[OpenSIPS-Users] Does OpenSIPS use SQLite configuration to hold on queries when database is locked?

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS users,


In my software, programmed in QT (framework for C++) and handling data with a 
SQLite database, I have used this:


pDb.setConnectOptions("QSQLITE_BUSY_TIMEOUT=6000");


That is, if SQLite complains that "database is locked" sometime when my 
software tries to register some datum there, such database keeps the query 
paused (hold on), and then after 6 seconds let the query execute. This 
mechanism is transparent for my software and certify that the query will be 
tried every 6 seconds until it complete successfully.


Is there something similar to it in OpenSIPS ? That is, does OpenSIPS uses some 
kind of configuration provided by SQLite?

If not, why the developers team decided not to use such mechanism?


Any comment will be very helpful!


Thanks a lot!



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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[OpenSIPS-Users] OpenSIPS 2.2 SIP ping not triggering

[SamyGo]

Hi All,

I've OpenSIPS 2.2 and doing some tests with it I can't seem to find any
OPTIONS triggering for the registered users.

I've following settings for usrloc module and nathelper

loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
"mysql://root:Q4FRX@localhost/opensips_2_2")


loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pin...@siptest.saevolgo.ca")
modparam("nathelper", "received_avp", "$avp(received)")
modparam("nathelper", "natping_tcp", 1)
modparam("nathelper", "ping_threshold", 10)
modparam("nathelper", "max_pings_lost", 5)

I force set NAT flag for all SIP packets either INVITE or REGISTER

route{

route(SIP_CHECK);
# Handles and adjusts packets for NAT'e clients
force_rport();
if (nat_uac_test("23")) {
if (is_method("REGISTER")) {
fix_nated_register();
setbflag(NAT);
} else {
fix_nated_contact();
setflag(NAT);
}
}
setflag(NAT);

and set SIP_PING_FLAG before save() function:


if ( proto==TCP || proto==TLS || 0 ) setflag(TCP_PERSISTENT);

if (isflagset(NAT)) {
setbflag(SIP_PING_FLAG);
}

if (!save("location"))
sl_reply_error();

If I look at online users the flags are seen applied to the extensions:

root@test1:/etc/opensips# opensipsctl ul show
Domain:: location table=512 records=1
AOR:: 1...@siptest.saevolgo.ca
Contact::
sip:1009@64.231.959.69:30514;rinstance=38d9f5f00287404f;transport=udp
Q=
Expires:: 3093
Callid::
1955a010ad311f31OGZjZmJjZWZhZTQxMjNjOGRhOTZiYjUxNWQ4Y2JiODg.
Cseq:: 2
User-agent:: eyeBeam release 1003s stamp 31159
State:: CS_SYNC
Flags:: 0
Cflags:: SIP_PING_FLAG
Socket:: udp:X.X.X.X:5060
Methods:: 5951


I waited for like 15 minutes but no OPTIONS triggered from opensips !

Any guidance/help will be appreciated.


Regards.
Sammy
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[OpenSIPS-Users] OpenSIPS @ ClueCon weekly call

[Bogdan-Andrei Iancu]

Hi,

Join us tomorrow, 29th of June 2016, 12:00 CST to the ClueCon weekly 
call where OpenSIPS is the main guest.


We will have discussion about the latest news/plans of OpenSIPS, about 
the integration with FreeSwitch and the joined training at ClueCon.


Where? https://conference.freeswitch.org/vc/ 
extension 888 See you 
tomorrow !!! Regards,


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

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Re: [OpenSIPS-Users] Preventing invalid packet parsing

[Bogdan-Andrei Iancu]

Hi,

Well, if the "message" got to the script level, it means is "looks" like 
a SIP message - to reach the script, the message must have a SIP valid 
first line and Top most VIA.


So, if your "bogus" messages do not have anything to do with SIP, 
considering actions at script level is a non sense as it is too late. 
What you can do is to rely on the size of the package in order to drop 
it (by default, OpenSIPS drops any UDP packet shorter than 20 bytes - 
see MIN_UDP_PACKET in config.h). You can change that value to match your 
needs. And recompile.


IF the incoming traffic looks like SIP packages (and it lands in your 
script), you should use the sipmsg_validate() to get ready of it asap. 
It is not about processing, it is more about avoid polluting the logs 
with errors :)


Regards,

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

On 28.06.2016 18:06, Owais Ahmad wrote:
Thats not the case Alex. But I am expecting a large number of UDP 
messages arriving on the same port as my udp listening socket.

Just want to be sure there is no wasteful work done parsing such packets.

On Tue, Jun 28, 2016 at 8:04 PM, Alex Balashov 
> wrote:


Are you under the impression that these consume a significant
amount of OpenSIPS resources? If so, how did you arrive at that
conclusion?

On 06/28/2016 11:02 AM, Owais Ahmad wrote:

Hello,

I am getting UDP (non-SIP) packets on opensips listening port.
How can I
drop those packets to ensure it only consumes the least
opensips resources.

