Re: [OpenSIPS-Users] opensips and syslog facility/level

[Muhammad Shahzad]

You can set opensips log level using "debug" parameter.

http://www.opensips.org/Documentation/Script-CoreParameters-1-11#toc35

Also make sure your syslog service is configured to send local2.* logs to
the log file(s) where you want to see them. i.e. to see all opensips logs
file named opensips.log set /etc/rsyslog.conf to have entry like this,

local2.*  /var/log/opensips.log

Then restart the service.

Hope this helps.

Thank you.




On Wed, Aug 6, 2014 at 10:49 AM, Oleksandr Kunytsia 
wrote:

> Hello,
>
> Logging of my opensips looks like the following:
>
> cut
> log_facility=LOG_LOCAL2
> log_stderror=no
> cut
>
> Opensips sends logging information as LOCAL2.error messages,
>
> How to configure to send messages with another log_level? e.g. local2.info
> ?
>
> ___
> 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] cachedb_redis :: cachedb_url => FQDN not working

[Muhammad Shahzad Shafi]

 

Possibly DNS error, can you try to telnet on this address on
opensips machine and see if you get connected. If not then it is DNS
problem, rather then opensips. 

Thank you. 

On 2014-07-10 22:00, Gary
Nyquist wrote: 

> I am setting the FQDN like this: 
>
modparam("cachedb_redis",
"cachedb_url","redis:cluster://rediscluster.vhrurm.0001.use1.cache.amazonaws.com:6379/")

> 
> No complains by "opensips -c" 
> On starting, the log gets filled
with error messages, starting with the following line: 
>
ERROR:cachedb_redis:redis_connect_node: failed to open redis connection
rediscluster.vhr�#030:6379 - Name or service not known 
> 
> BR 
> -Gary

> 
> SENT: Thursday, July 10, 2014 at 12:41 PM
> FROM: "Bogdan-Andrei
Iancu" 
> TO: "OpenSIPS users mailling list" , "Gary Nyquist" 
>
SUBJECT: Re: [OpenSIPS-Users] cachedb_redis :: cachedb_url => FQDN not
working 
> 
> Hi,
> 
> By not working, what do you mean ? do you get
some error ? simply not connecting ?
> 
> Regards, 
> 
> Bogdan-Andrei
Iancu
> OpenSIPS Founder and Developer
>
http://www.opensips-solutions.com
> On 10.07.2014 18:59, Gary Nyquist
wrote: 
> 
>> Hi, 
>> 
>> I am trying to use the FQDN for specifying the
'cachedb_url' like this: 
>> modparam("cachedb_redis",
"cachedb_url","redis:cluster://my_redis_server.com:6379/") 
>> But it is
not working; 
>> 
>> Whereas the following is working fine:
>>
modparam("cachedb_redis",
"cachedb_url","redis:cluster://10.0.0.228:6379/") 
>> 
>> What needs to
be done to make the FQDN to work? 
>> 
>> Thanks
>> -Gary 
>> 
>>
___
>> Users mailing list
>>
Users@lists.opensips.org
>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 
Mit
freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Avoid waiting in opensips.cfg

[Muhammad Shahzad Shafi]

 

Its simple, set invite try timer to appropriate value,


http://www.opensips.org/html/docs/modules/1.11.x/tm.html#fr_timeout


then send it to some impossible path, 

sip:$tU@127.0.0.1:5060 

after
try timer times out, you will get back the original invite in failure
route, where you can send it actual destination to send it again to
"go-around" trip to impossible destination. Of course while this is
happening, you should send fake "180 Ringing" message to caller so s/he
won't suffer from dead-air silence. 

Thank you. 

On 2014-07-10 22:17,
Gary Nyquist wrote: 

> Hi, 
> 
> I am looking for some idea for the
following use case: 
> 
> 1. opensips receives initial INVITE from the
caller.
> 2. finds that the callee is not registered.
> 3. sends a
non-SIP message to the callee device to register.
> 4. callee registers
and opensips routes the INVITE to the callee contact uri. 
> 
> The
calle device may take around 10 seconds to register back (after sending
the non-SIP message).
> Obvisously, it is not advisable to wait in the
opensips.cfg for such a long time. 
> Can anyone suggest a smart way to
handle this? 
> 
> Thanks
> -Gary

-- 
Mit freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] MediaProxy questions

[Muhammad Shahzad Shafi]

 

OK, let me try to explain. 

Mediaproxy consists of two components,
the dispatcher and the relay. The dispatcher is kind of controller for
relay component. It is usually installed on same machine as opensips and
can be used to control many relays running on different machine in
LAN/WAN. This design is so flexible that you can have relays running in
different geographical locations and you can control them all using
central dispatcher. 

Secondly installed of compiling media proxy
yourself, better to just use pre-built packages. Just add the apt
repository suitable for you OS at,


http://projects.ag-projects.com/projects/documentation/wiki/Repositories


and run this, 

sudo apt-get update
sudo apt-get install
mediaproxy-dispatcher mediaproxy-relay mediaproxy-web-sessions

For more
details, see
this,

http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/InstallationGuide

It
will install all dependencies along with relevant components of media
proxy.

Lastly, the ip forwarding is only need for at media relay, since
this component actually handles the media exchange. 

Hope this helps.


Thank you. 

On 2014-07-10 23:44, Gary Patton wrote: 

> Hello. Some
MediaProxy questions. 
> 
> I've been trying to follow these
installation how-tos and OpenSIPS integration pages:
http://mediaproxy.ag-projects.com,
http://voiprookie.blogspot.com/2009/04/blog-post.html,
http:www.smartvox.co.uk/serfaq_install_mediaproxy2.html, and
http://antechxxx.blogspot.com/2012/06/how-to-manual-installation-mediaproxy.html.
Those are useful in their own way (and I thank the authors), but I'm
still not 100% clear. 
> 
> First - it's terrible but I'm not sure which
is the dispatcher and which is the relay. I believe OpenSIPS is the
dispatcher and MediaProxy is the relay. 
> 
> Second - The VoIP Rookie
blog says 
> 
> In order to build and install, MediaProxy has the
following requirements: 
> - Linux (at least 2.6.18) with the following
features compiled in: 
> - netfilter support 
> - connection tracking
support 
> - connection tracking netlink interface 
> - connection
tracking event notification API 
> - netfilter "NOTRACK" target support

> - netfilter "CONNMARK" target support 
> - netfilter "connmark" match
support 
> - IPv4 connection tracking support 
> - IP tables support 
>
- IP tables Full NAT support 
> 
> I looked into compiling the Linux
kernel and it seems kind of crazy that I need to compile (or recompile)
my Linux distro in order to install/run MediaProxy. I've searched how to
compile the kernel with those modules/features and I can't find anything
specific, just general instructions on how to compile the kernel from
source. (If anyone knows of something specific to MediaProxy please
point me in the right direction!) 
> 
> Can't I just install netfilter,
iptables, conntrack, etc, support for MediaProxy via the "apt-get
install" method instead of compiling those into the kernel? 
> 
> Third
- I'm making a big assumption that you'd install OpenSIPS and MediaProxy
on separate boxes in a production network. If so, I'm not sure which box
I would permanently configure IP forwarding on? The dispatcher
(OpenSIPS) or the relay (MediaProxy)? Or is it both? 
> 
> I would
really appreciate some direction since it's very confusing to me. Thank
you very much. 
> 
> Regards 
> 
> Gary

-- 
Mit freundlichen
Grüßen
Muhammad Shahzad
---
CISCO Rich
Media Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Problem about the audio conference

[Muhammad Shahzad]

I think you have posted the problem on wrong mailing list. OpenSIPS is NOT
an MCU for A/V conferences. Also whatever problem you have seems to be
related to linphone, NOT opensips.

Thank you.



On Tue, Jun 10, 2014 at 4:00 AM, CC  wrote:

> Hi, everyone.
>
> Now I try to merge some calls into a conference, but I have some problems.
>
> The description of the problems is as flows:
>
> First
>
> In the class which named “FriendAdapter.class”, when I click the item of
> the listview which contains the information of friends, I wrote a method to
> call the selected friend’s sip: ”* LinphoneManager*.*getInstance*
> ().callTo(sipid);”.Then in the class which named “LinphoneManager.class”,
> I add the “addToConference(LinphoneCall)” in the method “callTo(String
> address)” .
>
> Then, I try to test the function. Connection can be established, but both
> the callers cannot hear to each other. And I make a LOG message in the code
> to show the size of the conference and to indicate whether the local user
> is part of the conference. The message shows the size of the conference is
> “1” and the indication whether the local user is part of the conference is
> “false”. But in the doc of Linphone, I see this:
>
> void *addToConference*(LinphoneCall
> 
>  call)
>
> Merge a call into a conference. If this is the first call that enters the
> conference, the virtual conference will be created automatically. If the
> local user was actively part of the call (ie not in paused state), then the
> local user is automatically entered into the conference. If the call was in
> paused state, then it is automatically resumed when entering into the
> conference.
>
> But in my test result, the local user wasn’t automatically entered into
> the conference. I don’t know why.
>
> Hope you put forward guidance to help me solve the problems, Thank you.
>
>
>
> ___
> 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] Fwd: RTPproxy project

[Muhammad Shahzad Shafi]

 

To be honest, i have stopped using rtpproxy for over 2 years now. It
is not evolving as fast as it should be, specially in the context of ICE
and WebRTC technologies. 

I would like to suggest that opensips team
should consider adding support for rtpengine from SIPWise,


https://github.com/sipwise/rtpengine 

For now mediaproxy from AG
Projects is the only good choice for handling media in opensips with ICE
support (though it still lacks WebRTC features). 

Thank you. 

On
2014-05-23 14:55, Bogdan-Andrei Iancu wrote: 

> Going for a public
exposure on this question to Maxim, maybe we will get an answer here.
>

>  Original Message  
> 
> SUBJECT:
> RTPproxy
project
> 
> DATE:
> Mon, 14 Apr 2014 15:03:31 +0300
> 
> FROM:
>
Bogdan-Andrei Iancu
> 
> TO:
> Maxim Sobolev
> 
> CC:
> Razvan Crainea
>

> Hello Maxim,
> 
> Long time, no talks, but I hope everything is fine
on your side.
> 
> I'm reaching you in order to ask about your future
plans in regards to 
> the rtpproxy project? We see no much activity
around it and other media 
> relays are popping around.
> 
> RTPP is an
essential component for us, we invested a lot of work, we 
> have many
patches (extensions) for it (which we want to push to the 
> public
tree, but there is no answer on this) and we are also looking for 
>
investing a lot into big future plans (as adding more
functionalities).
> 
> Now, my question is - what is your commitment and
disponibility for the 
> RTPP project ? depending on that we what to
re-position ourselves, as we 
> do not want to waste time and work on
things which are out of control.
> 
> Best regards,
> 
> -- 
>
Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
>
http://www.opensips-solutions.com

-- 
Mit freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Adjusting Headers

[Muhammad Shahzad]

TO header contains DNIS, so be careful when editing it. Anyways, the
easiest way is to use textops module,

http://www.opensips.org/html/docs/modules/1.10.x/textops.html#id249986

You can also use uac module, (recommended)

http://www.opensips.org/html/docs/modules/1.10.x/uac.html#id250102

Thank you.




On Thu, Feb 6, 2014 at 1:38 AM, Alectronic wrote:

> Hi All,
> Thanks my device is now able to register with other device.
>
> However I now have another problem which is the other end does not trying
> to
> authenticate when I send an Invite (It just hang there) I have compared a
> trace call of a freepbx and a SIP phone over wireshark connecting to the
> other endpoint and it would seem I need to manipulate the header so that
> the To header is that of the register IP/domain.
>
> What would it take to be able to do this manipulation?
>
> Thanks
>
> Alec
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/no-subject-tp7589434p7589472.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] call control module

[Muhammad Shahzad Shafi]

 

All SIP requests generated by OpenSIPS appear in local_route. So you
can filter BYE generated by opensips there. 

Also using is_direction
method you can determine who sent BYE (caller or callee).


http://www.opensips.org/html/docs/modules/devel/rr.html#id293720


Since BYE generated by opensips is sent in both directions, so you
probably want to do accounting only for one BYE (to avoid duplication).


Thank you. 

On 2014-01-27 23:39, Eddie Chan wrote: 

> Hi all, 
> 
>
I am having problem triggering CDR when the max call duration in call
control module timeout. 
> 
> For a normal call, I use setflag in the
main routing logic to trigger CDRs. 
> 
> route( 
> 
> … 
> 
> If
(is_method("BYE")) { 
> 
> … 
> 
> setflag(AAA_DO); 
> 
> } 
> 
>
However, when the call control timer expired, it will generate two BYE
messages to each endpoints. Since the BYE messages were not originated
by the endpoints, the main routing loop cannot detect the BYE message
and thus failed to generate CDR. 
> 
> Can anyone give me some idea on
this problem? Where should I put the setflag if the BYE messages were
originated by Opensip itself? 
> 
> Thanks, 
> 
> Eddie

-- 
Mit
freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SIP Registration in a Loadbalancing environment

[Muhammad Shahzad]

** ")
>>> modparam("auth_db", "load_credentials", "$avp(password)=password")
>>>
>>> # - module auth ---
>>> loadmodule "auth.so"
>>> # - auth params -
>>> modparam("auth","username_spec","$var(username)")
>>> modparam("auth","password_spec","$avp(password)")
>>> modparam("auth","calculate_ha1",1)
>>> *modparam("auth","disable_nonce_check", 1)*
>>
>>
>>
>>
>>
>>  if (is_method("REGISTER"))
>>> {
>>> xlog("L_INFO","$ci -- New REGISTER received from $si with
>>> Contact : $ct\n");
>>>
>>> if (!www_authorize("", "subscriber"))
>>> {
>>> if ($rc < 0)
>>> {
>>> switch ($rc)
>>> {
>>> case -5:
>>> xlog("L_INFO","$ci -- REGISTER Failed because of :
>>> Generic Error");
>>> break;
>>> case -4:
>>> xlog("L_INFO","$ci -- REGISTER Failed because of :
>>> No Credentials");
>>> break;
>>> case -3:
>>> xlog("L_INFO","$ci -- REGISTER Failed because of :
>>> Stale nonce");
>>> break;
>>> case -2:
>>> xlog("L_INFO","$ci -- REGISTER Failed because of :
>>> Valid User but Wrong Password");
>>> break;
>>> case -1:
>>> xlog("L_INFO","$ci -- REGISTER Failed because of :
>>> Invalid User");
>>> break;
>>>     }
>>> }
>>> www_challenge("", "0");
>>> exit;
>>> }
>>>
>>> if (!save("location"))
>>> {
>>> xlog("L_INFO","$ci -- error with save_location from $au\n");
>>> }
>>> else
>>> {
>>> xlog("L_INFO","$ci -- save_location is OK from $au\n");
>>> }
>>>
>>> exit;
>>> }
>>
>>
>>
>> So, as you can see, I configured the auth module with
>> "disable_nonce_check" parameter, because of my "loadbalanced" architecture
>> as it's said in the documentation (
>> http://www.opensips.org/html/docs/modules/1.9.x/auth.html#id250075) .
>>
>> But, when a SIP Phone tries to register, the first Register (without any
>> credentials) is sent to the 1st Registrar. It's answered with a 401
>> Unauthorized containing a nonce.
>> Then, the 2nd Register (with credentials, and the previously given nonce)
>> is sent to the 2nd Registrar; but it's still answered with a 401.
>>
>> Thanks to the return code of www_authorize, I see that it's for the
>> "Stale Nonce" reason, even if "disable_nonce_check" is set to 1 ...
>>
>> Maybe there's a misconfiguration, or a bug; so, I need your help :-)
>>
>> Thanks a lot,
>>
>>
>>
>> *Bien cordialement, Best Regards,  **Kevin MATHY* | Ingénieur VoIP
>>
>>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS not processing responses?

[Muhammad Shahzad Shafi]

g received by our registrar.
There 
>>> also
>>> doesn't appear to be any issues with the
retransmission by OpenSIPS 
>>> at
>>> the right intervals, nor does the
registrar appear to be introducing
>>> any delays. There is not much
other traffic going on at the same 
>>> time,
>>> but there are other
REGISTERs that are getting the same result.
>>> 
>>> Here are some logs
from OpenSIPS:
>>> Initial relay to our registrar:
>>> Nov 1 19:14:40
ip-10-72-7-129 rcs-opensips[13839]: RCS INFO:
>>> ROUTE[2] (Inbound and
Outbound) - REGISTER REQUEST relayed. Exiting.
>>> Nov 1 19:14:40
ip-10-72-7-129 rcs-opensips[13839]: DBG:tm:t_unref:
>>> UNREF_UNSAFE:
[0x7f8f8b8d6518] after is 0
>>> Nov 1 19:14:40 ip-10-72-7-129
rcs-opensips[13839]:
>>> DBG:core:destroy_avp_list: destroying list
(nil)
>>> 
>>> Retransmissions:
>>> Nov 1 19:14:40 ip-10-72-7-129
rcs-opensips[13836]:
>>> DBG:tm:retransmission_handler:
retransmission_handler : request
>>> resending (t=0x7f8f8b8d6518,
REGISTER ... )
>>> Nov 1 19:14:41 ip-10-72-7-129
rcs-opensips[13836]:
>>> DBG:tm:retransmission_handler:
retransmission_handler : request
>>> resending (t=0x7f8f8b8d6518,
REGISTER ... )
>>> Nov 1 19:14:43 ip-10-72-7-129
rcs-opensips[13836]:
>>> DBG:tm:retransmission_handler:
retransmission_handler : request
>>> resending (t=0x7f8f8b8d6518,
REGISTER ... )
>>> Nov 1 19:14:53 ip-10-72-7-129
rcs-opensips[13836]:
>>> DBG:tm:retransmission_handler:
retransmission_handler : request
>>> resending (t=0x7f8f8b8d6518,
REGISTER ... )
>>> Nov 1 19:15:04 ip-10-72-7-129
rcs-opensips[13836]:
>>> DBG:tm:retransmission_handler:
retransmission_handler : request
>>> resending (t=0x7f8f8b8d6518,
REGISTER ... )
>>> 
>>> Reply "received" (IP,s TNs, and domains
anonymized):
>>> Nov 1 19:15:08 ip-10-72-7-129 rcs-opensips[13831]:
>>>
DBG:tm:t_reply_matching: REF_UNSAFE:[0x7f8f8b8d6518] after is 1
>>> Nov
1 19:15:08 ip-10-72-7-129 rcs-opensips[13831]:
>>>
DBG:tm:t_reply_matching: reply matched (T=0x7f8f8b8d6518)!
>>> Nov 1
19:15:08 ip-10-72-7-129 rcs-opensips[13831]: DBG:tm:t_check:
>>>
end=0x7f8f8b8d6518
>>> Nov 1 19:15:08 ip-10-72-7-129
rcs-opensips[13831]:
>>> DBG:tm:reply_received: org. status uas=0,
uac[0]=0 local=0
>>> is_invite=0)
>>> Nov 1 19:15:08 ip-10-72-7-129
rcs-opensips[13831]: RCS INFO:
>>> PROCESSING ONREPLY ROUTE[2] (Reply
for Inbound Request)
>>> Nov 1 19:15:08 ip-10-72-7-129
rcs-opensips[13831]: RCS DEBUG:
>>> ONREPLY_ROUTE[1] (Reply for Inbound
Request) - Dump Request Info.
>>> Nov 1 19:15:08 ip-10-72-7-129
rcs-opensips[13831]: RCS DEBUG:
>>> = DUMP REPLY
=
>>> Nov 1 19:15:08 ip-10-72-7-129 rcs-opensips[13831]: RCS
DEBUG: SIP
>>> message buffer:#012SIP/2.0 401 Unauthorized#015#012Via:
SIP/2.0/UDP
>>>
x.x.x.x:6000;branch=z9hG4bKf439.62091482.0;i=c04c3,SIP/2.0/TCP
>>> 
>>>
x.x.x.x:40042;branch=z9hG4bK1490773660;received=x.x.x.x;rport=40042#015#012Record-Route:
>>>
#015#012Call-ID:
>>>
6d76ce9e-ed19-9701-1e75-f070eacd3400#015#012From:
>>>
;tag=330928025#015#012To:
>>> ;tag=Jmuh3YOvEmCA#015#012CSeq:
221716357
>>> REGISTER#015#012Contact: #015#012Require:
>>>
gruu#015#012WWW-Authenticate: Digest
>>> 
>>>
realm="domain.net",nonce="5273fda02d788dccba9301c8e1f68f078f8e95e5",qop="auth",opaque="004533235332435434ffac663e",algorithm=MD5#015#012Content-Length:
>>>
0#015#012P-Associated-URI:
>>> #015#012P-Preferred-Identity:
>>>
#015#012P-Access-Network-Info:
>>>
ADSL;utran-cell-id-3gpp=#015#012Privacy:
none#015#012#015#012
>>> 
>>> On the registrar the logs appear as
follows:
>>> [2013-Nov-01 19:14:40.185708] [14002] [6-inf] ===RECV==>
REGISTER
>>> (6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx =>
+xx)
>>> [2013-Nov-01 19:14:40.188331] [14002] [6-inf] 
>>>
+xx)
>>> [2013-Nov-01 19:14:40.721011] [14002] [6-inf]
===RECV==> REGISTER
>>>
(6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx => +xx)
>>>
[2013-Nov-01 19:14:40.721162] [14002] [6-inf] 
>>> +xx)
>>>
[2013-Nov-01 19:14:41.723160] [14002] [6-inf] ===RECV==> REGISTER
>>>
(6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx => +xx)
>>>
[2013-Nov-01 19:14:41.723262] [14002] [6-inf] 
>>> +xx)
>>> 
>>>
etc
>>> 
>>> This "bogging down" seems to happen on a regular basis,
and the only
>>> way we are able to resolve it at the moment is to
restart OpenSIPS.
>>> Until it is restarted all of the registrations are
essentially
>>> failing. I believe it is v1.8.2 that we are running.
>>>

>>> Anyone have any thoughts as to what might be happening?
>>> 
>>>
Thanks,
>>> 
>>> Gavin
> 
> -- 
> Gavin Murphy
> Vice President & CTO |
www.newpace.ca - Real Technology Solutions 
> (e)
gavin.mur...@newpace.ca
> (w) +1.902.406.8375 x 1002
> (m)
+1.902.401.9445
> (f) +1.902.406.8377

-- 
Mit freundlichen
Grüßen
Muhammad Shahzad
---
CISCO Rich
Media Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 

Links:
--
[1]
mailto:users-requ...@lists.opensips.org
[2]
http://www.opensips.org/Documentation/Script-CoreParameters-1-8#toc75
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS not processing responses?

[Muhammad Shahzad Shafi]

ows:
[2013-Nov-01 19:14:40.185708] [14002] [6-inf] ===RECV==> REGISTER
(6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx => +xx)
[2013-Nov-01 19:14:40.188331] [14002] [6-inf] <==SENT=== 401
Unauthorized (6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx =>
+xx)
[2013-Nov-01 19:14:40.721011] [14002] [6-inf] ===RECV==> REGISTER
(6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx => +xx)
[2013-Nov-01 19:14:40.721162] [14002] [6-inf] <==SENT=== 401
Unauthorized (6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx =>
+xx)
[2013-Nov-01 19:14:41.723160] [14002] [6-inf] ===RECV==> REGISTER
(6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx => +xx)
[2013-Nov-01 19:14:41.723262] [14002] [6-inf] <==SENT=== 401
Unauthorized (6d76ce9e-ed19-9701-1e75-f070eacd3400:+xx =>
+xx)

etc

This "bogging down" seems to happen on a regular basis, and the only
way we are able to resolve it at the moment is to restart OpenSIPS.
Until it is restarted all of the registrations are essentially
failing. I believe it is v1.8.2 that we are running.

