[OpenSIPS-Users] customizing opensips as webrtc gateway for handling SIP over websockets

[riko nir]

Hi all,

I would like to customize the opensips to  only act as a webrtc gateway
(for handling websocket messages). It has to parse the websocket message
and relay the message content to another library which actually handles
register, invite etc.

Also, I would like to use RTPEngine with opensips for getting support of
ICE and DTLS, but the media will be going through another library.

How can I customize opensips for this purpose? Anybody have done this
already?

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

Re: [OpenSIPS-Users] Generating 183 reply and Playing Early Media using rtpproxy_stream2uac()

[Faheem Muhammad]

Husnain,
The same type of question is already answered by Kristian F. Høgh on the
opensips mailing list. Try to search "*[OpenSIPS-Users] Playing caller a
file before dialing callee*"

Hope it will solve your problem.

Faheem


On Thu, Jan 7, 2016 at 10:37 AM, Hamid Hashmi 
wrote:

> Try the following example. Change connection IP and codec order
> accordingly.
>
> if (is_method("INVITE") && has_body("application/sdp")) {
> $var(Session_owner) = $rb[1];
> append_to_reply("Content-Type: 
> application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4 
> 10.130.130.114\r\nt=0 0\r\nm=audio 61896 RTP 0 8 3 101\r\na=rtpmap:0 
> pcmu/8000\r\na=rtpmap:8 pcma/8000\r\na=rtpmap:3 gsm/8000\r\na=rtpmap:101 
> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n")';
> t_reply_with_body("183", "Session Progress", "$var(body)");
> }
>
>
>
> *Hamid R. Hashmi*
> Software Engineer - VoIP
> Vopium A/S
>
>
> --
> Date: Wed, 6 Jan 2016 20:33:29 +0300
> From: husnain.tas...@gmail.com
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] Generating 183 reply and Playing Early Media
> using rtpproxy_stream2uac()
>
>
> Dear Users,
> I have a scenario where I want to Play an announcement as early media to
> the UAC before answering the call but I don't want to use any media server
> like asterisk/Freeswitch.
>
> When user agent sends an INVITE I am calling rtpproxy_offer() and sending
> INVITE to B party. On 100 Trying from B party I am
> calling rtpproxy_stream2uac() and streaming the file I can see that RTPs
> are going towards the UAC (caller) but softphone is not accepting those
> RTPs because 183 was not sent to the softphone so he don't know the media
> details of the rtpproxy. but as 200 Ok reaches to the softphone last part
> of the audio can be heard immediately after Answer.
>
> So I think on 100 Trying from B Part if I send 183 Session Progress to the
> softphone and then starting the RTP stream will work. So can you please
> tell me is there a way to generate 183 Session Progress with media details
> of RTPPROXY in opensips ? so that my scenario starts work.
>
> Regards,
> Husnain Taseer
> VoIP Developer
>
> ___ 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] Generating 183 reply and Playing Early Media using rtpproxy_stream2uac()

[Hamid Hashmi]

Try the following example. Change connection IP and codec order accordingly.
if (is_method("INVITE") && has_body("application/sdp")) {
$var(Session_owner) = $rb[1];
append_to_reply("Content-Type: 
application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4 
10.130.130.114\r\nt=0 0\r\nm=audio 61896 RTP 0 8 3 101\r\na=rtpmap:0 
pcmu/8000\r\na=rtpmap:8 pcma/8000\r\na=rtpmap:3 gsm/8000\r\na=rtpmap:101 
telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n")';
t_reply_with_body("183", "Session Progress", "$var(body)");
}

Hamid R. HashmiSoftware Engineer - VoIPVopium A/S

Date: Wed, 6 Jan 2016 20:33:29 +0300
From: husnain.tas...@gmail.com
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Generating 183 reply and Playing Early Media using
rtpproxy_stream2uac()

Dear Users,I have a scenario where I want to Play an announcement as early 
media to the UAC before answering the call but I don't want to use any media 
server like asterisk/Freeswitch. 
When user agent sends an INVITE I am calling rtpproxy_offer() and sending 
INVITE to B party. On 100 Trying from B party I am calling 
rtpproxy_stream2uac() and streaming the file I can see that RTPs are going 
towards the UAC (caller) but softphone is not accepting those RTPs because 183 
was not sent to the softphone so he don't know the media details of the 
rtpproxy. but as 200 Ok reaches to the softphone last part of the audio can be 
heard immediately after Answer. 
So I think on 100 Trying from B Part if I send 183 Session Progress to the 
softphone and then starting the RTP stream will work. So can you please tell me 
is there a way to generate 183 Session Progress with media details of RTPPROXY 
in opensips ? so that my scenario starts work.

Regards,Husnain TaseerVoIP Developer

___
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] external app listening to OpenSIPS/SIP events

[Julian Kay]

THANKS!! 

raising the event in the script got rid of the error.

 

But OpenSIPS is not calling the method I'm testing on a Tomcat XMLRPC 
serverlet. This is my actual event subscription code I'm using: 
subscribe_event("E_SIP_MESSAGE","xmlrpc:192.168.3.132:8080:ccurbiz.xmlrpc.OpenSIPSInterface.msgtests");

 

I've tried several (many) syntax variations. 

 

On the Tomcat server it always points to the root RPC2 and returns the error 
code 404.

 

 

ccurbiz -> is my  java project name

xmlrpc -> is the servelet mapping to the Java class OpenSIPSInterface

msgtest -> is the method I'm trying to call

 

I wanted to make sure the xmlrpc server was working, I'm able to successfully 
call the method with xmlrpc-test-tool

 

Any suggestion to be able to make this work with Tomcat server are greatly 
appreciated!!

 

Thanks!

Juls

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Wednesday, December 30, 2015 10:40 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP events

 

Hi, Julian!

Make sure you are raising that event from your script (i.e. 
raise_event("E_SIP_MESSAGE")).

http://www.opensips.org/Documentation/Tutorials-EventInterface#toc9

Best regards,
Răzvan

On 12/24/2015 03:39 AM, Julian Kay wrote:

Thanks for the input, but for now I need to work with XMLRPC.

 

is it possible to create a custom event? I've been trying to use in the 
startup_route: subscribe_event("E_SIP_MESSAGE", 
"xmlrpc:http://192.168.3.201:/msgtests";) 

 

I get the error: ERROR: core:evi_event_subscribe: invalid event name 


 

Any help much appreciated, thx!!! 

