Re: [OpenSIPS-Users] Release] OpenSIPS 2.2.0 Beta is out !

[Abdul Basit]

Interesting!!!
On 31 Mar 2016 23:48, "Hristo Donev"  wrote:

> Hello,
>
> Thanks for your great project.
> All working fine!
>
> Thank you !!!
>
> Hristo Donev
> VoIP support Center BG-Tel
> web: www.bg-tel.com
>
> 2016-03-31 19:13 GMT+03:00 Bogdan-Andrei Iancu :
>
>> Hi all,
>>
>> In the name of the entire OpenSIPS team, I'm proud to announce the
>> release of the OpenSIPS 2.2.0 Beta.
>>
>> This version continues the work from 2.1 and extends the async support in
>> other parts like RADIUS, LDAP, HEP or sleep(). In the same time it brings
>> new interesting and useful features like:
>> * build-in clustering support (data replication and sharing without
>> shared storage)
>> * advanced SIP Capturing and HEP proxying (powerful and flexible SIP
>> tracing, HEPv3 switching)
>> * WSS support
>> * SQL Cacher (ability to cache generic SQL tables)
>> * enhanced events support (text backend, load balancing & redundancy
>> in delivery)
>> * realtime monitoring of registered end-points
>> * in-dialog pinging with re-INVITEs
>> and many other :
>> http://www.opensips.org/About/Version-2-2-0
>>
>> The performance, the ability to debug and troubleshoot, the integration
>> effort, all were part of the work for OpenSIPS 2.2.0
>>
>> OpenSIPS 2.2 is the result of the whole community effort - there were
>> many people involved, actively contributing with code, helping with testing
>> and reporting - and we want to thank you all for making this release
>> possible:
>> https://github.com/OpenSIPS/opensips/blob/2.2/CREDITS
>>
>> Version 2.2 is a release candidate - the stable GA version is to be
>> release in the next  month, after more in depth testing .
>>
>> You can download OpenSIPS 2.2.0
>> http://opensips.org/pub/opensips/latest/
>> Also, thanks to Nick Altmann, the YUM and APT repos are populated with
>> the 2.2.0 packages:
>> yum.opensips.org
>> apt.opensips.org
>>
>> Enjoy,
>>
>> --
>> 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
>>
>
>
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Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

[Travis Manson-Drake]

Alex,

Thank you again for all your help!

I apologize on being so new to this. I went ahead and sent a linkedin request 
if that's ok.

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Alex Balashov
Sent: Thursday, March 31, 2016 11:51 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

On 03/31/2016 02:49 PM, Travis Manson-Drake wrote:

> I would be more than happy to send it to you privately if that's ok?

Of course!

-- 
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] What is the module to fix SDP ?

[Rodrigo Pimenta Carvalho]

Hi  Alex.

The project I´m currently working on was requested by a customer from the 
telecommunication market. I don´t know the business details as my manager, but 
what I know is :

-the entire system has to run in a ARM hardware. It is an embedded system, with 
limited hardware resources.
-due to custom reasons, it was decided not to use another hardware to run media 
relay. That is why they have decided to implement direct media.

Yes I know. 

Thank you very much for talking with me about this ideas.

Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
Brazil


De: users-boun...@lists.opensips.org  em nome 
de Alex Balashov 
Enviado: quinta-feira, 31 de março de 2016 15:15
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] What is the module to fix SDP ?

As long as the STUN client properly ascertains its external RTP port
through STUN as well, that should work fine.

Out of curiosity, what are your hardware limitations that would make
server-side RTP relay impractical? Also, you are aware that your RTP
relay doesn't have to be on the same hardware as your SIP proxy, correct?

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

[Alex Balashov]

On 03/31/2016 02:49 PM, Travis Manson-Drake wrote:


I would be more than happy to send it to you privately if that's ok?


Of course!

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

[Travis Manson-Drake]

   Alex I can, however it has the public IP of my PBX etc etc.

I would be more than happy to send it to you privately if that's ok?