Anyone have any thoughts as to what might be happening?

Thanks,

Gavin


--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com

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

Re: [OpenSIPS-Users] How to remove or uninstall opensips completely ?

[Muhammad Shahzad Shafi]

Nope, this command won't uninstall it completely. For example what are 
you going to do about osip* files?


However, personally i request opensips development team to add "make 
uninstall" command. I would be very handy in upgrade or downgrading 
opensips between major versions, since opensips is mostly installed 
manually, specially in production environments, rather then by package 
manager of respective OS.


Thank you.


On 2013-10-31 12:22, uservoip0 wrote:

Hi,

Follow below procedure:
Firstly run this on your bash prompt

find / -name opensip*

thereafter you will get all list of opensips files and Directory then 
remove

them by using rm command
and then you will delete these files :
osipsconsolerc
osipsconsole
osipsconfig
osipsconsole
these all files related to opensips you can also find them by 
following find

command and delete them
after this download to opensips source code and then compile it

it must work because I have followed this procedure many time and 
it's

worked for me.

if you are good in scripting then you can also write script by 
following to

those command whatever I have you , which will delete every file and
directly with in one run.

thanks





--
View this message in context:

http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-remove-or-uninstall-opensips-completely-tp7588292p7588322.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


--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com

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

Re: [OpenSIPS-Users] Help - WARNING: database engine not found - tried 'MYSQL'

[Muhammad Shahzad Shafi]

 

I think the message suggest issue with opensipsctl rather then
mysql. For some reason this script is unable to to find or execute
opensipsctl.mysql script that should be installed by "make install"
command. 

Can you run following command in parent folder where opensips
is installed? 

find . -name 'opensipsctl.mysql' 

This should find this
file and indicating its relative path to current folder, e.g. on my
debian x64 machine this script is installed at,


./lib64/opensips/opensipsctl/opensipsctl.mysql 

If nothing is found
then you should simply re-run "make install" and carefully read the
output of it to see if this script was now installed successfully or
not. 

A side note, since i am not following this thread from the very
beginning, so i am not sure which OS you are using. If you are happen to
be using any RPM class OS such as CentOS, Redhat, Fedora or Open Suse
etc. then make sure that you have the Devil's Whorx i.e. SELinux
disabled, otherwise it won't let opensips run in peace. 

Thank you.


On 2013-10-31 14:32, Luis Pérez Urteaga wrote: 

> Yes, but the
message WARNING: database engine not found - tried 'MYSQL' persist.
> 
>
can you helpme?
> 
>> Date: Thu, 31 Oct 2013 09:26:23 +0100
>> From:
raz...@opensips.org
>> To: users@lists.opensips.org
>> Subject: Re:
[OpenSIPS-Users] Help - WARNING: database engine not found - tried
'MYSQL'
>> 
>> Hi, Luis!
>> 
>> Have you configured OpenSIPS to use
mysql in the menuconfig tool? This 
>> should be done before installing
it.
>> 
>> Best regards,
>> 
>> Răzvan Crainea
>> OpenSIPS Core
Developer
>> http://www.opensips-solutions.com
>> 
>> On 10/30/2013
11:08 PM, Luis Pérez Urteaga wrote:
>> > Wilmar,
>> >
>> > This is my
list of modules:
>> >
>> > [root@SDEVCCLM47 /]# opensipsctl start
>> >
WARNING: database engine not found - tried 'MYSQL'
>> > INFO: Starting
OpenSIPS :
>> > INFO: started (pid: 5932)
>> > [root@SDEVCCLM47 /]#
opensipsctl stop
>> > WARNING: database engine not found - tried
'MYSQL'
>> > INFO: Stopping OpenSIPS :
>> > INFO: stopped
>> >
[root@SDEVCCLM47 /]# ls /usr/local/lib64/opensips/modules/
>> > acc.so
cachedb_sql.so dispatcher.so exec.so
>> > mi_fifo.so
presence_callinfo.so sipmsgops.so tm.so
>> > alias_db.so call_control.so
diversion.so gflags.so
>> > msilo.so presence_xcapdiff.so siptrace.so
uac_auth.so
>> > auth_aaa.so cfgutils.so dns_cache.so group.so
>> >
nathelper.so qos.so sl.so uac_redirect.so
>> > auth_db.so closeddial.so
domainpolicy.so imc.so
>> > nat_traversal.so ratelimit.so sms.so
uac_registrant.so
>> > auth_diameter.so db_cachedb.so domain.so
load_balancer.so
>> > options.so registrar.so speeddial.so uac.so
>> >
auth.so db_flatstore.so drouting.so mangler.so
>> > path.so rr.so sst.so
uri.so
>> > avpops.so db_mysql.so enum.so mathops.so
>> > pdt.so
rtpproxy.so statistics.so userblacklist.so
>> > b2b_entities.so
db_text.so event_datagram.so maxfwd.so
>> > peering.so seas.so stun.so
usrloc.so
>> > benchmark.so db_virtual.so event_route.so
mediaproxy.so
>> > permissions.so signaling.so textops.so
>> >
cachedb_local.so dialog.so event_xmlrpc.so mi_datagram.so
>> > pike.so
sipcapture.so tlsops.so
>> > [root@SDEVCCLM47 /]#
>> >
>> >
>> >
>> >
>>
>
>> > ___
>> > 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

-- 
Mit
freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenXcap Wiki Configuration Clarification

[Muhammad Shahzad Shafi]

The easiest way is to download the openxcap trunk. For this you will 
need to install "darcs" first.


apt-get install darcs

Then download openxcap trunk,

darcs get http://devel.ag-projects.com/repositories/openxcap

then switch to openxcap directory, where you can find everything you 
need.


Note, you don't need to install trunk version (specially on a 
production system), just copy whatever you need from it then delete it. 
Installation instructions can be found here,


http://openxcap.org/projects/openxcap/wiki/Installation

Thank you.


On 2013-10-24 14:11, Rob Taggart wrote:

I tried that command and nothing showed up...under what top level
directory does the openxcap directory live? Cant seem to find it

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of voip user
Sent: Thursday, October 24, 2013 4:57 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] OpenXcap Wiki Configuration 
Clarification


Hi ,

By following below steps you will get scriptes for table creation.

# cd openxcap-2.1.0/scripts/

[root@localhost scripts]# ls -l
total 12
-rwxr-xr-x. 1 1000 1000 580 Apr 26  2010 add-openxcap-user.py
-rw-r--r--. 1 1000 1000 776 Apr  6  2010 mysql-create-tables.sql
-rw-r--r--. 1 1000 1000 302 Apr 26  2010 mysql-create-user.sql
[root@localhost scripts]#





On 10/24/13, Rob Taggart  wrote:
I have followed the Wiki to install Openxcap and have now reached 
the

point in the config section to create a database since I do not use
OpenSips.  I am confused as to where to find the scripts directory
that contains the scripts to create the tables.  I am new to linux 
and

somewhat green but learning.  Would someone please point me in the
right direction as to where to find the scripts and how to launch 
them from the Ubuntu command line?






___
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


--
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com

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

Re: [OpenSIPS-Users] fr_timer not working properly

[Muhammad Shahzad Shafi]

 

This is pretty strange, since it never happened to me on an AMD
machine, though i have different model, AMD Athlon II X2 250e. 

Can you
try using fr_timer_avp? e.g. 

modparam("tm", "fr_timer_avp",
"$avp(10)") 
modparam("tm", "fr_inv_timer_avp", "$avp(20)") 

See more
info,


http://www.opensips.org/html/docs/modules/1.10.x/tm.html#id293753 [3]


Set these vars and print xlog when INVITE comes in. 

branch_route {

... 

 $avp(10) = 5; 
 $avp(20) = 60; 
 xlog("L_NOTICE",
"[$pr:$fU@$si:$sp]: Setting TRY timer to '$avp(10)' and RING timer to
'$avp(20)' for call from '$fu' to '$ru' n"); 
... 
} 

Then check in
failure route, 

failure_route { 
... 

 if (t_check_status("408")) { 

if ( t_local_replied("all") ) { 
 # local timeout with no reply received
-> fr_timer 
 xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Timeout on TRY timer
for '$rm' from '$fu' to '$ru' after '$avp(10)' seconds n"); 
 } else if
( t_local_replied("last") ) { 
 # timeout with replies received ->
fr_inv_timer 
 xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Timeout on RING
timer for '$rm' from '$fu' to '$ru' after '$avp(20)' seconds n"); 
 }
else { 
 # received timeout 
 xlog("L_NOTICE", "[$pr:$fU@$si:$sp]:
Timeout on T2 timer for '$rm' from '$fu' to '$ru' after '$avp(20)'
seconds n"); 
 }; 
 }; 
... 
} 

This will make things clearer if there
is any problem. 

Thank you. 

On 2013-10-25 09:35, mayamatakeshi wrote:


> On Tue, Oct 15, 2013 at 10:09 AM, mayamatakeshi
 wrote:
> 
>> Hello,
>> I have some servers
running opensips 1.10 commit bfd86ab25554082053167a82655e669ca1c0ea7c
>>

>> I am seeing an issue in them that i cannot reproduce in other
servers: the fr_timer is not respected.
>> I set tm fr_timer=5 and use
sipp to make a call and make opensips to send the call to an address
that is not being listened by any app like 127.0.0.1:5010 [1]. In these
servers, sometimes timeout happens after 6, 9, 12 seconds but never (so
far after) 5 seconds. But in other machines, fr_timer is consistently
respected. So I am trying to figure out what would be causing this. 
>>
Does anyone have any idea of configuration of the server that could
cause this? 
>> The only significant difference I see is that on the
problematic servers corosync was installed for HA (to control opensips
switchover) but I don't see how this could have such impact.
> 
> Just
in case someone else gets into this. 
> We verified the problem only
happens if the VM runs on a VM host using this processor:
> 
> model
name : AMD Opteron(tm) Processor 6344 
> 
> On the server where we have
this one: 
> 
> model name : Intel(R) Xeon(R) CPU X5550 @ 2.67GHz there
is no problem.
> 
> Regards,
> Takeshi

-- 
Mit freundlichen
Grüßen
Muhammad Shahzad
---
CISCO Rich
Media Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 

Links:
--
[1]
http://127.0.0.1:5010
[2] mailto:mayamatake...@gmail.com
[3]
http://www.opensips.org/html/docs/modules/1.10.x/tm.html#id293753
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] convert 180 to 183 after the fact

[Muhammad Shahzad Shafi]

 

Yes of course, you need to remove sdp as well while changing reply
from 183 to 180,


http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#id292832
[15] 

Thank you. 

On 2013-09-26 18:21, Jeff Pyle wrote: 

> Muhammad,

> 
> That makes sense. I think in my case I would have to strip the SDP
as well? Any thoughts on the media sent from the b-leg back to the a-leg
when it's not being expected (because there is no SDP)? 
> 
> - Jeff 
>

> On Tue, Sep 24, 2013 at 11:03 PM, Muhammad Shahzad Shafi
 wrote:
> 
>> Well, you have to sacrifice
183 Early Media, since converting 183 to 180 is far more easy and
convenient then converting 180 to 183 (since then you have to involve a
media server, which is not going to be so easy). 
>> 
>> Therefore, my
advice would be to change all 183 from that carrier to 180 response. You
can use change_reply_status method, 
>> 
>>
http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status
[11] 
>> 
>> Thank you. 
>> 
>> On 2013-09-25 03:05, Jeff Pyle wrote:

>> 
>>> No takers? :) 
>>> 
>>> I wonder if it's possible to script
this in a B2BUA scenario? I'm not sure how one would do detection of 180
without SDP versus 180/183 with SDP in B2B-land. Or, what to do from
there once it knew. 
>>> 
>>> - Jeff 
>>> 
>>> On Mon, Sep 23, 2013 at
10:43 AM, Jeff Pyle  wrote:
>>> 
>>>> Hi
Laszlo, 
>>>> 
>>>> Unfortunately the effect for the caller would be the
same - ringback would stop. 
>>>> 
>>>> Here's the whole flow. My
terminating gateway is SIP to ISDN PRI. Call terminates through the
gateway to a particular mobile switching office. I receive an ISDN
PROGRESS message with inband audio. This translates to the 183 with SDP.
Then I receive an ALERTING message with no inband audio. This translates
to the 180. When the MSO sends the ALERTING, it has stopped sending the
inband audio from the previous PROGRESS message. 
>>>> 
>>>> I'm
thinking I need to do something else in the onreply_route to connect to
the media server for a new 183. Since I've executed t_relay to route the
INVITE to the gateway, it seems my options are limited. 
>>>> 
>>>> -
Jeff 
>>>> 
>>>> -- 
>>>> Jeff Pyle 
>>>>
Director, Voice Engineering
>>>> Fidelity Voice and Data
>>>>
216-245-4106
>>>> www.fidelityvoice.com [8]
>>>> 
>>>> On Mon, Sep 23,
2013 at 8:57 AM, Laszlo  wrote:
>>>> 
>>>>>
What if you simply drop the 180 in the onreply_route?
>>>>> 
>>>>>
-Laszlo 
>>>>> 
>>>>> 2013/9/23 Jeff Pyle 
>>>>> 
>>>>>> Hello, 
>>>>>> 
>>>>>> I have one particular PSTN
call flow that causes a 183 with SDP, then a 180 without SDP prior to
200 OK. Some of my customer endpoints don't handle the 180 properly
after a 183 and they cease to hear ringback. 
>>>>>> 
>>>>>> I'm
thinking through how intercept the 180 and convert it to a 183 with SDP.
I have a media server available to generate the 183 and the media. I'm
struggling with how to relay the INVITE to the media server when the 180
arrives in the middle of the call setup. 
>>>>>> 
>>>>>> Any
recommendations are appreciated. 
>>>>>> 
>>>>>> Regards, 
>>>>>> Jeff

>>>>>> ___
>>>>>> Users
mailing list
>>>>>> Users@lists.opensips.org [1]
>>>>>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [2]
>>>>>

>>>>> -- 
>>>>> 
>>>>> -- Kind regards, Laszlo Bekesi
http://voipfreak.net [4] 
>>>>>
___
>>>>> Users mailing
list
>>>>> Users@lists.opensips.org [5]
>>>>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [6]
>> 
>> --

>> Mit freundlichen Grüßen
>> Muhammad Shahzad
>>
---
>> CISCO Rich Media Communication
Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell:
+49 176 99 83 10 85
>> MSN: shari_78...@hotmail.com
>> Email:
shaherya...@googlemail.com
>> 
>>
___
>> Users mailing list
>>
Users@lists.opensips.org [12]
>>
http://lists.opensips.org/cgi-bin/mailman/listin

Re: [OpenSIPS-Users] hangup call when receive 200 OK

[Muhammad Shahzad Shafi]

 

You should decided if call should be answered or not BEFORE
forwarding it to asterisk, OR do this decision on asterisk side. Instead
stopping it at answer event. 

Anyways, there a way to hangup running
calls (that are already established with 200 OK + ACK) using opensips
management interface, have a look at,


http://www.opensips.org/html/docs/modules/1.9.x/dialog.html#id295424
[1] 

Thank you. 

On 2013-09-25 06:12, nguyen khue wrote: 

> Hi all,

> 
> I use OpenSIPS as follow model: 
> 
> SIPClientOpenSIPSAsterisk.

> 
> when user make call to asterisk, asterisk return 200 OK and I can
receive this message in on_reply route block. If I want hangup this
call, how I can do it? Please guide me. 
> 
> Thanks & Best Regards, 
>
Khue.

-- 
Mit freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 

Links:
--
[1]
http://www.opensips.org/html/docs/modules/1.9.x/dialog.html#id295424
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] convert 180 to 183 after the fact

[Muhammad Shahzad Shafi]

 

Well, you have to sacrifice 183 Early Media, since converting 183 to
180 is far more easy and convenient then converting 180 to 183 (since
then you have to involve a media server, which is not going to be so
easy). 

Therefore, my advice would be to change all 183 from that
carrier to 180 response. You can use change_reply_status method,


http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status
[11] 

Thank you. 

On 2013-09-25 03:05, Jeff Pyle wrote: 

> No
takers? :) 
> 
> I wonder if it's possible to script this in a B2BUA
scenario? I'm not sure how one would do detection of 180 without SDP
versus 180/183 with SDP in B2B-land. Or, what to do from there once it
knew. 
> 
> - Jeff 
> 
> On Mon, Sep 23, 2013 at 10:43 AM, Jeff Pyle
 wrote:
> 
>> Hi Laszlo, 
>> 
>>
Unfortunately the effect for the caller would be the same - ringback
would stop. 
>> 
>> Here's the whole flow. My terminating gateway is SIP
to ISDN PRI. Call terminates through the gateway to a particular mobile
switching office. I receive an ISDN PROGRESS message with inband audio.
This translates to the 183 with SDP. Then I receive an ALERTING message
with no inband audio. This translates to the 180. When the MSO sends the
ALERTING, it has stopped sending the inband audio from the previous
PROGRESS message. 
>> 
>> I'm thinking I need to do something else in
the onreply_route to connect to the media server for a new 183. Since
I've executed t_relay to route the INVITE to the gateway, it seems my
options are limited. 
>> 
>> - Jeff 
>> 
>> -- 
>> Jeff Pyle

>> Director, Voice Engineering
>> Fidelity
Voice and Data
>> 216-245-4106
>> www.fidelityvoice.com [8]
>> 
>> On
Mon, Sep 23, 2013 at 8:57 AM, Laszlo 
wrote:
>> 
>>> What if you simply drop the 180 in the onreply_route?
>>>

>>> -Laszlo 
>>> 
>>> 2013/9/23 Jeff Pyle 
>>> 
>>>> Hello, 
>>>> 
>>>> I have one particular PSTN call flow
that causes a 183 with SDP, then a 180 without SDP prior to 200 OK. Some
of my customer endpoints don't handle the 180 properly after a 183 and
they cease to hear ringback. 
>>>> 
>>>> I'm thinking through how
intercept the 180 and convert it to a 183 with SDP. I have a media
server available to generate the 183 and the media. I'm struggling with
how to relay the INVITE to the media server when the 180 arrives in the
middle of the call setup. 
>>>> 
>>>> Any recommendations are
appreciated. 
>>>> 
>>>> Regards, 
>>>> Jeff 
>>>>
___
>>>> Users mailing
list
>>>> Users@lists.opensips.org [1]
>>>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [2]
>>> 
>>> --

>>> 
>>> -- Kind regards, Laszlo Bekesi http://voipfreak.net [4] 
>>>
___
>>> Users mailing
list
>>> Users@lists.opensips.org [5]
>>>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users [6]

-- 
Mit
freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 

Links:
--
[1]
mailto:Users@lists.opensips.org
[2]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[3]
mailto:jp...@fidelityvoice.com
[4] http://voipfreak.net
[5]
mailto:Users@lists.opensips.org
[6]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[7]
mailto:jp...@fidelityvoice.com
[8] http://www.fidelityvoice.com
[9]
mailto:las...@voipfreak.net
[10] mailto:jp...@fidelityvoice.com
[11]
http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Presence xml

[Muhammad Shahzad]

That's because presence_xml does not have db_url parameter anymore.

http://www.opensips.org/html/docs/modules/1.9.x/presence_xml.html

Also when asking for help, it would be better if you mention your opensips
version / revision. This will save everybody's time.

Thank you.




On Wed, Aug 28, 2013 at 12:58 AM, troxlinux  wrote:

> I have already the module
>
> /lib64/opensips/modules/presence_xml.so
>
>
>
> 2013/8/27 troxlinux 
>
>> Hi list , I tried to add presence to my sipserver
>>
>> check my config
>>
>> loadmodule "presence.so"
>> loadmodule "presence_xml.so"
>>
>> ## --presencia modulos
>> modparam("presence|presence_xml", "db_url", "mysql://ossips:@localhost
>> /opensips")
>> modparam("presence_xml", "force_active", 1)
>> modparam("presence", "server_address", "sip:172.x.x.x:5060" )
>>
>>
>> error log:
>>
>>
>> ERROR:core:set_mod_param_regex: parameter  not found in module
>> 
>> Aug 27 15:15:48 sipia opensips: CRITICAL:core:yyerror: parse error in
>> config file /etc/opensips/opensips.cfg, line 86, column 20-21: Parameter
>>  not found in module  - can't set
>> Aug 27 15:15:48 sipia opensips: ERROR:core:main: bad config file (1
>> errors)
>> Aug 27 15:17:45 sipia opensips: WARNING:core:warn: warning in config file
>> /etc/opensips/opensips.cfg, line 16, column 13-16: tls support not compiled
>> in
>> Aug 27 15:17:45 sipia opensips: ERROR:core:set_mod_param_regex: parameter
>>  not found in module 
>> Aug 27 15:17:45 sipia opensips: CRITICAL:core:yyerror: parse error in
>> config file /etc/opensips/opensips.cfg, line 86, column 20-21: Parameter
>>  not found in module  - can't set
>> Aug 27 15:17:45 sipia opensips: ERROR:core:main: bad config file (1
>> errors)
>>
>>
>> --
>> rickygm
>>
>> http://gnuforever.homelinux.com
>>
>
>
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] LCR Route Prefixes -> Gateways (From Scratch)

[Muhammad Shahzad]

Sorry, by mistake i wrote "flat-rate", what i actually meant was
"flat-base", it has nothing to do with billing interval, its mostly Dynamic
Routing and LCR stuff. Let me try to explain here.

Nearly in all countries of the world we have two mediums, namely Mobile
Line and Fixed Line and a subscriber's full number format structure is
quite simply, e.g.

Mobile Line => CC + Operator Code + Subscriber Number (SIM Card Number)
Fix Line => CC + City Code + Subscriber Number (can be expended as Exchange
Number + Actual Subscriber Number, generally insignificant for third party
operator)

The pattern remains constant regardless of time and space of the
subscriber, For example if my cell number is +49 176 12345678, then any one
who wants to call me can simply dial complete number and can reach me
regardless if temporarily,

I move from Wiesbaden to Frankfurt (change of city)
I move from Frankfurt to Berlin (change of state)
I move from Berlin to Paris (change of country)
i move from Paris to NY (change of continent)

But for South American countries the story is completely different, i try
to explain with a fake example. The full number format is something like
this,

+CC ABC... 12345678

Where,

A if present is operator code of 1 to say 5 digits,
B if present is state code of 1 to say 2 digits,
C if present is city code of 1 to say 3 digits,
and so on, many more such variable (both count and range is different in
each country), each changes with subscriber location and even on time of
day or day of week basis...

Also there is inter-city roaming, inter-state roaming, international
roaming and so on. And the worst of all, people can call you even without
any of these variables by simply dialing +CC 12345678. How you route this
number? Well, in this case you have to look on to caller-id to determine
values of A, B, C and so on, which is another painful story of its kind.

Hope this briefly explains the problem. So be prepared, flat base billing
that works perfect in 90% of the world, would not work in Spanish America,
at least not without good-will and assistance from a local operator (or its
partner), that manages all these complexities for you (at probably very
high cost).

Thank you.