 

Juls

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Tito Cumpen
Sent: Wednesday, December 16, 2015 6:19 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] external app listening to OpenSIPS/SIP events

 

Julian,

 

 

Look into using http://www.opensips.org/html/docs/modules/devel/event_rabbitmq 
This module allows you to spin up an event based at any moment in the script 
and pass variables in the process(Meaning it is a publisher.). Your app can 
subscribe as reader of the queue and do whatever task needs to be done. You can 
go as far as using this CDR as well.

 

Goodluck,

Tito

 

On Wed, Dec 16, 2015 at 6:14 PM, Julian Kay  wrote:

Hi;

 

I'm looking for some guidance or confirmation that I'm on the right path. If I 
want an external app to be able to listen to SIP events like SIP event 180 
(ringing), is registering a custom event with OpenSIPS MI and then use 
"raise_event" to fire a custom event a good way to get my external app to 
listen to OpenSIPS events?

 

Thanks!!

Juls


___
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





-- 
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Request Timeout on INVITE After Idle Registration

[Nabeel]

I managed to capture the SIP traffic with Wireshark.  It seems that the
party generating the 408 reply is OpenSIPS, not the callee.  OpenSIPS does
not seem to forward the call to the callee at all.

Below are traces showing a successful call and a call with Request Timeout.

The server IP is 162.249.6.110, the caller IP is 92.40.249.9, and the
callee IP is 188.29.165.24.

Trace for a successful call:

http://pastebin.com/2xn0bkEU

Trace for a call with Request Timeout:

http://pastebin.com/WR7BA6pj

Please advise what may be causing this.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] REG::opensipsctl with multiple opensips instances

[Ravitez Ravi]

Hello All,
Good Day and a Happy New Year.
I have two instances of openSips(V1.11) executable running say
"opensips1" and "opensips2"
when i execute opensipsctl it always picks up data from
"opensips2" is there a way that we can
choose which executable to pick.Thank you for the help.

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

[OpenSIPS-Users] Generating 183 reply and Playing Early Media using rtpproxy_stream2uac()

[Husnain Taseer]

Dear Users,
I have a scenario where I want to Play an announcement as early media to
the UAC before answering the call but I don't want to use any media server
like asterisk/Freeswitch.

When user agent sends an INVITE I am calling rtpproxy_offer() and sending
INVITE to B party. On 100 Trying from B party I am
calling rtpproxy_stream2uac() and streaming the file I can see that RTPs
are going towards the UAC (caller) but softphone is not accepting those
RTPs because 183 was not sent to the softphone so he don't know the media
details of the rtpproxy. but as 200 Ok reaches to the softphone last part
of the audio can be heard immediately after Answer.

So I think on 100 Trying from B Part if I send 183 Session Progress to the
softphone and then starting the RTP stream will work. So can you please
tell me is there a way to generate 183 Session Progress with media details
of RTPPROXY in opensips ? so that my scenario starts work.

Regards,
Husnain Taseer
VoIP Developer
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Processing calling-name(CNAM) from PRI

[Zahid Mehmood]

Hi,
INFO is generated in early stage.

The information I provided in my original post was not entirely accurate.
My setup was as follows:

Carrier PRI  - Siemens PBX (4000)  ---(tried qsiq and primary-ni) ---
Cisco 3845  PROXY

After more debugging we found out that Siemens was not passing the correct
information in the ISDN setup messages.  Specifically, it was missing the
following:

"Operation = InformationFollowing (calling_name)

   Name information in subsequent FACILITY message”


We then connected the carrier PRI directly into the media gateway and
that resolved the issue.  Cisco gateway was already configured to act
similar to how Jeff described the Adtran configuration.  We started
seeing the correct calling name as soon as the PRI was connected
directly.


Thank you for the followup.


Best regards,


--

Zahid




On Wed, Jan 6, 2016 at 4:49 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Zahid,
>
> When in the dialog is the INFO injected ? in the early stage ? or after
> the call is established ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 18.12.2015 17:12, Zahid Mehmood wrote:
>
> Hi Jeff,
>Thanks for your response.  >From what I understand from Cisco
> documentation, gateway seems to act that way when using H323 but, sadly,
> not for SIP.  I will push Cisco about this.
>
> Assuming the worst, Is there any thing I can try on the proxy side to get
> the desired results?
>
> Regards,
> --
> Zahid
>
>
> On Fri, Dec 18, 2015 at 9:44 AM, Jeff Pyle 
> wrote:
>
>> In Adtran TA900 series gateways (very Cisco-like) I'm able to configure
>> the PRI interface to wait for the FACILITY message before sending the
>> initial INVITE.  When the INVITE does leave the gateway towards the proxy,
>> it has full caller name information.  Perhaps something like this is
>> available on the Cisco.  I hope so, because if not, you're going to have a
>> difficult time integrating the INFO message.
>>
>>
>> - Jeff
>>
>>
>> On Thu, Dec 17, 2015 at 2:53 PM, Zahid Mehmood < 
>> z...@columbia.edu> wrote:
>>
>>> Hi,
>>>I am having trouble figuring out how to process the calling-name
>>> coming from the PRI. In my setup, PRI is connected to a Cisco media gateway
>>> which sends traffic to the proxy servers.  Calling name is not coming  in
>>> the ISDN setup message.  It is actually provided in a separate facility
>>> message [1].
>>>
>>> Cisco gateway processes this secondary messages and generates a INFO
>>> message.  Polycom phone sends the 200 ok message but there is no change in
>>> the visible caller id.
>>>
>>> Does anyone have a working example or suggestion of how this is supposed
>>> to work?
>>>
>>> Invite:
>>>
>>> U 2015/12/17 14:20:31.215540 10.10.1.1:50975 -> 10.10.2.2:5060
>>> INVITE sip:10301@10.10.2.2:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3A2284.
>>> Remote-Party-ID: "111222" < 
>>> sip:111222@10.10.1.1>;party=calling;screen=yes;privacy=off.
>>> From: "111222" < 
>>> sip:111222@10.10.1.1>;tag=5745CCC-1C72.
>>> To: .
>>> Date: Thu, 17 Dec 2015 19:20:31 GMT.
>>> Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1.
>>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
>>> Min-SE:  1800.
>>> Cisco-Guid: 0311776101-2754220517-2148597785-1445067520.
>>> User-Agent: Cisco-SIPGateway/IOS-12.x.
>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>> SUBSCRIBE, NOTIFY, INFO, REGISTER.
>>> CSeq: 101 INVITE.
>>> Max-Forwards: 70.
>>> Timestamp: 1450380031.
>>> Contact: .
>>> Expires: 180.
>>> Allow-Events: telephone-event.
>>> Content-Type: application/sdp.
>>> Content-Disposition: session;handling=required.
>>> Content-Length: 279.
>>> .
>>> v=0.
>>> o=CiscoSystemsSIP-GW-UserAgent 3918 6190 IN IP4 10.10.1.1.
>>> s=SIP Call.
>>> c=IN IP4 10.10.1.1.
>>> t=0 0.
>>> m=audio 18854 RTP/AVP 0 18 101.
>>> c=IN IP4 10.10.1.1.
>>> a=rtpmap:0 PCMU/8000.
>>> a=rtpmap:18 G729/8000.
>>> a=fmtp:18 annexb=no.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-16.
>>>
>>> Invite messages:
>>>
>>> U 2015/12/17 14:20:31.546310 10.10.1.1:50975 -> 10.10.2.2:5060
>>> INFO sip:10301@10.219.136.69:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3C1EC0.
>>> From: "111222" < 
>>> sip:111222@10.10.1.1>;tag=5745CCC-1C72.
>>> To: ;tag=1768D8EC-CCDB1323.
>>> Date: Thu, 17 Dec 2015 19:20:31 GMT.
>>> Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1.
>>> User-Agent: Cisco-SIPGateway/IOS-12.x.
>>> Max-Forwards: 70.
>>> Route: .
>>> Timestamp: 1450380031.
>>> CSeq: 103 INFO.
>>> Contact: .
>>> Remote-Party-ID: "WIRELESS CALLER" < 
>>> sip:111222@10.10.1.1>;party=calling;screen=no;privacy=off.
>>> Content-Length: 0.
>>> .
>>>
>>>
>>> Best Regards,
>>>
>>> --
>>> Zahid
>>>
>>> [1]
>>> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-isdn.html#GUID-53D5C9AB-AAC4-4178-8158-0DAEFB5BC33E