I do have mechanisms to prevent this on the firewall but just
want to be
sure that if such a network packet does reach the server, opensips
doesn't end up parsing it assuming its SIP.

I want to do this without using: sipmsg_validate





Regards,
Owais


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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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Re: [OpenSIPS-Users] Preventing invalid packet parsing

[Alex Balashov]

On 06/28/2016 11:06 AM, Owais Ahmad wrote:


Thats not the case Alex. But I am expecting a large number of UDP
messages arriving on the same port as my udp listening socket.
Just want to be sure there is no wasteful work done parsing such packets.


You can be reasonably sure that OpenSIPS "fingerprints" such packets 
quite efficiently and, if it doesn't type as a SIP packet, it won't be 
parsed.


For more details, see:

https://github.com/OpenSIPS/opensips/blob/master/parser/msg_parser.c#L543

As you can see, it parses the first line of the message first, and 
doesn't do any unnecessary work to parse the whole message if the first 
line doesn't suggest it's SIP.


I wouldn't worry about this "premature optimisation". :-)

--
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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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Re: [OpenSIPS-Users] Preventing invalid packet parsing

[Owais Ahmad]

Thats not the case Alex. But I am expecting a large number of UDP messages
arriving on the same port as my udp listening socket.
Just want to be sure there is no wasteful work done parsing such packets.

On Tue, Jun 28, 2016 at 8:04 PM, Alex Balashov 
wrote:

> Are you under the impression that these consume a significant amount of
> OpenSIPS resources? If so, how did you arrive at that conclusion?
>
> On 06/28/2016 11:02 AM, Owais Ahmad wrote:
>
>> Hello,
>>
>> I am getting UDP (non-SIP) packets on opensips listening port. How can I
>> drop those packets to ensure it only consumes the least opensips
>> resources.
>>
>> I do have mechanisms to prevent this on the firewall but just want to be
>> sure that if such a network packet does reach the server, opensips
>> doesn't end up parsing it assuming its SIP.
>>
>> I want to do this without using: sipmsg_validate
>> 
>> 
>>
>> Regards,
>> Owais
>>
>>
>> ___
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>> 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] Preventing invalid packet parsing

[Alex Balashov]

Are you under the impression that these consume a significant amount of 
OpenSIPS resources? If so, how did you arrive at that conclusion?


On 06/28/2016 11:02 AM, Owais Ahmad wrote:

Hello,

I am getting UDP (non-SIP) packets on opensips listening port. How can I
drop those packets to ensure it only consumes the least opensips resources.

I do have mechanisms to prevent this on the firewall but just want to be
sure that if such a network packet does reach the server, opensips
doesn't end up parsing it assuming its SIP.

I want to do this without using: sipmsg_validate



Regards,
Owais


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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] Preventing invalid packet parsing

[Owais Ahmad]

Hello,

I am getting UDP (non-SIP) packets on opensips listening port. How can I
drop those packets to ensure it only consumes the least opensips resources.

I do have mechanisms to prevent this on the firewall but just want to be
sure that if such a network packet does reach the server, opensips doesn't
end up parsing it assuming its SIP.

I want to do this without using: sipmsg_validate



Regards,
Owais
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Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

[Newlin, Ben]

No. I am using topology_hiding on my private nodes, so the intent of using 
set_advertised_address was to have the Contact and other headers contain the 
public node’s IP after topology_hiding. Instead I removed the 
set_advertised_address on the private nodes and I record_route on the public 
nodes. This makes the calls route properly as before, but it exposes the 
private node’s IP in Contact and other headers. It was not desired, but it’s 
not a big deal especially since the logic and routing is much cleaner this way.

Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Tuesday, June 28, 2016 at 5:45 AM
To: "Newlin, Ben" , "users@lists.opensips.org" 

Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong 
address in VIA header

Hi Ben,

You mean you used the record_route_preset ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.06.2016 20:02, Newlin, Ben wrote:
I have opened issue #917 on Github [1].

I should mention that I have worked around this problem by deciding to use 
Record-Routes instead of set_advertised_address. It is much cleaner, even 
though it does expose my private IPs. However, I have kept this configuration 
in case you need me to perform any tests or get tracing/logs. Thanks!

[1] https://github.com/OpenSIPS/opensips/issues/917

Ben Newlin

From: "Newlin, Ben" 
Date: Monday, June 27, 2016 at 11:06 AM
To: Bogdan-Andrei Iancu , 
"users@lists.opensips.org" 

Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong 
address in VIA header

Did you mean to say if you set it in request route?

I should clarify that when I am setting the advertised address the second time 
it is of course happening in failure_route as the first request has failed at 
that point. Perhaps that is the issue?

I will open a bug. Thanks.

Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Monday, June 27, 2016 at 10:41 AM
To: "Newlin, Ben" , 
"users@lists.opensips.org" 

Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong 
address in VIA header

Hi Ben,

If you set the advertised host / port in branch route, it will have impact over 
the entire transaction (all branches). So, any local replies (CANCEL and ACK) 
that are constructed by OpenSIPS (for any branch) will use the same set of 
advertised values. Which is of course wrong. Let us come up with the fix (as 
idea and code).

Could you open a  bug report on the GITHUB tracker, please ?

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.06.2016 15:45, Newlin, Ben wrote:
I always set the advertised address in request route.

Also as the original issue noted the second INVITE does go out with the correct 
advertised address in the VIA. It is only the local ACK for the failed second 
request that contains the wrong address in the VIA. So set_advertised_address 
appears to be working, but the local generated ACK is not using that address.


Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Monday, June 27, 2016 at 5:37 AM
To: "users@lists.opensips.org" 
, "Newlin, Ben" 

Subject: Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong 
address in VIA header

Hi Ben,

Where in the script do you do the first advertise_address ? In the request 
route or in a branch route ?

Regards,





Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.06.2016 03:41, Newlin, Ben wrote:
I have run into the same problem that was described in this previous post [1], 
however it doesn’t appear it was ever solved at the time.

I am using the dispatcher module to route calls to external carriers and I am 
using set_advertised_address to set the outgoing public address prior to 
sending the request. If the first destination returns failure, the ACK is sent 
correctly. Then I select a different destination and set a different public 
address using set_advertised_address. If this second call also fails, the ACK 
that is sent out uses the first advertised address, not the current on for the 
request.

Has anyone figured this out? I am using 1.11.6.

[1] http://lists.opensips.org/pipermail/users/2014-August/029779.html


Ben Newlin







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Re: [OpenSIPS-Users] running sip tls on 443

[Bogdan-Andrei Iancu]

Hi Tito,

If opensips crashes, were you able to extract a backtrace from the core 
file(s) ?


Thanks and regards,

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

On 27.06.2016 21:20, Tito Cumpen wrote:

Group,


I am experiencing strange behavior when configuring sip tls on port 
443. At time opensips crashes or stops accepting new connections. Here 
are the tcp configs I am using:


#disable_tcp=no

tcp_connection_lifetime=3600

tcp_connect_timeout=3

tcp_keepidle = 30

tcp_keepinterval = 5

tcp_keepalive = 1

tcp_keepcount = 5

tcp_max_msg_time = 8

tcp_children=10


Any idea what would case this? I am assuming there are probes out in 
the internet that eventually make opensips crash?


Thanks,
Tito


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Re: [OpenSIPS-Users] How to configure opensips to show passwords in column password from table subscriber?

[Bogdan-Andrei Iancu]

Hi Rodrigo,

IF you want to use plain text passwords, see 
http://www.opensips.org/Documentation/TipsFAQ#toc2 - in opensipsctl set 
the STORE_PLAINTEXT_PW to 1 (to store the text pwd too).  On the 
OpenSIPS side, to use the plain text pwd, enable the calculate_ha1 param 
in the auth module:

http://www.opensips.org/html/docs/modules/2.2.x/auth.html#id293572


Best regards,

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

On 27.06.2016 16:45, Rodrigo Pimenta Carvalho wrote:


Hi.



When I execute /opensipsctl add user password/, the column password in 
table subscriber remains empty. Ha1 has a kind of encrypted password.



How to configure opensips to show passwords in column password from 
table subscriber?



I have tried changing some parameters in module auth_db, but it didn't 
take effect. So, what is the correct configuration?



Any hint will be very helpful!


Regards.


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


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Re: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github?

[Rodrigo Pimenta Carvalho]

Hi Razvan


Thank you very much for the instructions and the alert!

I will fork and pull that.


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 Razvan Crainea 
Enviado: terça-feira, 28 de junho de 2016 04:19
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] I have a Patch that fixes memory leak on 
OpenSIPS. How to apply this path via github?


Hi, Rodrigo!


Please fork opensips and open a pull request on Github[1].

The idea is simple:

1. fork the repository[2]

2. Apply the patch, commit it and push it in your fork

3. Open a pull request[3]


[1] https://github.com/OpenSIPS/opensips/pulls

[2] https://help.github.com/articles/fork-a-repo/

[3] 
https://help.github.com/articles/using-pull-requests/#initiating-the-pull-request


PS: it is not a good idea to attach a file on a mailing list. Use 
gist.github.com, or pastebin.com next time :).


Thanks and regards,

Razvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/27/2016 09:05 PM, Rodrigo Pimenta Carvalho wrote:


Dear OpenSIPS users,


Daniel Fússia, from Inatel Competence Center 
(www.inatel.br) has discovered some 
issues related to the code in OpenSIPS 2.2 that handles some transactions in 
SQLite. He also has proposed the solution for such issues and his work is 
attached on this message.