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Alex Balashov
Sent: Thursday, March 31, 2016 10:50 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

Well, can we see the requests being sent to Asterisk? :)

On 03/31/2016 01:48 PM, Travis Manson-Drake wrote:

> Hello everyone.
>
> Hope your all doing well!
>
> I seem to be having an issue in which when a call is sent through 
> OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a 
> hang up Cause of 111/unrecognized sip header. I looked at the headers 
> of all my packets but can't find anything out of the norm. has anyone 
> experienced this before and ideas on what it might be or what I might check?
>
> I found a few article on asterisk forums mention NAT issues, but I've 
> implemented a NAT helper into my routing logic so that shouldn't be 
> the case.
>
> Thank you all for your time
>
> Travis Manson-Drake
> Voice Systems Analyst
>
> Simply Bits, LLC
> T:520.545.0311 F:520.545.7252
> E:trav...@simplybits.com 
> 5225 N. Sabino Canyon Road
> Tucson, AZ 85750
> Support Hotline: 520.545.0333
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] Release] OpenSIPS 2.2.0 Beta is out !

[Hristo Donev]

Hello,

Thanks for your great project.
All working fine!

Thank you !!!

Hristo Donev
VoIP support Center BG-Tel
web: www.bg-tel.com

2016-03-31 19:13 GMT+03:00 Bogdan-Andrei Iancu :

> Hi all,
>
> In the name of the entire OpenSIPS team, I'm proud to announce the release
> of the OpenSIPS 2.2.0 Beta.
>
> This version continues the work from 2.1 and extends the async support in
> other parts like RADIUS, LDAP, HEP or sleep(). In the same time it brings
> new interesting and useful features like:
> * build-in clustering support (data replication and sharing without
> shared storage)
> * advanced SIP Capturing and HEP proxying (powerful and flexible SIP
> tracing, HEPv3 switching)
> * WSS support
> * SQL Cacher (ability to cache generic SQL tables)
> * enhanced events support (text backend, load balancing & redundancy
> in delivery)
> * realtime monitoring of registered end-points
> * in-dialog pinging with re-INVITEs
> and many other :
> http://www.opensips.org/About/Version-2-2-0
>
> The performance, the ability to debug and troubleshoot, the integration
> effort, all were part of the work for OpenSIPS 2.2.0
>
> OpenSIPS 2.2 is the result of the whole community effort - there were many
> people involved, actively contributing with code, helping with testing and
> reporting - and we want to thank you all for making this release possible:
> https://github.com/OpenSIPS/opensips/blob/2.2/CREDITS
>
> Version 2.2 is a release candidate - the stable GA version is to be
> release in the next  month, after more in depth testing .
>
> You can download OpenSIPS 2.2.0
> http://opensips.org/pub/opensips/latest/
> Also, thanks to Nick Altmann, the YUM and APT repos are populated with the
> 2.2.0 packages:
> yum.opensips.org
> apt.opensips.org
>
> Enjoy,
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
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Re: [OpenSIPS-Users] What is the module to fix SDP ?

[Alex Balashov]

As long as the STUN client properly ascertains its external RTP port 
through STUN as well, that should work fine.


Out of curiosity, what are your hardware limitations that would make 
server-side RTP relay impractical? Also, you are aware that your RTP 
relay doesn't have to be on the same hardware as your SIP proxy, correct?


--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] What is the module to fix SDP ?

[Rodrigo Pimenta Carvalho]

Hi Alex.

That is the point. We have just decided what to do:

We will have a STUN server in the same hardware where is OpenSIPS and a 
softphone (this softphone will talk to others softphones from Internet). Such 
hardware will be connected to a switch and such switch will have a public IP.  
The switch will work as a NAT. OpenSIPS and the main softphone will be behind 
this NAT. So, when the main softphone communicates with other one from 
Internet, this one will have to fix the SDP (IP and PORT) so that the one from 
Internet will be able to send its media to the right end point. This work of 
fixing SDP will use the STUN + ICE.

Our intention is to implement direct media. We cannot implement a media relay 
in our hardware because it doesn't have sufficient resources for this kind of 
relay.

Tell me if it is clear and if it sounds good, please.

Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


De: users-boun...@lists.opensips.org  em nome 
de Alex Balashov 
Enviado: quinta-feira, 31 de março de 2016 14:31
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] What is the module to fix SDP ?

Hello Rodrigo,

If you're not going to use an outboard media relay, how do you propose
to "fix" media ports in the SDP offers & answers? What will you rewrite
them to if you have no RTP endpoint to give you a means of seeing where
media is coming from? :-)

-- Alex

--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] Asterisk Unrecognized sip header

[Alex Balashov]

Well, can we see the requests being sent to Asterisk? :)

On 03/31/2016 01:48 PM, Travis Manson-Drake wrote:


Hello everyone.

