Re: [OpenSIPS-Users] Registration permissions per username

[Pavel Eremin]

Hello why not regexp not working for you. I give you my example that works
fine.(but with 1.11)
to reload i use command opensipsctl fifo regex_reload

[config]
 REGEX
loadmodule "regex.so"
modparam("regex", "file", "/usr/out_isp/etc/opensips/regex_groups")

...
if (pcre_match_group("$fU", "0")) {
}
...

[content regex_group file]
[0]

^1000
^1001
^1003
^50065
^anyname_from_start
anyname_in_anywhere of $fU




вт, 2 июл. 2019 г. в 10:33, Alexey Kazantsev via Users <
users@lists.opensips.org>:

> Hi list,
>
> is it possible to filter REGISTER requests with permissions.so [1] module,
> based on username?
>
> It's written " Main purpose of the function is to prevent registration of
> "prohibited" IP addresses. " When speaking about IP filtering,
> I'd rather use check_address or check_source_address functions.
>
> But now I'd like to filter by userame, because users may register from
> random
> addresses.
>
> I tried to create pairs of regexps in register.allow and register.deny
> files,
> but no success. Maybe I've done something wrong.
>
>
> [1]
> https://opensips.org/html/docs/modules/3.0.x/permissions.html#sec-registration-permissions
>
> ---
> BR, Alexey
> http://alexeyka.zantsev.com/
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Registration permissions per username

[Alexey Kazantsev via Users]

Hi list,

is it possible to filter REGISTER requests with permissions.so [1] module,
based on username?

It's written " Main purpose of the function is to prevent registration of
"prohibited" IP addresses. " When speaking about IP filtering,
I'd rather use check_address or check_source_address functions.

But now I'd like to filter by userame, because users may register from random
addresses.

I tried to create pairs of regexps in register.allow and register.deny files,
but no success. Maybe I've done something wrong.


[1]  
https://opensips.org/html/docs/modules/3.0.x/permissions.html#sec-registration-permissions

---
BR, Alexey
http://alexeyka.zantsev.com/
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPs 3.0 Crash issue

[Dipteshkumar Patel]

Hello Virendra,

It was due to lack of shared memory. It gives you signal no. 6 which shows
that your heap section is going to overflow.
There will be two possible things
1. you are not providing sufficient shared memory according to your
traffic.
2. Possible memory leaks in opensips.

how many shared memory are you providing at run-time using -m options ? if
it is enough then enable memory debug and check memory leaks.
Please follow the below document for enable the memory debug logs.

https://www.opensips.org/Documentation/TroubleShooting-OutOfMem

*Diptesh Patel*
Jr. Software Developer
Ecosmob Technologies Ltd
Ahmedabad
Mo:*+919898962659*


On Mon, Jul 1, 2019 at 9:51 PM Virendra Bhati 
wrote:

> Dear Team,
>
> We have setup Opensips 3.0 stable version for stress testing. We are
> invoking calls from SIPp to Opensips. We noticed after 600 CC with 10 CPS
> Opensips crash. I have attached files for more details of DUMP.
>
> Below is the call flow
>
> (Leg A) SIPp (UAC)---> OpenSIPs(5070) ---> Freeswitch(5060)
> (Leg B) Freeswitch(5060) --> OpenSIPs(5070) ---> SIPp (UAS)
>
> OpenSIPs and Freeswitch is running on same machine with different ports.
>
> As per our initial understanding it seems there is an issue with Memory.
> Please help me on it,
> --
> Regards
> Virendra Bhati
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

-- 
*Disclaimer*
In addition to generic Disclaimer which you have agreed on our 
website, any views or opinions presented in this email are solely those of 
the originator and do not necessarily represent those of the Company or its 
sister concerns. Any liability (in negligence, contract or otherwise) 
arising from any third party taking any action, or refraining from taking 
any action on the basis of any of the information contained in this email 
is hereby excluded.