On Mon, Aug 5, 2013 at 10:58 PM, Nick Khamis  wrote:

> Muhammad,
>
> I'm sorry to do this, but by flat rate, are you referring to the
> per-second vs. per minute and per six seconds billing?
> I just don't want to assume or overlook anything when you mention
> countries with messed up telecom infrastructure.
>
> I am learning a lot from the few words that we are exchanging. I know
> you're busy, you can respond when you
> get a chance...
>
> >> A tractor and car both run on wheels, but this does not mean you can
> fit in a tractor's wheel in car or vice versa.
>
> I love proverbs. Especially ones from the east that are, or derive from
> ones, that are thousands of years old :)
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] LCR Route Prefixes -> Gateways (From Scratch)

[Muhammad Shahzad]

Well, for flat rate countries like Afghanistan, that will work, either way.
You can pre-process the data like you mentioned above or let some script do
it at run time. But think about countries with much messed up telecom
architecture like Mexico or Caribbeans countries or even worst Brazil. And
if you have inbound carriers as well, like DID / Access number providers,
GIS etc. then things like LCR would literally become hell. However if you
ignore Spanish America for now, then all is OK and you can move on with it,
either way, it doesn't matter much.

And don't worry about reinventing the wheel, A tractor and car both run on
wheels, but this does not mean you can fit in a tractor's wheel in car or
vice versa. ;-)

Thank you.




On Mon, Aug 5, 2013 at 7:53 PM, Nick Khamis  wrote:

> Correction,
>
> *dr_gateways*
>
>
>
> ++--+--+---+---++---+++
> | id | gwid | type | address| strip | pri_prefix |
> attrs | probe_mode | description |
>
> ++--+--+---+---++---+++
> |  1 | 0|2| carrier_address:5060 | 0 | 0011101| yes
> |  0 | Carrier 0   |
>
> ++--+--+---+---++---+++
> |  1 | 1|2| carrier_address:5060 | 0 | |
> yes   |  0 | Carrier 1   |
>
> ++--+--+---+---++---+++
> .
>
> N.
>
> ...
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] LCR Route Prefixes -> Gateways (From Scratch)

[Muhammad Shahzad]

Well, a very basic sign of good piece of software, that is driven by
database,is that it ALWAYS uses column names, instead of index numbers.
OpenSIPS not only confirm to this very basic standard but it further
improves it by allowing user to define table and column names as module
parameter that will be internally called by module through SQL queries. So,
yes adding more columns should not and will not effect its processing.
However, be sure to only add columns that are really need or at least fit
to the philosophy of the module.

In my personal view routing and rating are two distinct entities and for
sake of data independence they should not be mixed together in a single
table. Perhaps you should to be interested in call-control module or
something similar. If you are good in PERL or LUA you can even write up
your own module for rating and billing, and it would work like a charm even
for high loads, as long as you keep your database optimized. And yes, you
can make such rating / billing module on commercial license.

Thank you.




On Mon, Aug 5, 2013 at 6:00 PM, Nick Khamis  wrote:

> Hello Muhammad,
>
> I have no problem updating the DB script and relevant code using optimized
> indexing/views, and sharing that with the community. It's about time I
> contribute in some form or another... :)
>
> Furthermore, I would also like to share the final DB schema and script in
> charge of creating an OpenSIPS LCR instance with the community. And would
> be happy to see it posted on the OpenSIPS Tut section. I will have some
> questions to steer me to the right direction however, will keep them short
> and connect the dots without taking too much of the community's time.
>
> My first question will be, is it safe to add additional fields to the
> `dr_rules` table. What I will need are things like (bill_minimum,
> bill_increment, list/retail_rate etc..). Things we usually see in a rate
> deck ;). What I am asking is do the OpenSIPS code use index based queries
> or actual field names when querying the database. The latter would be safer
> and thus pose no problem when adding extra fields...
>
> PS I will use this email as a thread to this subject.
>
> Kind Regards,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Parallel forking - which branch answered

[Muhammad Shahzad]

Try $T_ruri?

http://www.opensips.org/html/docs/modules/1.8.x/tm.html#id294531

Make sure you have transaction enable for each branch.

Thank you.




On Fri, Aug 2, 2013 at 3:50 PM, John Quick wrote:

> Hi,
>
> I am using parallel forking with a destination set returned by exec_dset.
> OpenSIPS v1.8.2
> For branches that go to a gateway (no lookup used) I find it very difficult
> to identify which branch answered the call.
> I am trying to find a variable I can use in on_reply that identifies the
> branch's R-URI. All the variables seem to return the original R-URI, not
> the
> one for this branch. If I report $branch(uri) to my log file it has a NULL
> value.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Regarding opensips-cp 5.0

[Muhammad Shahzad]

The acc module params are missing flags for db, you only have flags for
logging, which actually logs acc data to syslog, also note that flags MUST
be quoted strings as integer flags have been deprecated in opensips 1.9.x.
See below link for details,

http://www.opensips.org/html/docs/modules/1.9.x/acc.html#id293425

Thank you.





On Wed, Jul 31, 2013 at 12:07 PM, Nandini madhu  wrote:

> Dear Shahzad,
>
> I have tried the above options to know the status. But still i am facing
> the same problem.
>
> I am giving my acc module configuration list please have a glance and
> advice me please.
>
> loadmodule "acc.so"
> modparam("acc", "failed_transaction_flag", 1)
> modparam("acc", "report_cancels", 0)
> ##modparam("acc", "report_ack", 0)
> modparam("acc", "early_media",0)
> modparam("acc", "log_level",  1)
> modparam("acc", "log_flag",   1)
> modparam("acc", "log_missed_flag",1)
> modparam("acc", "cdr_flag", 1)
> modparam("acc", "db_url", "mysql://opensips:opensipsrw@opensips/opensips")
> modparam("acc", "db_table_acc", "acc")
> modparam("acc", "acc_method_column", "method")
> modparam("acc", "acc_from_tag_column", "from_tag")
> modparam("acc", "acc_to_tag_column", "to_tag")
> modparam("acc", "acc_callid_column", "callid")
> modparam("acc", "acc_sip_code_column", "sip_code")
> modparam("acc", "acc_sip_reason_column", "sip_reason")
> modparam("acc", "acc_time_column", "time")
>
> i have given super privileges to opensips user in mysql database
>
> Please advice
>
> regards
> sermj
>
>
> On Wed, Jul 31, 2013 at 1:09 PM, Nandini madhu wrote:
>
>> Dear Shahzad,
>>
>> Thank you for your prompt response. i will follow your suggestions and
>> let you know thank you.
>>
>> regards
>> sermj
>>
>>
>> On Wed, Jul 31, 2013 at 11:48 AM, Muhammad Shahzad > > wrote:
>>
>>> Check if,
>>>
>>> 1. you have set required module parameters for acc module as well, most
>>> importantly accounting flags.
>>> 2. you have set those accounting flags in appropriate routes in
>>> opensips.cfg.
>>> 3. you acc table is accessible and permitted for opensips (check
>>> connection string, the user used in it is allowed to insert rows in acc
>>> table etc).
>>> 4. you can connect to mysql or whatever backend you are using, and see
>>> acc table is being populated by opensips during / end of calls.
>>> 5. your opensips-cp has correct db connection string set to access acc
>>> table.
>>>
>>> one of above should solve your problem.
>>>
>>> Thank you.
>>>
>>>
>>>
>>>
>>> On Wed, Jul 31, 2013 at 8:06 AM, Nandini madhu wrote:
>>>
>>>> Hi all,
>>>>
>>>> I am using opensips-cp 5.0 . domain andrtpproxy modules are working
>>>> fine on myserver (opensips-1.9.1-tls). But i am unable to see CDRviewer in
>>>> my GUI.
>>>>
>>>> I hav loaded required acc module parameters in opensips.cfg file.
>>>>
>>>> Please help me
>>>>
>>>> Regards,
>>>> sermj
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>> --
>>> Mit freundlichen Grüßen
>>> Muhammad Shahzad
>>> ---
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +49 176 99 83 10 85
>>> MSN: shari_78...@hotmail.com
>>> Email: shaherya...@googlemail.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
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Regarding opensips-cp 5.0

[Muhammad Shahzad]

Check if,

1. you have set required module parameters for acc module as well, most
importantly accounting flags.
2. you have set those accounting flags in appropriate routes in
opensips.cfg.
3. you acc table is accessible and permitted for opensips (check connection
string, the user used in it is allowed to insert rows in acc table etc).
4. you can connect to mysql or whatever backend you are using, and see acc
table is being populated by opensips during / end of calls.
5. your opensips-cp has correct db connection string set to access acc
table.

one of above should solve your problem.

Thank you.




On Wed, Jul 31, 2013 at 8:06 AM, Nandini madhu  wrote:

> Hi all,
>
> I am using opensips-cp 5.0 . domain andrtpproxy modules are working fine
> on myserver (opensips-1.9.1-tls). But i am unable to see CDRviewer in my
> GUI.
>
> I hav loaded required acc module parameters in opensips.cfg file.
>
> Please help me
>
> Regards,
> sermj
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] LCR Route Prefixes -> Gateways (From Scratch)

[Muhammad Shahzad]

Well, i would prefer and recommend drouting module, for a number of reasons,

1. its actively maintained and updated with new stuff, while carrierroute
isn't so active for a long time.
2. it has less overhead, especially with respect to db, this would matter a
lot if you have a lot of traffic and too many prefixes / rules.
3. Theoretically its designed for routing, regardless if its internal
(within your own network) or external routing (to carrier networks), so its
a good choice if you a large network with many VAS served on dedicated
servers, e.g. voicemail, ivr, transcoding and so on. Although carrierroute
can also do same, but its considers all servers as carrier network (even
ones within your own network).

Thank you.





On Wed, Jul 31, 2013 at 3:11 AM, Nick Khamis  wrote:

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


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] opensips tcp problem

[Muhammad Shahzad Shafi]

 

BTW, your SIP trace shows all communication is in UDP, whereas you
mentioned TCP in subject... 

Thank you. 

On 2013-07-31 01:29, Muhammad
Shahzad Shafi wrote: 

> Either destination is off line or you do not
have NAT keep-alive enabled, resulting relay failure. Make sure you have
NAT Helper and / or NAT Traversal module enabled and configured
correctly for NAT handing. 
> 
> Thank you. 
> 
> On 2013-07-30 17:17,
Jason Sia wrote: 
> 
>> I installed opensips 1.8. I used the default
configuration. I have two clients one is an android phone using native
sip client, and the other one x-lite. I can call x-lite to phone but not
the other way around. How do I fix it? 
>> 
>> Here is a sample of the
logs: I think it does not allow the TCP connection. Is NAT a problem
here? I use a vps to host the opensips.
>> 
>> U 2013/07/29
18:23:52.994331 49.144.184.97:41998 [1] -> 198.23.160.81:5060 [2]
>>
INVITE sip:639195015475@198.23.160.81 [3] SIP/2.0.
>> Via: SIP/2.0/UDP
192.168.0.107:41998;branch=z9hG4bK-d8754z- 
>>
2999840441d3d05a-1---d8754z-;rport.
>> Max-Forwards: 70.
>> Contact:
.
>> To:
.
>> From:
"639178864952";tag=1b3b463c.
>>
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
>> CSeq: 1
INVITE.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO.
>> Content-Type: application/sdp.
>>
Supported: replaces.
>> User-Agent: X-Lite release 4.5.3 stamp 70576.
>>
Content-Length: 357.
>> .
>> v=0.
>> o=- 1375107832761441 1 IN IP4
192.168.0.107.
>> s=X-Lite 4 release 4.5.3 stamp 70576.
>> c=IN IP4
192.168.0.107.
>> t=0 0.
>> m=audio 53084 RTP/AVP 123 9 0 8 97 100 98
101.
>> a=rtpmap:123 opus/48000/2.
>> a=fmtp:123 useinbandfec=1.
>>
a=rtpmap:97 speex/8000.
>> a=rtpmap:100 speex/16000.
>> a=rtpmap:98
ILBC/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>>
a=sendrecv.
>> 
>> U 2013/07/29 18:23:52.994422 198.23.160.81:5060 [7]
-> 49.144.184.97:41998 [8]
>> SIP/2.0 407 Proxy Authentication
Required.
>> Via: SIP/2.0/UDP
192.168.0.107:41998;received=49.144.184.97;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport=41998.
>>
To: ;tag=80403214130c49515b8f7d7842a4c119.ef65.
>> From:
"639178864952";tag=1b3b463c.
>>
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
>> CSeq: 1
INVITE.
>> Proxy-Authenticate: Digest realm="198.23.160.81",
nonce="51f67b1600fae8c70592af760c78fa39cd43f84c1a6b".
>> Server:
OpenSIPS (1.8.3-notls (x86_64/linux)).
>> Content-Length: 0.
>> .
>> 
>>
U 2013/07/29 18:23:53.299025 49.144.184.97:41998 [11] ->
198.23.160.81:5060 [12]
>> ACK sip:639195015475@198.23.160.81 [13]
SIP/2.0.
>> Via: SIP/2.0/UDP
192.168.0.107:41998;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport.
>>
Max-Forwards: 70.
>> To: ;tag=80403214130c49515b8f7d7842a4c119.ef65.
>> From:
"639178864952";tag=1b3b463c.
>>
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
>> CSeq: 1 ACK.
>>
Content-Length: 0.
>> .
>> 
>> U 2013/07/29 18:23:53.317842
49.144.184.97:41998 [16] -> 198.23.160.81:5060 [17]
>> INVITE
sip:639195015475@198.23.160.81 [18] SIP/2.0.
>> Via: SIP/2.0/UDP
192.168.0.107:41998;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport.
>>
Max-Forwards: 70.
>> Contact: .
>> To: .
>> From:
"639178864952";tag=1b3b463c.
>>
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
>> CSeq: 2
INVITE.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO.
>> Content-Type: application/sdp.
>>
Proxy-Authorization: Digest
username="639178864952",realm="198.23.160.81",nonce="51f67b1600fae8c70592af760c78fa39cd43f84c1a6b",uri="sip:639195015475@198.23.160.81
[22]",response="1605e4e6b20dbbd6be5917c94dd3961e",algorithm=MD5.
>>
Supported: replaces.
>> User-Agent: X-Lite release 4.5.3 stamp 70576.
>>
Content-Length: 357.
>> .
>> v=0.
>> o=- 1375107832761441 1 IN IP4
192.168.0.107.
>> s=X-Lite 4 release 4.5.3 stamp 70576.
>> c=IN IP4
192.168.0.107.
>> t=0 0.
>> m=audio 53084 RTP/AVP 123 9 0 8 97 100 98
101.
>> a=rtpmap:123 opus/48000/2.
>> a=fmtp:123 useinbandfec=1.
>>
a=rtpmap:97 speex/8000.
>> a=rtpmap:100 speex/16000.
>> a=rtpmap:98
ILBC/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>>
a=sendrecv.
>> 
>> U 2013/07/29 18:23:53.318191 198.23.160.81:5060 [23]
-> 49.144.184.97:41998 [24]
>> SIP/2.0 100 Giving a try.
>> Via:
SIP/2.0/UDP
192.168.

Re: [OpenSIPS-Users] opensips tcp problem

[Muhammad Shahzad Shafi]

-d8754z-;rport=41998.
>
To: ;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9.
> From:
"639178864952";tag=1b3b463c.
>
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2
INVITE.
> Server: OpenSIPS (1.8.3-notls (x86_64/linux)).
>
Content-Length: 0.
> .
> 
> U 2013/07/29 18:24:03.582094
49.144.184.97:41998 [31] -> 198.23.160.81:5060 [32]
> ACK
sip:639195015475@198.23.160.81 [33] SIP/2.0.
> Via: SIP/2.0/UDP
192.168.0.107:41998;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport.
>
Max-Forwards: 70.
> To: ;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9.
> From:
"639178864952";tag=1b3b463c.
>
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2 ACK.
>
Content-Length: 0.

-- 
Mit freundlichen Grüßen
Muhammad
Shahzad
---
CISCO Rich Media
Communication Specialist (CRMCS)
CISCO Certified Network Associate
(CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email:
shaherya...@googlemail.com
 

Links:
--
[1]
http://49.144.184.97:41998
[2] http://198.23.160.81:5060
[3]
mailto:sip%3A639195015475@198.23.160.81
[4]
http://sip:639178864952@49.144.184.97:41998
[5]
mailto:sip%3A639195015475@198.23.160.81
[6]
mailto:sip%3A639178864952@198.23.160.81
[7]
http://198.23.160.81:5060
[8] http://49.144.184.97:41998
[9]
mailto:sip%3A639195015475@198.23.160.81
[10]
mailto:sip%3A639178864952@198.23.160.81
[11]
http://49.144.184.97:41998
[12] http://198.23.160.81:5060
[13]
mailto:sip%3A639195015475@198.23.160.81
[14]
mailto:sip%3A639195015475@198.23.160.81
[15]
mailto:sip%3A639178864952@198.23.160.81
[16]
http://49.144.184.97:41998
[17] http://198.23.160.81:5060
[18]
mailto:sip%3A639195015475@198.23.160.81
[19]
http://sip:639178864952@49.144.184.97:41998
[20]
mailto:sip%3A639195015475@198.23.160.81
[21]
mailto:sip%3A639178864952@198.23.160.81
[22]
mailto:sip%3A639195015475@198.23.160.81
[23]
http://198.23.160.81:5060
[24] http://49.144.184.97:41998
[25]
mailto:sip%3A639195015475@198.23.160.81
[26]
mailto:sip%3A639178864952@198.23.160.81
[27]
http://198.23.160.81:5060
[28] http://49.144.184.97:41998
[29]
mailto:sip%3A639195015475@198.23.160.81
[30]
mailto:sip%3A639178864952@198.23.160.81
[31]
http://49.144.184.97:41998
[32] http://198.23.160.81:5060
[33]
mailto:sip%3A639195015475@198.23.160.81
[34]
mailto:sip%3A639195015475@198.23.160.81
[35]
mailto:sip%3A639178864952@198.23.160.81
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] local_route not being called

[Muhammad Shahzad]

There are many circumstances, for example,

1. You are using loadbalancer / dispatcher module, which send SIP keep
alive requests (e.g. OPTIONS or INFO message) to destination to determine
that are still available.
2. You are using media proxy / rtpproxy with media timeouts enabled, so if
media proxy / rtpproxy does not receives rtp and timeout occurs, it
triggers a BYE requests to end the call. Which you many want to track in
local_route to do accounting etc.
3. You are using dialog module with "B" flag enabled in create_dialog
method, which will result in BYE request to end points when dialog timeouts
(same way as above).
4. You are using B2BUA or topo-hiding  setup, which will result in all SIP
request duplicated but with somewhat different headers / parameters to send
to destination. All those messages will appear in local route where you can
do their accounting etc.

OpenSIPS does not generates any reply on its own, unless in special
circumstances, e.g.

1. It sends 100 Trying when you do t_relay, However, you can force opensips
NOT to send this reply automatically by specifying it a parameter to this
method.
2. It sends 408 upon T2 timer timeout for destination. You can track it in
reply_route to do whatever you want in that situation.

You can generate your own replies overriding the reply that is actually
received from destination in reply_route, or even from route block if you
do not want to (or you can not send to) an incoming request to a
destination at all.

Have a look at tutorials for more information,

http://www.opensips.org/Documentation/Tutorials

Thank you.




On Thu, Jul 25, 2013 at 6:08 PM, Jason Caulfield  wrote:

> Muhammad,
>
> Thanks for the reply.
>
> I guess I don't understand under which circumstances the TM module would
> send out a request message, presumably an INVITE.
>
> Can you please add clarification.
>
> Thanks,
> Jason
>
>
> On Wed, Jul 24, 2013 at 6:55 PM, Muhammad Shahzad 
> wrote:
>
>> Local route is invoked by opensips when a REQUEST is generated by TM
>> module internally, NOT the replies. See its description for details,
>>
>> http://www.opensips.org/Documentation/Script-Routes-1-9#toc6
>>
>> Thank you.
>>
>>
>>
>>
>> On Wed, Jul 24, 2013 at 11:42 PM, Jason Caulfield wrote:
>>
>>> Users,
>>>
>>> I am just getting started with opensips.  (version 1.7 from CentOS 6
>>> epel repo)
>>>
>>> Can you please help me understand why in my code (see below) local_route
>>> is not invoked.
>>>
>>> I would expect that local_route would be invoked when a "100 Trying" is
>>> issued, but it is not.
>>>
>>> Below is the config, log out, and network output.
>>>
>>> I am testing it with sipp.
>>>
>>>
>>> Config:
>>>
>>> ### Global Parameters #
>>> debug=0
>>> log_stderror=no
>>> log_facility=LOG_LOCAL0
>>> fork=yes
>>> children=12
>>> log_name="TEST"
>>> disable_tcp=yes
>>> port=5060
>>> ###
>>>
>>> ### Modules Section ###
>>> mpath="/usr/lib/opensips/modules"
>>>
>>> loadmodule "tm.so"
>>> loadmodule "textops.so"
>>> loadmodule "exec.so"
>>>
>>> modparam("tm" , "onreply_avp_mode", 1)
>>> modparam("tm" , "fr_timer", 2)  # Vendor timeout
>>> modparam("tm" , "fr_inv_timer", 2)
>>> modparam("tm" , "enable_stats", 0)
>>> modparam("tm" , "via1_matching", 0)
>>> modparam("tm" , "T1_timer", 500)  # Retransmit interval
>>> modparam("tm" , "T2_timer", 1000)  # Retransmit total duration
>>> ###
>>>
>>> ### Routing Logic #
>>> route {
>>> xlog("ROUTE");
>>> seturi("sip:55@10.0.1.27:9003");
>>> t_relay();
>>> }
>>>
>>> onreply_route {
>>> xlog("ONREPLY");
>>> }
>>> error_route {
>>> xlog("ERROR");
>>> }
>>>
>>> local_route {
>>> xlog("LOCAL");
>>> }
>>> ###
>>>
>>>
>>> Log:
>>>
>>> Jul 25 05:34:32 rmps-b TEST[12856]: ROUTE
>>> Jul 25 05:34:32 rmps-b TEST[12855]: ONREPLY
>>> Jul 25 05:34:32 rmps-b TEST[12857

Re: [OpenSIPS-Users] local_route not being called

[Muhammad Shahzad]

> Content-Type: application/sdp.
> Content-Length:   129.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 10.0.1.27.
> s=-.
> c=IN IP4 10.0.1.27.
> t=0 0.
> m=audio 6001 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
>
> #
> U 10.0.1.147:5060 -> 10.0.1.27:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-0.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>;tag=10195SIPpTag014.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 1 INVITE.
> Contact: .
> Content-Length: 0.
> .
>
> #
> U 10.0.1.147:5060 -> 10.0.1.27:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-0.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>;tag=10195SIPpTag014.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 1 INVITE.
> Contact: .
> Content-Type: application/sdp.
> Content-Length:   129.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 10.0.1.27.
> s=-.
> c=IN IP4 10.0.1.27.
> t=0 0.
> m=audio 6001 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
>
> #
> U 10.0.1.27:5060 -> 10.0.1.147:5060
> ACK sip:55@10.0.1.147:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-5.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 1 ACK.
> Contact: sip:sipp@10.0.1.27:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 10.0.1.147:5060 -> 10.0.1.27:9003
> ACK sip:55@10.0.1.27:9003 SIP/2.0.
> Via: SIP/2.0/UDP 10.0.1.147;branch=z9hG4bK580f.8ddd3891.2.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-5.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 1 ACK.
> Contact: sip:sipp@10.0.1.27:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 10.0.1.27:5060 -> 10.0.1.147:5060
> BYE sip:55@10.0.1.147:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-7.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 2 BYE.
> Contact: sip:sipp@10.0.1.27:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 10.0.1.147:5060 -> 10.0.1.27:9003
> BYE sip:55@10.0.1.27:9003 SIP/2.0.
> Via: SIP/2.0/UDP 10.0.1.147;branch=z9hG4bK280f.81db1714.0.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-7.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 2 BYE.
> Contact: sip:sipp@10.0.1.27:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 10.0.1.27:9003 -> 10.0.1.147:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.0.1.147;branch=z9hG4bK280f.81db1714.0, SIP/2.0/UDP
> 10.0.1.27:5060;branch=z9hG4bK-10199-4-7.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 2 BYE.
> Contact: .
> Content-Length: 0.
> .
>
> #
> U 10.0.1.147:5060 -> 10.0.1.27:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.0.1.27:5060;branch=z9hG4bK-10199-4-7.
> From: sipp ;tag=10199SIPpTag004.
> To: sut <55@10.0.1.147:5060>.
> Call-ID: 4-10199@10.0.1.27.
> CSeq: 2 BYE.
> Contact: .
> Content-Length: 0.
>
>
> Thanks,
> Jason
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] BUG: Shutdown timeout

[Muhammad Shahzad]

Khalid, this does not make sense to result in opensips crash. I guess the
problem is something else. You need to enable crash dump and send backtrace
of it.