Hope your all doing well!

I seem to be having an issue in which when a call is sent through
OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a
hang up Cause of 111/unrecognized sip header. I looked at the headers of
all my packets but can’t find anything out of the norm. has anyone
experienced this before and ideas on what it might be or what I might check?

I found a few article on asterisk forums mention NAT issues, but I’ve
implemented a NAT helper into my routing logic so that shouldn’t be the
case.

Thank you all for your time

Travis Manson-Drake
Voice Systems Analyst

Simply Bits, LLC
T:520.545.0311 F:520.545.7252
E:trav...@simplybits.com 
5225 N. Sabino Canyon Road
Tucson, AZ 85750
Support Hotline: 520.545.0333



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--
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1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[OpenSIPS-Users] Asterisk Unrecognized sip header

[Travis Manson-Drake]

Hello everyone.

Hope your all doing well!

I seem to be having an issue in which when a call is sent through OpenSIPS to 
my Asterisk PBX asterisk with eventually send a BYE with a hang up Cause of 
111/unrecognized sip header. I looked at the headers of all my packets but 
can't find anything out of the norm. has anyone experienced this before and 
ideas on what it might be or what I might check?

I found a few article on asterisk forums mention NAT issues, but I've 
implemented a NAT helper into my routing logic so that shouldn't be the case.

Thank you all for your time


Travis Manson-Drake
Voice Systems Analyst


Simply Bits, LLC
T: 520.545.0311  F: 520.545.7252
E: trav...@simplybits.com
5225 N. Sabino Canyon Road
Tucson, AZ 85750
Support Hotline: 520.545.0333



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Re: [OpenSIPS-Users] What is the module to fix SDP ?

[Rodrigo Pimenta Carvalho]

Hi Carlos.


That is what I'm looking for and now trying to understand if it is sufficient 
for may case.

With this function I will be able to fix IP. But I still have to fix Ports.


If there are NATs, so I suppose there is some kind of Ports mapping that has to 
be taken into account when solving the NAT traversal.  I and a coworker are 
investigating how to solve the port mapping. Maybe the solution will demand ICE 
and STUN. We can't use media proxy module, because our hardware will not be 
able to do media relay. So, We can have a media relay in the same place as 
OpenSIPs.



Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Carlos Eduardo 
Enviado: quinta-feira, 31 de março de 2016 11:43
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the module to fix SDP ?

Hello Rodrigo,

Maybe the function fix_nated_sdp, from nathelper module, would help.

1.5.2.  fix_nated_sdp(flags [, ip_address])

Alters the SDP information in orer to facilitate NAT traversal. What changes to 
be performed may be controled via the "flags" paramter.

Meaning of the parameters is as follows:

flags - the value may be a bitwise OR of the following flags:
0x01 - adds "a=direction:active" SDP line;
0x02 - rewrite media IP address (c=) with source address of the message or the 
provided IP address (the provide IP address take precedence over the source 
address).
0x04 - adds "a=nortpproxy:yes" SDP line;
0x08 - rewrite IP from origin description (o=) with source address of the 
message or the provided IP address (the provide IP address take precedence over 
the source address).

ip_address - IP to be used for rewriting SDP. If not specified, the received 
signalling IP will be used. The parameter allows pseudo-variables usage. NOTE: 
For the IP to be used, you need to use 0x02 or 0x08 flags, otherwise it will 
have no effect.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, 
BRANCH_ROUTE.

Example:
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};


2016-03-30 14:57 GMT-03:00 Rodrigo Pimenta Carvalho 
>:

Hi.


I have a case when a peer answering a call (INVITE) is behind a NAT.

So, in its SIP OK message I would like to see the SDP containing a valid IP and 
media Port valid to receive audio from the caller. That is, the caller need to 
know a valid IP and Port where he/she can send his/her audio packets.


1 - Is it possible to "fix" SDP content for such objective?

2 - Can OpenSIPS do something for this idea works or must I to use something 
more like a stun server?

3 - What is the OpenSIPS module that can help me with this task?