*Confidentiality*
This communication (including any 
attachment/s) is intended only for the use of the addressee(s) and contains 
information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, 
dissemination, distribution, or copying of this communication is 
prohibited. Please inform originator if you have received it in error.



*Caution for viruses, malware etc.*
This communication, including any 
attachments, may not be free of viruses, trojans, similar or new 
contaminants/malware, interceptions or interference, and may not be 
compatible with your systems. You shall carry out virus/malware scanning on 
your own before opening any attachment to this e-mail. The sender of this 
e-mail and Company including its sister concerns shall not be liable for 
any damage that may incur to you as a result of viruses, incompleteness of 
this message, a delay in receipt of this message or any other computer 
problems. 
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] problem in dialplan

[Richard Revels]

Try enclosing the 1 in quotes

dp_translate("1","$ruri.user/$var(rU)")



[image: BandwidthMaroon.png]



Richard Revels  •  System Architect II

900 Main Campus Drive, Suite 100, Raleigh, NC 27606



m: 919-578-3421  •  o: 919-727-4614

e: rrev...@bandwidth.com


On Mon, Jul 1, 2019 at 5:28 AM johan de clercq  wrote:

> Hi,
>
> Using latest opensips 3.0,  there seems to be a problem in dialplan
>
>
>
>   1 09:23:25 hendrix /data/opensips/sbin/opensips[4197]:
> ERROR:core:get_cmd_fixups: Variable in param [1] is not an integer
>
> Jul  1 09:23:25 hendrix /data/opensips/sbin/opensips[4197]:
> ERROR:core:do_action: Failed to get fixups for command 
>
>
>
> This the code :
>
> if (dp_translate(1,"$ruri.user/$var(rU)"))
>
> {
>
> xlog("callid=$ci: Route[normalizeforinbound]: we dropped 0,00,+
> from $rU, result is var(rU) $var(rU)");
>
> }
>
>
>
> [image: cid:F3100D46-F00D-4610-87ED-3E91DA790A82]
>
> Johan De Clercq, Managing Director
> Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke
>
> Tel +3256980990 – GSM +32478720104
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] I need some help in websocket connection error .

[Sasmita Panda]

I am sure the client supports TLSv1.2 version . That was confirmed .

I am not sure about the ciphers . I have to ask them .

modparam("tls_mgm", "ciphers_list", "AES256-GCM-SHA384,AES256-
SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,CAMELLIA128-SHA,RC4-SHA")

Is this the list of whitelisted ciphers ?

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*


On Fri, Jun 28, 2019 at 1:33 PM Răzvan Crainea  wrote:

> Hi, Sasmita!
>
> I see that you require TLSv1.2 authentication method - are you sure your
> clients do support this version? A similar problem can be with the
> ciphers, are you sure your clients support the whitelisted ciphers? As
> you do not enforce anything, this might be true, but this is something
> you should double-check. Besides that, I don't have any other ideas.
>
> Best regards,
> Răzvan
>
> On 6/27/19 9:02 AM, Sasmita Panda wrote:
> > Hi,
> >
> > SSL miss configuration in client side or in opensips side . I think I
> > have done the configuration right .
> >
> > listen=wss:192.168.143.20:443 
> > loadmodule "tls_mgm.so"
> > modparam("tls_mgm", "tls_method", "tlsv1_2")
> > modparam("tls_mgm", "verify_cert", "0")
> > modparam("tls_mgm", "require_cert", "0")
> > modparam("tls_mgm", "certificate",
> > "/usr/local/etc/opensips/tls/3ccloudwebrtc2019.crt")
> > modparam("tls_mgm", "private_key",
> > "/usr/local/etc/opensips/tls/3ccloud.key")
> > modparam("tls_mgm", "ca_list",
> > "/usr/local/etc/opensips/tls/rootCA/cacert.pem")
> > loadmodule "proto_wss.so"
> > modparam("proto_wss", "wss_port", 443)
> >
> > This is for wss . Is there anything I am missing in configuration ?
> >
> >
> > */Thanks & Regards/*
> > /Sasmita Panda/
> > /Senior Network Testing and Software Engineer/
> > /3CLogic , ph:07827611765/
> >
> >
> > On Wed, Jun 26, 2019 at 8:10 PM Răzvan Crainea  > > wrote:
> >
> > TBH, all I can see in the logs you sent is that a connection was
> > terminated (without even being started), and a connection that was
> > started, but closed by the client. So in order to understand what's
> > happening, you need to understand why the client is closing the
> > connection. Check logs, documentation, anything, but this doesn't
> seem
> > to be related to OpenSIPS, it looks like some SSL misconfiguration.
> >
> > Best regards,
> > Răzvan
> >
> > On 6/26/19 4:24 PM, Sasmita Panda wrote:
> >  > Is there any update on this issue . How I can solve this error
> > message
> >  > from my opensips logs .
> >  >
> >  >
> >  > */Thanks & Regards/*
> >  > /Sasmita Panda/
> >  > /Senior Network Testing and Software Engineer/
> >  > /3CLogic , ph:07827611765/
> >  >
> >  >
> >  > On Tue, Jun 25, 2019 at 3:48 PM Sasmita Panda  > 
> >  > >> wrote:
> >  >
> >  > I have tried to take ssldump in the webrtc server in run time
> .
> >  >
> >  > New TCP connection #19: 192.168.1.y(48530) <->
> 192.168.0.x(443)
> >  > 190.0011 (0.0011)  C>S  TCP FIN
> >  > 190.0013 (0.0001)  S>C  TCP FIN
> >  >
> >  > New TCP connection #20: 192.168.0.y(52975) <->
> 192.168.0.x(443)
> >  > 20 1  0.0006 (0.0006)  C>S  Handshake  ClientHello
> >  > 20 2  0.0008 (0.0002)  S>C  Handshake  ServerHello
> >  > 20 3  0.0008 (0.)  S>C  Handshake  Certificate
> >  > 20 4  0.0008 (0.)  S>C  Handshake  ServerHelloDone
> >  > 20 5  0.0020 (0.0011)  C>S  Handshake  ClientKeyExchange
> >  > 20 6  0.0020 (0.)  C>S  ChangeCipherSpec
> >  > 20 7  0.0020 (0.)  C>S  Handshake
> >  > 20 8  0.0036 (0.0015)  S>C  Handshake20 9  0.0036 (0.)
>  S>C
> >  >   ChangeCipherSpec
> >  > 20 10 0.0036 (0.)  S>C  Handshake
> >  > 20 11 0.0042 (0.0006)  C>S  Alert
> >  > 200.0042 (0.)  C>S  TCP FIN
> >  > 200.0043 (0.)  S>C  TCP FIN
> >  >
> >  > The portion I marked in red whenever appear there is error in
> >  > opensips logs  . For below portion the connection was
> accepted  .
> >  >
> >  > I am not even getting any error  in my browser side .  How I
> will
> >  > debug this ? please help .
> >  >
> >  > */Thanks & Regards/*
> >  > /Sasmita Panda/
> >  > /Senior Network Testing and Software Engineer/
> >  > /3CLogic , ph:07827611765/
> >  >
> >  >
> >  > On Fri, Jun 14, 2019 at 2:51 PM Callum Guy
> > mailto:callum@x-on.co.uk>
> >  >  > >> wrote:
> >  >
> >  > You might find that a tcpdump is the only way to get to
> grips
> >  > with the underlying issue.
> >  >
> > 

[OpenSIPS-Users] OpenSIPs with mutual TLS and client CA lists

[Phil Whitener]

I have looked into using OpenSIPS with optional mutual TLS.  In short, using 
verify_cert=1 & require_cert=0.  In this case, the OpenSIPs acting as a server 
sends the TLS "Certificate Request" during the handshake and based on the 
response the OpenSIPs server decides whether to continue (as either server-only 
TLS or mutual TLS) or terminate the connection.  I have experienced more 
failures than expected as some remote endpoints are attempting to satisfy the 
certificate request by sending any potential certificate that meets the 
requested criteria.