Thank you.



On Fri, Jul 19, 2013 at 2:47 PM, M.Khaled W Chehab wrote:

> I checked the call capture for the  call that crashes my opensips and it
> was due to:
>
> ** **
>
> The user send a second invite with the same URI  and  same call-ID in the
> same second after  it  send  an ACK for  the 200 ok.
>
>  How can I block the second invite in a call ,I tried to usethe below
> function but  it didn’t works please advice 
>
>if (has_totag()) {
>
> ** **
>
>if (is_method("INVITE")) {
>
>if (t_check_trans()) {
>
>xlog("has to tag and t_check trans
> t_relay---$ci \n");
>
>#t_relay();
>
>}
>
>xlog("INVITE  ---has to tag ---$ci\n");
> 
>
>send_reply("403","Multi Invites");
>
> exit;
>
>  }
>
> ** **
>
> Please advice
>
> Regards
>
> ** **
>
> ** **
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *M.Khaled W Chehab
> *Sent:* Friday, July 19, 2013 10:58 AM
> *To:* users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] BUG: Shutdown timeout
>
> ** **
>
> Hello,
>
> ** **
>
> ** **
>
> My opensips stops suddenly with the below error in log file, how to find
> and fix this bug  
>
> ** **
>
> 2013-07-18T21:28:41.301961+00:00 opensips opensips[14336]:
> ERROR:tm:t_check: reply cannot be parsed
>
> 2013-07-18T21:33:37.056151+00:00 opensips opensips[14333]: --Debug
> Customer IP:xx.xx.xx.xx-Permission Denied for INVITE from
> sip:213659236...@xx.xx.xx.xx to sip:962776482...@xx.xx.xx.xx
> 275579741-0-4233428028@62.190.146.31
>
> 2013-07-18T21:33:37.757086+00:00 opensips opensips[14323]: --Debug
> Customer IP:xx.xx.xx.xx-Permission Denied for INVITE from
> sip:213659236...@xx.xx.xx.xx to sip:962776482...@xx.xx.xx.xx
> 161768799-0-1203493098@62.190.146.30
>
> 2013-07-18T22:21:50.702682+00:00 opensips opensips[14318]: --Debug
> Customer IP:xx.xx.xx.xx-Permission Denied for INVITE from
> sip:213696613...@xx.xx.xx.xx to sip:962776483...@xx.xx.xx.xx
> 161798679-0-1206385818@62.190.146.30
>
> 2013-07-18T22:21:51.165865+00:00 opensips opensips[14320]: --Debug
> Customer IP:xx.xx.xx.xx-Permission Denied for INVITE from
> sip:213696613...@xx.xx.xx.xx to sip:962776483...@xx.xx.xx.xx
> 275610902-0-4236321928@62.190.146.31
>
> 2013-07-18T22:22:48.351683+00:00 opensips opensips[14320]: --Debug
> Customer IP:xx.xx.xx.xx-Permission Denied for INVITE from
> sip:213696613...@xx.xx.xx.xx to sip:962776483...@xx.xx.xx.xx
> 161799195-0-1206443498@62.190.146.30
>
> 2013-07-18T22:22:48.983005+00:00 opensips opensips[14344]: --Debug
> Customer IP:xx.xx.xx.xx-Permission Denied for INVITE from
> sip:213696613...@xx.xx.xx.xx to sip:962776483...@xx.xx.xx.xx
> 275611430-0-4236379748@62.190.146.31
>
> 2013-07-19T06:16:26.873502+00:00 opensips opensips[14453]:
> CRITICAL:dialog:unref_dlg: bogus ref -1 with cnt 1 for dlg 0x7f93a95e8878
> [4011:185119809] with clid '162005786-0-1234838388@62.190.146.30' and
> tags 'sansay321252821rdb4676' '4089d96837c81bfcf1773f74421dd1cd'
>
> 2013-07-19T06:17:28.673757+00:00 opensips opensips[14309]:
> CRITICAL:core:sig_alarm_abort: BUG - shutdown timeout triggered, dying...*
> ***
>
> ** **
>
> ** **
>
> Please advice
>
> Regards
>
> ** **
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OK message inspection inside the dialog

[Muhammad Shahzad]

In reply_route where all replies from endpoints are receive, you should be
able to filter your desired replies and do whatever you want to do with
them. It should always work, that's what reply_route is designed to do...!

Thank you.


On Sat, Jul 13, 2013 at 9:33 PM, Maciej Bylica  wrote:

> Hello,
>
> I have a problem to verify and change headers in OK message that Opensips
> is receiving within the dialog by using insert_hf and search functions.
> The problem is not with these functions but to catch OK that is a part of
> the sip dialog.
> Any changes are applied to INVITE unfortunately.
>
> Is there any way to get into 180/183/200 messages?
>
> Thanks,
> Mac
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] NAT - Unable to solve RTP Problem

[Muhammad Shahzad]

Look at rtpproxy module documentation,

http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html#id249057

It mentions about a patch regarding timeout notification. Though this patch
can also be applied to latest rtpproxy code but opensips also maintains a
patched version of rtpproxy in its file repository.

I am not sure the reason for high CPU usage due to rtpproxy, in fact it
never happened to me, but then all my deployments are on real machines with
public IP. However, many fellows has faced this problem on VMs behind NAT.
I suspect this problem is more related to netfilter on bridge rather then
rtpproxy itself. Since rtpproxy consumes the greatest part of data traffic
that comes into VM in most of the cases, therefore it is blames for the
problem, while i think it is just a victim.

Thank you.




On Sun, Jun 23, 2013 at 3:48 PM, Nick Khamis  wrote:

>
> Hello Muhammad,
>
> Where did the "opensips rtpproxy" come from? Are the original creators no
> longer maintaining
> the project? We would really like to use MediaProxy in a vm environment
> however, stuck on
> the "public ip" requirement. A while back I did come across the following:
>
> http://lists.sip-router.org/pipermail/users/2008-January/015111.html
>
> Not sure how the hack would hold up. Maybe the AG people could chime in?
>
> Kind Regards,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] NAT - Unable to solve RTP Problem

[Muhammad Shahzad]

I have heard original rtpproxy has high CPU problem on VMs, But the patched
one from opensips works fine. For media proxy, its really good and easy to
use but unfortunately it requires public ip.


On Sun, Jun 23, 2013 at 2:47 PM, Nick Khamis  wrote:

> Hello Everyone,
>
> Sorry to chime in here but we are kind of experiencing the same problem
> (RTPProxy 99% CPU usage). Flavio we would love to use
> mediaproxy. Is it possible in a VM setup behind nat?
>
> Kind Regards,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] timer_route

[Muhammad Shahzad]

route_timer is periodic scheduler with respect to start of OpenSIPS
service. You can not call it on per transaction basis as it does not
processing a message. See the docs,

http://www.opensips.org/Documentation/Script-Routes-1-9#toc8

Thank you.




On Sun, Jun 16, 2013 at 9:25 AM, Dragomir Haralambiev wrote:

> Hello,
>
> I try to start timer_route when receive INVITE.
>
> if (is_method("INVITE")) {
>  .
>  timer_route(gw_update);
>
> }
>
>  timer_route[gw_update, 10] {
>xlog("L_ERR", "timer_route ");
>   }
>
>
> What I do wrong?
>
> Best regards,
> PlayMen
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSips and firewall

[Muhammad Shahzad]

Are you using force_send_socket in opensips.cfg? If so, it is incorrectly
configured. Also check your sip client settings, it should send SIP packets
to TCP 5061 server port instead of 5060.

Thank you.



On Fri, Jun 7, 2013 at 10:15 AM, ivan smiljkovic  wrote:

> Muhammad Shahzad,
> I tried your solution and now can establish connection between clients via
> udp. But TLS is problem.
> Here is problematic part of log:
>
>
>
> SIP/2.0 100 Giving a try
> Via: SIP/2.0/TLS 10.78.90.72:52829
> ;received=89.216.yy.yy;rport=36024;branch=z9hG4bK750366181
> From: ;tag=1190324263
> To: 
> Call-ID: 22475986@10.78.90.72
> CSeq: 401 INVITE
> Server: OpenSIPS (1.9.1-tls (x86_64/linux))
> Content-Length: 0
>
>
> Jun  7 09:47:25 [2564] DBG:tm:_reply_light: reply sent out.
> buf=0x7f56c69f7fc8: SIP/2.0 1..., shmem=0x7f56b38faf40: SIP/2.0 1
> Jun  7 09:47:25 [2564] DBG:tm:_reply_light: finished
> Jun  7 09:47:25 [2564] DBG:core:buf_init: initializing...
> new branch at sip:1005@10.78.90.72:50753;transport=tcp
> Jun  7 09:47:25 [2564] DBG:core:mk_proxy: doing DNS lookup...
> Jun  7 09:47:25 [2564] DBG:core:get_send_socket: force_send_socket of
> different proto (2)!
> Jun  7 09:47:25 [2564] WARNING:core:get_send_socket: protocol/port mismatch
> Jun  7 09:47:25 [2564] DBG:core:parse_headers: flags=2000
> Jun  7 09:47:25 [2564] DBG:core:build_req_buf_from_sip_req: id added:
> <;i=a>, rcv proto=3
> Jun  7 09:47:25 [2564] DBG:core:tcp_send: no open tcp connection found,
> opening new one
> Jun  7 09:47:25 [2564] DBG:core:probe_max_sock_buff: getsockopt: snd is
> initially 16384
> Jun  7 09:47:25 [2564] DBG:core:probe_max_sock_buff: trying : 32768
> Jun  7 09:47:25 [2564] DBG:core:probe_max_sock_buff: setting snd:
> set=32768,verify=65536
> Jun  7 09:47:25 [2564] DBG:core:probe_max_sock_buff: trying : 65536
> Jun  7 09:47:25 [2564] DBG:core:probe_max_sock_buff: setting snd:
> set=65536,verify=131072
> Jun  7 09:47:25 [2564] DBG:core:probe_max_sock_buff: trying : 131072
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Serious cancel problem

[Muhammad Shahzad]

There is a console version of wireshark available for debian at least, it
is called "tshark". Install it on remote server, do something like this,

tshark -i  -w .pcap

you can read its man page or "tshark -h" to see all available options, e.g.
you can specify capture filters to dump only interesting traffic to the
file. Also you can read a captured call from console too. However, it won't
be as easily and neat as GUI version.

Thank you.




On Thu, Jun 6, 2013 at 11:44 PM, Davide Dal Frà  wrote:

>  Hi Nick,
>
> No problem for this. You could do a live dump on remote server using
> tcpdump over ssh.
> Something like :
>
>- first: make a fifo with mkfifo /tmp/capture
> - ssh user@host tcpdump -i yourethinterface -U -s0 -w - 'udp 5060' >
>/tmp/capture (you could personalize the filter on tcpdump delimited between
>->'<- )
>- Open wireshark->Capture->Interface->Options
>- Mange Interface->new-> browse or digit directly the path of the fifo
>begin created
>- Save
>- Start dumping & enjoy!
>
> If you have Signaling on a server and media on another one there are no
> problem. You could dump signaling as described before, and make another
> fifo and dump in the same way the media from the other server.
>
> On Wireshark side add both fifo interface (make sure that after you have
> saved the interface you have both selected into the menu) and start the
> live dump!
>
> Maybe coul seem complicated, but you can automate all in a bash script!
>
> Khaled, sorry again!
>
> BR
> Davide
>
> On 06/06/13 22:24, Nick Khamis wrote:
>
> The problem is, wireshark is running on my computer, but the voip traffic
> is on the servers, also within the network. Khaled, sorry for the hijack!
>
>  N.
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Serious cancel problem

[Muhammad Shahzad]

The screenshot seems to be from wireshark, which is in fact a pretty neat
tool.

Thank you.




On Thu, Jun 6, 2013 at 6:09 PM, Nick Khamis  wrote:

> Khaled, what are you using for capture. I would like to use this tool as
> well.
>
> Kind Regards,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSips and firewall

[Muhammad Shahzad]

Oh, i see. I thought the opensips has a public IP interface and you were
listening only on private IP interface. Anyways, try this,

listen=tls:10.78.90.71:5061
advertised_address=89.216.xx.xx:5061
alias=89.216.xx.xx:5061





On Thu, Jun 6, 2013 at 10:20 AM, ivan smiljkovic  wrote:

> and there is log
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSips and firewall

[Muhammad Shahzad]

Change your opensips.cfg to something like this,

disable_tls=no
listen=tls:10.78.90.71:5061
listen=tls:89.216.xx.xx:5061

and remove alias and port settings.

Thank you.




On Wed, Jun 5, 2013 at 1:15 PM, ivan smiljkovic  wrote:

> @*Muhammad Shahzad*
>
> *adress od my server is 10.78.90.71 and public address is 89.216.xx.xx*
> *
> *
> *cfg:*
> *
> *
> *listen=tls.10.78.90.71:5061*
> *alias=10.78.90.71*
> *alias=89.216.xx.xx*
> *
> *
> *disable_tls=no*
> *port=5061*
>
>
> On Tue, Jun 4, 2013 at 10:59 AM, ivan smiljkovic wrote:
>
>> Hi,
>> I configured and started Opensips server in my local network. Everything
>> was ok, TLS, Mikey key exchange between clients and calls.
>> But, now, clients are on 3g network (internet) and OpenSips is behind
>> firewall. Firewall is configured to bypass: staticIP:5061 to
>> openSipsIP:5061.
>>
>> I can register clients but cannot start ringing and start call.
>> I dont have idea where to start with solving this problem. Here is
>> difference in log between working local network and not working internet
>> solution:
>>
>> May 31 12:10:44 [2461] DBG:tm:_reply_light: reply sent out.
>> buf=0x7f93d5fb72c0: SIP/2.0 1..., shmem=0x7f93c2c39c18: SIP/2.0 1
>> May 31 12:10:44 [2461] DBG:tm:_reply_light: finished
>> new branch at sip:1004@10.245.199.188:38809;transport=tcp
>> May 31 12:10:44 [2461] DBG:core:mk_proxy: doing DNS lookup...
>> May 31 12:10:44 [2461] DBG:core:get_send_socket: force_send_socket of
>> different proto (2)!
>> May 31 12:10:44 [2461] WARNING:core:get_send_socket: protocol/port
>> mismatch
>> May 31 12:10:44 [2461] DBG:core:parse_headers: flags=2000
>> May 31 12:10:44 [2461] DBG:core:build_req_buf_from_sip_req: id added:
>> <;i=84>, rcv proto=3
>> May 31 12:10:44 [2461] DBG:core:tcp_send: no open tcp connection found,
>> opening new one
>> May 31 12:10:44 [2461] DBG:core:probe_max_sock_buff: getsockopt: snd is
>> initially 16384
>> May 31 12:10:44 [2461] DBG:core:probe_max_sock_buff: trying : 32768
>>
>> Difference is DBG:core:tcp_send: no open tcp connection found, opening
>> new one and down.
>>
>> What to do, where is solution for this problem?
>>
>>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSips and firewall

[Muhammad Shahzad]

Check your "listen" variable in opensips.cfg. Are you listening on WAN
interface?

Thank you.



On Tue, Jun 4, 2013 at 10:59 AM, ivan smiljkovic  wrote:

> Hi,
> I configured and started Opensips server in my local network. Everything
> was ok, TLS, Mikey key exchange between clients and calls.
> But, now, clients are on 3g network (internet) and OpenSips is behind
> firewall. Firewall is configured to bypass: staticIP:5061 to
> openSipsIP:5061.
>
> I can register clients but cannot start ringing and start call.
> I dont have idea where to start with solving this problem. Here is
> difference in log between working local network and not working internet
> solution:
>
> May 31 12:10:44 [2461] DBG:tm:_reply_light: reply sent out.
> buf=0x7f93d5fb72c0: SIP/2.0 1..., shmem=0x7f93c2c39c18: SIP/2.0 1
> May 31 12:10:44 [2461] DBG:tm:_reply_light: finished
> new branch at sip:1004@10.245.199.188:38809;transport=tcp
> May 31 12:10:44 [2461] DBG:core:mk_proxy: doing DNS lookup...
> May 31 12:10:44 [2461] DBG:core:get_send_socket: force_send_socket of
> different proto (2)!
> May 31 12:10:44 [2461] WARNING:core:get_send_socket: protocol/port mismatch
> May 31 12:10:44 [2461] DBG:core:parse_headers: flags=2000
> May 31 12:10:44 [2461] DBG:core:build_req_buf_from_sip_req: id added:
> <;i=84>, rcv proto=3
> May 31 12:10:44 [2461] DBG:core:tcp_send: no open tcp connection found,
> opening new one
> May 31 12:10:44 [2461] DBG:core:probe_max_sock_buff: getsockopt: snd is
> initially 16384
> May 31 12:10:44 [2461] DBG:core:probe_max_sock_buff: trying : 32768
>
> Difference is DBG:core:tcp_send: no open tcp connection found, opening new
> one and down.
>
> What to do, where is solution for this problem?
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] A few simple general question

[Muhammad Shahzad]

Hi,

1. Try $dd, which should give you destination of packet.
2. Depends on your need, I have seen more setups of DR then CR. But this by
no means indicate DR is better then CR.

Thank you.



On Mon, May 27, 2013 at 1:52 AM, Nick Khamis  wrote:

> Hello Everyone,
>
> Just a few quick questions:
>
> 1) Is there any core variable that returns the IP destination of where
> the SIP packet is going. For example, 192.168.2.20 in:
>
> U 2013/05/26 19:26:14.807995 192.168.2.10:5060 -> 192.168.2.20:5060
>
> $si: 192.168.2.10:5060
> $di: Return the destination of the incoming packets (i.e., the
> interface of OpenSIPS)
>
> 2) When to use CR vs. DR, is one a subset of the other. And finally,
> which will be more likely to stand the test of time?
>
> Thank you for your time,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Mediaproxy relay on CentOS 6 VPS

[Muhammad Shahzad]

Humm, strange i never had to do that. If you can check which VMS you have,
may be i can get it tested in local data center. Following three commands
may give you some idea,

cat /proc/cpuinfo
cat /etc/issue
uname -a

OR you can always fallback to Debian. :-)

Thank you.




On Mon, May 20, 2013 at 6:45 PM, John Quick wrote:

> Muhammed,
>
>
> Thanks for responding so quickly. I already have iptables-devel package
> installed, but I used yum to install it.
>
> I did some research the last time it happened and think it concerns Linux
> loadable modules and options that may require a re-build of the kernel.***
> *
>
> On that occasion, I chickened out and changed to Ubuntu.
>
> ****
>
> John
>
> ** **
>
> *From:* Muhammad Shahzad [mailto:shaherya...@gmail.com]
> *Sent:* 20 May 2013 17:37
> *To:* John Q; OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Mediaproxy relay on CentOS 6 VPS
>
> ** **
>
> I think you need to install iptables-devel package.
>
> ** **
>
> http://rpm.pbone.net/index.php3?stat=3&search=iptables-devel&srodzaj=3
>
> ** **
>
> Thank you.
>
> ** **
>
> ** **
>
> On Mon, May 20, 2013 at 5:56 PM, John Quick 
> wrote:
>
> On two different Virtual Private Servers running CentOS 6 I have hit the
> same problem.
> It is that when I try to start mediaproxy relay, it errors with:
> FATAL: Module ip_tables not found.
>
> Unfortunately, as it is a third party VPS, I do not know what
> virtualization
> environment is being used.
> Does anyone know how to debug this problem (before I re-image it with
> Debian/Ubuntu)?
>
> Mediaproxy version is 2.5.2 and I used my own instructions as outlined at
> http://kb.smartvox.co.uk/opensips/install-mediaproxy-centos-6/
> The same procedure has worked fine on many other servers running CentOS 6.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
>
> _______
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> 
>
> ** **
>
> -- 
>
> Mit freundlichen Grüßen
>
> Muhammad Shahzad
> -------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com 
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Regular expression matching problem

[Muhammad Shahzad]

OR to simplify more, you just want to match last four characters as digits,
so you can try this as well,

if ($rU =~ "[0-9]{4}$") {
xlog("L_WARN", ">>>>>>>>>>>>> MATCHED
<<<<<<<<<<<<<< \n");
} else {
xlog("L_WARN", ">>>>>>>>>>>>> NOT MATCHED
<<<<<<<<<<<<<< \n");
}

Thank you.




On Mon, May 20, 2013 at 6:41 PM, Bogdan-Andrei Iancu wrote:

> **
> Hi Diego,
>
> The REGEXPs in OpenSIPS are POSIX compliant, so \d are not supported. 
> Try:"^(.)?[0-9]{4}$"
> .
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/20/2013 06:57 PM, Diego Barberio wrote:
>
>   Hi All,
>
>  I'm having a really stupid issue with opensips 1.8.1. I need to do
> different logic if the request line user is a number of 4 digits that can
> be preceded by any character. So I created this regular expression:
> ^(.)?\d{4}$
>  I've tested it on www.regular-expressions.info/javascriptexample.htmland 
> works perfect, however on opensips it never matches.
>
>  I've made the following test script:
>
>
> if($rU =~ '^(.)?\d{4}$') {
> xlog("MATCHES $rU\n");
> }else{
> xlog("NOT MATCHES $rU\n");
> }
>
>  And I always get "NOT MATCHES":
>
> May 20 11:54:33 localhost /usr/local/sbin/opensips[22628]: NOT MATCHES
> *5522
> May 20 11:54:42 localhost /usr/local/sbin/opensips[22629]: NOT MATCHES
> 5522
>
>  What am I doing wrong?
>
>  Thanks
>  Diego
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Mediaproxy relay on CentOS 6 VPS

[Muhammad Shahzad]

I think you need to install iptables-devel package.

http://rpm.pbone.net/index.php3?stat=3&search=iptables-devel&srodzaj=3

Thank you.



On Mon, May 20, 2013 at 5:56 PM, John Quick wrote:

> On two different Virtual Private Servers running CentOS 6 I have hit the
> same problem.
> It is that when I try to start mediaproxy relay, it errors with:
> FATAL: Module ip_tables not found.
>
> Unfortunately, as it is a third party VPS, I do not know what
> virtualization
> environment is being used.
> Does anyone know how to debug this problem (before I re-image it with
> Debian/Ubuntu)?
>
> Mediaproxy version is 2.5.2 and I used my own instructions as outlined at
> http://kb.smartvox.co.uk/opensips/install-mediaproxy-centos-6/
> The same procedure has worked fine on many other servers running CentOS 6.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-Devel] Status of OpenSIPS repository

[Muhammad Shahzad]

OK, many thanks. I will file bugs at source forge hopefully tonight.

Thank you.