Re: [OpenSIPS-Users] Request Timeout on INVITE After Idle Registration

[Nabeel]

Hi Bogdan,

I'm trying to use the siptrace module to capture SIP traffic, as I am using
TLS.

In my config file I have the following:

...

loadmodule "siptrace.so"
modparam("siptrace", "db_url", "mysql://user:passwd@host/dbname")
modparam("siptrace", "trace_flag", "TRACE_FLAG")

...

route{
sip_trace();
setflag(TRACE_FLAG);

...

I also did 'opensipsctl fifo sip_trace on', but nothing is captured in the
sip_trace table of the database.

I want to capture all traffic.  What am I missing?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Query for opensips-1.11 crash due to mongodb connection break .

[Sasmita Panda]

Hi Andrei ,

   Opensips version is : 1.11.2-notls

 Bellow is the back trace .


GNU gdb (GDB) 7.6.1-ubuntu
Copyright (C) 2013 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 /root/opensips-1.11/sbin/opensips...done.
[New LWP 16965]

warning: Can't read pathname for load map: Input/output error.
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `/root/opensips-1.11//sbin/opensips -m 1024 -f
/root/opensips-1.11//etc/opensips'.
Program terminated with signal 11, Segmentation fault.
#0  bson_iterator_next (i=i@entry=0x70fca8d0) at src/bson.c:346
346 return ( bson_type )( *i->cur );
(gdb) bt
#0  bson_iterator_next (i=i@entry=0x70fca8d0) at src/bson.c:346
#1  0x7f077dd56244 in mongo_db_insert_trans (con=,
table=, _k=, _v=,
_n=)
at cachedb_mongodb_dbase.c:1704
#2  0x7f077d722e60 in db_cachedb_insert (_h=,
_k=, _v=, _n=) at dbase.c:149
#3  0x7f077aef3407 in insert_siptrace_flag (keys=0x7f077b10bd20
, vals=0x7f077b10be20 , msg=0x7f077b10b710 ) at
siptrace.c:468
#4  save_siptrace (avp=avp@entry=0x0, first_val=first_val@entry=0x70fcacc0,
msg=0x7f0739db73d0, vals=0x7f077b10be20 , keys=0x7f077b10bd20
)
at siptrace.c:560
#5  0x7f077aef97af in trace_onreply_in (t=,
type=, ps=) at siptrace.c:1212
#6  0x7f077d2cabbd in run_trans_callbacks (type=type@entry=2,
trans=0x7f073a1606d8, req=, rpl=rpl@entry=0x7f077ee222c0,
code=)
at t_hooks.c:212
#7  0x7f077d2d1e2c in t_reply_matching (p_msg=p_msg@entry=0x7f077ee222c0,
p_branch=) at t_lookup.c:827
#8  0x7f077d2d25bf in t_check (p_msg=p_msg@entry=0x7f077ee222c0,
param_branch=param_branch@entry=0x70fcaee0) at t_lookup.c:898
#9  0x7f077d2f1ec6 in reply_received (p_msg=0x7f077ee222c0) at
t_reply.c:1525
#10 0x00437af6 in forward_reply (msg=msg@entry=0x7f077ee222c0) at
forward.c:575
#11 0x00486da6 in receive_msg (buf=, len=, rcv_info=rcv_info@entry=0x70fcb0b0) at receive.c:214
#12 0x00501f8d in udp_rcv_loop () at udp_server.c:424
#13 0x00414b01 in main_loop () at main.c:942
#14 main (argc=, argv=) at main.c:1634



Actually , I am not getting anything from this . What else you want to find
the problem let me know .

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

On Wed, Jan 6, 2016 at 2:49 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Sasmita,
>
> If mysql server is down, opensips will continue to work, failing to
> perform any DB operation (of course) - definitely it will not stop or so.
>
> Now, back to the mongo issue - you have siptrace configured to insert into
> mongo via db_cachedb ; and if you stop the mondo daemon, the next insert it
> will crash opensips.
> I see in your logs that a core file was generated - can you extract and
> post here the backtrace ?
>
> Also, what is the exact version of opensips (do opensips -V)
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 06.01.2016 07:59, Sasmita Panda wrote:
>
> Hi ,
>
> Yes , it seems like this .
>
>   My concern is , If I am stopping mysql of the machine and my
> opensips is using mysql , then also opensips is not getting stopped . But
> why this is happening with mongodb  ?
>
> This is the logs I am getting during opensips get stopped .
>
>  translated to  2
>  request domain,
>  translated to  2
>  ACC: transaction answered:
> timestamp=1452059689;method=BYE;from_tag=as2a07e122;to_tag=85d3d0e795bb4a57ab6a76dd95d7ec4e;call_id=8f01c197bf4f498ab9379a4f589a4d6b;code=200;reason=OK
>  ERROR:cachedb_mongodb:mongo_db_insert_trans: Failed to run query. Err =
> 2, 32 , 0
>  opensips segfault at 4 ip 7fb1eb1c3fbc sp 7fff267b0660 error 4 in
> libmongoc.so.0.6.0[7fb1eb1bd000+13000]
>  ACC: transaction answered:
> timestamp=1452059690;method=BYE;from_tag=as45b598fe;to_tag=988c2c314cc545cabb60b0c7c09c1aed;call_id=751c426ebacc4a03aba5ea3e326dcb91;code=200;reason=OK
>  ERROR:cachedb_mongodb:mongo_db_insert_trans: Failed to run query. Err =
> 2, 32 , 0
>  segfault at 4 ip 7fb1eb1c3fbc sp 7fff267b0660 error 4 in
> libmongoc.so.0.6.0[7fb1eb1bd000+13000]
>  CRITICAL:core:receive_fd: EOF on 15
>  INFO:core:handle_sigs: child process 20023 exited by a signal 11
>  INFO:core:handle_sigs: core was generated
>  INFO:core:handle_sigs: terminating due to SIGCHLD
>  INFO:core:sig_usr: signal 15 received
>  INFO:core:sig_usr: signal 15 received
>  INFO:core:sig_usr: signal 15 received
>  INFO:core:sig_usr: signal 15 received
>  