I guess I will have to fix 2 fiels in SDP:

Media Description, name and address (m): audio 55142 RTP/AVP 8 101  ( to 
fix the port)
Connection Information (c): IN IP4 192.168.100.156  
( to fix the IP)


P.S.: I have already a solution (opensips.cfg) that let SIP messages cross NATs 
without problems. Only SDP has to be fixed.


Any hint will be very helpful!

Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979

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[OpenSIPS-Users] Release] OpenSIPS 2.2.0 Beta is out !

[Bogdan-Andrei Iancu]

Hi all,

In the name of the entire OpenSIPS team, I'm proud to announce the 
release of the OpenSIPS 2.2.0 Beta.


This version continues the work from 2.1 and extends the async support 
in other parts like RADIUS, LDAP, HEP or sleep(). In the same time it 
brings new interesting and useful features like:
* build-in clustering support (data replication and sharing without 
shared storage)
* advanced SIP Capturing and HEP proxying (powerful and flexible 
SIP tracing, HEPv3 switching)

* WSS support
* SQL Cacher (ability to cache generic SQL tables)
* enhanced events support (text backend, load balancing & 
redundancy in delivery)

* realtime monitoring of registered end-points
* in-dialog pinging with re-INVITEs
and many other :
http://www.opensips.org/About/Version-2-2-0

The performance, the ability to debug and troubleshoot, the integration 
effort, all were part of the work for OpenSIPS 2.2.0


OpenSIPS 2.2 is the result of the whole community effort - there were 
many people involved, actively contributing with code, helping with 
testing and reporting - and we want to thank you all for making this 
release possible:

https://github.com/OpenSIPS/opensips/blob/2.2/CREDITS

Version 2.2 is a release candidate - the stable GA version is to be 
release in the next  month, after more in depth testing .


You can download OpenSIPS 2.2.0
http://opensips.org/pub/opensips/latest/
Also, thanks to Nick Altmann, the YUM and APT repos are populated with 
the 2.2.0 packages:

yum.opensips.org
apt.opensips.org

Enjoy,

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


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Re: [OpenSIPS-Users] SIPTrace not sending to duplicate_uri

[Kneeoh]

Apparently setting the trace flag is required before trace_dialog() contrary to 
the docs. I've set that and am seeing packets egress (although I don't think 
I'm seeing the full dialog trace. Only 3 packets on a call setup) It looks like 
i'm not getting the replys. Do I need to arm trace_dialog() again in the 
on_reply route? My understanding was that you only have to set it once. 

Issue 2: SIP traffic goes across eth0, my public facing interface. I want the 
siptrace module to shoot trace data out of eth1, my private interface. The 
kernel has a static route that says to get to my duplicate_uri host, use eth1. 
Which works at the kernel level. I can issue a ping to the duplicate_uri host 
and I see requests going out the private and replys, etc.
However, when running tcpdump and placing a call via this opensips box, I see 
that eth0 is the egress interface that is being used to try to reach the 
duplicate_uri host. I've tried adding a separate listen statement for that 
private interface, but that didn't do anything. I thought perhaps the 
trace_local_ip parameter controlled, this but was informed by Liviu that this 
was not the case. Any idea on how to solve this problem or the problem with 
missing replys when using trace_dialog()?
Thanks in advance! (I'm on IRC if anyone has any more detailed questions...or 
just ask here.)
 

On Wednesday, March 30, 2016 11:49 AM, Kneeoh  wrote:
 

 Hi Liviu, I did confirm that trace_on was already set to 1. I did define the 
trace flag as you suggested and tried it with both trace_dialog() and 
sip_trace() but am still not seeing any packets egress from the source host to 
the dest host. Here's what I have set up. I had to fill in a bogus DB (not 
using it anyway) because it wouldn't let me start opensips if it wasn't 
defined. My goal is to have this sip_trace node send captures to a sipcapture 
(homer) node. 

 SIP Trace module
loadmodule "siptrace.so"
modparam("siptrace", "db_url", "mysql://user:passwd@host/dbname")
modparam("siptrace", "duplicate_uri", "sip:192.168.2.142:9060")
modparam("siptrace", "duplicate_with_hep", 1)
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_flag", "TRACE_FLAG")
modparam("siptrace", "hep_version", 2)

INSIDE MAIN ROUTE BLOCK:
    # create dialog with timeout and hide topology
    if ( !create_dialog("B") ) {
        xlog("L_INFO", "Unable to create dialog \n");
        send_reply("500","Internal Server Error");
        exit;

    } else {

        # We have a dialog, Lets hide the topology from where the call 
originated
        topology_hiding();
    }

    # Trace this dialog
    setflag(TRACE_FLAG);
    sip_trace();
    #trace_dialog();


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Re: [OpenSIPS-Users] dialplan module

[Johan De Clercq]

There is a nice tutorial on this on the opensips website.

