Re: [OpenSIPS-Users] Opensips crash

2017-02-05 Thread Denis via Users
Hello, Bogdan! Could you, please, tell me how can i checked, that the version of Opensips is latest? Thank you. -- С уважением, Денис.Best regards, Denis   05.02.2017, 19:05, "Bogdan-Andrei Iancu" :Hi Denis,First, be sure you use the latest 2.2 from GIT.Now, the crash indicates a memory corruption, so please activate the memory debugger:  Run 'make menuconfig' from the main source dir -> Configure Compile Options -> Configure Compile Flags -> check QM_MALLOC and DBG_MALLOC. Then hit 'q' to go back to the previous menu, and hit 'Save Changes'.Regards,Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.comOn 02/02/2017 10:03 AM, Denis via Users wrote:Hello! Server:: OpenSIPS (2.2.2 (x86_64/linux)) Information from the core file: https://yadi.sk/i/IfhdM1Y83CQvPk Thank you for any help. -- С уважением, Денис.Best regards, Denis___
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Re: [OpenSIPS-Users] IPv6 causing problems with HEP in siptrace

2017-02-05 Thread Bogdan-Andrei Iancu

Hi Jeff,

I suspect that this report is related to the second one you made - a 
wrong traced SRC interface for the replies will generate a mixture of 
IPv4 and IPv6 for same packet.

Please apply that fix and let's see if it fixes this one too.

Regards,

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

On 02/05/2017 01:12 AM, Jeff Pyle wrote:

Hi Bogdan,

Yes, but they do ingress and egress different families.

In this case the UAC's src IP is IPv6, talking to the proxy's IPv6 
interface address (no NAT64).  In the script there is a sethostport() 
to an IPv4 address, and a force_send_socket() to the proxy's IPv4 
interface address. Then a t_relay().


I have a sip_trace("tid", "d") right after create_dialog() for initial 
invites.  In a pure IPv4 flow this captures all ingress and egress 
traffic (except for t_relay's '100 Giving a try').  In this mixed IPv6 
-> IPv4 flow, I see the error, and capture only the egress IPv4 leg.


The HEP target is IPv4 at the moment.  The HEP capture server is also 
dual-stack, if this is relevant.



- Jeff


On Sat, Feb 4, 2017 at 12:58 PM, Bogdan-Andrei Iancu 
> wrote:


Hi Jeff,

That error is generated when the src and dst IP have different AF
families - which of course is impossible :)

I guess you get the error when you have an incoming SIP packet
over IPv6 (so both SRC and DST should be IPv6), right ?

Best regards,

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

On 02/04/2017 03:58 AM, Jeff Pyle wrote:
Hello, I have a simple dual-stack OpenSIPS 2.2.2 configuration. 
It includes a registrar and basic routing between locally

registered users.  These users have a mix of IPv4 and IPv6
contacts.  All is well.  I recently added Homer to the mix.  The
siptrace module is acting as a capture agent, duplicating
messages over HEPv3 to an upstream Homer capture server. When one
of the clients is IPv6, duplication fails.  I see this message:
ERROR:siptrace:trace_send_hep_duplicate: ERROR:
trace_send_hep_duplicate: interworking detected ?
ERROR:siptrace:save_siptrace: Failed to duplicate with hep to <:>
I'd like to duplicate messages regardless whether they've IPv4 or
IPv6, as many dialogs are both.  The Homer capture server is
IPv6-enabled and can receive HEP messages either (I think).  I've
tried to configure the siptrace trace_id with an IPv6 address but
it doesn't look like it's able to parse it.
What am I missing?
- Jeff

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Re: [OpenSIPS-Users] SIP message relay order

2017-02-05 Thread Bogdan-Andrei Iancu

Hi Stas,

Such races may happen at application level or even at network level 
(when using UDP) - so if you have 2 packets very close as time, they may 
swap. That is SIP :)


The full guilt is in the UAC device, IMHO - it should let some time gap 
between the ACK and re-INVITE, to eliminate any possible races.