How could I resquet to the OpenSIPS development team to apply this fix? That 
is, can someone here give me the instructions on how to use github and request 
that fix? I? very new on github.


Any hint will be very helpful!


Thanks alot!


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









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Re: [OpenSIPS-Users] Testing Reason Header

[Bogdan-Andrei Iancu]

Hi Alain,

In failure route, the processed SIP message is the original INVITE - see 
http://www.opensips.org/Documentation/Script-Routes-2-2#toc3


And you want to get the "Reason" from the reply message , so you should 
do $(hdr(Reason)) - see 
http://www.opensips.org/Documentation/Script-CoreVar-2-2 .


Best regards,

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

On 28.06.2016 12:41, Alain Bieuzent wrote:


Hi Nick, thanks for response.

Allready test without « : », doesn’t work also

thanks

*De : * au nom de Nick Altmann 


*Répondre à : *OpenSIPS users mailling list 
*Date : *mardi 28 juin 2016 11:18
*À : *OpenSIPS users mailling list 
*Objet : *Re: [OpenSIPS-Users] Testing Reason Header

Use is_present_hf("Reason"), but not is_present_hf("Reason:")


--
Nick

2016-06-28 12:14 GMT+03:00 Alain Bieuzent >:


Hi all,

I need to test if some particular value exist in Reason header on
a 404 reply.

I receive :

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK7454

From: "+331" >;tag=as005364a4

To: >;tag=SDufib599-019e007499e0

Call-ID: 323ca6b15f8404cd3d05d53014336311@1.2.3.4:5060


CSeq: 102 INVITE

Content-Length: 0

Reason: Q.850;cause=001

I’m trying using this code, but is_present_hf() function return
always false.

if ( t_check_status("404") ) {

if (is_present_hf("Reason:"))
xlog("L_WARN","Reason exist");

xlog("L_WARN","Reason not exist");

}

Any Idea ?


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Re: [OpenSIPS-Users] opensips 2.3 crash

[Bogdan-Andrei Iancu]

Hi Tito,

Are you sure there is no error in your local installation ? Starting 
with 2.2 there is no DBG_QM_MALLOC option as per your snapshot. Be sure 
you do not have mixture between different OpenSIPS versions.


Also, in this link : 
http://www.opensips.org/Documentation/TroubleShooting-OutOfMem, in "How 
to handle it" section, you have special paragraph for >= "2.2" .


Regards,

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

On 27.06.2016 21:29, Tito Cumpen wrote:

Bogdan,


The QM_malloc option specified in this document does not exist.Inline 
image 1

Was it replaced with


On Tue, May 3, 2016 at 1:11 PM, Bogdan-Andrei Iancu 
> wrote:


Hi Tito,

You should have reported it sooner:)
Could you compile in the memory debugger (as the backtrace you
posted here points to a memory corruption). Use DBG_MALLOC flag in
combination with F_ / HP_ / QM_ memeory managers :
http://www.opensips.org/Documentation/TroubleShooting-OutOfMem

Regards,

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

On 03.05.2016 20:00, Tito Cumpen wrote:

Group,


I get about two nightly crashes in the latest opensips head this
has been happening consistently for about a week and half now.
All crashes are identical to:

Using host libthread_db library "/lib64/libthread_db.so.1".

Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u
opensips -g opensips -M 1024 -f /etc'.

Program terminated with signal 11, Segmentation fault.

#0  0x00512512 in fm_remove_free (n=0x7f384a06b680,
qm=0x7f3849fc8010) at mem/f_malloc.c:187

187*pf=n->u.nxt_free;

Missing separate debuginfos, use: debuginfo-install
cyrus-sasl-lib-2.1.26-20.el7_2.x86_64
glibc-2.17-106.el7_2.4.x86_64 gmp-6.0.0-12.el7_1.x86_64
gnutls-3.3.8-14.el7_2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64
krb5-libs-1.13.2-12.el7_2.x86_64 libcom_err-1.42.9-7.el7.x86_64
libcurl-7.29.0-25.el7.centos.x86_64 libffi-3.0.13-16.el7.x86_64
libgcc-4.8.5-4.el7.x86_64 libgcrypt-1.5.3-12.el7_1.1.x86_64
libgpg-error-1.12-3.el7.x86_64 libidn-1.28-4.el7.x86_64
libmicrohttpd-0.9.33-2.el7.x86_64 librabbitmq-0.5.2-1.el7.x86_64
libselinux-2.2.2-6.el7.x86_64 libssh2-1.4.3-10.el7_2.1.x86_64
libstdc++-4.8.5-4.el7.x86_64 libtasn1-3.8-2.el7.x86_64
nettle-2.7.1-4.el7.x86_64 nspr-4.11.0-1.el7_2.x86_64
nss-3.21.0-9.el7_2.x86_64
nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64
openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64
pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64
xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64

