[OpenSIPS-Users] Professional OpenSips Support

[Albert Ofertas]

Hello, good morning,

We think that we have identified a bug on OpenSips and we would like to contact 
the official Opensips support team.

The thread where we have asked to the user group is this:
Problem: INVITE port mismatch

http://lists.opensips.org/pipermail/users/2015-July/032006.html

On the other hand, we have tried to contact two times in last week with 
Opensips-Solutions.com team, but we have not received any answer.

Could someone give us correct indications to receive professional support over 
OpdnSips?

Best regards

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Re: [OpenSIPS-Users] Simple questions before adopting OpenSIPS.

[Rodrigo Pimenta Carvalho]

Hi Sammy.


Thank you again for the reply today.

For my case, I will use direct media. So, I don't intend to install media-proxy.

Now I'm going to study more about OpenSIPS, with the documentation or videos.


Best regards




RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9300 (Brasil)

De: users-boun...@lists.opensips.org  em nome 
de SamyGo 
Enviado: quarta-feira, 15 de julho de 2015 11:54
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] Simple questions before adopting OpenSIPS.

See reply inline.

On Wed, Jul 15, 2015 at 10:43 AM, Rodrigo Pimenta Carvalho 
mailto:pime...@inatel.br>> wrote:




Dear OpenSIPS-users,

This is my first contact in this mailing list.

Yesterday I was watching the video OpenSIPS Kick Start, from Youtube, and I 
have collected all needed instructions to install and configure it.
But, before deciding to adopt OpenSIPS, I have to be sure that such SIP Proxy 
will be enough to my current requirements. So, tell me whether the following 
statements are true or false, please:

1 - The OpenSIPS has some kind of script, as an Asterisk dialplan, to let the 
programmer handle calls and SIP messages. True or false?

True

2 - By using OpenSIPS, it is possible to get information from a SIP message 
(EX: SIP MESSAGE body content, or SIP 183 message header field, etc), to save 
it in a database, or text file. True or false?

Yes

3 - By using OpenSIPS, it is possible to create a sort of ring group 
implementation where users are dialed and first one to answer will get the 
call. True or false?

True, but for media connectivity you need additional media-proxy, or the end 
points can contact directly.

4 - By using OpenSIPS, it is possible to let one callee answer the call, in a 
ring group implementation, and immediately cancel the ring of the others 
callees. True or false?

True


5 - Let's suppose all callees, in a ring group implementation, are softphones 
that reply with SIP 183 after receiving SIP INVITE. By using OpenSIPS, during a 
sort of ring group implementation where users are dialled at same time, the 
OpenSIPS will forward each SIP 183 message to the caller. So, if the caller 
calls N callees, it will receive N SIP 183 messages. True or false?

True

6 - OpenSIPS can use MySQL. It can use SQLite too. True or false?

True


7 - OpenSIPS has some integration or provide some way to use ICE. True or false.

True


Thanks.

Best Regards.




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

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Re: [OpenSIPS-Users] Simple questions before adopting OpenSIPS.

[SamyGo]

See reply inline.

On Wed, Jul 15, 2015 at 10:43 AM, Rodrigo Pimenta Carvalho <
pime...@inatel.br> wrote:

>
>
>
>
> Dear OpenSIPS-users,
>
> This is my first contact in this mailing list.
>
>
> Yesterday I was watching the video OpenSIPS Kick Start, from Youtube, and
> I have collected all needed instructions to install and configure it.
> But, before deciding to adopt OpenSIPS, I have to be sure that such SIP
> Proxy will be enough to my current requirements. So, tell me whether the
> following statements are true or false, please:
>
> 1 - The OpenSIPS has some kind of script, as an Asterisk dialplan, to let
> the programmer handle calls and SIP messages. True or false?
>
True

>
> 2 - By using OpenSIPS, it is possible to get information from a SIP
> message (EX: SIP MESSAGE body content, or SIP 183 message header field,
> etc), to save it in a database, or text file. True or false?
>
Yes

>
> 3 - By using OpenSIPS, it is possible to create a sort of ring group
> implementation where users are dialed and first one to answer will get the
> call. True or false?
>
True, but for media connectivity you need additional media-proxy, or the
end points can contact directly.