On Mon, May 20, 2013 at 5:57 PM, Răzvan Crainea  wrote:

> Hi, Muhammad!
>
> We are working on this as we speak. According to our schedule[1], the
> ticketing interface will be complete migrating until 24th of May. Until
> further notice, you should continue to use the SourceForge interface[2].
> We will be back later with further updates.
>
> [1] 
> http://www.opensips.org/About/**GitHub-Migration<http://www.opensips.org/About/GitHub-Migration>
> [2] 
> http://sourceforge.net/p/**opensips/_list/tickets<http://sourceforge.net/p/opensips/_list/tickets>
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
>
> On 05/20/2013 12:45 PM, Muhammad Shahzad wrote:
>
>> OpenSIPS repo is frozen since May 08, 2013. Meanwhile i did couple of
>> important production upgrades to version 1.9 rev. 10024 and found a
>> couple of critical bugs. I just wonder if we still have source-forge
>> interface to report bugs? And for how long repository will remain locked
>> for migration to Github? The bugs i found and want to report are, (very
>> brief summary, will provide detailed info in bug reporting interface),
>>
>> 1. OpenSIPS crashes when Linphone on Android with push notification
>> enabled registers. Crash happens around add_rcv_param("1") method,
>> possibly due to memory corruption.
>>
>> 2. RTPProxy (offer / answer mechanism) does not work on serialized
>> branches.
>>
>> Thank you.
>>
>>
>> --
>> Mit freundlichen Grüßen
>> Muhammad Shahzad
>> --**-
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_78...@hotmail.com 
>> <mailto:shari_786pk@hotmail.**com
>> >
>> Email: shaherya...@googlemail.com 
>> <mailto:shaheryarkh@**googlemail.com
>> >
>>
>>
>> __**_
>> Devel mailing list
>> de...@lists.opensips.org
>> http://lists.opensips.org/cgi-**bin/mailman/listinfo/devel<http://lists.opensips.org/cgi-bin/mailman/listinfo/devel>
>>
>>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Status of OpenSIPS repository

[Muhammad Shahzad]

OpenSIPS repo is frozen since May 08, 2013. Meanwhile i did couple of
important production upgrades to version 1.9 rev. 10024 and found a couple
of critical bugs. I just wonder if we still have source-forge interface to
report bugs? And for how long repository will remain locked for migration
to Github? The bugs i found and want to report are, (very brief summary,
will provide detailed info in bug reporting interface),

1. OpenSIPS crashes when Linphone on Android with push notification enabled
registers. Crash happens around add_rcv_param("1") method, possibly due to
memory corruption.

2. RTPProxy (offer / answer mechanism) does not work on serialized branches.

Thank you.


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] firewall configuration

[Muhammad Shahzad]

The default port range media proxy uses is UDP 5 - 6. You can
modify it in its config.ini file. So this port range must be open in
firewall.

Thank you.



On Sun, May 19, 2013 at 8:35 PM, Nicholas Papadakos wrote:

>
> Hello,
>
> I have the following setup :
>
> Internet --->BSD firewall>   Opensips
> 78.xx.xx.xx 78.xx.xx.xx
>
>
>
> Both the firewall and the opensips machine have live internet ips.
>
> How should I configure opensips with mediaproxy ?
> The firewall is blocking all ports except the ones I explicitly allow.
>
> Kind Regards,
>
> Nicholas Papadakos
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] sipclient registration with domain name.

[Muhammad Shahzad]

Either you use same DNS server in clients or make your DNS server public
serving a unique DNS domain.

Thank you.



On Wed, May 15, 2013 at 2:10 PM, sermj 2012  wrote:

> i installed opensips,server sucessfully.
> in my pc i have installed Domain name server.
> my pc is working fine with DNS.
> but iam unable to register the clients with domain name.
> instead of i can register the clients with IP address.
>
> please help me
>
>
> nandini
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [OpenSIPS-Devel] Migrating OpenSIPS to GitHub

[Muhammad Shahzad]

Though i don't like git much as a user, perhaps because i am too much
addicted to subversion. But anyways, Congrats!


On Thu, May 9, 2013 at 2:32 AM, Nick Khamis  wrote:

> Congratulations on the successful migration.
>
> Best Regards,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

[Muhammad Shahzad]

Hi Nick,

I haven't used Radius with CDRTool so not very sure how the patch effects
OpenSIPS accounting events triggered to Radius. We are using
much sophisticated business logic with integrated VLR and HRL along with
call billing with respect to national and international roaming cases.
Anyhow all you need to do to get failed transactions events in Radius
server, is to set failed_transaction_flag on those transactions, e.g.

modparam("acc", "aaa_flag", 1)
modparam("acc", "failed_transaction_flag", 3)

...

if ((is_method("INVITE") && !has_totag()) || (is_method("BYE"))) {
setflag(1);
setflag(3);

};

...


Then you should get failed transaction events on Radius regardless of
failure reason, e.g. no-answer (480), cancel (487), busy (486) or even
request timeout (408) and so on. This things works out of the box without
patching Radius and / or OpenSIPS.

Thank you.


On Sat, May 4, 2013 at 3:35 PM, Nick Khamis  wrote:

> Hello Everyone,
>
> Sorry to chime in however, we are also working on failed route
> accounting using radius.
> My impression was that accounting failed sessions was supported by
> Radius when patching
> the server using CDRTool. Would we still need the above script along
> with the failed packets:
>
> http://cdrtool.ag-projects.com/projects/cdrtool/wiki/Installation_Guide
>
> Kind Regards,
>
> Nick.
>
> On 5/2/13, Muhammad Shahzad  wrote:
> > Something like this,
> >
> > if (t_check_status("408")) {
> > if ( t_local_replied("all") ) {
> > # local timeout with no reply received ->
> fr_timer
> > } else if ( t_local_replied("last") ) {
> > # timeout with replies received -> fr_inv_timer
> > } else {
> > # received timeout
> > };
> > };
> > ...
> >
> > Thank you.
> >
> >
> >
> >
> > On Thu, May 2, 2013 at 12:29 PM, qasimak...@gmail.com
> > wrote:
> >
> >> Hi,
> >>
> >> Thanks Bogdan for your reply.
> >>
> >> Now for my question, I want to keep my STOP event on reply as i have
> >> faced
> >> issues when generating event on request time. The thing is how should i
> >> cater the fact that the device has gone offline and there is no response
> >> generated and hence no accounting STOP event.
> >>
> >> Regards,
> >> Qasim
> >>
> >>
> >> On Tue, Apr 30, 2013 at 2:26 PM, Bogdan-Andrei Iancu
> >> wrote:
> >>
> >>> **
> >>> Hello,
> >>>
> >>> All accounting triggers (START/STOP or CDR based) are on replies, so
> >>> when
> >>> the transaction is completed. Of course, all transactions are
> terminated
> >>> in
> >>> OpenSIPS  - either by received replies, either by a timeout (if no
> reply
> >>> received).
> >>>
> >>> If you want to generate the STOP event at BYE request time (versus BYE
> >>> reply time), you can manually do it from script via the acc function
> >>> acc_db_request() (instead of setting the acc flag and letting the acc
> >>> module to generate automatically the event) - the generate event is the
> >>> same. See:
> >>>
> >>> http://www.opensips.org/html/docs/modules/1.9.x/acc.html#id294346
> >>>
> >>> Regards,
> >>>
> >>> Bogdan-Andrei Iancu
> >>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
> >>>
> >>>
> >>> On 04/30/2013 08:00 AM, qasimak...@gmail.com wrote:
> >>>
> >>>  I have tried this scenario. Still if the User B is behind a NAT or is
> >>> unreachable the opensips generates the BYE with retransmitted BYE's and
> >>> the
> >>> dialog is closed but there is no response to BYE received from that
> user
> >>> hence no radius acct request.
> >>>
> >>>  Regards,
> >>> Qasim
> >>>
> >>>
> >>> On Mon, Apr 29, 2013 at 8:36 PM, Muhammad Shahzad
> >>> wrote:
> >>>
> >>>> Per my understanding, accounting event is sent when BYE completes,
> >>>> whether if destination replies with 200 OK or BYE re-transmission
> times
> >>>> out
> >>>> and opensips responds with 408 Request timeout. In each case SIP
> >>>> res

Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

[Muhammad Shahzad]

Something like this,

if (t_check_status("408")) {
if ( t_local_replied("all") ) {
# local timeout with no reply received -> fr_timer
} else if ( t_local_replied("last") ) {
# timeout with replies received -> fr_inv_timer
} else {
# received timeout
};
};
...

Thank you.




On Thu, May 2, 2013 at 12:29 PM, qasimak...@gmail.com
wrote:

> Hi,
>
> Thanks Bogdan for your reply.
>
> Now for my question, I want to keep my STOP event on reply as i have faced
> issues when generating event on request time. The thing is how should i
> cater the fact that the device has gone offline and there is no response
> generated and hence no accounting STOP event.
>
> Regards,
> Qasim
>
>
> On Tue, Apr 30, 2013 at 2:26 PM, Bogdan-Andrei Iancu 
> wrote:
>
>> **
>> Hello,
>>
>> All accounting triggers (START/STOP or CDR based) are on replies, so when
>> the transaction is completed. Of course, all transactions are terminated in
>> OpenSIPS  - either by received replies, either by a timeout (if no reply
>> received).
>>
>> If you want to generate the STOP event at BYE request time (versus BYE
>> reply time), you can manually do it from script via the acc function
>> acc_db_request() (instead of setting the acc flag and letting the acc
>> module to generate automatically the event) - the generate event is the
>> same. See:
>> http://www.opensips.org/html/docs/modules/1.9.x/acc.html#id294346
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>> On 04/30/2013 08:00 AM, qasimak...@gmail.com wrote:
>>
>>  I have tried this scenario. Still if the User B is behind a NAT or is
>> unreachable the opensips generates the BYE with retransmitted BYE's and the
>> dialog is closed but there is no response to BYE received from that user
>> hence no radius acct request.
>>
>>  Regards,
>> Qasim
>>
>>
>> On Mon, Apr 29, 2013 at 8:36 PM, Muhammad Shahzad 
>> wrote:
>>
>>> Per my understanding, accounting event is sent when BYE completes,
>>> whether if destination replies with 200 OK or BYE re-transmission times out
>>> and opensips responds with 408 Request timeout. In each case SIP response
>>> code is set appropriately and you should use stop time as accounting end
>>> time rather then the time your receive account stop request on radius (they
>>> both may differ, e.g. under high load scenarios).
>>>
>>>  Thank you.
>>>
>>>
>>>
>>>  On Mon, Apr 29, 2013 at 3:27 PM, qasimak...@gmail.com <
>>> qasimak...@gmail.com> wrote:
>>>
>>>>Hi,
>>>>
>>>>  I wanted to confirm if radius accounting requests are generated on a
>>>> successful transaction or it can be generated on a received BYE only. To
>>>> elaborate my question you can look at 2 diagrams below. Is first scenario
>>>> correct based on RFC's? My next question is that if scenario A is correct
>>>> then how can we account the call if say user B has gone offline and we do
>>>> not receive 200 OK of the BYE sent?
>>>>
>>>> Can we send a manual accounting request to Radius with acc_aaa_request
>>>> in accounting module?
>>>>
>>>>  *Scenario A:*
>>>>  User AOpenSIPsRadius   User B
>>>>
>>>> |---BYE--->|  |
>>>> |
>>>> |-BYE>|
>>>>  |   |---acct-BYE--->|
>>>>
>>>> *Scenario B:*
>>>> User AOpenSIPsRadius   User B
>>>> |---BYE--->|
>>>> |   |
>>>> |
>>>> |-BYE>|
>>>>  |   |<---200 OK
>>>> -|
>>>>  |<200 OK -|
>>>> |   |---acct-BYE--->|
>>>>
>>>>
>>>>  Regards,
>>>> Qasim Ayyaz Khan
>>>>
>>>>  ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>&g

Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

[Muhammad Shahzad]

Per my understanding, accounting event is sent when BYE completes, whether
if destination replies with 200 OK or BYE re-transmission times out and
opensips responds with 408 Request timeout. In each case SIP response code
is set appropriately and you should use stop time as accounting end time
rather then the time your receive account stop request on radius (they both
may differ, e.g. under high load scenarios).

Thank you.



On Mon, Apr 29, 2013 at 3:27 PM, qasimak...@gmail.com
wrote:

> Hi,
>
> I wanted to confirm if radius accounting requests are generated on a
> successful transaction or it can be generated on a received BYE only. To
> elaborate my question you can look at 2 diagrams below. Is first scenario
> correct based on RFC's? My next question is that if scenario A is correct
> then how can we account the call if say user B has gone offline and we do
> not receive 200 OK of the BYE sent?
>
> Can we send a manual accounting request to Radius with acc_aaa_request in
> accounting module?
>
> *Scenario A:*
> User AOpenSIPsRadius   User B
> |---BYE--->|  |
> |   |-BYE>|
> |   |---acct-BYE--->|
>
> *Scenario B:*
> User AOpenSIPsRadius   User B
> |---BYE--->|  |   |
> |   |-BYE>|
> |   |<---200 OK -|
> |<200 OK -|
> |   |---acct-BYE--->|
>
>
> Regards,
> Qasim Ayyaz Khan
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS, OpenXCAP and Blink: SIP Presence Tutorial

[Muhammad Shahzad]

Thanks, really useful stuff.

Thank you.


On Wed, Apr 17, 2013 at 12:27 PM, Adrian Georgescu wrote:

> Here is the list with OpenSIPS tutorials updated:
>
> http://www.opensips.org/Resources/DocsTutorials#toc25
>
> Regards,
> Adrian
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [RFC] Distributed User Location

[Muhammad Shahzad]

resources hit the performance problem
>> related to the child being blocked while processing a request and the
>> farther away the database or relay the worse it gets.  I think that by
>> addressing the async issue, will automagically create a multitude of
>> solutions for better distribution and load balancing.
>>
>> Adrian
>>
>> On Apr 9, 2013, at 10:17 AM, Bogdan-Andrei Iancu wrote:
>>
>>  Hi,
>>>
>>> Putting together what you said and what Adrian and Muhammad said :
>>>
>>> Actually we may have a distributed USRLOC for 2 purposes: geo
>>> distribution and load distribution - how they are approach it is a bit
>>> different.
>>>
>>> But first let's look into the common part (for the 2 cases) : IMHO, in
>>> both cases we should have the SIP part (opensips) storing the actual full
>>> registration in a certain location (via USRLOC) and an upper layer,
>>> distributed, to keep a mapping between users (AORs) and the location(s)
>>> they are registered with. So:
>>> - local level - OpenSIPS doing classing registrations (a node)
>>> - distributed level - some other tool to keep (in a distributed
>>> fashion) the mapping of AORs on the nodes
>>>
>>> Now, here comes the difference.
>>>
>>> If you do geo distribution, you want to keep registration as closes as
>>> possible to the user. So the registration will be kept on the OpenSIPS node
>>> which was contacted by the user. In this case Chord does not work (at
>>> distributed level) as it has its own alg to distributed data across nodes;
>>> in our case we want to control the distribution and to say what
>>> data/registration stays on what node/opensips.
>>>
>>> If you do load distribution, you want to balance all received
>>> registrations across all existing nodes/opensips - in this case a Chord
>>> like approach will help (as it will do the load distribution for you).
>>>
>>>
>>> As I see the solution : have the 2 layers (local and distributed) as
>>> built in in OpenSIPS and additionally to be able to use different
>>> algorithms to do the mapping between registrations and OpenSIPS nodes.
>>>
>>>
>>> Is the above a good approach ??
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>>>
>>>
>>> On 04/05/2013 04:45 PM, Rudy wrote:
>>>
>>>> Everyone,
>>>>
>>>>   Before we get too off topic, I think the goal should be to design
>>>> something truly distributed. This would be more like what Adrian
>>>> suggested and less like a super node / slave node scenario. The nodes
>>>> should be able to coordinate amongst themselves, again, similar to the
>>>> docs Adrian shared.
>>>>
>>>>   One thing we will need is a consistent hashing alg. Adrian suggested
>>>> Chord, another that works well for us in our implementations is
>>>> Ketama. Either way, it needs to be able to have consistent hashing, so
>>>> that additions / removals of nodes do not change the location of home
>>>> proxy of each registered user.
>>>>
>>>> http://en.wikipedia.org/wiki/**Consistent_hashing<http://en.wikipedia.org/wiki/Consistent_hashing>
>>>>
>>>> Thanks in advance,
>>>> --Rudy
>>>> Dynamic Packet
>>>> Toll-Free: 888.929.VOIP ( 8647 )
>>>>
>>>>
>>>> On Fri, Apr 5, 2013 at 9:39 AM, Muhammad Shahzad
>>>>  wrote:
>>>>
>>>>> Well, i am not much familiar with internals of opensips, i.e. its core
>>>>> and
>>>>> modules and how they interact with each other. But as an abstract
>>>>> idea, i
>>>>> suggest that both Base Node and Super Node should be opensips modules.
>>>>> No
>>>>> change in standard registrar or usrloc modules are actually needed.
>>>>>
>>>>> In the Super Node module, we will have,
>>>>>
>>>>> 1. one db table to store base node addresses for monitoring the Event.
>>>>> 2. one db table to store data received from the Event, lets call it
>>>>> "Event
>>>>> Table".
>>>>> 3. one process to manage "Event Table", pretty much the same way
>>>>> locati

Re: [OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

[Muhammad Shahzad]

Samy, just for research purpose can you share the load details, e.g how
much cps, how many concurrent calls, no. of opensips children, latency
between dispatcher and relays etc. This might help others in designing
large setups.

Thank you.


On Mon, Apr 8, 2013 at 12:34 PM, Adrian Georgescu wrote:

> The best optimization would be to make OpenSIPS core work asynchronous so
> that a request won't block the processing of new requests. Until this
> happens is little you can optimize if you want distributed infra with high
> cps. The lowest hanging fruit today to achieve highest CPS is to colocate
> all external components on the same LAN with OpenSIPS.
>
> Adrian
>
> On Apr 8, 2013, at 11:32 AM, SamyGo wrote:
>
> Hi AG,
> I understand your point, so in this matter we should not only consider the
> latency between dispatcher and relay but also consider the processing
> capability of both components as high CPS might require significant
> resources on relays as well.
>
> Are there any particular areas which you can point out for me to optimize
> this?
>
> Thanks
> Sammy
> On Apr 7, 2013 6:40 PM, "Adrian Georgescu"  wrote:
>
>> As OpenSIPS core is not async, the whole chain of processing a message
>> can cause this. Practically, summing up all RTT for all your database
>> queries, DNS lookups, Radius requests, media reservations can cause this.
>>
>> The farther away each component is the poorest the performance as that
>> child cannot process any new packet until is done with the previous.
>>
>> Adrian
>>
>> On Apr 7, 2013, at 3:27 PM, SamyGo wrote:
>>
>> Hi Sir,
>> Yes, we've optimized our opensips to have enough shared memory as well as
>> the number of children have been increased as well but this situation is
>> still the same.
>>
>> What I can logically think why this is happening is as follow:
>>
>> 1- Media-relays are in another DC,
>> 2- Dispatcher queries relays to get the port info to update the SDP with
>>
>> The delay or slowness of relays might be something causing overall queue
>> length to shoot up.
>>
>> Let me know what you think,
>>
>> --
>> Sammy
>>
>>
>>
>> On Sat, Apr 6, 2013 at 2:43 AM, Muhammad Shahzad 
>> wrote:
>>
>>> Well, if you have very high CPS then typically need more opensips
>>> children processes to handle load. So you need to,
>>>
>>> 1. increase no. of children.
>>> 2. optimize per child and shared memory sizes.
>>>
>>> BTW what does it has to do with media proxy as you are reporting SIP UDP
>>> port only gets hanged?
>>>
>>> Thank you.
>>>
>>>
>>>
>>> On Fri, Apr 5, 2013 at 8:42 PM, SamyGo  wrote:
>>>
>>>> Hello,
>>>>
>>>> I'm working with opensips with heavy CPS, recently I added
>>>> mediaproxy-dispatcher on the server and couple of relays on different
>>>> servers in different Data Center. Everything worked fine until we observed
>>>> that opensips SIP UDP port 5060  got huge queued packets.
>>>>
>>>> That obviously impacts the call processing. I want to know if this is a
>>>> usual behavior? or can this be treated/tuned ?
>>>>
>>>> Thanks,
>>>> Sammy
>>>>
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>> --
>>> Mit freundlichen Grüßen
>>> Muhammad Shahzad
>>> ---
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +49 176 99 83 10 85
>>> MSN: shari_78...@hotmail.com
>>> Email: shaherya...@googlemail.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
>>
>>
>>
>> ___
>> 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
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] MediaProxy behaviour on Heavy call volume.

[Muhammad Shahzad]

Well, if you have very high CPS then typically need more opensips children
processes to handle load. So you need to,

1. increase no. of children.
2. optimize per child and shared memory sizes.

BTW what does it has to do with media proxy as you are reporting SIP UDP
port only gets hanged?

Thank you.



On Fri, Apr 5, 2013 at 8:42 PM, SamyGo  wrote:

> Hello,
>
> I'm working with opensips with heavy CPS, recently I added
> mediaproxy-dispatcher on the server and couple of relays on different
> servers in different Data Center. Everything worked fine until we observed
> that opensips SIP UDP port 5060  got huge queued packets.
>
> That obviously impacts the call processing. I want to know if this is a
> usual behavior? or can this be treated/tuned ?
>
> Thanks,
> Sammy
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [RFC] Distributed User Location

[Muhammad Shahzad]

Well, i am not much familiar with internals of opensips, i.e. its core and
modules and how they interact with each other. But as an abstract idea, i
suggest that both Base Node and Super Node should be opensips modules. No
change in standard registrar or usrloc modules are actually needed.

In the Super Node module, we will have,

1. one db table to store base node addresses for monitoring the Event.
2. one db table to store data received from the Event, lets call it "Event
Table".
3. one process to manage "Event Table", pretty much the same way location
table is managed by usrloc module.
4. some scripting functions for opensips.cfg, to look up in "Event Table"
and do SIP redirect.
5. some MI functions to manually manage base node table and event table.

In the Base Node module, we will have,

1. module parameters to define address of Super Node and event advertise
socket (Super Node will connect to this socket to receive events).
2. a process to monitor usrloc table, such that as soon as a new user
registers, it advertise this to event socket.
3. some scripting functions for opensips.cfg, to send call to Super Node if
lookup function (from registrar module) fails and in reply route to handle
SIP Redirect to send call to destination base node returned by Super Node.
4. some MI functions etc.

Thank you.