(gdb) bt full

#0  0x00512512 in fm_remove_free (n=0x7f384a06b680,
qm=0x7f3849fc8010) at mem/f_malloc.c:187

pf = 0x0

hash = 2051

#1  fm_malloc (qm=0x7f3849fc8010, size=size@entry=65592) at
mem/f_malloc.c:415

frag = 0x7f384a06b680

n = 

hash = 2051

__FUNCTION__ = "fm_malloc"

#2  0x7f383e018652 in tcp_handle_req
(_max_msg_chunks=, con=,
req=) at ../../net/proto_tcp/tcp_common.h:459

local_rcv = {src_ip = {af = 760673648, len = 32766, u =
{addrl = {16, 20}, addr32 = {16, 0, 20, 0}, addr16 = {16, 0, 0,
0, 20, 0, 0, 0}, addr =
"\020\000\000\000\000\000\000\000\024\000\000\000\000\000\000"}},
dst_ip = {af = 1, len = 1, u = {addrl = {

139878494896234, 5869476}, addr32 = {106, 32568,
5869476, 0}, addr16 = {106, 0, 32568, 0, 36772, 89, 0, 0}, addr =
"j\000\000\000\070\177\000\000\244\217Y\000\000\000\000"}},
src_port = 0, dst_port = 0, proto = 0, proto_reserved1 = 0,

  proto_reserved2 = 0, src_su = {s = {sa_family = 62576,
sa_data = "V-\376\177\000\000\001\000\000\000\000\000\000"}, sin
= {sin_family = 62576, sin_port = 11606, sin_addr = {s_addr =
32766}, sin_zero = "\001\000\000\000\000\000\000"}, sin6 = {

  sin6_family = 62576, sin6_port = 11606,
sin6_flowinfo = 32766, sin6_addr = {__in6_u = {__u6_addr8 =
"\001\000\000\000\000\000\000\000\200\364V-\376\177\000",
__u6_addr16 = {1, 0, 0, 0, 62592, 11606, 32766, 0}, __u6_addr32 =
{1, 0, 760673408, 32766}}},

  sin6_scope_id = 24}}, bind_address =
0x4a03e631}

msg_buf = 

msg_len = 

c = 

size = 

#3  tls_read_req (con=0x7f38481398c0, bytes_read=0x7ffe2d56f550)
at proto_tls.c:441

bytes = 

total_bytes = 0

req = 

__FUNCTION__ = "tls_read_req"

#4  0x0059c478 in handle_io (fm=fm@entry=0x7f384a03e630,
idx=idx@entry=3, event_type=event_type@entry=1) at
net/net_tcp_proc.c:205

Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

[Bogdan-Andrei Iancu]

Hi Ben,

You mean you used the record_route_preset ?

Regards,

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

On 27.06.2016 20:02, Newlin, Ben wrote:


I have opened issue #917 on Github [1].

I should mention that I have worked around this problem by deciding to 
use Record-Routes instead of set_advertised_address. It is much 
cleaner, even though it does expose my private IPs. However, I have 
kept this configuration in case you need me to perform any tests or 
get tracing/logs. Thanks!


[1] https://github.com/OpenSIPS/opensips/issues/917

Ben Newlin

*From: *"Newlin, Ben" 
*Date: *Monday, June 27, 2016 at 11:06 AM
*To: *Bogdan-Andrei Iancu , 
"users@lists.opensips.org" 
*Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address 
contains wrong address in VIA header


Did you mean to say if you set it in request route?

I should clarify that when I am setting the advertised address the 
second time it is of course happening in failure_route as the first 
request has failed at that point. Perhaps that is the issue?


I will open a bug. Thanks.

Ben Newlin

*From: *Bogdan-Andrei Iancu 
*Date: *Monday, June 27, 2016 at 10:41 AM
*To: *"Newlin, Ben" , "users@lists.opensips.org" 

*Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address 
contains wrong address in VIA header


Hi Ben,

If you set the advertised host / port in branch route, it will have 
impact over the entire transaction (all branches). So, any local 
replies (CANCEL and ACK) that are constructed by OpenSIPS (for any 
branch) will use the same set of advertised values. Which is of course 
wrong. Let us come up with the fix (as idea and code).


Could you open a  bug report on the GITHUB tracker, please ?

Regards,



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

On 27.06.2016 15:45, Newlin, Ben wrote:

I always set the advertised address in request route.

Also as the original issue noted the second INVITE does go out
with the correct advertised address in the VIA. It is only the
local ACK for the failed second request that contains the wrong
address in the VIA. So set_advertised_address appears to be
working, but the local generated ACK is not using that address.

Ben Newlin

*From: *Bogdan-Andrei Iancu 

*Date: *Monday, June 27, 2016 at 5:37 AM
*To: *"users@lists.opensips.org" 
 ,
"Newlin, Ben"  
*Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address
contains wrong address in VIA header

Hi Ben,

Where in the script do you do the first advertise_address ? In the
request route or in a branch route ?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 25.06.2016 03:41, Newlin, Ben wrote:

I have run into the same problem that was described in this
previous post [1], however it doesn’t appear it was ever
solved at the time.