During the "Certificate Request" there is an optional parameter allowing the 
trusted certificate authority distinguished name to be provided in the request. 
 This is defined in OpenSSL's SSL_CTX_set_client_CA_list.  Without this 
directive defined the remote client may choose to send a client certificate 
that meets the only defined parameter (Certificate types); however, in many 
cases OpenSIPs may reject the client selected certificate.  It does not appear 
that OpenSIPs controls this optional parameter.

I may have missed this definition in OpenSIPs.  This may be a potential feature 
request.  If it has been omitted, I feel that when OpenSIPs is acting as a TLS 
server, the existing parameter CA_LIST could be defined in the server domain to 
provide a set of trusted certificate authorities to pass along as the 
Certificate Request distinguished name.  In this case the remote client peer 
that is not able to satisfy the scoped Certificate Request can then choose to 
proceed without mutual authentication and continue the handshake without 
offering a client certificate.

RFC5246 7.4.6 Client Certificate 
https://tools.ietf.org/html/rfc5246#section-7.4.6

TLSv1 Record Layer: Handshake Protocol: Multiple Handshake Messages
Content Type: Handshake (22)
Version: TLS 1.0 (0x0301)
Length: 14
Handshake Protocol: Certificate Request
Handshake Type: Certificate Request (13)
Length: 6
Certificate types count: 3
Certificate types (3 types)
Certificate type: RSA Sign (1)
Certificate type: DSS Sign (2)
Certificate type: ECDSA Sign (64)
Distinguished Names Length: 0
Handshake Protocol: Server Hello Done
Handshake Type: Server Hello Done (14)
Length: 0

OpenSSL SSL_CTX_set_client_CA_list

https://www.openssl.org/docs/man1.0.2/man3/SSL_CTX_set_client_CA_list.html


Thank you for your review,

Phil Whitener
phil.white...@genesys.com

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

[OpenSIPS-Users] OpenSIPs 3.0 Crash issue

[Virendra Bhati]

Dear Team,

We have setup Opensips 3.0 stable version for stress testing. We are
invoking calls from SIPp to Opensips. We noticed after 600 CC with 10 CPS
Opensips crash. I have attached files for more details of DUMP.

Below is the call flow

(Leg A) SIPp (UAC)---> OpenSIPs(5070) ---> Freeswitch(5060)
(Leg B) Freeswitch(5060) --> OpenSIPs(5070) ---> SIPp (UAS)

OpenSIPs and Freeswitch is running on same machine with different ports.