On Fri, Apr 5, 2013 at 1:37 PM, Bogdan-Andrei Iancu wrote:

> **
> Hello Muhammad,
>
> Your approach is the correct one (from SIP perspective) IMHO. But there
> are questions on the implementation side too - like the "Super Node" is
> just a storage or it should have SIP capabilities? How much of this
> behavior should be hardcoded in the registrar + usrloc module ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 04/05/2013 04:57 AM, Muhammad Shahzad wrote:
>
> Well at 5 am in the morning while thinking on this topic the only thing
> ringing in my mind is a mechanism similar to IP to IP Gateway. Here is the
> main concept.
>
>  1. We have number of SIP servers running, say sip1.mydomain.com,
> sip2.mydomain.com ... sipN.mydomain.com, each serving domain mydomain.comand 
> a SIP client A can select any one of these servers through DNS look-up
> (or whatever way possible) and registers to that server. Lets call these
> servers as Base Nodes.
>
>  2. Upon successful registration of client A to server sip1.mydomain.com,
> this Registrar Node fires an Event, which can be subscribed by a back-end
> SIP server, lets call it Super Node. This event will only contain following
> things,
>
> a). User part of all Contact URIs of client A with Expiry.
>b). Registrar Node information e.g. its IP address + Port.
>c). SIP domain of client A. (in case of multi-domain setup)
>
>  3. Super Node stores this information in some db back-end (memcache,
> redis, mysql etc.). This is sort of back-to-back register process but
> without SIP or authentication, since that has already been handled on Based
> Node anyway. The Super Node only needs to know which user is registered on
> which Base Node e.g. user 1001 is registered on node sip1.mydomain.com,
> user 1203 is registered on sip6.mydomain.com and so on.
>
>  4. When a SIP client B tries to send INVITE or MESSAGE or SUBSCRIBE to
> SIP client A. The SIP request will arrive on Base Node it is currently
> registered with, say sip2.mydomain.com. This node will first do local
> look-up for location of client A. Upon failure it will forward request to
> Super Node, which will do a look-up on Event database and finds that client
> A is registered on node sip1.mydomain.com, so it will send SIP redirect
> response 302 to requester Base Node. Now the request source node knows the
> address of request destination node, where it will send request next and
> they both, while acting as independent SIP servers, establish SIP session
> between caller and callee. This should work regardless if both nodes serve
> same or different SIP domains.
>
>  5. The Super Node will also give us global presence of all users
> currently registered to all Base Nodes, which may be useful for management
> and monitoring etc.
>
>  Pros:
> 1. Completely independent of network topology and SIP.
> 2. Will work seamlessly for multi and federated domains.
> 3. Scale-able in every direction.
> 4. Minimal overhead for session establishment. Once supper node return
> destination base node address in SIP redirect response, session will
> establish directly between source and destination base node. Further
> optimizations are possible, e.g. base node can cache destination base node
> returned by supper node for any particular user and avoid querying super
> node for recurring SIP sessions.
>
>

Re: [OpenSIPS-Users] [RFC] Distributed User Location

[Muhammad Shahzad]

Well at 5 am in the morning while thinking on this topic the only thing
ringing in my mind is a mechanism similar to IP to IP Gateway. Here is the
main concept.

1. We have number of SIP servers running, say sip1.mydomain.com,
sip2.mydomain.com ... sipN.mydomain.com, each serving domain
mydomain.comand a SIP client A can select any one of these servers
through DNS look-up
(or whatever way possible) and registers to that server. Lets call these
servers as Base Nodes.

2. Upon successful registration of client A to server sip1.mydomain.com,
this Registrar Node fires an Event, which can be subscribed by a back-end
SIP server, lets call it Super Node. This event will only contain following
things,

   a). User part of all Contact URIs of client A with Expiry.
   b). Registrar Node information e.g. its IP address + Port.
   c). SIP domain of client A. (in case of multi-domain setup)

3. Super Node stores this information in some db back-end (memcache, redis,
mysql etc.). This is sort of back-to-back register process but without SIP
or authentication, since that has already been handled on Based Node
anyway. The Super Node only needs to know which user is registered on which
Base Node e.g. user 1001 is registered on node sip1.mydomain.com, user 1203
is registered on sip6.mydomain.com and so on.

4. When a SIP client B tries to send INVITE or MESSAGE or SUBSCRIBE to SIP
client A. The SIP request will arrive on Base Node it is currently
registered with, say sip2.mydomain.com. This node will first do local
look-up for location of client A. Upon failure it will forward request to
Super Node, which will do a look-up on Event database and finds that client
A is registered on node sip1.mydomain.com, so it will send SIP redirect
response 302 to requester Base Node. Now the request source node knows the
address of request destination node, where it will send request next and
they both, while acting as independent SIP servers, establish SIP session
between caller and callee. This should work regardless if both nodes serve
same or different SIP domains.

5. The Super Node will also give us global presence of all users currently
registered to all Base Nodes, which may be useful for management and
monitoring etc.

Pros:
1. Completely independent of network topology and SIP.
2. Will work seamlessly for multi and federated domains.
3. Scale-able in every direction.
4. Minimal overhead for session establishment. Once supper node return
destination base node address in SIP redirect response, session will
establish directly between source and destination base node. Further
optimizations are possible, e.g. base node can cache destination base node
returned by supper node for any particular user and avoid querying super
node for recurring SIP sessions.

Cons:
1. Well, the key problem i can guess is of course the Event database size
and speed, as it will have information on every user registered to every
Base Node. I suggest memory cache db such as Redis would be idle for this
storage.
2. Bandwidth consumed in Event transport. We can apply compression and make
event queues as optimization.

Comments and suggestions are highly welcome.

Thank you.




On Thu, Apr 4, 2013 at 2:05 PM, Vlad Paiu  wrote:

> Hello all,
>
> We would like to get suggestions and help on the matter of distributing
> the user location information.
> Extending the User Location with a built-in distributed support is not
> straight forward - it is not about simply sharing data - as it is really
> SIP dependent and network limited
>
> While now, by using the OpenSIPS trunk, it is possible to just share the
> actual usrloc info ( by using the db_cachedb module and storing the
> information in a MongoDB cluster ), you can encounter real-life scenarios
> where just sharing the info is not enough, like :
> - NAT-ed clients, where only the initial server that received the
> Register has the pin-hole open, and thus is the only server that can relay
> traffic back to the respective client
> - the user has a SIP client that only accepts traffic from the server
> IP that it's currently registered against, and thus would reject direct
> traffic from other IPs ( due to security reasons )
>
> We would like to implement a true general solution for this issue, and
> would appreciate your feedback on this. Also we'd appreciate if you could
> share the needs that you would have from such a distributed user location
> feature, and the scenarios that you would use such a feature in real-life
> setups.
>
>
> Best Regards,
>
> --
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
>
> __**_
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman

Re: [OpenSIPS-Users] Latest version suitable for production use

[Muhammad Shahzad]

Well, both 1.8 and 1.9 branches are stable now. But i have mostly 1.8
running in production without any problem. You can use 1.9 if you need
fairly new features and modules that are not present in 1.8 like support
for Web Socket VIA header parsing etc.

Thank you.


On Thu, Apr 4, 2013 at 2:30 PM, Telecube - John wrote:

> Hi,
>
> Can someone confirm the current suggested version of opensips to use in a
> production environment please?
>
> Thanks,
> John
>
> __**_
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS and Prepaid accounts

[Muhammad Shahzad]

Check the billing_* tables in radius database. There are tables for rates,
customers and profiles you need to fill in. See the sample data to get the
idea.

Thank you.



On Mon, Apr 1, 2013 at 2:14 PM, Leonardo Uzcudun  wrote:

> That's clear but how do I configure for example to the SIP user
> t...@example.com to have a prepaid billing and its credit?
>
>
>   ------
> *Da:* Muhammad Shahzad 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Lunedì 1 Aprile 2013 14:47
> *Oggetto:* Re: [OpenSIPS-Users] OpenSIPS and Prepaid accounts
>
> Well, all you need to do is configure the rating engine to bill  the calls,
>
> http://callcontrol.ag-projects.com/projects/callcontrol/wiki/Installation
>
> If this does not work, then please give exactly what problem you observe
> with respective logs.
>
> Thank you.
>
>
>
> On Sun, Mar 31, 2013 at 6:53 PM, leo  wrote:
>
> Hello Davide:
>
> Can i ask you how you've got "/I'm playing with OpenSIPS and prepaid
> accounts, and for each single sips
> users it's working fine./" working?
> I've already the Opensips server, mediaproxy, radius, cdr and callcontrol
> but i'm blocked on how to set the prepaid billing, how do you assign to the
> SIP user?
> Thanks a lot.
> Leo.
>
>
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-and-Prepaid-accounts-tp7583796p7585568.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
>
>
>
>
> --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.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
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] CDRTool

[Muhammad Shahzad]

Well again, you do radius setup ONLY when you need to bill the the calls
using rating engine. You can skip it if you just want CDRTool without call
rating.

Thank you.


On Mon, Apr 1, 2013 at 2:09 PM, leo  wrote:

> Of course with database only it works.
>
> But following the CDRTool install guide
> (http://cdrtool.ag-projects.com/projects/cdrtool/wiki/Installation_Guide)
> it
> says that for the mediaproxy the accounting must be done with radius and
> database.
>
> So, should it be only database or both?
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-tp7585554p7585572.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
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS and Prepaid accounts

[Muhammad Shahzad]

Well, all you need to do is configure the rating engine to bill  the calls,

http://callcontrol.ag-projects.com/projects/callcontrol/wiki/Installation

If this does not work, then please give exactly what problem you observe
with respective logs.

Thank you.



On Sun, Mar 31, 2013 at 6:53 PM, leo  wrote:

> Hello Davide:
>
> Can i ask you how you've got "/I'm playing with OpenSIPS and prepaid
> accounts, and for each single sips
> users it's working fine./" working?
> I've already the Opensips server, mediaproxy, radius, cdr and callcontrol
> but i'm blocked on how to set the prepaid billing, how do you assign to the
> SIP user?
> Thanks a lot.
> Leo.
>
>
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-and-Prepaid-accounts-tp7583796p7585568.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
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] CDRTool

[Muhammad Shahzad]

Disable radius unless you are using it and have it configured. To do this
set,

accounting = database

Make sure you have created the database and user with appropriate
permissions (if you are a newbie then just grant all permissions) as
mentioned in database section of config.ini.

Thank you.



On Sun, Mar 31, 2013 at 11:56 PM, leo  wrote:

> Hello:
>
> Nobody has an idea about it?
> All the posts i've found in the forum say they have fixed installing
> python-pyrad 1.1 but considering the AG site it seems that the
> mediaproxy-dispatcher was fixed to work with python-pyrad 1.2 too.
> Is this problem about the python-pyrad version still alive?
>
> I'm still getting "Apr  1 00:39:54 sip-dev media-dispatcher[6761]: fatal
> error: cannot read the RADIUS configuration file"
>
> Of course the file exists (radius configuration file:
> /etc/opensips/radius/client.conf) and it is configured.
>
> Thanks.
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-tp7585554p7585569.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
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Parsing null parameter

[Muhammad Shahzad]

This is a database inconsistency issue, check gw_attrs field for the
gateway which gives this error, most likely there is only one value there
where as your opensips script expect at least 2 comma separated values.

Thank you.


On Tue, Mar 26, 2013 at 10:46 AM, M.Khaled W Chehab wrote:

> Hi,
>
> ** **
>
> I got the below error while $var(prefix) = null  ,how to avoid this error
> 
>
> $var(prefix) = $(avp(gw_attrs){csv.value,1});
>
> ** **
>
> ERROR:core:tr_eval_csv: Index out of bounds
>
> ERROR:core:do_assign: no value in right expression
>
> ERROR:core:do_assign: error at line: 877
>
> ** **
>
> Regards
>
> ** **
>
> ** **
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Muhammad Shahzad]

If you are unfamiliar with rtp proxy and how it works, then it would be
better for you to use engage_rtp_proxy rather then offer / answer model.
Also RTP Proxy requires public IP address, its likely not to work on
private subnets (unless you have all SIP entities on same LAN, in which
case theoretically it should work but i have never tested it myself).

Thank you.


On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis  wrote:

> I am not sure if this is the correct place to post OpenSIPS+RTPProxy
> questions however, I tried to subscribing to the RTP proxy mailing
> list and never heard from them since. If it is ok to post RTP proxy
> related questions here I am trying to test OpenSIPS with RTP proxy
> with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
> RTPPoxy) just for testing.
>
> The code I am using is:
>
> route {
>  force_rport();
> }
> route[1] {
> if (is_method("INVITE")) {
> t_on_branch("1");
> t_on_reply("1");
> t_on_failure("1");
>
> if (has_body("application/sdp"))  rtpproxy_offer();
> }
> else if (is_method("BYE|CANCEL")) {
> unforce_rtp_proxy();
> }
>
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
> onreply_route[1] {
>  if (has_body("application/sdp")) rtpproxy_answer();
> }
>
>
> There is no way audio using RTP proxy, but audio is fine between the
> UA without including the RTP proxy related script. Looking at the log
> I found that RTP is prefilling the callers address twice, but not the
> callees address.
>
>
> INFO:main: rtpproxy started, pid 7287
> INFO:handle_command: new session
> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
> d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
> INFO:handle_command: new session on a port 35010 created, tag
> d23f0168-538e-e211-8550-001b7700a65b;1
> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5062
> INFO:handle_command: new session
> ae450168-538e-e211-8550-001b7700a65b@oakville, tag
> d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
> INFO:handle_command: new session on a port 22982 created, tag
> d23f0168-538e-e211-8550-001b7700a65b;2
> INFO:handle_command: pre-filling caller's address with 192.168.2.101:5064
> INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 35010/0 is cleaned up
> INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
> INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
> relayed, 0 dropped
> INFO:remove_session: session on ports 22982/0 is cleaned up
>
> Is it possible to test RTP relaying with everything on the same network?
>
> Thanks in Advance,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Load Balancer module for REGISTER as well as INVITE?

[Muhammad Shahzad]

Load balancer depends on SIP address only for routing, so just about any
SIP server and just about any SIP method, you can use this module for load
balancing. However, you need to plan carefully how on-net calls and
presence will work in such an architecture!

Thank you.


On Mon, Mar 18, 2013 at 5:56 PM, Tuomas Kaikkonen <
tuomas.kaikko...@twistpair.com> wrote:

> Can the Load Balancer module be configured to balance REGISTERs as well as
> the INVITES so that the above mentioned setup would work? OR is the Load
> Balancer module just useful for balancing RTP Proxy / media server
> resources for INVITEs?
>
> ** **
>
> I’m new to the Load Balancer module of OpenSIPs. I am running OpenSIPs
> stable branch 1.7 – just by looking examples from the documentation it
> looks like INVITE load balancing is supported.
>
> ** **
>
> Tuomas Kaikkonen
>
> ** **
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK Loop when changing contact on_reply: Please Help!!!

[Muhammad Shahzad]

Yup, that's expected to happen. ACK is sent to Contact header of 200 OK.
So, if you mess up with it, then unexpected results will happen, as in your
case you are perhaps setting Contact address of 200 OK pointing to opensips
itself, instead of destination party, so ACK will obviously loop as
expected.

Thank you.


On Mon, Mar 18, 2013 at 5:55 PM, Nick Khamis  wrote:

> Hello Everyone,
>
> We are changing the "Contact" header in the on_reply to a public ip
> address using:
>
> onreply_route[1] {
> xlog("incoming reply\n");
> if (has_body("application/sdp")) {
> remove_hf("Contact");
> append_hf("Contact:
> \r\n");
> append_hf("P-hint: Onreply-route -
> fixcontact \r\n");
>
> }
> }
>
> When doing so, ACK is going into a loop:
>
> U 2013/03/18 13:42:11.021017 75.15.201.2:5060 -> 192.168.2.5:5060
> ACK sip:75.15.201.2;lr;did=b03.4af9f8f3 SIP/2.0.
> Call-ID: VQUK2UGSQBCPHEW27UN5NBJIQM@81.201.86.45.
> CSeq: 102 ACK.
> From: "15178334003" ;tag=91641.
> To: ;tag=2643FD58-346926A7.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 75.15.201.2;branch=z9hG4bKcd1e.d68abdc.2.
> Via: SIP/2.0/UDP 7
>
>
> Your help is greatly appreciated,
>
> Nick.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips RTP issue

[Muhammad Shahzad]

ethods")
> modparam("usrloc", "flags_column", "flags")
> modparam("usrloc", "cflags_column", "cflags")
> modparam("usrloc", "user_agent_column", "user_agent")
> modparam("usrloc", "received_column", "received")
> modparam("usrloc", "socket_column", "socket")
> modparam("usrloc", "path_column", "path")
>
>
> ##
> ## NAT Traversal Module Parameters
> ##
> modparam("nat_traversal", "keepalive_interval", 60)
> modparam("nat_traversal", "keepalive_method", "OPTIONS")
> modparam("nat_traversal", "keepalive_from", "sip:keepal...@xx.xx.xx.xx
> :5060")
> modparam("nat_traversal", "keepalive_state_file",
> "/tmp/opensips_keepalive_state")
>
> ##
> ## Dispatcher Module Parameters
> ##
> modparam("dispatcher", "db_url", "mysql://opensips:opensipsrw@localhost
> /opensips")
> modparam("dispatcher", "flags", 2)
> modparam("dispatcher", "use_default", 0)
> modparam("dispatcher", "force_dst", 1)
> modparam("dispatcher", "dst_avp", "$avp(271)")
> modparam("dispatcher", "attrs_avp", "$avp(272)")
> modparam("dispatcher", "grp_avp", "$avp(273)")
> modparam("dispatcher", "cnt_avp", "$avp(274)")
> modparam("dispatcher", "hash_pvar", "$avp(273)")
> modparam("dispatcher", "ds_ping_method", "OPTIONS")
> modparam("dispatcher", "ds_ping_from", "sip:sipch...@xx.xx.xx.xx:5060")
> modparam("dispatcher", "ds_ping_interval", 10)
> modparam("dispatcher", "ds_probing_threshhold", 3)
> modparam("dispatcher", "ds_probing_mode", 1)
> modparam("dispatcher", "options_reply_codes", "501,403,404,400,200")
>
> ##
> ## MI-FIFO Module Parameters
> ##
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>
>
> ##
> ## Multiple Module Parameters
> ##
>
> ##
> ## Main Request Routing
> #
> route{
>
> if (is_method("INVITE"))
> {
>   if (nat_uac_test("16"))
>   {
>   fix_nated_contact();
>   force_rport();
> };
>
>
> ds_select_dst("1", "0");
> forward();
> exit;
>
> }
> }
>
>
> Thanks
> Jagadish
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK Timer

[Muhammad Shahzad]

No Khaled, loose_route is not inside ACK if condition, it should be where
it is in your script. Also the INVITE if  he mentioned is in else of
has_totag, not loose_route. Read his reply carefully and you will get the
idea. ;-)

Thank you.


On Sat, Mar 16, 2013 at 10:24 PM, M.Khaled W Chehab wrote:

> Dear Bogdan,
>
> ** **
>
> Please can you confirm  the changes I mark down with  red color or please
> correct it if its wrong .
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> if (has_totag()) {
>
> ** **
>
>   
>
> *if (is_method("ACK")) {*
>
> * $avp(timeout2) = 3540; *
>
> * *
>
> * *
>
> *###should I add here loose_route();*
>
> *  }*
>
>   # sequential request withing a dialog should
>
>   # take the path determined by record-routing
>
> ** **
>
>   if (loose_route()) {
>
>  # validate the sequential request against dialog
>
>  #if ( $DLG_status!=NULL && !validate_dialog() ) {
>
>  #  xlog("In-Dialog $rm from $si (callid=$ci) is
> not valid according to dialog\n");
>
>  #  #exit;
>
>  #}
>
> ** **
>
>  if (is_method("BYE")) {
>
> end_media_session();
>
> setflag(1); # do accounting ...
>
> setflag(3); #transaction falis
>
> ** **
>
>  } else if (is_method("INVITE")) {
>
> # even if in most of the cases is useless, do
> RR for
>
> # re-INVITEs alos, as some buggy clients do
> change route set
>
> # during the dialog.
>
> record_route();
>
>  }
>
> ** **
>
>  if (check_route_param("nat=yes")) {
>
> setflag(5);
>
>  }
>
> ** **
>
>  # route it out to whatever destination was set by
> loose_route()
>
>  # in $du (destination URI).
>
>  route(1);
>
> ** **
>
>   } else {
>
> ** **
>
> *if ( is_method("INVITE")) {**
> **$avp(timeout2) = 3; **
> **}***
>
>  if ( is_method("ACK") ) {
>
> if ( t_check_trans() ) {
>
># non loose-route, but stateful ACK;
> must be an ACK after 
>
># a 487 or e.g. 404 from upstream server
> 
>
>t_relay();
>
>exit;
>
> } else {
>
># ACK without matching transaction ->**
> **
>
># ignore and discard
>
>exit;
>
> }
>
>  }
>
>  sl_send_reply("404","Not here");
>
>   }
>
>   exit;
>
>}
>
> Regards
>
> ** **
>
> *From:* Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> *Sent:* Thursday, March 14, 2013 6:59 PM
> *To:* OpenSIPS users mailling list
> *Cc:* M.Khaled W Chehab; 'Brito Nicolas'
>
> *Subject:* Re: [OpenSIPS-Users] ACK Timer
>
> ** **
>
> Hi Khaled,
>
> your mistake here is to set the timeout for INVITE under the
> has_totag() branch - initial INVITEs do not have TO tags.
>
>
> Try:
>
>
> if (has_totag()) {
>  if (is_method("ACK")) {
>  $avp(timeout2) = 3540;
>   }
>   . (loose_route)
>
> } else {
>
> if ( is_method("INVITE")) {
> $avp(timeout2) = 3;
> }
>
> ...
>
> }
>
>
>
> Regards,
> Bogdan
>
> 
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>
> http://www.opensips-solutions.com
>
>
>
> 
>
> ** **
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Record-routing & failover (drouting)

[Muhammad Shahzad]

No, I think the serial forked invite contains same RR and in general all
changes you did to original invite just before creating the transaction (by
calling t_newtrans or t_relay or any t_* function that creates transaction).

Thank you.


On Thu, Mar 14, 2013 at 5:15 PM, Max Mühlbronner  wrote:

>  Hi,
>
> I am not sure about record-routing in combination with failover of
> drouting. Maybe someone knows for sure :)
>
> If i got a configuration where i am record_routing on inital invite, but
> later there is a failover (use_next_gw() returns true) and the call is sent
> to the next gateway. But the serial forked call (second INVITE) is missing
> the Record-route header?
>
>
> Does this mean i just have to explicitly call record routing again on
> failover? But to me it seems like this can't be right, or is this
> correct/expected behaviour?
>
>  if (use_next_gw()) {
> ...
> record_route();
>
> }
>
>
>
> Best Regards
>
> --
> Max Mühlbronner
>
> 42com Telecommunication GmbH
> Straße der Pariser Kommune 12-16
> 10243 Berlin
>
> E-Mail: m...@42com.com
> Web: www.42com.com
>
> Firmenangaben/Company information:
> Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
> Umsatzsteuer-ID/VAT-ID: DE223812306
> Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig
>
> Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese 
> sind möglicherweise vertraulich und ausschließlich für den Adressaten 
> bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise 
> erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder 
> per E-Mail.
>
> This message is intended only for the use of the individual or entity to 
> which it is addressed. If you have received this message by mistake, please 
> notify us immediately.
>
>
> _______
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Mit freundlichen Grüßen
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Fwd: Re: How to protect OpenSIPS from undesidered requests (DoS attack?)

[Muhammad Shahzad]

While this will work for small setups but i have feeling that this won't be
suitable for high load productive systems, since it does same thing as
fail2ban but runs INLINE, blocking other sip requests till it is finished.

Thank you.