[OpenSIPS-Users] sdp answer

[riko nir]

Hi all,

How is the communication flow happens between opensips and rtpengine incase
of a call through SIP over websocket?
When opensips receives a sdp offer from a web-based sip client, opensips is
creating sdp answer or rtpengine is creating sdp anwer or rtpengine just
updating ice information only in sdp and other than that, media related
info is created by opensips?

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

[OpenSIPS-Users] Is NATHELPER module enough in my case? Help please.

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS-users,


First of all, happy new year.

I'm still reading about NATs, SIP proxies and SIP, to understand how to solve a 
question in my project. I'm not expert in TCP, IP, routers, NATs and networks. 
That is why I would like to get some help here.  I guess I'm almost getting the 
point to solve it.


In my current network topology case I have:




 Mobile Phone 1 on Internet


   / \ |


|  |


|  SIP |


|  \/ Real IP  
(Internet)


 NAT  


/\ |  'Nated' 
IP (Wan)


|  |


| SIP   |


|  |


|  |


|  \/

   ROUTER   



/\ |  (Lan)


|   SIP |


|  |


  SIP Proxy OPENSIPS


|  |


|   SIP|


|  |


Mobile Phone 2 on local network



When Phone 2 calls Phone 1, everything is ok. Phone 1 is registered on OpenSIPS 
with 'Nated' IP, by someway. So, Phone 2 sends INVITE to such IP. When Phone 1 
answers with SIP OK, the Contact header filed has the same 'Nated' IP. Then, 
Phone 2 can send the ACK to the correct path.


However, when Phone 1 calls Phone 2 and Phone 2 answer with SIP OK, Phone 1 
can't send the ACK, because  the Contact header field from SIP OK has the local 
IP for Phone 2.

It is true because when Phone 2 registers itself on OpenSIPS, the local IP is 
recorded in the local database table.


So I have the following questions:


1 - Can this problem be solved without using the Nat Traversal Module?


2 - Can NATHELPER module fix the Contact header field (as a I saw this module 
has functions related with it), when Phone 2 sends SIP OK to Phone 1?  If yes, 
how can such module determine a 'good' IP to put in this header field?


3 - If Phone 2 uses a stun server, will such phone be registered on OpenSIP 
with a real IP, won't be? Could it be a solution too?


Any hint will be very helpful!


Thanks a lot!







RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] MenuConfig Compile Fail on Solaris Sparc

[Nathaniel L. Keeling III]

here is the end of the output from running "NICER=0 make"

Compiling net/net_tcp.c
gcc -m64 -Wcast-align  -mcpu=ultrasparc -I/usr/include/json 
-DPKG_MALLOC  -DSHM_MMAP  -DUSE_MCAST -DDISABLE_NAGLE  -DSTATISTICS  
-DHAVE_RESOLV_RES  -DF_MALLOC -DEXTRA_DEBUG   -DNAME='"opensips"' 
-DVERSION='"2.1.1"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 
4.9.2"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_yes 
-DCFG_DIR='"/opt//etc/opensips/"'  -DVERSIONTYPE='"git"' 
-DTHISREVISION='"7df8a1a"' -DFAST_LOCK -DADAPTIVE_WAIT 
-DADAPTIVE_WAIT_LOOPS=1024  -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H 
-DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION 
-D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c net/net_tcp.c 
-o net/net_tcp.o

In file included from net/../mem/../locking.h:66:0,
 from net/../mem/../statistics.h:143,
 from net/../mem/../pt.h:34,
 from net/../mem/../dprint.h:55,
 from net/../mem/mem.h:38,
 from net/net_tcp.c:42:
net/../mem/../lock_alloc.h: In function 'lock_set_alloc':
net/../mem/../lock_alloc.h:70:13: warning: cast increases required 
alignment of target type [-Wcast-align]

ls->locks=(gen_lock_t*)((char*)ls+sizeof(gen_lock_set_t));
 ^
In file included from net/../globals.h:30:0,
 from net/net_tcp.c:44:
net/../ip_addr.h: In function 'sockaddr2ip_addr':
net/../ip_addr.h:224:25: warning: cast increases required alignment of 
target type [-Wcast-align]

memcpy(ip->u.addr, &((struct sockaddr_in*)sa)->sin_addr, 4);
 ^
net/../ip_addr.h:229:25: warning: cast increases required alignment of 
target type [-Wcast-align]

memcpy(ip->u.addr, &((struct sockaddr_in6*)sa)->sin6_addr, 16);
 ^
In file included from net/../reactor.h:41:0,
 from net/net_tcp.c:50:
net/../io_wait_loop.h: In function 'io_wait_loop_devpoll':
net/../io_wait_loop.h:426:4: error: 'e' undeclared (first use in this 
function)

e = get_fd_map(h, h->fd_array[r].fd);
^
net/../io_wait_loop.h:426:4: note: each undeclared identifier is 
reported only once for each function it appears in

In file included from net/../globals.h:30:0,
 from net/net_tcp.c:44:
net/net_tcp.c: In function 'tcp_connect_blocking':
net/../ip_addr.h:168:26: warning: cast increases required alignment of 
target type [-Wcast-align]

   _port_no = su_getport( (union sockaddr_union*)_su); \
  ^
net/net_tcp.c:270:4: note: in expansion of macro 'get_su_info'
get_su_info( servaddr, ip, port);
^
net/../ip_addr.h:168:26: warning: cast increases required alignment of 
target type [-Wcast-align]

   _port_no = su_getport( (union sockaddr_union*)_su); \
  ^
net/net_tcp.c:300:4: note: in expansion of macro 'get_su_info'
get_su_info( servaddr, ip, port);
^
net/../ip_addr.h:168:26: warning: cast increases required alignment of 
target type [-Wcast-align]

   _port_no = su_getport( (union sockaddr_union*)_su); \
  ^
net/net_tcp.c:319:5: note: in expansion of macro 'get_su_info'
 get_su_info( servaddr, ip, port);
 ^
Makefile.rules:25: recipe for target 'net/net_tcp.o' failed
make: *** [net/net_tcp.o] Error 1

thanks

nathaniel


On 1/6/16 3:32 AM, Răzvan Crainea wrote:

can you run 'NICER=0 make"

Thanks,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 07:55 PM, Nathaniel L. Keeling III wrote:

No. The compile just stops.