I am using the dispatcher module to route calls to external
carriers and I am using set_advertised_address to set the
outgoing public address prior to sending the request. If the
first destination returns failure, the ACK is sent correctly.
Then I select a different destination and set a different
public address using set_advertised_address. If this second
call also fails, the ACK that is sent out uses the first
advertised address, not the current on for the request.

Has anyone figured this out? I am using 1.11.6.

[1]
http://lists.opensips.org/pipermail/users/2014-August/029779.html

Ben Newlin






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Re: [OpenSIPS-Users] ACK after set_advertised_address contains wrong address in VIA header

[Bogdan-Andrei Iancu]

Hi Ben,

Thanks for the clarification. I labeled this as a bug, as whatever you 
do in the script, you cannot get the desired (which is valid) behavior. 
I got the ticker, we will try to get it fixed shortly.


Regards,

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

On 27.06.2016 18:06, Newlin, Ben wrote:


Did you mean to say if you set it in request route?

I should clarify that when I am setting the advertised address the 
second time it is of course happening in failure_route as the first 
request has failed at that point. Perhaps that is the issue?


I will open a bug. Thanks.

Ben Newlin

*From: *Bogdan-Andrei Iancu 
*Date: *Monday, June 27, 2016 at 10:41 AM
*To: *"Newlin, Ben" , "users@lists.opensips.org" 

*Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address 
contains wrong address in VIA header


Hi Ben,

If you set the advertised host / port in branch route, it will have 
impact over the entire transaction (all branches). So, any local 
replies (CANCEL and ACK) that are constructed by OpenSIPS (for any 
branch) will use the same set of advertised values. Which is of course 
wrong. Let us come up with the fix (as idea and code).


Could you open a  bug report on the GITHUB tracker, please ?

Regards,


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

On 27.06.2016 15:45, Newlin, Ben wrote:

I always set the advertised address in request route.

Also as the original issue noted the second INVITE does go out
with the correct advertised address in the VIA. It is only the
local ACK for the failed second request that contains the wrong
address in the VIA. So set_advertised_address appears to be
working, but the local generated ACK is not using that address.

Ben Newlin

*From: *Bogdan-Andrei Iancu 

*Date: *Monday, June 27, 2016 at 5:37 AM
*To: *"users@lists.opensips.org" 
 ,
"Newlin, Ben"  
*Subject: *Re: [OpenSIPS-Users] ACK after set_advertised_address
contains wrong address in VIA header

Hi Ben,

Where in the script do you do the first advertise_address ? In the
request route or in a branch route ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 25.06.2016 03:41, Newlin, Ben wrote:

I have run into the same problem that was described in this
previous post [1], however it doesn’t appear it was ever
solved at the time.

I am using the dispatcher module to route calls to external
carriers and I am using set_advertised_address to set the
outgoing public address prior to sending the request. If the
first destination returns failure, the ACK is sent correctly.
Then I select a different destination and set a different
public address using set_advertised_address. If this second
call also fails, the ACK that is sent out uses the first
advertised address, not the current on for the request.

Has anyone figured this out? I am using 1.11.6.

[1]
http://lists.opensips.org/pipermail/users/2014-August/029779.html

Ben Newlin





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Re: [OpenSIPS-Users] Testing Reason Header

[Alain Bieuzent]

Hi Nick, thanks for response.

 

Allready test without « : », doesn’t work also

 

thanks

 

 

 

De :  au nom de Nick Altmann 

Répondre à : OpenSIPS users mailling list 
Date : mardi 28 juin 2016 11:18
À : OpenSIPS users mailling list 
Objet : Re: [OpenSIPS-Users] Testing Reason Header

 

Use is_present_hf("Reason"), but not is_present_hf("Reason:")


--
Nick

 

2016-06-28 12:14 GMT+03:00 Alain Bieuzent :

Hi all,

 

I need to test if some particular value exist in Reason header on a 404 reply. 

 

I receive : 

 

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK7454

From: "+331" ;tag=as005364a4

To: ;tag=SDufib599-019e007499e0

Call-ID: 323ca6b15f8404cd3d05d53014336311@1.2.3.4:5060

CSeq: 102 INVITE

Content-Length: 0

Reason: Q.850;cause=001

 

I’m trying using this code, but is_present_hf() function return always false.

 

if ( t_check_status("404") ) {

if (is_present_hf("Reason:")) xlog("L_WARN","Reason exist");

xlog("L_WARN","Reason not exist");

}

 

Any Idea ?