On Wed, Mar 6, 2013 at 8:48 PM, Hubert Mickael  wrote:

>  Hi,
>  Pike module to stop flood ?
> I have add perl script at pike to add firewall rule in my freebsd.
>
> Example opensips conf:
>
> #--- module pike ---
> loadmodule "pike.so"
> #--- pike params 
> modparam("pike", "sampling_time_unit", 2)
> modparam("pike", "reqs_density_per_unit", 80)
> modparam("pike", "remove_latency", 130)
> modparam("pike", "pike_log_level", -1)
>
> in script:
>
> *if(!pike_check_req())**
> **{**
> **if(perl_exec("pikesendmail"))**
> **{**
> **xlog("L_INFO","Fonction perl_exec PIKE OK");**
> **}**
> **xlog("L_WARN","PIKE_CHECK_REQ banned IP $si because of
> flooding requests");**
> **exit;**
> **}*
>
> perl script:
>
>
> sub pikesendmail
> {
> MIME::Lite->send('smtp', 'smtp.');
>
> my $serverIP = OpenSIPS::AVP::get("serverIP");
> my $sourceIP = OpenSIPS::AVP::get("sourceIP");
>
> my @exceptions = (@my IP);
>
> my $logfile = "/var/log/pikemodule.log";
> my $date = localtime();
> open LOGFILE, ">>$logfile" or die "cannot open logfile $logfile
> for append: $!";
>
> my $subject = "IP $sourceIP blocked by server $serverIP";
> my @body ;
> my @argsbash ;
> my @listeIP ;
> my $maxid ;
> my $newid ;
> my $reglepresente = 0 ;
> my $inhib = 0 ;
> my $i=0;
>
> for $i (@exceptions)
> {
> if($sourceIP eq $i)
> {
> $inhib = 1 ;
> last ;
> }
> }
>
> foreach(`ipfw list | grep ^005 | awk -F" " {'print \$5'}`)
> {
> push (@listeIP,$_) ;
> }
>
> for(@listeIP){
> print "$_";
> if($_ =~ $sourceIP){
> #print "regle deja presente\n";
> $reglepresente = 1 ;
> }
> }
>
> if($reglepresente == 0 && $inhib == 0){
>
> $maxid=`ipfw list | grep ^005 | tail -n1 | awk -F" "
> {'print \$1'} | sed "s/^00//"`;
> if ($maxid eq ''){
> $newid = 500 ;
> }else{
> $newid = $maxid+1 ;
> }
>
>
> @argsbash = ("ipfw", "add $newid deny ip from $sourceIP to
> me");
> if(system(@argsbash) == 0
> or die "system @argsbash failed: $?"){
> print LOGFILE "$date INFO : Nouveau blocage pour
> SIP flooding \n";
> print LOGFILE "$date INFO : Regle IPFW appliquee
> ID $newid \n";
> log(L_INFO, "SIP Flooding, IP $sourceIP blocked
> with IPFW rule $newid\n");
> }
>
> open(EMAILB,"/usr/local/libexec/templ_email.tpl") || die
> ("Erreur d'ouverture de EMAILB") ;
> while () {
> $_ =~ s/PARA1/$sourceIP/g;
> $_ =~ s/PARA2/$serverIP/g;
> $_ =~ s/PARA3/$newid/g;
> push (@body,$_);
> }
>
> close(EMAILB);
>
> # Création d'un objet MIME::Lite avec les en-têtes du
> message
> my $message = MIME::Lite->new(
> From   => 'OpenSIPS 
> 
> ',
> To => '',
> Subject=> "$subject",
> "X-Mailer" => 'OpenSIPS',
>Type   => 'text/html',
> Data   => "@body",
> );
>
> if($message->send()){
> print LOGFILE "$date INFO : Mail envoye pour
> blocage IP $sourceIP\n";
> log(L_INFO, "SIP Flooding, mail has been sent\n");
> }
>
> close LOGFILE ;
> }
>
> return 1;
> }
>
> bye
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] How to protect OpenSIPS from undesidered requests (DoS attack?)

[Muhammad Shahzad]

e matching and is only an
> alias for
> #  (?:::f{4,6}:)?(?P\S+)
> # Values:  TEXT
> #
>
> failregex = Auth error for .* from  cause -[0-9]
>
> # Option:  ignoreregex
> # Notes.:  regex to ignore. If this regex matches, the line is ignored.
> # Values:  TEXT
> #
> ignoreregex =
>
> and on /etc/fail2ban/jail.conf
>
> [opensips]
> enabled  = true
> filter   = opensips
> action   = iptables-allports[name=opensips, protocol=all]
> sendmail-whois[name=opensips, dest=[hidden email],
> sender=[hidden email]]
> logpath  = /var/log/opensips.log
> maxretry = 3
> bantime = 7200
>
>
> Regards
>
>
> ___
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> --
>  If you reply to this email, your message will be added to the discussion
> below:
>
> http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-protect-OpenSIPS-from-undesidered-requests-DoS-attack-tp7585091p7585097.html
>  To unsubscribe from How to protect OpenSIPS from undesidered requests
> (DoS attack?), click here.
> NAML<http://opensips-open-sip-server.1449251.n2.nabble.com/template/NamlServlet.jtp?macro=macro_viewer&id=instant_html%21nabble%3Aemail.naml&base=nabble.naml.namespaces.BasicNamespace-nabble.view.web.template.NabbleNamespace-nabble.view.web.template.NodeNamespace&breadcrumbs=notify_subscribers%21nabble%3Aemail.naml-instant_emails%21nabble%3Aemail.naml-send_instant_email%21nabble%3Aemail.naml>
>
>
>
> --
> View this message in context: Re: How to protect OpenSIPS from
> undesidered requests (DoS 
> attack?)<http://opensips-open-sip-server.1449251.n2.nabble.com/How-to-protect-OpenSIPS-from-undesidered-requests-DoS-attack-tp7585091p7585123.html>
>
> Sent from the OpenSIPS - Users mailing list 
> archive<http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html>at
>  Nabble.com.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] segfault in db_mysql.so and others..

[Muhammad Shahzad]

I am not sure but it seems that the data in one or more column is larger
then column width provided by MySQL, resulting in buffer overflow. Can you
check which SQL query might have caused this problem? There are two
possibilities,

1. Table encoding is multi-byte char and opensips is somehow considering it
single-byte char. But in this case it should ALWAYS crash, not randomly as
you said.
2. You imported larger amount of data in column with lesser width while
size check constraints were disabled. You need to find such records and
increase respective column width to match data size. You may also consider
to shorten the data in respective columns of those records.

Thank you.


On Wed, Mar 6, 2013 at 1:54 PM, Brett Nemeroff  wrote:

> Any ideas here? :/
>
>
> On Tue, Mar 5, 2013 at 10:10 AM, Brett Nemeroff wrote:
>
>> Hey All,
>> I'm running opensips 1.8.2 svnrevision: 2:9628M
>>
>> I'm getting some random crashes. Load isn't terribly high on these boxes
>> and I can't figure out what specifically is causing the crash.
>>
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration

[Muhammad Shahzad]

Great stuff Maruzzelli. Thanks for sharing.

Thank you.


On Mon, Mar 4, 2013 at 7:05 PM, Giovanni Maruzzelli wrote:

> Ciao VOIPers,
>
> it's my pleasure to bring to your attention a new tutorial on realtime
> integration between OpenSIPS and FreeSWITCH.
>
> It's a cut and paste tutorial, so you can test it right away, eg on a
> virtual machine, and when confident customize it and put it in
> production.
>
> The stack is Debian Squeeze 6.x, OpenSIPS 1.8.x, FreeSWITCH 1.2.x,
> OpenSIPS-CP as GUI, MySQL as database.
>
> You can find the tutorial at URL:
> http://www.opensips.org/Resources/DocsTutFreeSwitch with all required
> files.
>
> Please let us know what do you think about it, and what other
> tutorials you would like to read (at the moment I'm thinking at an HA
> install of FusionPBX+FreeSWITCH+OpenSIPS, but other requests will be
> taken into account too).
>
> See below for a small excerpt of this tutorial:
>
> =
> 1.1  Scope
>
> This tutorial can be used as a cut and paste complete and working
> installation. Please follow strictly all the steps, in the order
> given.
>
> This tutorial presents the concept and implementation of a realtime
> integration of OpenSIPS SIP server and FreeSWITCH media server.
>
> OpenSIPS is used a SIP server - users are registering with it, it
> routes calls, etc - while the purpose of FreeSWITCH is to provide a
> full set of media services - like voicemail, conference,
> announcements, etc.
>
> It is a realtime integration because both OpenSIPS and FreeSWITCH are
> provisioned in the same time when comes to user accounts - when
> creating a new OpenSIPS user, automatically FreeSWITCH will learn
> about it an provide and configure all necessary media services for it.
>
> Both OpenSIPS and FreeSWITCH will be provisioned (for user accounts)
> via a shared mysql database.
>
> All FreeSWITCH functionalities will be available to OpenSIPS users by
> prefixing "*" (eg: star) to the extension dialed. *1234 will be passed
> to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as
> *1234
>
> 
>
> 1.2  Setup presentation
>
> This tutorial can be used as a cut and paste complete and working
> installation. Please follow strictly all the steps, in the order
> given.
>
> The following services will be offered by FreeSWITCH by this
> integrated configuration:
>
> voicemail - users will get access to their mailbox; authentication
> will be done by OpenSIPS; while FreeSWITCH will only provide voicemail
> IVR (with access PIN);
> conference' - OpenSIPS will detect and forward calls related to
> conference service (based on prefixes) to FreeSWITCH, which will
> provide access (pin based) to the conference rooms;
> all functionalities - OpenSIPS users will prefix * to reach the
> corresponding extension in FreeSWITCH (*1234 will be passed to
> FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as
> *1234)
>
> =
>
> ciao for now,
>
> -giovanni
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] NAT

[Muhammad Shahzad]

Setting modparam ice_candidate to low-priority will enable ICE for all
calls in media proxy with media proxy's own address as TURN service with
low-priority. However if you want to control priority on per call basis
then then you can define ice_candidate_avp and set priority of media proxy
relay ICE candidates at run time on per call basis. You will just need to
set specified AVP within route definition in opensips.cfg.

Thank you.


On Fri, Mar 1, 2013 at 3:34 PM, Leonardo Uzcudun  wrote:

> Hello saul:
>
> I've now running the OpenSIPS server with the Mediaproxy.
> Some natted UA are working and others no. Those that can't, thay can call
> but no audio/video.
> I would like to enable the ICE (cause the UA also support it) but i'm
> really stuck on the wiki page you've recomended. Could you help me on how
> to enable ICE?
> I think i've to add the following lines to the opensips.conf file:
>
> modparam("mediaproxy", "ice_candidate", "low-priority")
> modparam("mediaproxy", "ice_candidate_avp", "$avp(ice_candidate)")
>
> but i don't understand what should i do with the followings ones:
>
> route {
> ...
> $avp(s:ice_candidate) := "low-priority";
> ...
> }
>
> and
>
> 1 if ((ice_candidate != none and ice_candidate_avp != none) 2 and SDP 
> offer contains a=ice-pwd 3 and a=ice-ufrag and a=candidate line(s) 
> then: 4 append to the SDP the following line:5 
> a=candidate:R(random string) 1 UDP PRIORITY MP_IP MP_PORT typ relay6 
> a=candidate:R(random string) 2 UDP PRIORITY MP_IP MP_RTCP_PORT typ relay
>
> Thanks for your help.
> BR,
>
> Leo.
>
>   --
> *Da:* Saúl Ibarra Corretgé 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Giovedì 28 Febbraio 2013 12:07
> *Oggetto:* Re: [OpenSIPS-Users] NAT
>
>
> On Feb 28, 2013, at 11:47 AM, leo wrote:
>
> > Thanks Adrian.
> >
> > There was a sense in my question, having the RTP traffic P2P would be
> > translated in:
> > 1) better user experience, less jitter and latency.
> > 2) saving to install one or more mediaproxies. Considering that all RTP
> > traffic will pass on them, they will need to have high bandwidth
> (symmetric
> > up/down) and maybe CPU/RAM.
> >
>
> Fixing the signaling is not enough because the media flows through
> different ports and with symmetric NATs you don't know what ports those
> will be. To (try to) use P2P media you need your endpoints to use ICE.
> MediaProxy is ICE aware, so if it detects that media is flowing P2P it will
> bail out. You can find more information about this here:
> http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/ICE
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
> ___
> 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
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] NAT

[Muhammad Shahzad]

You are missing one fundamental fact, that is you have to handle NAT for
both signalling and media. From your description it looks signalling is
going perfect (NAT is correctly handled), since you are able to establish
call between two clients successfully, clients can register, make call,
accept call and hangup call with your server. So main goal of NAT Traversal
module is achieved.

However, there is no media on call, so media NAT is not handled. NAT
Traversal and / or NAT Helper modules may try to fix media NAT issues as
well by manipulating SDP but in so many case they will be simply NOT enough
for this purpose. Especially in case of 3g and corporate networks, which
may have very very complex network typology with multiple layers of NAT (so
called Nested NAT). So rtp / media proxy is the ONLY solution that can
handle media across such complex networks.

If you have really good sip clients with support for STUN / TURN / ICE etc.
and you somewhat control over client data network environment, them you may
fix media NAT issues up to 90% but in about 5-10% cases you will still need
a media relay.

Thank you.


On Mon, Feb 25, 2013 at 11:51 PM, leo  wrote:

> Hello,
>
> Unfortunately after reading the forum i've to open a new post about NAT
> because i couldn't find a clear solution and information for my problem.
> I've also read the NAT Traversal module documentation.
>
> I've an OpenSIPS server (version 1.8.2) on a Debian system (6.0.7 -
> 2.6.32-5-686).
> OpenSIPS was installed by the apt-get install using the apt.opensips.org
> repository and configured with osipsconfig (residential script with
> ALIASES,
> AUTH, DBACC, DBUSRLOC and DIALOG).
>
> The UAs can register to the OpenSIPS server. They can place the call but i
> 've no audio no video.
> The OpenSIPS server has a public IP address (so, no natted).
> The UAs could be natted or with public ip thru 3G.
>
> I wouldn't like to use rtproxy or mediaproxy cause the rtp traffic would be
> passing by those servers (am i correct?) adding jitter and latency.
> I would set up the system in the way the the rtp traffic would be P2P.
> Would
> NAT Traversal be the solution? How it should be configured (i've already
> enabled the required modules too)?
>
> Thanks a lot.
>
> Leo.
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/NAT-tp7584918.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
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK/BYE delivery

[Muhammad Shahzad]

Remove all workarounds and set OpenSIPS as your outbound proxy in your
VOIPSwitch. This should do the trick.

Thank you.


On Fri, Feb 22, 2013 at 11:20 PM, M.Khaled W Chehab wrote:

> Hi,
>
> ** **
>
> I am using Resedential.conf file with permission module
>
> I have a problem that when a call behind nat send  a call to voipswitch and 
> voipswitch forward the call to opensips -àtrunk,
>
> The 100 trying ,180 ringing and the 200 Ok work fine unless the ACK reply for 
> the 200 ok will be directly forwarded from voipswitch to trunk ,without 
> passing by opensips .
>
> As I can see in the capture that when opensips send the 200 OK to Voipswitch 
> the 
>
> ;tag=9302846936345048301
>
> Contact:  :5060;user=phone;transport=udp;nat=yes>
>
> ** **
>
> We try to do some work around as but it didn’t succeed as  the call ends with 
> forbidden 
>
> ** **
>
> 
> if(!(is_method("INVITE|ACK|CANCEL|BYE|UPDATE|OPTIONS|INFO"))) {
>
>  xlog("L_INFO", ">>>>> Rejecting 
> method '$rm' from '$fu' to '$ru' - User-Agent: $ua <<<<<\n");
>
>  sl_send_reply("405","Method Not 
> Allowed");
>
>  exit;
>
> } else {
>
>  if (is_method("REGISTER")) {
>
>  
> fix_nated_register();
>
>  } else {
>
>  
> fix_nated_contact();
>
>  }
>
>  setflag(5);
>
>  setbflag(6);
>
> }
>
> ** **
>
> route[1] {
>
> ** **
>
> ** **
>
> if (subst_uri('/(sip:.*);nat=yes/\1/')) {
>
> setbflag(6);
>
> }
>
> ** **
>
> if (isflagset(5)) {
>
> search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
>
> }
>
> ** **
>
> ** **
>
> onreply_route[2] {
>
> search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
>
> ** **
>
> if (isbflagset(6)) {
>
> fix_nated_contact();
>
> };
>
> ** **
>
> ** **
>
> if (t_check_status("200") ) {
>
>  if (is_method("INVITE")) {
>
>  if 
> (subst('/^Contact: \r/ig')) {
>
>   
>xlog("L_INFO","-- Contact modified!");
>
>  };
>
>  }
>
> ** **
>
> Please advice
>
> ** **
>
> Regards
>
> 
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Save function with "s" flag

[Muhammad Shahzad]

Make sure you set sock_hdr_name,

http://www.opensips.org/html/docs/modules/devel/registrar.html#id250401

Thank you.


On Mon, Feb 18, 2013 at 1:34 PM, microx  wrote:

> Hi all,
>
> From registrar module, I find that the save function with "s" flag is able
> to store the socket specified in the REGISTER
> request. I use insert_hf("Sock: udp:127.0.0.1:5060") and
> insert_hf("udp:127.0.0.1:5060") in the header
> of REGISTER requests. When a SIP server receives such REGISTER requests, it
> invokes the save("location", "s")
> function. However, the SIP server still stores the received socket into the
> location table rather than "udp:127.0.0.1".
> How can I let the SIP server store "udp:127.0.0.1" when receiving a
> REGISTER
> request?
> Please help me resolve this issue. Thanks so much in advance.
>
> Best wishes,
> Chen-Che
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Save-function-with-s-flag-tp7584789.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
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] RTPProxy nortpproxy_str issue

[Muhammad Shahzad]

Yes, you can use this method,

http://www.opensips.org/html/docs/modules/devel/textops.html#id250333

e.g. something like,

if (has_body("application/sdp") && replace_body_atonce("a=schipmangle:yes",
""))
   xlog("Removed a=schipmangle:yes from carrier xxx");

Thank you.


On Fri, Feb 15, 2013 at 2:53 AM, Seth Schultz wrote:

>  Muhammad,
>
> I don't know what the remote carrier is using for their RTP.  I set a
> custom nortpproxy_str to try and avoid this (instead of leaving it as the
> default a=nortpproxy:yes).  Is it correct for them to leave our custom
> a=schipmangled:yes record in the SDP?  I have had problems with the "f"
> flag and failover routing (basically rewrites the IP in the SDP twice like
> this yyy.yyy.yyy.yy.yyy.yyy.yyy).  Is there an easy way for me to just
> remove the a=schipmangle:yes in my onreply_route?
>
> Thanks,
> Seth
>
>
> On 2/14/2013 8:28 PM, Muhammad Shahzad wrote:
>
> You mean both you and your carrier are using their own rtp-proxy? If so,
> then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will
> allow you can you carrier to create a chain of rtp-proxy together. See
> flags description here,
>
> http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744
>
> Thank you.
>
>
> On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz wrote:
>
>> Hello,
>>
>> I am having a problem with RTPProxy where in the reply, the remote
>> carrier is sending the "nortpproxy_str" in the reply SDP (example below).
>>  I would like to know what the best way is to detect this, and remove it
>> from the sip message before calling rtpproxy_answer function, because
>> rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP.
>>
>> Thanks in advance,
>> Seth
>>
>> U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
>> INVITE sip:1999...@xxx.xxx.xxx.xxx SIP/2.0
>> Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
>> Max-Forwards: 69
>> From: "Unknown" 
>> 
>> ;tag=33XjNy6SQZrQS
>> To:  
>> Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
>> CSeq: 40108106 INVITE
>> Contact: 
>> User-Agent: FS1
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 247
>> P-Call-Type: Notification
>> X-FS-Support: update_display,send_info
>> Remote-Party-ID: "Unknown" 
>> 
>> ;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
>> s=FreeSWITCH
>> c=IN IP4 yyy.yyy.yyy.yyy
>> t=0 0
>> m=audio 40562 RTP/AVP 0 8 3 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=schipmangled:yes  <--- rtpproxy added this on initial invite
>>
>> ...
>>
>> U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK2d9e.187ebf5.0
>> From: "Unknown" 
>> 
>> ;tag=33XjNy6SQZrQS
>> To:  
>> ;tag=SDs07f299-gK0e9f2e8d
>> Call-ID: 004c5840-f1aa-1230-9c93-6320dec8e883
>> CSeq: 40108106 INVITE
>> Accept: application/sdp, application/isup, application/dtmf,
>> application/dtmf-relay,  multipart/mixed
>> Contact: 
>> Allow:
>> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
>> Require: timer
>> Supported: timer
>> Session-Expires: 7200;refresher=uas
>> Content-Length: 259
>> Content-Disposition: session; handling=required
>> Content-Type: application/sdp
>>
>> v=0
>> o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
>> s=SIP Media Capabilities
>> c=IN IP4 xxx.xxx.xxx.xxx
>> t=0 0
>> m=audio 29772 RTP/AVP 0 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=schipmangled:yes  <--- they sent this back in the 200 OK reply
>> a=ptime:20
>> a=sendrecv
>>
>>
>> _______
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRM

Re: [OpenSIPS-Users] RTPProxy nortpproxy_str issue

[Muhammad Shahzad]

You mean both you and your carrier are using their own rtp-proxy? If so,
then simply add "f" flag to rtpproxy_offer | rtpproxy_answer. Which will
allow you can you carrier to create a chain of rtp-proxy together. See
flags description here,

http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id292744

Thank you.


On Fri, Feb 15, 2013 at 2:18 AM, Seth Schultz wrote:

> Hello,
>
> I am having a problem with RTPProxy where in the reply, the remote carrier
> is sending the "nortpproxy_str" in the reply SDP (example below).  I would
> like to know what the best way is to detect this, and remove it from the
> sip message before calling rtpproxy_answer function, because
> rtpproxy_answer will fail if the nortpproxy_str already exists in the SDP.
>
> Thanks in advance,
> Seth
>
> U 2013/02/14 19:32:21.142567 yyy.yyy.yyy.yyy:5060 -> xxx.xxx.xxx.xxx:5060
> INVITE sip:1999...@xxx.xxx.xxx.**xxx SIP/2.0
> Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=**z9hG4bK2d9e.187ebf5.0
> Max-Forwards: 69
> From: "Unknown" ;tag=33XjNy6SQZrQS
> To: 
> Call-ID: 004c5840-f1aa-1230-9c93-**6320dec8e883
> CSeq: 40108106 INVITE
> Contact: 
> User-Agent: FS1
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 247
> P-Call-Type: Notification
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "Unknown"  yyy>;party=calling;screen=yes;**privacy=off
>
> v=0
> o=FreeSWITCH 1360855702 1360855703 IN IP4 yyy.yyy.yyy.yyy
> s=FreeSWITCH
> c=IN IP4 yyy.yyy.yyy.yyy
> t=0 0
> m=audio 40562 RTP/AVP 0 8 3 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=schipmangled:yes  <--- rtpproxy added this on initial invite
>
> ...
>
> U 2013/02/14 19:32:37.425606 xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=**z9hG4bK2d9e.187ebf5.0
> From: "Unknown" ;tag=33XjNy6SQZrQS
> To: ;tag=SDs07f299-gK0e9f2e8d
> Call-ID: 004c5840-f1aa-1230-9c93-**6320dec8e883
> CSeq: 40108106 INVITE
> Accept: application/sdp, application/isup, application/dtmf,
> application/dtmf-relay,  multipart/mixed
> Contact: 
> Allow: INVITE,ACK,CANCEL,BYE,**REGISTER,REFER,INFO,SUBSCRIBE,**
> NOTIFY,PRACK,UPDATE,OPTIONS
> Require: timer
> Supported: timer
> Session-Expires: 7200;refresher=uas
> Content-Length: 259
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
>
> v=0
> o=Sonus_UAC 7607 20874 IN IP4 xxx.xxx.xxx.xxx
> s=SIP Media Capabilities
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 29772 RTP/AVP 0 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=schipmangled:yes  <--- they sent this back in the 200 OK reply
> a=ptime:20
> a=sendrecv
>
>
> __**_
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Textops Module Integration with presence. Changes not visible--> replace_body_all

[Muhammad Shahzad]

First your opensips configuration indicates you are using a very old
version of opensips (1.6.x perhaps). There are many major changes in
textops modules in newer version which fix tons of problems.