>
> 4 - By using OpenSIPS, it is possible to let one callee answer the call,
> in a ring group implementation, and immediately cancel the ring of the
> others callees. True or false?
>
True

>
> 5 - Let's suppose all callees, in a ring group implementation, are
> softphones that reply with SIP 183 after receiving SIP INVITE. By using
> OpenSIPS, during a sort of ring group implementation where users are
> dialled at same time, the OpenSIPS will forward each SIP 183 message to the
> caller. So, if the caller calls N callees, it will receive N SIP 183
> messages. True or false?
>
True

>
> 6 - OpenSIPS can use MySQL. It can use SQLite too. True or false?
>
True

>
>  7 - OpenSIPS has some integration or provide some way to use ICE. True
> or false.
>
True

>
>  Thanks.
>
> Best Regards.
>
>
>
>
>   RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Simple questions before adopting OpenSIPS.

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS-users,

This is my first contact in this mailing list.

Yesterday I was watching the video OpenSIPS Kick Start, from Youtube, and I 
have collected all needed instructions to install and configure it.
But, before deciding to adopt OpenSIPS, I have to be sure that such SIP Proxy 
will be enough to my current requirements. So, tell me whether the following 
statements are true or false, please:

1 - The OpenSIPS has some kind of script, as an Asterisk dialplan, to let the 
programmer handle calls and SIP messages. True or false?

2 - By using OpenSIPS, it is possible to get information from a SIP message 
(EX: SIP MESSAGE body content, or SIP 183 message header field, etc), to save 
it in a database, or text file. True or false?

3 - By using OpenSIPS, it is possible to create a sort of ring group 
implementation where users are dialed and first one to answer will get the 
call. True or false?

4 - By using OpenSIPS, it is possible to let one callee answer the call, in a 
ring group implementation, and immediately cancel the ring of the others 
callees. True or false?

5 - Let's suppose all callees, in a ring group implementation, are softphones 
that reply with SIP 183 after receiving SIP INVITE. By using OpenSIPS, during a 
sort of ring group implementation where users are dialled at same time, the 
OpenSIPS will forward each SIP 183 message to the caller. So, if the caller 
calls N callees, it will receive N SIP 183 messages. True or false?

6 - OpenSIPS can use MySQL. It can use SQLite too. True or false?


7 - OpenSIPS has some integration or provide some way to use ICE. True or false.


Thanks.

Best Regards.




RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)
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[OpenSIPS-Users] Opensips 1.10 crash

[dpa]

Hello!

 

Opensips1.10.1. Working with rtpproxy2.0.

Today Opensips has crashed. Core file was generated. 

Information from core file is in attachment.

 

The question is why Opensips has crashed?

 

Thank you for any help.

 

 

 

gdb /usr/local/opensips1.10.1/sbin/opensips /opensipscore/core
GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04
Copyright (C) 2012 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later 
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
For bug reporting instructions, please see:
...
Reading symbols from /usr/local/opensips1.10.1/sbin/opensips...done.
[New LWP 9791]

warning: Can't read pathname for load map: Îøèáêà ââîäà/âûâîäà.
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".