Now, what you can do is to use the dialog module and to check the dialog 
state when receiving the re-invite. If $DLG_status is /3/ (Confirmed by 
a final reply but no ACK received yet), drop with no reply the 
re-INVITEs (to force a later retransmission) :

http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400

Regards,

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

On 02/02/2017 10:31 PM, Stas Kobzar wrote:

Hello List,

My call flow has initial INVITE and re-INVITE to update RTP IP/port.
Usually everything works well, but sometimes OpenSIPS come up with 
following example:


UA OpenSIPS  PSTN GW
---
INV(CSeq: 100) -> | ---> INV(CSeq: 100)
< 200 OK  | <--- 200 OK

(UA sends ACK then new INVITE)

ACK(CSeq: 100) -> |
reINV(Cseq: 101) ---> |

(OpenSIPS relays first INVITE then ACK)
  | ---> reINV(CSeq: 101)
  | --->   ACK(CSeq: 100)

When PSTN gateway receives re-INVITE before ACK for previous INVITE
it responds 500 with Retry-After header.
This is correct behaviour which conforms to the RFC 3261 section 14.2

My question is:
Is it possible to assure order of received and relayed messages within 
the same SIP session? Is there any configuration parameter?


Thank you,
--

Stas Kobzar

Developeur VoIP / VoIP Developer


Modulis­.ca Inc.

# Bureau / Office: 514-284-2020 x 246

Email: s tas.kob...@modulis.ca

https://www.modulis.com 



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Re: [OpenSIPS-Users] Rtpproxy bug

2017-02-05 Thread Bogdan-Andrei Iancu

Hi Robert,

It looks like a bug to me, but you have to check and report this to 
someone having commit rights on the rtpproxy repository.


Best regards,

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

On 02/04/2017 01:59 AM, Robert Dyck wrote:

I am reporting here because I don't know how to leave a bug report with Sippy
Software.

When I installed opensips and configured it use use rtpproxy and the unix
socket, opensips would not start. It turned out to be an issue with
permissions. It was simple to change to the loopback instead but I was curious
and wanted to get to the root of the problem.

I experimented with the -w command line option of rtpproxy. Combined with the
-u option this is supposed to allow one to set the permissions for the socket.
The default mode for the socket is 755 and I tried setting it to 775. Setting
-w 775 resulted in something very different and further changes did not yield
predictable results.

I looked into the code and it uses the atoi function to convert the mode
string into an integer rather than an octal number. Working backwards from 775
I get 0x1FD and 509 decimal. If I set the mode to 509 I get the desired
result.

Version info
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup command
Extension 20081224: Support for session timeout notifications
Extension 20090810: Support for automatic bridging
Extension 20140323: Support for tracking/reporting load
Extension 20140617: Support for anchoring session connect time
Extension 20141004: Support for extendable performance counters

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Re: [OpenSIPS-Users] Run a command in the timeroute

2017-02-05 Thread Bogdan-Andrei Iancu

Hi Jehanzaib,

IF you run some external command, there is no other way rather than 
using the exec module.


Regards,

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

On 02/02/2017 11:30 PM, J E H A N Z A I B wrote:

Hi folks,

Is there any way i can run a command "opensipsctl fifo ds_reload" in 
the timer_route ?
I think i can use exec module but i dont want to use, if there is 
anyother way please?


Thank you.

--
Regards,
Jehanzaib


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Re: [OpenSIPS-Users] Opensips crash

2017-02-05 Thread Bogdan-Andrei Iancu

Hi Denis,

First, be sure you use the latest 2.2 from GIT.

Now, the crash indicates a memory corruption, so please activate the 
memory debugger:  Run 'make menuconfig' from the main source dir -> 
Configure Compile Options -> Configure Compile Flags -> check QM_MALLOC 
and DBG_MALLOC. Then hit 'q' to go back to the previous menu, and hit 
'Save Changes'.


Regards,

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

On 02/02/2017 10:03 AM, Denis via Users wrote:

Hello!
Server:: OpenSIPS (2.2.2 (x86_64/linux))
Information from the core file: https://yadi.sk/i/IfhdM1Y83CQvPk
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis


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Re: [OpenSIPS-Users] siptrace picks incorrect source socket

2017-02-05 Thread Bogdan-Andrei Iancu

Hi Jeff,

Thank you for detailed report. I was able to reproduce and fix it. 
Please see:

https://github.com/OpenSIPS/opensips/commit/b30af734cdb84991e1f906e3920a94e87c33ea04

If you confirm the fix, I will do the backporting to 2.2 too.

Thanks and Regards,

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

On 02/04/2017 04:41 AM, Jeff Pyle wrote:

Hello,

I recently enabled siptrace on an OpenSIPS 2.2.2 system acting as a 
registrar and a proxy.  It has one IPv4 address with several ports for 
UDP, TCP and TLS.  In a case where the INVITE is relayed from TLS to 
UDP, the replies to the UAC are incorrectly being reported as coming 
from the UDP socket.