2016-03-31 15:18 GMT+02:00 Francjos <35...@heb.be>:

> Please, which configurations do i have to make in Opensips and FreeSwitch
> servers in order calls from clients that come through my provider (OVH) to
> land in FreeSwitch server via the opensips server that acts as a proxy?
>
> Please, i need help, i'm blocked!!!
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/dialplan-module-tp7602459p7602475.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] What is the module to fix SDP ?

[Carlos Eduardo]

Hello Rodrigo,

Maybe the function fix_nated_sdp, from nathelper module, would help.

1.5.2.  fix_nated_sdp(flags [, ip_address])

Alters the SDP information in orer to facilitate NAT traversal. What
changes to be performed may be controled via the “flags” paramter.

Meaning of the parameters is as follows:

flags - the value may be a bitwise OR of the following flags:
0x01 - adds “a=direction:active” SDP line;
0x02 - rewrite media IP address (c=) with source address of the message or
the provided IP address (the provide IP address take precedence over the
source address).
0x04 - adds “a=nortpproxy:yes” SDP line;
0x08 - rewrite IP from origin description (o=) with source address of the
message or the provided IP address (the provide IP address take precedence
over the source address).

ip_address - IP to be used for rewriting SDP. If not specified, the
received signalling IP will be used. The parameter allows pseudo-variables
usage. NOTE: For the IP to be used, you need to use 0x02 or 0x08 flags,
otherwise it will have no effect.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE,
BRANCH_ROUTE.

Example:
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};


2016-03-30 14:57 GMT-03:00 Rodrigo Pimenta Carvalho :

> Hi.
>
>
> I have a case when a peer answering a call (INVITE) is behind a NAT.
>
> So, in its SIP OK message I would like to see the SDP containing a valid
> IP and media Port valid to receive audio from the caller. That is, the
> caller need to know a valid IP and Port where he/she can send his/her audio
> packets.
>
>
> 1 - Is it possible to "fix" SDP content for such objective?
>
> 2 - Can OpenSIPS do something for this idea works or must I to use
> something more like a stun server?
>
> 3 - What is the OpenSIPS module that can help me with this task?
>
>
> I guess I will have to fix 2 fiels in SDP:
>
> Media Description, name and address (m): audio 55142 RTP/AVP 8 101  (
> to fix the port)
> Connection Information (c): IN IP4 192.168.100.156
>  ( to fix the IP)
>
> P.S.: I have already a solution (opensips.cfg) that let SIP messages cross
> NATs without problems. Only SDP has to be fixed.
>
>
> Any hint will be very helpful!
>
> Best regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
> ___
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> Users@lists.opensips.org
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>
>


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*Tecnólogo em Telecomunicações, OCP, dCAA*

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Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 1.11.6

[Louis Rochon]

Anybody have any ideas?


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Thursday, March 24, 2016 3:16 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

I recompiled with the 1.11.6, and did not resolve the issue. Traces show 
behaviour identical between 1.11.5 and 1.11.6.

Looking at the release notes for 1.11.6, there are some fixes that are closely 
related, but don't fix my specific issue. i.e. 95f5f79, 26a0a62. 

Any suggestions? Is there some yet-to-be documented B2BLogic module parameter 
that needs to be set? 

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
Sent: Monday, March 21, 2016 2:22 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Thanks Liviu,

I will recompile and give it a try.

Louis


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Monday, March 21, 2016 9:31 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in 
1.11.5

Hi Louis,

There was a similar discussion back in August [1], concluding with a fix [2], 
made between the 1.11.5 and 1.11.6 releases. Updating to the latest OpenSIPS 
1.11 will most likely solve this problem.