warning: no loadable sections found in added symbol-file system-supplied DSO at 
0x7fff78bfe000
Core was generated by `/usr/local/opensips1.10.1/sbin/opensips -P 
/sock/opensips.pid -u opensips -w /o'.
Program terminated with signal 11, Segmentation fault.
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
175 n->u.nxt_free->prev = pf;
(gdb) bt
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
#1  fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383
#2  0x7fcf685860a5 in shm_malloc_unsafe (size=) at 
../../mem/shm_mem.h:248
#3  shm_malloc (size=) at ../../mem/shm_mem.h:258
#4  add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367
#5  0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260
#6  0x7fcf685715a4 in add_rule (rule=, prefix=, grplst=, rdata=) at dr_load.c:188
#7  dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, 
drd_table=, drc_table=, drr_table=0x7fcf687939f0) 
at dr_load.c:512
#8  0x7fcf685811c0 in dr_reload_data () at drouting.c:425
#9  dr_reload_cmd (cmd_tree=, param=) at 
drouting.c:813
#10 0x7fcf68baf504 in run_mi_cmd (param=0x25e5aa0, f=, 
t=0x0, cmd=) at ../../mi/mi.h:109
#11 mi_fifo_server (fifo_stream=0x25e8e30) at fifo_fnc.c:490
#12 0x7fcf68bb0bef in fifo_process (rank=) at mi_fifo.c:213
#13 0x004b9de5 in start_module_procs () at sr_module.c:586
#14 0x0041475a in main_loop () at main.c:840
#15 main (argc=, argv=) at main.c:1598
(gdb) bt full
#0  fm_remove_free (n=0x7fcf529e6840, qm=0x7fcf44a9b000) at mem/f_malloc.c:175
pf = 0x7fcf44a9b640
hash = 2096
#1  fm_malloc (qm=0x7fcf44a9b000, size=704) at mem/f_malloc.c:383
frag = 0x7fcf529e6840
n = 
hash = 
__FUNCTION__ = "fm_malloc"
#2  0x7fcf685860a5 in shm_malloc_unsafe (size=) at 
../../mem/shm_mem.h:248
p = 
#3  shm_malloc (size=) at ../../mem/shm_mem.h:258
p = 0x2c0
#4  add_rt_info (pn=0x7fcf4e465d18, r=0x7fcf4f6bc9f8, rgid=127) at routing.c:367
trg = 
rtl_wrp = 
rtlw = 0x0
i = 
__FUNCTION__ = "add_rt_info"
#5  0x7fcf68581f3c in add_prefix (ptree=0x7fcf4e465cf8, prefix=, r=0x7fcf4f6bc9f8, rg=127) at prefix_tree.c:260
tmp = 
res = 0
__FUNCTION__ = "add_prefix"
#6  0x7fcf685715a4 in add_rule (rule=, prefix=, grplst=, rdata=) at dr_load.c:188
tmp = 
t = 
ep = 0x2aa79cc ",130,139,142,145,148"
n = 
#7  dr_load_routing_info (dr_dbf=0x7fcf687941a0, db_hdl=0x7fcf6ac053b0, 
drd_table=, drc_table=, drr_table=0x7fcf687939f0) 
at dr_load.c:512
int_vals = {1522874, , 0, }
str_vals = {0x2aa7948 
"19,20,23,25,27,31,33,35,37,29,40,44,47,50,12,53,56,59,62,65,66,69,72,75,78,81,84,97,88,91,94,100,103,106,109,112,115,118,121,124,127,130,139,142,145,148",
  , , 0x2aa79ed "13", , 
}
tmp = {s = 0x2aa79e1 "81061881", len = 8}
columns = {0x7fcf68793400, 0x7fcf68793410, 0x7fcf68793420, 
0x7fcf68793430, 0x7fcf68793440, 0x7fcf68793450, 0x7fcf68793460, 0x7fcf68793390}
res = 0x7fcf6c24dd48
row = 
ri = 0x7fcf4f6bc9f8
rdata = 0x7fcf4955e898
time_rec = 
i = 4018
n = 29697
no_rows = 
__FUNCTION__ = "dr_load_routing_info"
#8  0x7fcf685811c0 in dr_reload_data () at drouting.c:425
new_data = 
old_data = 
#9  dr_reload_cmd (cmd_tree=, param=) at 
drouting.c:813
__FUNCTION__ = "dr_reload_cmd"
#10 0x7fcf68baf504 in run_mi_cmd (param=0x25e5aa0, f=, 
t=0x0, cmd=) at ../../mi/mi.h:109
ret = 
#11 mi_fifo_server (fifo_stream=0x25e8e30) at fifo_fnc.c:490
mi_cmd = 
mi_rpl = 0x7fcf6abe03c8
hdl = 0x0
line_len = 1
file_sep = 
---Type  to continue, or q  to quit---

[OpenSIPS-Users] Which TLS methoud should I use?