In the below scenario the UAC is at 64.65.66.67, and its ephemeral TCP 
client port for this call is 61235.  The OpenSIPS proxy is at 
131.132.133.134 listening on UDP 5060 and TLS 5061.  Also on 
131.132.133.134 there is a Freeswitch media server listening on UDP 
5080.  The UAC sends an INVITE to the proxy over TLS which routes it 
to the media server over UDP. The replies relayed to the UAC are 
reported as having come from port 5060 over UDP when in reality they 
have come from port 5061 over TCP (TLS).


My config:

listen=udp:131.132.133.134:5060 
listen=tls:131.132.133.134:5061 
listen=hep_udp:127.0.0.1:9030 

loadmodule "siptrace.so"
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_id",
"[hep]uri=hep:127.0.0.1:9060;transport=udp;")



Debugs:


INVITE in from UAC over TLS
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 22, host 131.132.133.134 , port 5061

INVITE out to media server over UDP
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080

100 Trying in from media server over UDP
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24650]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24650]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060

180 Ringing in from media server over UDP
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060

180 Ringing out to UAC over TLS (even though it reports UDP)
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235

200 OK in from media server over UDP
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060

200 OK out to UAC over TLS (even though it reports udp)
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235

ACK in from UAC over TLS
Feb  3 21:20:23 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235
Feb  3 21:20:23 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 22, host 131.132.133.134 , port 5061

ACK out to media server over UDP
Feb  3 21:20:23 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
Feb  3 21:20:23 testproxy /usr/sbin/opensips[24673]:
DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080


Everything routes properly, but it isn't reported by siptrace 
properly.  Is this a bug or am I doing something wrong?




- Jeff





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[OpenSIPS-Users] Run a command in the timeroute

2017-02-05 Thread J E H A N Z A I B
Hi folks,

Is there any way i can run a command "opensipsctl fifo ds_reload" in the
timer_route ?
I think i can use exec module but i dont want to use, if there is anyother
way please?

Thank you.

-- 
Regards,
Jehanzaib
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[OpenSIPS-Users] SipTrace Module - failed to get an id for "sip" tracing!

2017-02-05 Thread Victor Bogatyryev
Hi All, opensips 2.3 opensips.cfg  SipTrace Module loadmodule 
"siptrace.so" modparam("siptrace", "trace_on", 1) modparam("siptrace", 
"trace_id", 
"[tid]uri=mysql://opensips:opensipsrw@localhost/opensips;table=sip_trace;") 
MariaDB [opensips]> DESCRIBE sip_trace; 
+-+--+--+-+-++ 
| Field | Type | Null | Key | Default | Extra | 
+-+--+--+-+-++ 
| id | int(10) unsigned | NO | PRI | NULL | auto_increment | | 
time_stamp | datetime | NO | MUL | 1900-01-01 00:00:01 | | | callid | 
char(255) | NO | MUL | | | | trace_attrs | char(128) | YES | MUL | NULL 
| | | msg | text | NO | | NULL | | | method | char(32) | NO | | | | | 
status | char(128) | YES | | NULL | | | from_proto | char(5) | NO | | 
NULL | | | from_ip | char(50) | NO | MUL | | | | from_port | int(5) 
unsigned | NO | | NULL | | | to_proto | char(5) | NO | | NULL | | | 
to_ip | char(50) | NO | | | | | to_port | int(5) unsigned | NO | | NULL 
| | | fromtag | char(64) | NO | | | | | direction | char(4) | NO | | | | 
+-+--+--+-+-++ 
15 rows in set (0.00 sec) # opensipsctl start INFO: Starting OpenSIPS : 
ERROR: PID file /var/run/opensips/opensips.pid does not exist -- 
OpenSIPS start failed Degug: /local/sbin/opensips[2061]: 
ERROR:siptrace:mod_init: failed to get an id for "sip" tracing! 
/sbin/opensips[2061]: ERROR:core:init_mod: failed to initialize module 
siptrace /sbin/opensips[2061]: ERROR:core:main: error while initializing 
modules /sbin/opensips[2061]: INFO:core:cleanup: cleanup 
/local/sbin/opensips[2061]: NOTICE:core:main: Exiting Regards, 
Victor Bogatyryev


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