As per our initial understanding it seems there is an issue with Memory.
Please help me on it,
--
Regards
Virendra Bhati
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
Inside INVITE Call si=192.168.1.25 , sp=5080 , oP=udp ,avp=2
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
call FS
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: NOW 
rd=192.168.1.98 , rp=5060 , rP=udp, ru=sip:1046@192.168.1.98:5060 
du=sip:192.168.1.98:5060
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: new 
branch at sip:1046@192.168.1.98:5060
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
inside => Handle_nat Reply Status:200 and User Agent: 
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24815]: 
inside => Handle_nat Reply Status:200 and User Agent: 
FreeSWITCH-mod_sofia/1.8.5~64bit
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
Called [ACK] with [-2]
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
Called [ACK] with [-2]
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
inside => Handle_nat Reply Status:200 and User Agent: 
FreeSWITCH-mod_sofia/1.8.5~64bit
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:core:fm_malloc: not enough free shm memory (357200 bytes left, need 
6712), please increase the "-m" command line parameter!
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
INFO:core:fm_malloc: attempting defragmentation...
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
INFO:core:fm_malloc: unable to alloc a big enough fragment!
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:tm:t_uac: short of cell shmem
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:pua:send_publish_int: failed to send PUBLISH
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:pua_dialoginfo:dialog_publish: sending publish failed for pres_uri 
[sip:1740@192.168.1.98:5061] to server [sip:192.168.1.25:5070]
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:core:fm_malloc: not enough free shm memory (356864 bytes left, need 
6712), please increase the "-m" command line parameter!
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
INFO:core:fm_malloc: attempting defragmentation...
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
INFO:core:fm_malloc: unable to alloc a big enough fragment!
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:tm:t_uac: short of cell shmem
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:pua:send_publish_int: failed to send PUBLISH
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
ERROR:pua_dialoginfo:dialog_publish: sending publish failed for pres_uri 
[sip:1240@192.168.1.25:5070] to server [sip:192.168.1.25:5070]
Jun 26 14:31:13 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24812]: 
Called [ACK] with [-2]
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
inside => Handle_nat Reply Status:500 and User Agent: 
FreeSWITCH-mod_sofia/1.8.5~64bit
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
Called [BYE] with [-1]
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
Called [BYE] with [-1]
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24815]: 
inside => Handle_nat Reply Status:500 and User Agent: 
FreeSWITCH-mod_sofia/1.8.5~64bit
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24815]: 
Called [BYE] with [-1]
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24811]: 
inside => Handle_nat Reply Status:200 and User Agent: 
FreeSWITCH-mod_sofia/1.8.5~64bit
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
Called [BYE] with [-1]
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
ERROR:core:fm_malloc: not enough free shm memory (349984 bytes left, need 
6712), please increase the "-m" command line parameter!
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 /usr/local/sbin/opensips[24810]: 
INFO:core:fm_malloc: attempting defragmentation...
Jun 26 14:31:14 cloudconnect-PowerEdge-T30 

[OpenSIPS-Users] testing with opensips-cli

[johan de clercq]

Dears, 

Following up my dialplan troubles, I decided to test a bit with
opensips-cli. 

I created below config file : 

[localhost]

prompt_name: opensips-cli_localhost

prompt_intro: OpenSIPS CLI for localhost instance

communication_type: http

url: http://127.0.0.1:/json

 

and saved it as /etc/opensips-cli.cfg 

 

When I rung opensips-cli, I have below output 

root@hendrix:/# opensips-cli 

Welcome to OpenSIPS Command Line Interface!

(opensips-cli):

 

The prompt is still opensips-cli, so it seems to me that it doesn't find the
config file. 

 

However : 

root@hendrix:/# cat /etc/opensips-cli.cfg 

[localhost]

prompt_name: opensips-cli_localhost

prompt_intro: OpenSIPS CLI for localhost instance

communication_type: http

url: http://127.0.0.1:/json

 

The documentation states this as one of the default locations where it will
search for opensips-cli.cfg. 

Please advice on how to fix this. 

 

Best regards, 

 

 

 



Johan De Clercq, Managing Director
Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke

Tel +3256980990 - GSM +32478720104

 

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

[OpenSIPS-Users] URL encoding for rest_client

[Callum Guy]

Hi All,

My config integrates with an external routing API such that the call ID,
source and URI data (etc) are all provided to a service in URL parameters
via rest_get. The service returns some routing information such as revised
target URI and other options.

To improve this service and protect against injection attacks I wish to URL
encode the individual parameters in accordance with
https://www.ietf.org/rfc/rfc3986.txt

I'm sure this question will have been asked before but I can't find a
relevant discussion. The core OpenSIPs transformations
(i.e. {s.escape.user}) performs SIP specific replacements which is not
suitable. Specifically I'd like to see & and + being replaced.