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Re: [OpenSIPS-Users] Testing Reason Header

[Nick Altmann]

Use is_present_hf("Reason"), but not is_present_hf("Reason:")

--
Nick

2016-06-28 12:14 GMT+03:00 Alain Bieuzent :

> Hi all,
>
>
>
> I need to test if some particular value exist in Reason header on a 404
> reply.
>
>
>
> I receive :
>
>
>
> SIP/2.0 404 Not Found
>
> Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK7454
>
> From: "+331" ;tag=as005364a4
>
> To: ;tag=SDufib599-019e007499e0
>
> Call-ID: 323ca6b15f8404cd3d05d53014336311@1.2.3.4:5060
>
> CSeq: 102 INVITE
>
> Content-Length: 0
>
> Reason: Q.850;cause=001
>
>
>
> I’m trying using this code, but is_present_hf() function return always
> false.
>
>
>
> if ( t_check_status("404") ) {
>
> if (is_present_hf("Reason:")) xlog("L_WARN","Reason
> exist");
>
> xlog("L_WARN","Reason not exist");
>
> }
>
>
>
> Any Idea ?
>
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Testing Reason Header

[Alain Bieuzent]

Hi all,

 

I need to test if some particular value exist in Reason header on a 404 reply. 

 

I receive : 

 

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK7454

From: "+331" ;tag=as005364a4

To: ;tag=SDufib599-019e007499e0

Call-ID: 323ca6b15f8404cd3d05d53014336311@1.2.3.4:5060

CSeq: 102 INVITE

Content-Length: 0

Reason: Q.850;cause=001

 

I’m trying using this code, but is_present_hf() function return always false.

 

if ( t_check_status("404") ) {

    if (is_present_hf("Reason:")) xlog("L_WARN","Reason exist");

    xlog("L_WARN","Reason not exist");

}

 

Any Idea ?

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Re: [OpenSIPS-Users] How to invok lookup() and get attr from the caller, without side effects?

[Rodrigo Pimenta Carvalho]

Hi Johan and Ben.


Yes. AVPops is a easy solution. However, it is easy for dada stored in DB.

What about data stored in RAM?


I'm using db_mode = 0 for module usrloc (so user location is always in RAM). 
So, if AVPops could extract data from the RAM too, as it does with queries and 
DB, it would be very easy.


I have looked for a solution using avp_db_query, but it works only over DB, not 
over RAM.

That is why I started trying to use function lookup() and avp attr, to get 
caller specific information.


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 johan de clercq 
Enviado: quarta-feira, 22 de junho de 2016 07:35
Para: 'OpenSIPS users mailling list'; 'sevpal'
Assunto: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?


Ben is correct.  In my opinion, a very easy solution.



From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Newlin, Ben
Sent: Tuesday, June 21, 2016 5:24 PM
To: sevpal ; OpenSIPS users mailling list 

Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?



It also seems like AVPOPS module [1] may be a good solution here as it has 
functions to pull data from a database into AVPs based by user.



[1] http://www.opensips.org/html/docs/modules/2.2.x/avpops.html



Ben Newlin



From: 
> on 
behalf of sevpal >
Reply-To: sevpal >, OpenSIPS users 
mailling list >
Date: Tuesday, June 21, 2016 at 11:20 AM
To: OpenSIPS users mailling list 
>
Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?



Hi, have you tried/considered running a simple query on the database and 
parsing for the information you need?



From: Rodrigo Pimenta Carvalho

Sent: Tuesday, June 21, 2016 10:39 AM

To: OpenSIPS users mailling list

Subject: Re: [OpenSIPS-Users] How to invok lookup() and get attr from the 
caller, without side effects?



Hi Răzvan.



I have tried that idea. But that didn't work. The SIP INVITE message is being 
changed by the OpenSIPS in a wrong way, in my point of view.

Do you know some way to save the entire SIP INVITE message before calling 
lookup() and then make the saved message take place after the lookup() 
execution?



My original message is:



INVITE sip:6...@mydomain.com.br SIP/2.0
Via: SIP/2.0/TCP 192.168.21.40:5090;rport;branch=z9hG4bK876727215
From: ;tag=179920819
To: 
Call-ID: 1410250893
CSeq: 21 INVITE
Contact: 
Proxy-Authorization: Digest username="crdphmacl_SPnuV5xqtnSX", 
realm="localhost", nonce="5769458c01cc263a7c0d6995dc48d42288ec6f8e4048", 
uri="sip:6...@mydomain.com.br", response="0f4c122d2a0a28dea6194c235cd77430", 
algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   227



This is being changed to:



INVITE 
sip:crdphmacl_SPnuV5xqtnSX@131.221.240.204:60672;transport=tcp;line=c6356a7d87d6f81
 SIP/2.0
Record-Route: 
Via: SIP/2.0/TCP myDomain.com.br:5060;branch=z9hG4bKe2db.49d54587.0;i=1
Via: SIP/2.0/TCP 
192.168.21.40:5090;received=xxx.yyy.240.204;rport=60672;branch=z9hG4bK716249970
From: ;tag=12586028
To: 
Call-ID: 1106771604
CSeq: 21 INVITE
Contact: 
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Subject: Phone call
Content-Length:   224



So, the caller is receiving its own SIP INVITE.