Thanks

Nathaniel

On 1/5/16 6:58 AM, Răzvan Crainea wrote:
I doubt you are missing something on your side. Are there any other 
errors below that?


Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 02:33 PM, Nathaniel L. Keeling III wrote:

Razvan,

Changing to -lcurses solved that problem. Thanks. Ran into another 
compile problem when compiling opensips 2.1. Here is the error:


Makefile.rules:25: recipe for target 'net/net_tcp.o' failed
make[1]: *** [net/net_tcp.o] Error 1
make[1]: Leaving directory '/usr/local/src/opensips/opensips_2_1

Is there something else that I am missing?


Thanks

Nathaniel

On 1/5/16 4:56 AM, Răzvan Crainea wrote:

Hi, Nathaniel!

It seems to be a common problem on Solaris[1] - some symbols are 
defined in libncurses, some in libcurses. Can you try to replace 
the '-lncurses' with '-lcurses' in menuconfig/Makefile (line 8)?
If that doesn't work, try 'LD_EXTRA_OPTS=-lcurses make menuconfig' 
and post the output.


[1] https://gcc.gnu.org/ml/gcc-help/1999-11n/msg00020.html

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 07:18 AM, Nathaniel L. Keeling III wrote:
I am trying to install Opensips 2.1 on Solaris Sparc 10 and I am 
getting this error when I perform "make menuconfig":


# make menuconfig
Usage: grep -hblcnsviw pattern file . . .
/bin/sh: proto_: not found
make -C menuconfig
make[1]: Entering directory 
'/usr/local/src/opensips

Re: [OpenSIPS-Users] opensips-1.11.4: exec_msg_threshold warning logged showing delayes in unexpected places

[Michael Ulitskiy]

Hello,

So I wrote a script that watches opensips log for log_expiry warning and 
collects per-process
cpu utilization (using 'top' command) when such entry appears in the log.
Here's what it found:

LOG:
Dec 14 19:50:41 csp-p1n1 csp1[19109]: WARNING:core:log_expiry: threshold 
exceeded : msg processing took too long - 148363 us.Source : REGISTER 
sip:csp1.example.com:5060 SIP/2.0
Dec 14 19:50:41 csp-p1n1 csp1[19109]: WARNING:core:log_expiry: #1 is a module 
action : subst - 147916us - line 1308
Dec 14 19:50:41 csp-p1n1 csp1[19109]: WARNING:core:log_expiry: #2 is a module 
action : t_replicate - 78us - line 1310
Dec 14 19:50:41 csp-p1n1 csp1[19109]: WARNING:core:log_expiry: #3 is a module 
action : save - 61us - line 1291
Dec 14 19:50:41 csp-p1n1 csp1[19109]: WARNING:core:log_expiry: #4 is a module 
action : avp_subst - 34us - line 907
Dec 14 19:50:41 csp-p1n1 csp1[19109]: WARNING:core:log_expiry: #5 is a module 
action : pv_www_authorize - 15us - line 1254

TOP:
top - 19:50:41 up 350 days,  2:31,  0 users,  load average: 0.53, 0.33, 0.27
Tasks: 196 total,   3 running, 193 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.5%us,  0.5%sy,  0.0%ni, 98.7%id,  0.3%wa,  0.0%hi,  0.0%si,  0.0%st
Mem:   4005248k total,  3776444k used,   228804k free,   286420k buffers
Swap:  2097144k total,40960k used,  2056184k free,  1850068k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
19109 opensips  20   0  696m  19m  16m S   30  0.5  26:28.04 opensips   
19104 opensips  20   0  696m  19m  16m S2  0.5  26:33.08 opensips   
19105 opensips  20   0  696m  19m  16m S2  0.5  26:28.10 opensips   
1 root  20   0  4356  572  492 S0  0.0  13366:15 init   
2 root  20   0 000 S0  0.0   0:06.93 kthreadd   
3 root  20   0 000 S0  0.0  24:34.15 ksoftirqd/0
5 root   0 -20 000 S0  0.0   0:00.00 kworker/0:0H   
7 root  RT   0 000 S0  0.0   1:56.92 migration/0   

opensips itself utilized 30% of cpu time.

here's another example:

LOG:
Dec 14 21:07:29 csp-p1n1 csp1[19110]: WARNING:core:log_expiry: threshold 
exceeded : msg processing took too long - 148509 us.Source : NOTIFY 
sip:nswasser002@192.168.36.132:5061 SIP/2.0
Dec 14 21:07:29 csp-p1n1 csp1[19110]: WARNING:core:log_expiry: #1 is a module 
action : avp_subst - 63us - line 907
Dec 14 21:07:29 csp-p1n1 csp1[19110]: WARNING:core:log_expiry: #2 is a module 
action : t_relay - 57us - line 1116
Dec 14 21:07:29 csp-p1n1 csp1[19110]: WARNING:core:log_expiry: #3 is a core 
action : 69 - 15us - line 941
Dec 14 21:07:29 csp-p1n1 csp1[19110]: WARNING:core:log_expiry: #4 is a module 
action : sipmsg_validate - 14us - line 1002
Dec 14 21:07:29 csp-p1n1 csp1[19110]: WARNING:core:log_expiry: #5 is a module 
action : lookup - 13us - line 1631