My current approach will be to perform a series of specific replacements
using transformation {re.subst,reg_exp} however this seems overkill for
this purpose as multiple replacements will be required for each parameter.

So, before I get too far into this is anyone able to offer an alternative
approach? We have considered using JSON POST requests to circumvent the
issue however we'd like to keep the changes to the OpenSIPs side if
possible. I suppose it would be convenient to have an exported "encode()"
function in the rest_client module however this might be a longer term
option.

Many thanks,

Callum

-- 


 


*0333 
332   |  www.x-on.co.uk   |   ** 
    
   *


X-on
is a trading name of Storacall 
Technology Ltd a limited company registered in
England and Wales.


Registered Office : Avaland House, 110 London Road, Apsley, Hemel 
Hempstead,
Herts, HP3 9SD. Company Registration No. 2578478.

The 
information in this e-mail is confidential and for use by the addressee(s)

only. If you are not the intended recipient, please notify X-on immediately 
on +44(0)333 332  and delete the
message from your computer. If you are 
not a named addressee you must not use,
disclose, disseminate, distribute, 
copy, print or reply to this email. Views
or opinions expressed by an 
individual
within this email may not necessarily
reflect the views of X-on 
or its associated companies. Although X-on routinely
screens for viruses, 
addressees should scan this email and any attachments
for
viruses. X-on 
makes no representation or warranty as to the absence of viruses
in this 
email or any attachments.










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

[OpenSIPS-Users] Opensips 3.0 Crash issue

[Virendra Bhati]

Dear Team,

We have setup Opensips 3.0 stable version for stress testing. We are
invoking calls from SIPp to Opensips. We noticed after 600 CC with 10 CPS
Opensips crash. I have attached link for more details of DUMP.

Below is the call flow

(Leg A) SIPp (UAC)---> OpenSIPs(5070) ---> Freeswitch(5060)
(Leg B) Freeswitch(5060) --> OpenSIPs(5070) ---> SIPp (UAS)

OpenSIPs and Freeswitch is running on same machine with different ports.

As per our initial understanding it seems there is an issue with Memory.
Please help me on it.

https://pastebin.com/k0mtjjPh
--
Regards
Virendra Bhati
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] problem in dialplan

[johan de clercq]

In addition, when I use $var(i) I have more output 

 

Jul  1 09:32:39 hendrix /data/opensips/sbin/opensips[4593]:
DBG:dialplan:dp_translate_f: dpid is 1 partition is default

Jul  1 09:32:39 hendrix /data/opensips/sbin/opensips[4593]:
DBG:dialplan:dp_translate_f: input is +32478720101/

Jul  1 09:32:39 hendrix /data/opensips/sbin/opensips[4593]:
DBG:dialplan:dp_translate_f: no information available for dpid 1

Jul  1 09:32:39 hendrix /data/opensips/sbin/opensips[4593]:
callid=bmWolfo6-vYkkNdSG0sYBA..: Route[normalizeforinbound]: dp_translate
failed!, we drop the call

 

When I look into the db : 

 

MySQL [test]> select * from dialplan;

++--++--+--+-++-
-+-+--+---+

| id | dpid | pr | match_op | match_exp| match_flags | subst_exp  |
repl_exp | timerec | disabled | attrs |

++--++--+--+-++-
-+-+--+---+

|  4 |1 |  1 |1 | ^(00|\+|0).* |   0 | ^(00|\+|0)(.*) |
\2   | NULL|0 |   |

|  5 |   33 |  1 |1 | ^(0).*   |   1 | ^(0)(.*)   |
33\2 | NULL|0 |   |

|  6 |   33 |  1 |1 | ^(00|\+).*   |   1 | ^(00|\+)(.*)   |
\2  | NULL|0 |   |

| 11 |   32 |  1 |1 | ^(00|\+).*   |   1 | ^(00|\+)(.*)   |
\2   | NULL|0 |   |

| 13 |   32 |  1 |1 | ^(0).* 1 |   1 | ^(0)(.*)   |
32\2 | NULL|0 |   |

++--++--+--+-++-
-+-+--+---+

 

So there are 2 problems : 

*   Dp_translate when the first parameter is an integer the function
does not terminate normally. 
*   Please explain the error above. 