That is why when A calls B, is A that rings, not B.



It is becoming a bit complicated. So, I suspect I'm going to the incorrect 
direction



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 Răzvan Crainea >

Re: [OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

And are you forcing RTPengine to act as an ice light client?  It looks 
like you are gettin a single ICE candidate in the answer back from 
freeswitch which would indicate that you are.


I'd check your chrome webrtc statistics to see if tis failed to do do 
ice/stun negotiation on the 183.  In general the signalling looks good.


I think you may have an error on your Freeswitch side - some thing that 
is trying to force it to use SRTP all the time, even though the 
signalling has requested plain RTP (to freeswitch).


I think you should ask in #freeswitch on freenode at this point.

-Eric

On 06/23/2016 01:42 PM, John Nash wrote:
Actually the issue is i hear no audio on either side and just after 
session progress (I guess when media starts coming from remote media 
server) i see error  "SRTP output wanted, but no crypto suite was 
negotiated"


I had also checked media logs i could see RTP packets being sent from 
freeswitch to RTPengine IP but there was no packet at all just after 
that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine 
should send that packet to browser using wss?


On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme > wrote:


So - i dont see a problem here - Chrome is getting
UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP.  Freeswitch
responded to the offer in the invite with an answer in the 183,
and in the 200.  What is the failure you are seeing, and where is
it happening (in freeswitch? in the browser?)

The only thing that looks bad is that you are retransmitting the
ACK which FS either ... doesnt like, or is never getting,  because
it keeps retransmitting the 200, which is why you get a 481 when
you send BYE.

-Eric


On 06/23/2016 01:24 PM, John Nash wrote:

OK here is the log
https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744

Sorry took me a while to convert wireshark trace to text file.

My freeswitch is running on private IP (127.0.0.1) and opensips I
run on both public and private so that for outside world opensips
is the only public IP they see. In proxy log I pasted Opensips
===> Freeswitch logs and back.






On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme > wrote:

No - it's annoying to look at a trace that's had information
removed and try and piece together whats happening.  Your
paranoid side is wrong, sorry.

-Eric


On 06/23/2016 01:06 PM, Patrick Wakano wrote:

my paranoic side would recommend to hide/change private
informations, specially any authentication line that might
appear... this is certainly a sort of social engineering
threat we should worry...
better be safe than sorry


On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme
> wrote:

I mean you can use a private gist, but you will be
publishing the link in a public email list. In general I
personally dont believe revealing ip addresses etc. is
any problem - to put my money where my mouth is here is
a gist link to an unaltered SIP trace on my server :)

https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52

-Eric


On 06/23/2016 12:23 PM, John Nash wrote:

Ok i am ready with logs. About gist may I use private
option as traces have our IPs, user

On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme
> wrote:

Hey John,

Please paste a full UNALTERED sip trace into a gist
(gist.github.com ) from the
proxy servers perspective and provide a link so
that we can see what comes in, and what goes out
from both sides.

EG: ngrep -qtd any -W byline port 5060

This will show us the traffic that is leaving the
proxy destined for the Freeswitch box, and what the
freeswitch box sends back.

Also - you can look in your browsers console log
and provide the SIP trace from there in a seperate
gist, so that we can see what opensips sends back
up to your browser.

-Eric



Am I using correct sip.js example? I copied it to
my server and accessing it using https: (used
letsencrypt)

On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme
> wrote:

1. I would suggest using SIP.js -
https://github.com/onsip/SIP.js it is a much
more active project that sipml5.

2. Im 

Re: [OpenSIPS-Users] I have a Patch that fixes memory leak on OpenSIPS. How to apply this path via github?

[Răzvan Crainea]

Hi, Rodrigo!


Please fork opensips and open a pull request on Github[1].

The idea is simple:

1. fork the repository[2]

2. Apply the patch, commit it and push it in your fork

3. Open a pull request[3]


[1] https://github.com/OpenSIPS/opensips/pulls

[2] https://help.github.com/articles/fork-a-repo/

[3] 
https://help.github.com/articles/using-pull-requests/#initiating-the-pull-request



PS: it is not a good idea to attach a file on a mailing list. Use 
gist.github.com, or pastebin.com next time :).



Thanks and regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/27/2016 09:05 PM, Rodrigo Pimenta Carvalho wrote:



Dear OpenSIPS users,


Daniel Fússia, from Inatel Competence Center (www.inatel.br 
) has discovered some issues related to the code 
in OpenSIPS 2.2 that handles some transactions in SQLite. He also has 
proposed the solution for such issues and his work is attached on this 
message.



How could I resquet to the OpenSIPS development team to apply this 
fix? That is, can someone here give me the instructions on how to use 
github and request that fix? Iḿ very new on github.



Any hint will be very helpful!


Thanks alot!


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






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