[Carlos Oliva]

Hi list,

We are implementing TLS and have a doubt about wich TLS method should we use.

Now we are using SSLv23 as suggested in
http://www.opensips.org/html/docs/modules/devel/proto_tls.html#id293824
but we think is a very old method (and not RFC3261 conformant) and
TLS1.2 should be modern and most secure. An aditional problem is that
we are using VoiPMonitor sniffer and in the current version they don't
support SSLv2 because is old and insecure, and I think are good
reasons to not support it.

We are using different clients hardphones from Linksys/Cisco, Snom,
Grandstream, Polycom and Yealink and softphones from Zoiper (versions
2 and 3) Jitsi, Blink, Cloud Softphone (Android), CSipSimple (Android)
and Bria (iPhone) as you see lots of different clients.

I'm asking about opinions. Based in your experience what method do you
think should be better?


thanks and Regards,

Carlos Oliva

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Re: [OpenSIPS-Users] Problem: INVITE port mismatch

[Albert Vallespi Ofertas]

Hello again,

I have discovered that we can use the command t_relay with parameters, and
we have started using with this option.

t_relay("tcp:REMOTE_IP:5060")

The problem persists also with this command.
Opensips starts working well, but after some minutes it degrades and start
sending the INVITE with TCP wrong ports. (I supose opensips is using
destination port as source and viceversa)

With this situation we have seen some strange logs that could be related
with the issue.
Please see these logs below.

Could anybody help us to solve this issue?

Jul 15 04:55:12  /usr/sbin/opensips[28821]: WARNING:core:tcpconn_add_alias:
possible port hijack attempt
Jul 15 04:55:12  /usr/sbin/opensips[28821]: WARNING:core:tcpconn_add_alias:
alias already present and points to another connection (6 : 5060 and 38 :
5060)
Jul 15 04:55:12  /usr/sbin/opensips[28821]: WARNING:core:receive_msg: tcp
alias failed
Jul 15 04:55:12  /usr/sbin/opensips[28821]: WARNING:core:tcpconn_add_alias:
possible port hijack attempt
Jul 15 04:55:12  /usr/sbin/opensips[28821]: WARNING:core:tcpconn_add_alias:
alias already present and points to another connection (6 : 5060 and 38 :
5060)
Jul 15 04:55:12  /usr/sbin/opensips[28821]: WARNING:core:receive_msg: tcp
alias failed


2015-07-14 16:52 GMT+02:00 Albert Vallespi Ofertas <
avallespi.ofer...@gmail.com>:

> Hello again,
>
> I have also configures $du and $dp to the correct one.
>
> I am logging the $du before the t_relay and the log shows the correct port
> in the $du, but if I take traces I observe that the ports are not correct.
> INVITE is sent from opensips tcp port 5060 to a 64XXX port of the
> destination. It should be viceversa.
>
> Normally it takes some minutes working well with the correct ports, but
> after some minutes it degrades and opensips starts to send the INVITES with
> the TCP ports incorrects.
>
> Could anybody please help me?
>
> I have seen this thread, but I'm not sure if could be some related.
> https://github.com/OpenSIPS/opensips/issues/420
>
> Best regards
>
> 2015-07-13 13:54 GMT+02:00 Podrigal, Aron :
>
>> Make sure you have the correct value set for $du /$dp most likely you
>> have some function overriding that.
>> On Jul 13, 2015 7:47 AM, "Albert Vallespi Ofertas" <
>> avallespi.ofer...@gmail.com> wrote:
>>
>>> Hello Aron,
>>>
>>> Yes, we have tested also with force_send_socket("tcp:eth2_IP:5060") and
>>> it happens the same.
>>> Do you think I should use the command this way?
>>>
>>> After a volume of correct calls, opensips degrades and start sending TCP
>>> INVITES to a different port that the indicated in the r-uri.
>>> We have test in so many ways with some of these instructions, but always
>>> happened the same.
>>>
>>> Thanks a lot for the answer
>>>
>>>
>>>
>>> 2015-07-13 13:39 GMT+02:00 Podrigal, Aron :
>>>
 Have you tried using force_send_socket()?
 On Jul 13, 2015 5:35 AM, "Albert Vallespi Ofertas" <
 avallespi.ofer...@gmail.com> wrote:

> Hello again,
>
> We have re-checked the case, and I think there is no relation with
> "user=phone" parameter in the r-uri.
>
> We have verified that in our scenario, forwarding from UDP to TCP
> opensips sends INVITE in the TCP side to a wrong port.
> I think it changes origin and destination ports. This seems a opensips
> bug.
>
> - Network1 is using UDP, and opensips listens at port UDP:eth1_IP:5060
> - Network2 is using TCP, and opensips listens at port TCP:eth2_IP:5060
>
> 2015-07-10 19:46 GMT+02:00 Albert Ofertas  >:
>
>> Hi to all,
>>
>> We have a new opensips r.2.1 in a production environment where it is
>> configured between two different networks.
>> We are using always private networks, therefore there is not any NAT
>> or similar.
>>
>> - Network1 is using UDP, and opensips listens at port UDP:eth1_IP:5060
>> - Network2 is using TCP, and opensips listens at port TCP:eth2_IP:5060
>>
>> We have observed that when we forward an INVITE from Network1 to
>> Network2 sometimes there is a port mismatch in the outgoing INVITE.
>>
>> This INVITE should go from opensipts TCP:eth2_IP:6 (for example
>> 63445) to the remote peer that uses TCP:REMOTE_IP:5060.
>> What we have observed is that this INVITE many times and without a
>> logical explanation mixes the ports.
>> I mean, the r-uri is correct (example:  
>> XX@REMOTE_IP:5060;transport=tcp;user=phone),
>> but the message is sent via TCP with the ports crossed.
>> The wrong INVITE is going from  TCP:eth2_IP:5060 to
>> TCP:REMOTE_IP:6.
>>
>> We have observed that this incorrect behaviour in opensips is
>> happening when we are using the parameter "user=phone" in the request 
>> uri.
>>
>> We have tested some minutes without the "user=phone" and we have not
>> observed the port mismatch then.
>>
>> We must use the parameter "user=phone". This is a mandatory 

[OpenSIPS-Users] Load static registrations on startup

[Edwin]

On startup (opensips 1.11) we like to load the static registrations from the
location database, after this we like to have registrations just in memory
for fast performance with usrloc dbmode 0.

Is it posible to start opensips with dbmode 2, load the static registrations
in memory (maybe in startup_route) and than change to dbmode 0?

Maybe I should think different? Don't hesitate to share your thoughts, all
suggestions are welcome!



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[OpenSIPS-Users] How many users can register with the same username simultaneously?

[microx]

Hi,

I'm testing the SIP forking functionality. I set up a number of SIP clients
using the same username to do registration. From the registrar module, I
think that the number of concurrent clients with the same username is
limited only by the parameter ``max_contact''. However, when I set
modparam("registrar", "max_contacts", 30), the OpenSIPS instance stops when
the number of SIP clients using the same username is 20. The error log is
shown below.
/usr/sbin/opensips[15878]: CRITICAL:core:receive_fd: EOF on 12
/usr/sbin/opensips[15754]: ERROR:nat_traversal:save_keepalive_state: failed
to open keepalive state file for writing: Permission denied
/usr/sbin/opensips[15754]: CRITICAL:core:sig_alarm_abort: BUG - shutdown
timeout triggered, dying...

I use OpenSIPS 1.9.1. If any other configuration is required, please kindly
let me know. Thanks.

Best regards,
Chen-Che



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