[1]: http://lists.opensips.org/pipermail/users/2015-August/032239.html
[2]: https://github.com/OpenSIPS/opensips/commit/95f5f79b9250

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 21.03.2016 14:55, Louis Rochon wrote:
> Anybody?
>
> Louis
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Louis Rochon
> Sent: Wednesday, March 16, 2016 10:02 AM
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] SDP missing in ACK indialog reinvite only in
> 1.11.5
>
> SDP in ACK lost in OpenSIPS 1.11.5
>
> This is something that used to work in OpenSIPS 1.8.1, but seems to have been 
> broken 1.11.5.
>
> Using the B2BUA facilities, I make the second leg of the call using 
> b2b_init_request. Then, if required, I move the call to another user agent 
> via a bash shell script, which issues a opensipsctl fifo b2b_bridge.
>
> For this in-dialog reinvite to work, there has to be an SDP in the ACK, which 
> is missing the trace produced using OpenSIPS 1.11.5. It's there in 1.8.1.
>
> Anyway of reintroducing the SDP in the ACK in 1.11.5?
>
> Trace Explanation
> -
> Objects: 10.10.8.103: source of initial call. Recipipient of reinvite.
>   10.10.10.205: CentOS running OpenSIPS with B2BUA. In one case, it's 
> CentOS 5.11 running OpenSIPs 1.8.1, the other (broken) is CentOS 6.7 running 
> OpenSIPs 1.11.5.
>   
> Call Flow in Trace:
> 10.10.10.205 10.10.8.103
>--Invite-->Indialog reinvite, so SDP
> <--Trying--
> <--200 OK--With SDP
> --ACK -->  With SDP in 1.8.1, no SDP in 1.11.5
>   
> OpenSIPS 1.8.1:
> ---
> INVITE sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 INVITE
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS (1.8.1-notls (x86_64/linux))
> Contact: 
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.ce759b97.0
> From: ;tag=B2B.483.37
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> Call-ID: 113c305d-89188638-113c3058-a664e63f@10.10.8.103
> CSeq: 2 INVITE
> Contact: 
> Content-Type: application/sdp
> Content-Length:   185
>
> v=0
> o=VOIPSIL_SIP 416521046 416521046 IN IP4 10.10.8.103 s=Sip Call c=IN
> IP4 10.10.8.33
> t=0 0
> m=audio 21260 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
>
> ACK sip:4507710011@10.10.8.103;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.205;branch=z9hG4bK109d.de759b97.0
> To: "4507710011" 
> ;tag=89188638-a664e63f-ver5d
> From: ;tag=B2B.483.37
> CSeq: 2 ACK
> Call-ID: 

Re: [OpenSIPS-Users] dialplan module

[Bogdan-Andrei Iancu]

Hi,

Then you have to install also the opensips-dialplan package from the 
same repo.


Regards,

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

On 31.03.2016 12:03, Francjos wrote:

I do not install opensips from source, i've done a deposit from opensips
repository to /etc/apt/sources.list.
So i dont see how i can recompile via "make menuconfig" because there is no
Makefile.



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Re: [OpenSIPS-Users] dialplan module

[Francjos]

I do not install opensips from source, i've done a deposit from opensips
repository to /etc/apt/sources.list.
So i dont see how i can recompile via "make menuconfig" because there is no
Makefile.



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Re: [OpenSIPS-Users] dialplan module

[Nagorny, Dimitry]

Francjos,

You just have to recompile via "make menuconfig" within your local opensips-git 
folder and confiure it to compile with dialplan module activ (I don't know if 
there's a shortcut).


Best Regards
Dimitry Nagorny

/*-Ursprüngliche Nachricht-
/*Von: users-boun...@lists.opensips.org [mailto:users-
/*boun...@lists.opensips.org] Im Auftrag von Francjos
/*Gesendet: Donnerstag, 31. März 2016 10:26
/*An: users@lists.opensips.org
/*Betreff: [OpenSIPS-Users] dialplan module
/*
/*Hello,
/*
/*I'm using Opensips 2.1.2 but the dialplan module is not present in the
/*directory /usr/lib/opensips/modules. Because of this, the module can't be
/*loaded.
/*Is there a way to include the module?
/*
/*Thanks
/*
/*
/*
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[OpenSIPS-Users] dialplan module

[Francjos]

Hello,

I'm using Opensips 2.1.2 but the dialplan module is not present in the
directory 
/usr/lib/opensips/modules. Because of this, the module can't be  loaded.
Is there a way to include the module?

Thanks



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