TOP:
top - 21:07:29 up 350 days,  3:48,  0 users,  load average: 0.15, 0.29, 0.31
Tasks: 196 total,   1 running, 195 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.5%us,  0.5%sy,  0.0%ni, 98.7%id,  0.3%wa,  0.0%hi,  0.0%si,  0.0%st
Mem:   4005248k total,  3773700k used,   231548k free,   285912k buffers
Swap:  2097144k total,40960k used,  2056184k free,  1854896k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
19110 opensips  20   0  696m  19m  16m S   30  0.5  26:31.94 opensips   
9 root  20   0 000 S2  0.0 184:25.29 rcu_sched  
19104 opensips  20   0  696m  19m  16m S2  0.5  26:39.33 opensips   
19112 opensips  20   0  696m  19m  16m S2  0.5  26:27.47 opensips   
19115 opensips  20   0  696m  19m  16m S2  0.5  26:33.44 opensips   
19118 opensips  20   0  696m  19m  16m S2  0.5  26:33.73 opensips   
25899 root  19  -1  193m  24m  11m S2  0.6   6487:27 corosync   
1 root  20   0  4356  572  492 S0  0.0  13366:15 init   
2 root  20   0 000 S0  0.0   0:06.93 kthreadd   
3 root  20   0 000 S0  0.0  24:35.13 ksoftirqd/0
5 root   0 -20 000 S0  0.0   0:00.00 kworker/0:0H  

again opensips itself utilized 30% of cpu time. in addition log_expiry message 
doesn't show a single action that would take long, just the whole message 
processing takes ~150ms.

it looks like opensips for no obvious reasons randomly spikes its cpu 
utilization. it's not periodic, doesn't correspond to any periodic process in 
the system and looks completely random.

i checked with MI 'ps' command and those cpu-spiking processes are all "SIP 
receiver" processes.

I'm at loss. What could be causing opensips process to spike its cpu usage like 
that?

I'd appreciate any ideas.

Thanks,

Michael


On Wednesday, December 02, 2015 07:19:06 PM Liviu Chircu wrote:
> Hi Michael,
> 
> The subst() warning is very interesting, since it

Re: [OpenSIPS-Users] how opensips server load balance ( active active HA)

[Jerry Kendall]

Did you get it working? Overplayed with opensips but could not get it to work. 
Would you be willing to share your settings?

Sent from my iPhone

> On Dec 28, 2015, at 8:26 PM, chiu ching cheng  wrote:
> 
> I want to install a opensips in itsp , I want to design the infrastructure as 
> following ;
> 
> one mysql cluster with virtual ip , one opensips server cluster , several 
> mediaproxy server ,  But I don't find document about how to do opensips 
> cluster .
> 
> I find one document that two opensips can connect one mysql cluster , but not 
> about how opensips cluster .
> 
> Does opensips can be a cluster infrastructure ?
> ___
> 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] opensips 1.11.5-tls , SNMP not working - No Such Object available on this agent at this OID

[Bogdan-Andrei Iancu]

Hi Igor,

I will try to reproduce and revert. I assume you use the SNMP setup 
described in the SNMP tutorial from the README file, right ?


Regards,

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

On 22.12.2015 08:10, Игорь Павлов wrote:

Hi all,

I'm trying to get SNMP statistic from opensips with snmpstat-module. 
For each request with snmpwalk or snmpget I get reply:


/No Such Object available on this agent at this OID/ or /No entries/
/
/
For example:

snmpget -v 2c -c public localhost openserSIPRegStats
OPENSER-SIP-SERVER-MIB::openserSIPRegStats = No Such Object available 
on this agent at this OID


snmptable -Ci -v 2c -c public localhost openserSIPRegUserTable
OPENSER-SIP-SERVER-MIB::openserSIPRegUserTable: No entries

snmpwalk -v 2c -c public localhost .1.3.6.1.4.1.27483
OPENSER-REG-MIB::openser = No Such Object available on this agent at 
this OID


Also I have openSER_SNMPAgent.txt in /tmp directory which is always 
zero size.


ls -l /tmp/openSER_SNMPAgent.txt
-rw-r--r-- 1 opensips opensips 0 dec 22 08:45 /tmp/openSER_SNMPAgent.txt


Best regards,
Igor Pavlov


___
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] Processing calling-name(CNAM) from PRI

[Bogdan-Andrei Iancu]

Hi Zahid,

When in the dialog is the INFO injected ? in the early stage ? or after 
the call is established ?


Best regards,

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

On 18.12.2015 17:12, Zahid Mehmood wrote:

Hi Jeff,
   Thanks for your response.  >From what I understand from Cisco 
documentation, gateway seems to act that way when using H323 but, 
sadly, not for SIP.  I will push Cisco about this.


Assuming the worst, Is there any thing I can try on the proxy side to 
get the desired results?


Regards,
--
Zahid


On Fri, Dec 18, 2015 at 9:44 AM, Jeff Pyle 
mailto:jeff.p...@fidelityvoice.com>> wrote:


In Adtran TA900 series gateways (very Cisco-like) I'm able to
configure the PRI interface to wait for the FACILITY message
before sending the initial INVITE.  When the INVITE does leave the
gateway towards the proxy, it has full caller name information. 
Perhaps something like this is available on the Cisco.  I hope so,

because if not, you're going to have a difficult time integrating
the INFO message.


- Jeff


On Thu, Dec 17, 2015 at 2:53 PM, Zahid Mehmood mailto:z...@columbia.edu>> wrote:

Hi,
   I am having trouble figuring out how to process the
calling-name coming from the PRI. In my setup, PRI is
connected to a Cisco media gateway which sends traffic to the
proxy servers.  Calling name is not coming  in the ISDN setup
message.  It is actually provided in a separate facility
message [1].

Cisco gateway processes this secondary messages and generates
a INFO message. Polycom phone sends the 200 ok message but
there is no change in the visible caller id.

Does anyone have a working example or suggestion of how this
is supposed to work?

Invite:

U 2015/12/17 14:20:31.215540 10.10.1.1:50975 -> 10.10.2.2:5060
INVITE sip:10301@10.10.2.2:5060
 SIP/2.0.
Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3A2284.
Remote-Party-ID: "111222" mailto:sip%3A111222@10.10.1.1>>;party=calling;screen=yes;privacy=off.
From: "111222" mailto:sip%3A111222@10.10.1.1>>;tag=5745CCC-1C72.
To: mailto:sip%3A10301@10.10.2.2>>.
Date: Thu, 17 Dec 2015 19:20:31 GMT.
Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1
.
Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
Min-SE:  1800.
Cisco-Guid: 0311776101-2754220517-2148597785-1445067520.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE,
REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 70.
Timestamp: 1450380031.
Contact: http://sip:111222@10.10.1.1:5060>>.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 279.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 3918 6190 IN IP4 10.10.1.1.
s=SIP Call.
c=IN IP4 10.10.1.1.
t=0 0.
m=audio 18854 RTP/AVP 0 18 101.
c=IN IP4 10.10.1.1.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.