BR, 

 

 

From: johan de clercq  
Sent: Monday, July 1, 2019 11:28 AM
To: 'OpenSIPS users mailling list' 
Subject: problem in dialplan 

 

Hi, 

Using latest opensips 3.0,  there seems to be a problem in dialplan 

 

  1 09:23:25 hendrix /data/opensips/sbin/opensips[4197]:
ERROR:core:get_cmd_fixups: Variable in param [1] is not an integer

Jul  1 09:23:25 hendrix /data/opensips/sbin/opensips[4197]:
ERROR:core:do_action: Failed to get fixups for command 

 

This the code : 

if (dp_translate(1,"$ruri.user/$var(rU)"))

{

xlog("callid=$ci: Route[normalizeforinbound]: we dropped 0,00,+ from
$rU, result is var(rU) $var(rU)");

}

 



Johan De Clercq, Managing Director
Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke

Tel +3256980990 - GSM +32478720104

 

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

[OpenSIPS-Users] problem in dialplan

[johan de clercq]

Hi, 

Using latest opensips 3.0,  there seems to be a problem in dialplan 

 

  1 09:23:25 hendrix /data/opensips/sbin/opensips[4197]:
ERROR:core:get_cmd_fixups: Variable in param [1] is not an integer

Jul  1 09:23:25 hendrix /data/opensips/sbin/opensips[4197]:
ERROR:core:do_action: Failed to get fixups for command 

 

This the code : 

if (dp_translate(1,"$ruri.user/$var(rU)"))

{

xlog("callid=$ci: Route[normalizeforinbound]: we dropped 0,00,+ from
$rU, result is var(rU) $var(rU)");

}

 



Johan De Clercq, Managing Director
Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke

Tel +3256980990 - GSM +32478720104

 

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

[OpenSIPS-Users] Load balancer stops to count load over time

[Igor Pavlov]

Hi all,

I encountered a problem with load balancer module, it stops to count 
load over time, 5-6 hours or day may pass after restart.
|opensipsctl fifo lb_list| always shows |load=0|, but at the same time 
balancer do his work correctly - it balance calls.


Here is debug log:

|Jun 30 10:57:34 DP-SIP-CORE-1 OpenSIP-CORE-1[9819]: 
DBG:load_balancer:w_lb_start: pick a random destination among all 
selected dsts with equal load Jun 30 10:57:34 DP-SIP-CORE-1 
OpenSIP-CORE-1[9819]: DBG:load_balancer:lb_route: initial call of LB - 
found requested 1/1 resource [calls] Jun 30 10:57:34 DP-SIP-CORE-1 
OpenSIP-CORE-1[9819]: DBG:load_balancer:lb_route: initial call of LB - 
destination 2  selected for LB set with free=1000 Jun 
30 10:57:34 DP-SIP-CORE-1 OpenSIP-CORE-1[9819]: 
DBG:load_balancer:lb_route: initial call of LB - destination 3 
 selected for LB set with free=1000 Jun 30 10:57:34 
DP-SIP-CORE-1 OpenSIP-CORE-1[9819]: DBG:load_balancer:lb_route: initial 
call of LB - destination 4  selected for LB set with 
free=1000 Jun 30 10:57:34 DP-SIP-CORE-1 OpenSIP-CORE-1[9819]: 
DBG:load_balancer:lb_route: initial call of LB - winning destination 3 
 selected for LB set with free=1000 ||It always shows 'free=1000' I have opened issue 
https://github.com/OpenSIPS/opensips/issues/1748 |


--
--
Kind regards,
Igor Pavlov

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

[OpenSIPS-Users] opensips-cp 30

[johan de clercq]

Hello, 

 

Is there already more news on the new opensips-cp for 3.0 ?

