 25.04.2014 15:58, Cyrill Gremaud wrote:

Hello,

thanks for you response.

Ok I added below if(is_method(REGISTER)) the following code :

if(!ldap_search(….))

and after ?

On 02 Apr 2014, at 10:29, Bogdan-Andrei Iancu  wrote:


Hello,

Yes, use the ldap module just to fetch the password (plaintext or HA1) in 
combination with auth module to perform the actula auth via the 
pv_proxy_authorize() function - see 
http://www.opensips.org/html/docs/modules/1.11.x/auth.html#id294332

Regards,

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

On 10.03.2014 13:42, Cyrill Gremaud wrote:

Hello,

For my first post to this mailing list, I just want to know what I need to 
implement to use a LDAP remote directory with my OpenSIPS server ?
After some researches, I founded that I need to use LDAP module and Auth Module 
right ?

Thanks for your help.

cyrill gremaud
___
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

Re: [OpenSIPS-Users] rtpproxy_offer() adds IP and Port twice in SDP

[Bogdan-Andrei Iancu]

Hi,

Somehow is hard to believe that a single call to rtpproxy_offer() will 
insert 2 ips. So :

- what are the params you pass to this function ?
- place an xlog("*** Before rtpproxy_offer() \n"); before the 
function call and check how many times you get it for a single reply 
processing.


Regards,

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

On 09.06.2014 08:43, kaushik parmar wrote:

Hi,

I am using opensips 1.11.1-notls (x86_64/linux). If i dont write 
rtpproxy_offer() not modify SDP and send address of voip switch. If i 
write rtpproxy_offer() write in write onreply_route[] then it adds IP 
and Ports two times.



On Sun, Jun 8, 2014 at 8:17 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi,

Your script logic should take care of avoiding multiple
rtpproxy_offer() - if logic is too complex, use a flag to remember
if set or not.

BTW, what opensips version are you using ?

Regards,

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

On 07.06.2014 08:42, kaushik parmar wrote:

Hello,

Thank you for reply. I have checked it. Is there any way to check
if ip address already set to xx.xx.xx.xx or not in opensips? If
set then do nothing else apply rtpproxy_offer().


On Fri, Jun 6, 2014 at 10:45 PM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hello,

Check your script, for sure you do the rtpproxy_offer twice
for that SIP message.

Regards,

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

On 06.06.2014 19:03, kaushik parmar wrote:

Hello,

rtpproxy_offer() set IP and Port numbers twice in SDP. My
code and SDP output are as below.

*Opensips.cfg*

onreply_route[handle_nat] {
if (is_method("INVITE") &&
has_body("application/sdp"))
{
 rtpproxy_offer();
}
}



*SDP Packet : *

*SDP Before rtpproxy_offer(); *
*
*
v=0
o=Sippy 44829008 1 IN IP4  yy.yy.yy.yy
s=-
t=0 0
m=audio 46946 RTP/AVP 110 0 8 3 101
c=IN IP4 yy.yy.yy.yy
a=rtpmap:110 speex/8000
a=fmtp:110 mode=3;vbr=on
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off - - - -

*SDP After rtpproxy_offer(); **
*

v=0
o=Sippy 44829008 1 IN IP4 xx.xx.xx.xx
s=-
t=0 0
m=audio 40240 40240 RTP/AVP 110 0 8 3 101
c=IN IP4 xx.xx.xx.xx xx.xx.xx.xx
a=rtpmap:110 speex/8000
a=fmtp:110 mode=3;vbr=on
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off - - - -
a=oldmediaip:yy.yy.yy.yy
a=oldmediaip:yy.yy.yy.yy
a=nortpproxy:yes
a=nortpproxy:yes

Please help to resolve issue for Double IP and Port in SDP.


-- 
Kind regards,


Kaushik Parmar


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





-- 
Kind regards,


Kaushik Parmar





--
Kind regards,

Kaushik Parmar


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

Re: [OpenSIPS-Users] Second call on transferred with B2B refer scenario

[Bogdan-Andrei Iancu]

Hi Santi,

Indeed there is no new Refer-To hdr in the resulting INVITE (frame 38). 
As you can reproduce it, could you please run in full debug (debug=4) 
and send me the logs corresponding to the REFER processing ?


Thank and Regards,

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

On 11.06.2014 13:06, Santi Antón wrote:

Hi,

Nobody else had found the same issue trying refer scenario with b2b module? 
Someone has a valid opensips.cfg to compare it with mine?