Invite messages:

U 2015/12/17 14:20:31.546310 10.10.1.1:50975 -> 10.10.2.2:5060
INFO sip:10301@10.219.136.69:5060
 SIP/2.0.
Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3C1EC0.
From: "111222" mailto:sip%3A111222@10.10.1.1>>;tag=5745CCC-1C72.
To: mailto:sip%3A10301@10.10.2.2>>;tag=1768D8EC-CCDB1323.
Date: Thu, 17 Dec 2015 19:20:31 GMT.
Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E@10.10.1.1
.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Max-Forwards: 70.
Route: .
Timestamp: 1450380031.
CSeq: 103 INFO.
Contact: http://sip:111222@10.10.1.1:5060>>.
Remote-Party-ID: "WIRELESS CALLER" mailto:sip%3A111222@10.10.1.1>>;party=calling;screen=no;privacy=off.
Content-Length: 0.
.


Best Regards,

-- 
Zahid


[1]

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-isdn.html#GUID-53D5C9AB-AAC4-4178-8158-0DAEFB5BC33E
(figure 2 is close to what we are seeing)

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



_

Re: [OpenSIPS-Users] Getting NOTIFY on UDP while using transport as TCP

[Bogdan-Andrei Iancu]

Hi Surya,

As the NOTIFY is an in-dialog request (for the dialog created by the 
SUBSCRIBE request), it will be routed by OpenSIPS to the SIP contact 
advertised in th SUBSCRIBER request - does the SIP URI in the Contact 
header advertise the TCP transport protocol ?


Regards,

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

On 22.12.2015 07:27, surya wrote:

Hi,


I am testing opensips presence server 1.8.3 using sipp. I am using TCP as
the transport. When I send SUBSCRIBE, 200 OK is received fine but the NOTIFY
is sent on UDP.

When testing from 2 SIP clients it works fine on TCP. Is there something
special to take care while testing TCP from SIPP?

Below is the command and scenario file.
*./sipp 192.168.192.131 -t t1 -sf sn.xml -m 1 -i 192.168.192.128*
-


   
 
   


   
   

   
   


 
   



-

I am not able to figure out what the problem is. I suspect it is something
to do with sipp but not sure.
I couldn't find anything on internet, hence, posting here.

Thanks,
Surya





--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/Getting-NOTIFY-on-UDP-while-using-transport-as-TCP-tp7600288.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.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

Re: [OpenSIPS-Users] DR module fail over on multiple carriers with multiple gateways

[Bogdan-Andrei Iancu]

Hi Jim,

The prefix matching does not work like that, please see:
http://www.opensips.org/html/docs/modules/1.11.x/drouting.html#id293629

For what you what to achieve, the DR rule should look like:
prefix=216555, gwlist=#carrier2,#carrier1

Regards,

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

On 05.01.2016 22:33, Jim DeVito wrote:

Hi All,

I'm trying to get this going but am having no luck. Basically I would 
like the dynamic routing module to cycle through the list of gateways 
associated with a carrier AND then move on to the next matching prefix 
with a lower priority. All the reading and testing seem to say you 
can't do that.


dr_carriers
carrier1,'carrier1GW1=50,carrier1GW2=50'
carrier2,'carrier2GW1=50,carrier2GW2=50'

dr_rules
prefix=216555,pri=20,gwlist=#carrier1
prefix=216555,pri=10,gwlist=#carrier2

So as it stands the call goes (carrier1GW1 -> carrier2GW1) I would 
like it to go (carrier1GW1 -> carrier1GW2 -> carrier2GW1 -> carrier2GW2)


I call do_routing() with WF and F but the behavior is the same.

Any thoughts on what I am doing wrong would be great!

Thanks!!




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

Re: [OpenSIPS-Users] MenuConfig Compile Fail on Solaris Sparc

[Răzvan Crainea]

can you run 'NICER=0 make"

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 01/05/2016 07:55 PM, Nathaniel L. Keeling III wrote:

No. The compile just stops.

Thanks

Nathaniel

On 1/5/16 6:58 AM, Răzvan Crainea wrote:
I doubt you are missing something on your side. Are there any other 
errors below that?


Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 02:33 PM, Nathaniel L. Keeling III wrote:

Razvan,

Changing to -lcurses solved that problem. Thanks. Ran into another 
compile problem when compiling opensips 2.1. Here is the error:


Makefile.rules:25: recipe for target 'net/net_tcp.o' failed
make[1]: *** [net/net_tcp.o] Error 1
make[1]: Leaving directory '/usr/local/src/opensips/opensips_2_1

Is there something else that I am missing?


Thanks

Nathaniel

On 1/5/16 4:56 AM, Răzvan Crainea wrote:

Hi, Nathaniel!

It seems to be a common problem on Solaris[1] - some symbols are 
defined in libncurses, some in libcurses. Can you try to replace 
the '-lncurses' with '-lcurses' in menuconfig/Makefile (line 8)?
If that doesn't work, try 'LD_EXTRA_OPTS=-lcurses make menuconfig' 
and post the output.


[1] https://gcc.gnu.org/ml/gcc-help/1999-11n/msg00020.html

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/05/2016 07:18 AM, Nathaniel L. Keeling III wrote:
I am trying to install Opensips 2.1 on Solaris Sparc 10 and I am 
getting this error when I perform "make menuconfig":


# make menuconfig
Usage: grep -hblcnsviw pattern file . . .
/bin/sh: proto_: not found
make -C menuconfig
make[1]: Entering directory 
'/usr/local/src/opensips/opensips_2_1/menuconfig'
gcc -o configure -m64 -Wcast-align  -mcpu=ultrasparc -g -Wall 
-DMENUCONFIG_CFG_PATH=\"menuconfig/configs/\" 
-DMENUCONFIG_GEN_PATH=\"etc/\" -DMENUCONFIG_HAVE_SOURCES=1  cfg.o 
curses.o items.o commands.o menus.o parser.o main.o -lncurses

Undefined   first referenced
 symbol in file
initscr32   main.o
w32attron   curses.o
w32attroff  curses.o
ld: fatal: Symbol referencing errors. No output written to configure
Makefile:12: recipe for target 'all' failed
make[1]: *** [all] Error 1
make[1]: Leaving directory 
'/usr/local/src/opensips/opensips_2_1/menuconfig'

Makefile:720: recipe for target 'comp_menuconfig' failed

The version of ncurses install is 5.9 and I have gnu m4 version 
1.4.17 installed.