 

BR, 

 



Johan De Clercq, Managing Director
Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke

Tel +3256980990 - GSM +32478720104

 

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

Re: [OpenSIPS-Users] rtpengine_offer error

[Olle Frimanson]

We solved this by adding info in Register

Another option could be if you know if avp/ savp is tied to transport protocol 

So TLS implies savp 

Br Olle

Skickat från min iPhone

> 1 juli 2019 kl. 09:11 skrev Dragomir Haralambiev :
> 
> 
> Hi Alexej,
> 
> Yes. You are right if I know if the client has encryption (RTP / SAVP) or 
> encryption (RTP / AVP) turned off.
> Example:
> MicroSIP1 --> Opensips 
> -->MicroSIP2 
> Media encryption disableRtpEngine 
> Media encryption Mandatory ((RTP / SAVP)
> 
> OpenSips receive INVITE from MicroSIP1 - I can get from body  RTP / SAVP.
> 
> OpenSisp has no information whether MicroSIP2 it supports or not (RTP/SAVP)
> 1. If I setup:
> tpengine_offer("RTP/AVP replace-session-connection replace-origin 
> ICE=remove");  
> MicroSip2 rejected call because not supported  RTP/AVP
> 
> 2. If I setup:
> tpengine_offer("RTP/SAVP replace-session-connection replace-origin 
> ICE=remove");  
> MicroSip1 rejected call because not supported  RTP/SAVP
> 
> Best regards,
> Dragomir
> 
> На пн, 1.07.2019 г. в 8:58 ч. Alexey Vasilyev  
> написа:
>> Hi Dragomir.
>> 
>> What do you expect RTPEngine to do? You tell rtpengine_offer("RTP/AVP
>> RTP/SAVP replace-session-connection replace-origin ICE=remove"); So you tell
>> it, that you want to get as result unencrypted body (RTP/AVP) AND encrypted
>> body (RTP/SAVP). But it can be either encrypted or unencrypted.
>> 
>> You should choose, what you want. Use only one profile as parameter.
>> 
>> 
>> 
>> 
>> -
>> ---
>> Alexey Vasilyev
>> --
>> Sent from: 
>> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>> 
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] rtpengine_offer error

[Dragomir Haralambiev]

Hi Alexej,

Yes. You are right if I know if the client has encryption (RTP / SAVP) or
encryption (RTP / AVP) turned off.
Example:
MicroSIP1 --> Opensips
-->MicroSIP2
*Media encryption* disableRtpEngine
 *Media encryption* Mandatory ((RTP / SAVP)

OpenSips receive INVITE from MicroSIP1 - I can get from body  RTP / SAVP.

OpenSisp has no information whether MicroSIP2 it supports or not (RTP/SAVP)
1. If I setup:
tpengine_offer("RTP/AVP replace-session-connection replace-origin
ICE=remove");
MicroSip2 rejected call because not supported  RTP/AVP

2. If I setup:
tpengine_offer("RTP/SAVP replace-session-connection replace-origin
ICE=remove");
MicroSip1 rejected call because not supported  RTP/SAVP

Best regards,
Dragomir

На пн, 1.07.2019 г. в 8:58 ч. Alexey Vasilyev 
написа:

> Hi Dragomir.
>
> What do you expect RTPEngine to do? You tell rtpengine_offer("RTP/AVP
> RTP/SAVP replace-session-connection replace-origin ICE=remove"); So you
> tell
> it, that you want to get as result unencrypted body (RTP/AVP) AND encrypted
> body (RTP/SAVP). But it can be either encrypted or unencrypted.
>
> You should choose, what you want. Use only one profile as parameter.
>
>
>
>
> -
> ---
> Alexey Vasilyev
> --
> Sent from:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users