Regards,

Santi Antón



-Mensaje original-
De: Santi Antón
Enviado el: miércoles, 04 de junio de 2014 11:04
Para: OpenSIPS users mailling list; 'Bogdan-Andrei Iancu'
Asunto: RE: [OpenSIPS-Users] Second call on transferred with B2B refer scenario

Hi Bogdan,

Do you need my opensips.cfg? It seems very strange that this issue could be 
general, I think that maybe I'm making a mistake in opensips configuration.

Regards,

  
Santi Antón



-Mensaje original-
De: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 
En nombre de Santi Antón Enviado el: miércoles, 28 de mayo de 2014 9:35
Para: 'Bogdan-Andrei Iancu'; OpenSIPS users mailling list
Asunto: Re: [OpenSIPS-Users] Second call on transferred with B2B refer scenario

Hi Bogdan,

Yes, it is.

Thanks in advance,

  
Santi Antón




-Mensaje original-
De: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Enviado el:
miércoles, 28 de mayo de 2014 8:15
Para: OpenSIPS users mailling list; Santi Antón
Asunto: Re: [OpenSIPS-Users] Second call on transferred with B2B refer
scenario

Hi,

You mean the frame 38 in your trace, right ? Indeed the TO should have only the 
URI, not the REPLACE stuff I will look into.

Regards,

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

On 27.05.2014 17:10, Santi Antón wrote:
Hello,

I noticed that INVITE with Replaces header is built like this:

INVITE sip:9442@172.16.50.131:5063 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 172.16.53.59:5060;branch=z9hG4bK4867.fe529da6.0
Via: SIP/2.0/UDP 172.16.53.59:5060;branch=z9hG4bK4867.ee529da6.0
To:
sip:9442@172.16.53.59?Replaces=377952018%40172.16.51.138%3Bto-tag%3DB2
B.124.358%3Bfrom-tag%3D741375670
From:
;tag=fde67a95c41efd29d5eb8ddaecdb
abf0-9cb9
CSeq: 4 INVITE
Call-ID: B2B.345.6691153
Max-Forwards: 69
Content-Length: 376
User-Agent: OpenSIPS (1.10.1-notls (x86_64/linux))
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Contact: 


The replaces header is inside "To" header, is it correct? The examples I saw 
the Replaces header has an independent header like next:

F6 INVITE Transferee -> Transfer Target

INVITE sips:482n4z24...@chicago.example.com;gr=8594958 SIP/2.0
Via: SIP/2.0/TLS 192.0.2.4;branch=z9hG4bKnaslu82
Max-Forwards: 70
To: 
From: ;tag=954
Call-ID: kmzwdle3dl3d08
CSeq: 41 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: gruu, replaces, tdialog
Contact: 
Replaces: 592435881734450904;to-tag=9m2n3wq;from-tag=763231
Content-Type: application/sdp
Content-Length: ...

Regards,

Santi Antón



-Mensaje original-
De: Santi Antón
Enviado el: lunes, 26 de mayo de 2014 9:46
Para: 'users@lists.opensips.org'; 'Bogdan-Andrei Iancu'
Asunto: RV: [OpenSIPS-Users] Second call on transferred with B2B
refer scenario

Oh, I forget to attach the file. Next you can find a link to pcap.

https://drive.google.com/file/d/0B3db2bUGbNDQWFIzSjhoVFFKVUk/edit?usp
=sharing

Regards,
   
Santi Antón



-Mensaje original-
De: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Enviado el:
jueves, 22 de mayo de 2014 18:29
Para: users@lists.opensips.org; Santi Antón
Asunto: Re: [OpenSIPS-Users] Second call on transferred with B2B
refer scenario

Hello Santi,

There is nothing attached and it is not a good practice to have large 
attachments on the mailing lists (as you are flooding the inboxes).
Better uploade your trace on a pastebin like side (or store pcap on ftp/http 
servers) and just post the link.

Regards,

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

On 22.05.2014 13:28, Santi Antón wrote:
Hello,

I'm trying to implement attendant transfer capability to our Opensips server 
with the b2b module and refer.xml scenario.
Attended transfer works fine except the transferred receives a new call rather 
than update the ongoing call with transferor.
Looking for the cause I saw that from tag is different between the initial call 
(INVITE) to transferred and the from tag in Replaces header (INVITE). I think 
the two from tag have to be the same to consider the second INVITE an update 
and not a new call.
Attached you can find a capture with an isolated example (9443 calls 9441 and 
does an attended transfer to 9442).
Maybe I'm not understanding the module behavior but I couldn't find any clue in 
module's documentation.

[OpenSIPS-Users] sip forwarding

[Mike Claudi Pedersen]

when exactly does the package get forwarded, just from the default config ?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users