Thanks

Nathaniel


___
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




___
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] why tls not enable

[Bogdan-Andrei Iancu]

Try

"netstat -lnp | grep opensips"

Regards,

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

On 06.01.2016 04:26, chiu ching cheng wrote:

Dear Bogdan:

thanks your reply , but as following , the sip udp is enable , 
opensips is enable ,and log no error . but tls not enable


I ever install another opensips server which is behind the firewall 
and tls work fine , but this server is one interface public ip , one 
interface private , I just config one public ip . I don't know if this 
caused failure ( opensips at lease need two ip ?) .


thanks !

root@125-227-130-1:~# netstat -tuln | grep 5061
tcp0  0 0.0.0.0:25061  0.0.0.0:* 
  LISTEN

root@125-227-130-1:~# netstat -tuln | grep 5060
tcp0  0 0.0.0.0:25060  0.0.0.0:* 
  LISTEN

udp0  0 127.0.0.1:5060   0.0.0.0:*
root@125-227-130-1:~# ps auxw | grep opensips
root  1466  0.0  0.0  70052  4248 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1467  0.0  0.0  70056   892 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1468  0.4  0.0  70052   464 ?S  10:12   0:01 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1469  0.0  0.0  70052   464 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1470  0.0  0.0  70052   656 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1471  0.0  0.0  70052   656 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1472  0.0  0.0  70052   656 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1473  0.0  0.0  70052   896 ?S  10:12   0:00 
/usr/local/sbin/opensips -P /var/run/opensips.pid
root  1731  0.0  0.0  11744   924 pts/0R+ 10:18   0:00 grep 
--color=auto opensips


On Tue, Jan 5, 2016 at 9:54 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi,

Are you sure opensips is actually started ? check logs file for
error or do "ps auxw | grep opensips"

Regards,

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

On 28.12.2015 11:19, chiu ching cheng wrote:

Dear man :

I install opensips 2.1.1 . It's seems ok . But I find tls service
not start . why ? It's strange . Any one can give a explain .
Thanks .



lsb_release -a
No LSB modules are available.
Distributor ID:Ubuntu
Description:Ubuntu 14.04.1 LTS
Release:14.04
Codename:trusty
--
opensipsctl restart

INFO: Restarting OpenSIPS :
INFO: stopped

INFO: Starting OpenSIPS :
INFO: started (pid: 2706)


openssips.cfg

listen=tls:X.X.X.X:5061   # CUSTOMIZE ME
rev_dns=no

advertised_address="X.X.X.X"
alias=X.X.X.X


loadmodule "proto_udp.so"


loadmodule "proto_tls.so"
modparam("proto_tls","verify_cert", "0")
modparam("proto_tls","require_cert", "0")
modparam("proto_tls","tls_method", "TLSv1")
modparam("proto_tls", "ciphers_list", "NULL")
modparam("proto_tls","certificate",
"/usr/local/etc/opensips/tls/user/user-cert.pem")
modparam("proto_tls","private_key",
"/usr/local/etc/opensips/tls/user/user-privkey.pem")
modparam("proto_tls","ca_list",
"/usr/local/etc/opensips/tls/user/user-calist.pem")




netstat -tuln | grep 5061
tcp0  0 0.0.0.0:25061 
  0.0.0.0:*   LISTEN




___
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] Query for opensips-1.11 crash due to mongodb connection break .

[Bogdan-Andrei Iancu]

Hi Sasmita,

If mysql server is down, opensips will continue to work, failing to 
perform any DB operation (of course) - definitely it will not stop or so.


Now, back to the mongo issue - you have siptrace configured to insert 
into mongo via db_cachedb ; and if you stop the mondo daemon, the next 
insert it will crash opensips.
I see in your logs that a core file was generated - can you extract and 
post here the backtrace ?


Also, what is the exact version of opensips (do opensips -V)

Best regards,

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

On 06.01.2016 07:59, Sasmita Panda wrote:

Hi ,

Yes , it seems like this .

  My concern is , If I am stopping mysql of the machine and my 
opensips is using mysql , then also opensips is not getting stopped . 
But why this is happening with mongodb  ?


This is the logs I am getting during opensips get stopped .

 translated to  2
 request domain,
 translated to  2
 ACC: transaction answered: 
timestamp=1452059689;method=BYE;from_tag=as2a07e122;to_tag=85d3d0e795bb4a57ab6a76dd95d7ec4e;call_id=8f01c197bf4f498ab9379a4f589a4d6b;code=200;reason=OK
 ERROR:cachedb_mongodb:mongo_db_insert_trans: Failed to run query. Err 
= 2, 32 , 0
 opensips segfault at 4 ip 7fb1eb1c3fbc sp 7fff267b0660 error 
4 in libmongoc.so.0.6.0[7fb1eb1bd000+13000]
 ACC: transaction answered: 
timestamp=1452059690;method=BYE;from_tag=as45b598fe;to_tag=988c2c314cc545cabb60b0c7c09c1aed;call_id=751c426ebacc4a03aba5ea3e326dcb91;code=200;reason=OK
 ERROR:cachedb_mongodb:mongo_db_insert_trans: Failed to run query. Err 
= 2, 32 , 0
 segfault at 4 ip 7fb1eb1c3fbc sp 7fff267b0660 error 4 in 
libmongoc.so.0.6.0[7fb1eb1bd000+13000]

 CRITICAL:core:receive_fd: EOF on 15
 INFO:core:handle_sigs: child process 20023 exited by a signal 11
 INFO:core:handle_sigs: core was generated
 INFO:core:handle_sigs: terminating due to SIGCHLD
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:sig_usr: signal 15 received
 INFO:core:cleanup: cleanup
 NOTICE:db_cachedb:destroy: destroy module db_cachedb ...
 NOTICE:cachedb_mongodb:destroy: destroy module cachedb_mongodb ...


 Please help me if I am doing something wrong . Thank you in advance .


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

On Tue, Jan 5, 2016 at 7:22 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Sasmita,

you say when your mongos daemon is stopped then opensips is
crashing (on the next attempt of a db query) ?

Regards,

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

On 30.12.2015 14:28, Sasmita Panda wrote:

Hi All,

  I am using opensips-1.11 with cachedb_mongodb and
db_cachedb integrated on it .  Mongodb and Opensips is running in
different machines .

  Whenever mongodb is getting stopped opensips used to
crash . Is this a drawback or I am doing any mistake in my
configuration .

Bellow is the configuration I have done :
loadmodule "cachedb_mongodb.so"
loadmodule "db_cachedb.so"

modparam("cachedb_mongodb",

"cachedb_url","mongodb:instance1://opensips:opensipsmongo@x.x.x.x:27017/db.CCinfo"
)


modparam("db_cachedb","cachedb_url","mongodb:instance1://opensips:opensipsmongo@x.x.x.x:27017/db.db"
)

modparam("siptrace","db_url","cachedb://mongodb:instance1")


   I am using it for siptrace module and storing some data
through cache_store function . Please let me know if I can
overcome this problem or also let me know if connection with
 mongodb is always needed for successful running of opensips or
what .


Any kind of information is really appreciated .

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


___
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