Secondly i doubt if you will be able to see changes through ngrep. This
tool will show you the original message received on ethernet interface,
before it is handed over to opensips for processing. And since your
opensips is the end-point from where the message is not routed any further
(to another server), so changes you do will be invisible to ngrep.

Thank you.


On Thu, Feb 14, 2013 at 12:25 PM, garima sharma
wrote:

> Hi
>
> I am trying to use textops modules with standalone presence server. The
> opensips server starts without error but the replace_body change done in
> is_method(PUBLISH) is not reflected during testing.
>
> The changes in PUBLISH messages seen in ngrep is not changed. kindly help
>
> #
> # $Id$
> #
> # simple quick-start config script - Stand-alone presence server
> #
>
> # --- global configuration parameters 
>
> debug=3 # debug level (cmd line: -dd)
> fork=yes
> log_stderror=no # (cmd line: -E)
> children=4
>
> listen=udp:192.168.1.100:5065
> port=5065
>
> #dns=no
> #rev_dns=no
>
> # -- module loading --
>
> #set module path
> mpath="/usr/lib/opensips/modules/"
>
> loadmodule "db_mysql.so"
> loadmodule "sl.so"
> loadmodule "signaling.so"
> loadmodule "maxfwd.so"
> loadmodule "textops.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "exec.so"
> loadmodule "presence.so"
> loadmodule "presence_xml.so"
> loadmodule "avpops.so"
> loadmodule "mi_fifo.so"
> loadmodule "usrloc.so"
> loadmodule "pua.so"
> loadmodule "pua_usrloc.so"
> loadmodule "registrar.so"
>
>
>
> # - setting module-specific parameters ---
>  modparam("usrloc", "db_mode", 2)
> modparam("usrloc", "db_url", "mysql://opensips:opensipsrw@localhost
> /opensips")
>
> # -- presence params --
> modparam("presence|presence_xml", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips")
>
> modparam("presence_xml", "force_active", 1)
>
> modparam("presence", "server_address", "sip:192.168.1.100:5065")
>
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>
>
> modparam("pua", "db_url", "mysql://opensips:opensipsrw@localhost
> /opensips")
> modparam("pua_usrloc", "default_domain", "192.168.1.100")
> modparam("pua_usrloc", "presence_server", "sip:192.168.1.100:5065")
>
> # - request routing logic ---
>
> # main routing logic
>
> route{
> # initial sanity checks -- messages with
> # max_forwards==0, or excessively long requests
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> };
>
>
>  if (!is_method("SUBSCRIBE|PUBLISH")) {
>     sl_send_reply("488", "Not Acceptable Here");
> exit;
> }
>
>
>
>* if ((is_method("PUBLISH")) && ($fU=="bob")) {
>
> replace_body_all("closed", "open");
> }*
>
>
>
> # presence handling
> if (! t_newtran())
> {
> sl_reply_error();
> exit;
>  };
>
> if(is_method("PUBLISH"))
> {
>
> handle_publish();
>
> }
>
> else
> if( is_method("SUBSCRIBE"))
> {
>
> handle_subscribe();
>
> };
>
> exit;
> }
>
>
> --
> Thanks and Regards
> Garima Sharma
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Crash in db_mysql

[Muhammad Shahzad]

I had this core file overwriting problem about a year ago and i fixed it by
setting linux kernel core pattern. It really helps me in identifying the
correct core file to debug. You can do it too by setting something like
below in sysctl.conf,

kernel.core_pattern=/tmp/core.%e.%p.%h.%t

OR run below command,

echo "/tmp/cores/core.%e.%p.%h.%t" > /proc/sys/kernel/core_pattern

Interesting switches that can be used are,

%p: pid
%u: uid
%g: gid
%s: signal number
%t: UNIX time of dump
%h: hostname
%e: executable filename

Thank you.


On Thu, Feb 14, 2013 at 12:18 PM, Brett Nemeroff  wrote:

> 1.8.2-notls (x86_64/linux)
> svnrevision: 2:9777M
>
>
> On Thu, Feb 14, 2013 at 2:01 AM, Muhammad Shahzad 
> wrote:
>
>> Please specify opensips version / revision.
>>
>> Thank you.
>>
>>
>> On Thu, Feb 14, 2013 at 8:13 AM, Brett Nemeroff wrote:
>>
>>> Hey all,
>>> I'm running into a crash. Same thing hits over and over. Not sure what's
>>> causing it.. Here's a bt:
>>>
>>> Program terminated with signal 11, Segmentation fault.
>>> #0  0x7f33ce3339b9 in db_mysql_val2bind (v=0x7f330fa18cc0,
>>> binds=0x7f33ce88cf00, i=33) at val.c:297
>>> 297 *(binds[i].is_null) = 0;
>>> Missing separate debuginfos, use: debuginfo-install
>>> glibc-2.12-1.80.el6_3.7.x86_64 hiredis-0.10.1-3.el6.x86_64
>>> json-c-0.10-2.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64
>>> krb5-libs-1.9-33.el6_3.3.x86_64 libcom_err-1.41.12-12.el6.x86_64
>>> libmemcached-0.31-1.1.el6.x86_64 libselinux-2.0.94-5.3.el6.x86_64
>>> mysql-libs-5.1.67-1.el6_3.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64
>>> openssl-1.0.0-25.el6_3.1.x86_64 zlib-1.2.3-27.el6.x86_64
>>> (gdb) bt full
>>> #0  0x7f33ce3339b9 in db_mysql_val2bind (v=0x7f330fa18cc0,
>>> binds=0x7f33ce88cf00, i=33) at val.c:297
>>> t = 
>>> mt = 
>>> __FUNCTION__ = "db_mysql_val2bind"
>>> #1  0x7f33ce32d34c in db_mysql_do_prepared_query
>>> (conn=0x7f33ce88cd68, v=, n=36, uv=>> out>, un=0, query=0x7f33ce5477b0) at dbase.c:584
>>> i = 33
>>> j = 
>>> code = 
>>> cols = 
>>> pq_ptr = 0x7f33ce88cec0
>>> ctx = 0x7f33ce88e310
>>> mysql_bind = 0x7f33ce88cf00
>>> start = {tv_sec = 139860479483003, tv_usec = 5334081}
>>> buffered_rows = 0x0
>>> #2  0x7f33ce32fcae in db_mysql_insert (_h=0x7f33ce88cd68,
>>> _k=0x7f330d73197b, _v=0x7f330fa188a0, _n=36) at dbase.c:1028
>>> ret = 
>>> #3  0x00517729 in flush_query_list () at db/db_insertq.c:133
>>> it = 0x7f330d7318a0
>>> my_ps = 0x7f33ce88cec0
>>> i = 
>>> __FUNCTION__ = "flush_query_list"
>>> #4  0x0051799c in handle_ql_shutdown () at db/db_insertq.c:170
>>> No locals.
>>> #5  0x0042c2c1 in cleanup (show_status=1) at main.c:347
>>> __FUNCTION__ = "cleanup"
>>>
>>>
>>> and in case you need it:
>>> (gdb) print *binds
>>> $1 = {length = 0x7f33ce88dd70, is_null = 0x7f33ce88dd78 "", buffer =
>>> 0x7f330fa18d20, error = 0x0, row_ptr = 0x0, store_param_func = 0,
>>> fetch_result = 0, skip_result = 0, buffer_length = 0, offset = 0,
>>>   length_value = 0, param_number = 0, pack_length = 0, buffer_type =
>>> MYSQL_TYPE_STRING, error_value = 0 '\000', is_unsigned = 0 '\000',
>>> long_data_used = 0 '\000', is_null_value = 0 '\000',
>>>   extension = 0x0}
>>>
>>> (gdb) print binds[33]
>>> $2 = {length = 0x6, is_null = 0x0, buffer = 0x18, error = 0x0, row_ptr =
>>> 0x16 , store_param_func = 0, fetch_result =
>>> 0x25, skip_result = 0, buffer_length = 3, offset = 0,
>>>   length_value = 2, param_number = 0, pack_length = 0, buffer_type = 40,
>>> error_value = 0 '\000', is_unsigned = 0 '\000', long_data_used = 0 '\000',
>>> is_null_value = 0 '\000', extension = 0x0}
>>> (gdb)
>>>
>>>
>>> Please let me know if you need any further testing. Thanks!!!
>>> -Brett
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> --
>> Muhammad Shahzad
>> ---
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.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
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Crash in db_mysql

[Muhammad Shahzad]

Please specify opensips version / revision.

Thank you.


On Thu, Feb 14, 2013 at 8:13 AM, Brett Nemeroff  wrote:

> Hey all,
> I'm running into a crash. Same thing hits over and over. Not sure what's
> causing it.. Here's a bt:
>
> Program terminated with signal 11, Segmentation fault.
> #0  0x7f33ce3339b9 in db_mysql_val2bind (v=0x7f330fa18cc0,
> binds=0x7f33ce88cf00, i=33) at val.c:297
> 297 *(binds[i].is_null) = 0;
> Missing separate debuginfos, use: debuginfo-install
> glibc-2.12-1.80.el6_3.7.x86_64 hiredis-0.10.1-3.el6.x86_64
> json-c-0.10-2.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64
> krb5-libs-1.9-33.el6_3.3.x86_64 libcom_err-1.41.12-12.el6.x86_64
> libmemcached-0.31-1.1.el6.x86_64 libselinux-2.0.94-5.3.el6.x86_64
> mysql-libs-5.1.67-1.el6_3.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64
> openssl-1.0.0-25.el6_3.1.x86_64 zlib-1.2.3-27.el6.x86_64
> (gdb) bt full
> #0  0x7f33ce3339b9 in db_mysql_val2bind (v=0x7f330fa18cc0,
> binds=0x7f33ce88cf00, i=33) at val.c:297
> t = 
> mt = 
> __FUNCTION__ = "db_mysql_val2bind"
> #1  0x7f33ce32d34c in db_mysql_do_prepared_query (conn=0x7f33ce88cd68,
> v=, n=36, uv=, un=0,
> query=0x7f33ce5477b0) at dbase.c:584
> i = 33
> j = 
> code = 
> cols = 
> pq_ptr = 0x7f33ce88cec0
> ctx = 0x7f33ce88e310
> mysql_bind = 0x7f33ce88cf00
> start = {tv_sec = 139860479483003, tv_usec = 5334081}
> buffered_rows = 0x0
> #2  0x7f33ce32fcae in db_mysql_insert (_h=0x7f33ce88cd68,
> _k=0x7f330d73197b, _v=0x7f330fa188a0, _n=36) at dbase.c:1028
> ret = 
> #3  0x00517729 in flush_query_list () at db/db_insertq.c:133
> it = 0x7f330d7318a0
> my_ps = 0x7f33ce88cec0
> i = 
> __FUNCTION__ = "flush_query_list"
> #4  0x0051799c in handle_ql_shutdown () at db/db_insertq.c:170
> No locals.
> #5  0x0042c2c1 in cleanup (show_status=1) at main.c:347
> __FUNCTION__ = "cleanup"
>
>
> and in case you need it:
> (gdb) print *binds
> $1 = {length = 0x7f33ce88dd70, is_null = 0x7f33ce88dd78 "", buffer =
> 0x7f330fa18d20, error = 0x0, row_ptr = 0x0, store_param_func = 0,
> fetch_result = 0, skip_result = 0, buffer_length = 0, offset = 0,
>   length_value = 0, param_number = 0, pack_length = 0, buffer_type =
> MYSQL_TYPE_STRING, error_value = 0 '\000', is_unsigned = 0 '\000',
> long_data_used = 0 '\000', is_null_value = 0 '\000',
>   extension = 0x0}
>
> (gdb) print binds[33]
> $2 = {length = 0x6, is_null = 0x0, buffer = 0x18, error = 0x0, row_ptr =
> 0x16 , store_param_func = 0, fetch_result =
> 0x25, skip_result = 0, buffer_length = 3, offset = 0,
>   length_value = 2, param_number = 0, pack_length = 0, buffer_type = 40,
> error_value = 0 '\000', is_unsigned = 0 '\000', long_data_used = 0 '\000',
> is_null_value = 0 '\000', extension = 0x0}
> (gdb)
>
>
> Please let me know if you need any further testing. Thanks!!!
> -Brett
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Help trace source of error: unknown URI param list excedeed

[Muhammad Shahzad]

When reporting issues please make sure to add opensips version you are
using including operating system details. This helps in diagnosing problem
better.

Anyways here are the answers for your question,

1. This error means an invalid SIP packet is received which opensips was
unable to parse.
2. Yes you can inspect the packet in detail and do appropriate actions,
have a look at below URL to learn more

http://www.opensips.org/Resources/DocsCoreRoutes#toc5

3. Yes, these could be symptom of possible DOS / DDOS attack OR may be some
valid user / trunk has really buggy device which creating bad SIP packets.
Take appropriate measures.

Thank you.


On Wed, Feb 13, 2013 at 11:05 PM, Adam Raszynski wrote:

> Hi
>
> Recently I have discovered increasing amount of the following errors in my
> logs:
>
> Feb 13 09:36:59 node1 /usr/sbin/opensips[10458]: ERROR:core:parse_uri:
> unknown URI param list exceeded
>
> I see this error is repeated many thousands of times in my log
>
> Questions:
> - What does this error mean?
> - How to add source IP address of SIP request that caused error to log? Or
> how to add full SIP request body causing that error to log?
> - Is it a sign of any type of DOS attack?
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Why is OpenSipS sending a 407

[Muhammad Shahzad]

; Content-Length: 379.
> .
> 
>  state="full" entity="sip:51810...@nyc-02.mydomain.net">
> 
> terminated
> 
> sip:51810...@nyc-02.mydomain.net
> 
> 
> 
> 
> 
> 
>
>
>
>
> U 2013/02/11 12:04:12.300172 203.144.218.9:5060 -> 67.198.80.143:22413
> NOTIFY sip:51810401@10.0.0.102 SIP/2.0.
> Max-Forwards: 10.
> Record-Route: .
> Via: SIP/2.0/UDP 203.144.218.9;branch=z9hG4bK9509.edab2244.0.
> Via: SIP/2.0/UDP 127.0.0.1;rport=35600;received=203.144.218.9.
> From: ;tag=13606022526514.
> To: ;tag=892B04F3-2D3963A0.
> Contact: .
> Call-ID: 5a00cffc-a02c6e6d-59dbeb42@10.0.0.102.
> CSeq: 480931448 NOTIFY.
> User-Agent: Enswitch presence server.
> Event: dialog.
> Subscription-State: active;expires=119.
> Content-Type: application/dialog-info+xml.
> Content-Length: 379.
> X-Enswitch-RURI: sip:51810401@203.144.218.9:5060.
> X-Enswitch-Source: 203.144.218.9:35600.
> .
> 
>  state="full" entity="sip:51810...@nyc-02.mydomain.net">
> 
> terminated
> 
> sip:51810...@nyc-02.mydomain.net
> 
> 
> 
> 
> 
> 
>
>
>
>
> U 2013/02/11 12:04:12.321247 67.198.80.143:22413 -> 203.144.218.9:5060
> SIP/2.0 481 Call Leg/Transaction Does Not Exist.
> Via: SIP/2.0/UDP 203.144.218.9;branch=z9hG4bK9509.edab2244.0.
> Via: SIP/2.0/UDP 127.0.0.1;rport=35600;received=203.144.218.9.
> From: ;tag=13606022526514.
> To: ;tag=892B04F3-2D3963A0.
> CSeq: 480931448 NOTIFY.
> Call-ID: 5a00cffc-a02c6e6d-59dbeb42@10.0.0.102.
> Record-Route: .
> Event: dialog.
> User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.3.0069.
> Accept-Language: en.
> Content-Length: 0.
> .
>
>
>
>
> U 2013/02/11 12:04:12.321354 203.144.218.9:5060 -> 203.144.218.9:35600
> SIP/2.0 481 Call Leg/Transaction Does Not Exist.
> Via: SIP/2.0/UDP 127.0.0.1;rport=35600;received=203.144.218.9.
> From: ;tag=13606022526514.
> To: ;tag=892B04F3-2D3963A0.
> CSeq: 480931448 NOTIFY.
> Call-ID: 5a00cffc-a02c6e6d-59dbeb42@10.0.0.102.
> Record-Route: .
> Event: dialog.
> User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.3.3.0069.
> Accept-Language: en.
> Content-Length: 0.
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK Timer

[Muhammad Shahzad]

Ah, again typo, in last line of previous email, i meant to recover loss of
few seconds between 200 OK and CANCEL from caller, (not ACK from caller,
since ACK establishes a billable call).

Thank you.


On Tue, Feb 12, 2013 at 2:54 PM, Muhammad Shahzad wrote:

> Sorry i didn't see you are setting dialog timeout for sequential INVITE,
> instead of initial INVITE, so this timeout would actually have no effect on
> new call, it will effect only established call (e.g. when caller or callee
> sets call on hold etc.) and when that sequential INVITE comes in the
> destination has to accept it within 3 seconds, which is OK, since call is
> already established, only its state being changed.
>
> Secondly, a call is not considered established till ACK arrives from
> caller party. Since caller never sends ACK, so destination should end call
> (after 32 seconds per RFC 3261) even if it does not receives CANCEL from
> caller. And if destination receives CANCEL, then call should end anyway.
> Such call can not be billed, since it was never established.
>
> From billing the caller prospective, you should start billing upon
> receiving 200 OK from destination but you must discard it if CANCEL comes
> from caller instead of ACK. In fact i have seen some billing systems that
> actually start billing upon receiving ACK from caller, rather 200 OK from
> destination. To overcome the loss of few seconds (between 200 OK from
> destination and ACK from caller), they use a different billing head called
> "connection charges".
>
> Thank you.
>
>
> On Tue, Feb 12, 2013 at 2:24 PM, M.Khaled W Chehab wrote:
>
>> Please can you show me by code  the better way(ensure Cancel)  since this
>> is a critical issue and I am relaying the cancel and after that trunk  send
>> me more than 8 time 200 Ok  ,as is there a way to stop/hangup the call
>> since I receive the cancel from the client 
>>
>>  
>>
>> 2-what do you mean destination must be very quick ( since 200 oK is
>> received  the normal reply (ACK) takes  millsec or I am wrong ?
>>
>> ** **
>>
>> ** **
>>
>> Regards
>>
>> 
>>
>> ** **
>>
>> *From:* Muhammad Shahzad [mailto:shaherya...@gmail.com]
>> *Sent:* Tuesday, February 12, 2013 3:05 PM
>> *To:* M.Khaled W Chehab
>> *Cc:* users@lists.opensips.org; bog...@opensips.org;
>> users-boun...@lists.opensips.org; Muhammad Shahzad
>> *Subject:* Re: ACK Timer
>>
>> ** **
>>
>> This may work, only if you create dialog with 'B' flag, also 3 seconds
>> look very short, destination must be very quick to ACK the call.
>>
>> ** **
>>
>> I think there is a better way to achieve this, you only need to ensure
>> CANCEL is received at destination.
>>
>>
>> Thank you.
>>
>> ** **
>>
>> On Tue, Feb 12, 2013 at 1:45 PM, M.Khaled W Chehab 
>> wrote:
>>
>> Hi ,
>>
>>  
>>
>> I face a lot of scenarios where the customer send a cancel while the
>> trunk send 200 OK and start the billing from its side ,so the client call
>> will be canceled from his side and goes to the max call duration from my
>> side and trunk side .
>>
>> Si I draw this function below 
>>
>> loadmodule "dialog.so"
>>
>> modparam("dialog", "timeout_avp", "$avp(timeout2)")
>>
>>  
>>
>> if (has_totag()) {
>>
>>   if ( is_method("INVITE")) {
>>
>>  $avp(timeout2) = 3; 
>>
>>   } else if (is_method("ACK")) {
>>
>>  $avp(timeout2) = 3540; 
>>
>>   }
>>
>>  ****
>>
>> Do this function effect on my calls or cause  any problem
>>
>>  
>>
>> Regards
>>
>>  
>>
>>  
>>
>>  
>>
>> Khaled Chehab
>>
>> Senior NGN Engineer
>>
>> [image: Description: icucall]
>>
>> Operations Office - Lebanon
>>
>> Office: +961 1 515155 ext 300
>>
>> Mobile  : +961 3 045212
>>
>> E-mail: kche...@icucall.com****
>>
>> MSN ID :khalidche...@hotmail.com 
>>
>> Skype: k_chehab 
>>
>> Web Site: http://www.icucall.com
>>
>>  http://www.allohi.com
>>
>>  
>>
>>
>>
>> 
>>
>> ** **
>>
>> --
>> Muh

Re: [OpenSIPS-Users] ACK Timer

[Muhammad Shahzad]

Sorry i didn't see you are setting dialog timeout for sequential INVITE,
instead of initial INVITE, so this timeout would actually have no effect on
new call, it will effect only established call (e.g. when caller or callee
sets call on hold etc.) and when that sequential INVITE comes in the
destination has to accept it within 3 seconds, which is OK, since call is
already established, only its state being changed.

Secondly, a call is not considered established till ACK arrives from caller
party. Since caller never sends ACK, so destination should end call (after
32 seconds per RFC 3261) even if it does not receives CANCEL from caller.
And if destination receives CANCEL, then call should end anyway. Such call
can not be billed, since it was never established.

>From billing the caller prospective, you should start billing upon
receiving 200 OK from destination but you must discard it if CANCEL comes
from caller instead of ACK. In fact i have seen some billing systems that
actually start billing upon receiving ACK from caller, rather 200 OK from
destination. To overcome the loss of few seconds (between 200 OK from
destination and ACK from caller), they use a different billing head called
"connection charges".

Thank you.


On Tue, Feb 12, 2013 at 2:24 PM, M.Khaled W Chehab wrote:

> Please can you show me by code  the better way(ensure Cancel)  since this
> is a critical issue and I am relaying the cancel and after that trunk  send
> me more than 8 time 200 Ok  ,as is there a way to stop/hangup the call
> since I receive the cancel from the client 
>
>  
>
> 2-what do you mean destination must be very quick ( since 200 oK is
> received  the normal reply (ACK) takes  millsec or I am wrong ?
>
> ** **
>
> ** **
>
> Regards
>
> 
>
> ** **
>
> *From:* Muhammad Shahzad [mailto:shaherya...@gmail.com]
> *Sent:* Tuesday, February 12, 2013 3:05 PM
> *To:* M.Khaled W Chehab
> *Cc:* users@lists.opensips.org; bog...@opensips.org;
> users-boun...@lists.opensips.org; Muhammad Shahzad
> *Subject:* Re: ACK Timer
>
> ** **
>
> This may work, only if you create dialog with 'B' flag, also 3 seconds
> look very short, destination must be very quick to ACK the call.
>
> ** **
>
> I think there is a better way to achieve this, you only need to ensure
> CANCEL is received at destination.
>
>
> Thank you.
>
> ** **
>
> On Tue, Feb 12, 2013 at 1:45 PM, M.Khaled W Chehab 
> wrote:
>
> Hi ,
>
>  
>
> I face a lot of scenarios where the customer send a cancel while the
> trunk send 200 OK and start the billing from its side ,so the client call
> will be canceled from his side and goes to the max call duration from my
> side and trunk side .
>
> Si I draw this function below 
>
> loadmodule "dialog.so"
>
> modparam("dialog", "timeout_avp", "$avp(timeout2)")
>
>  
>
> if (has_totag()) {
>
>   if ( is_method("INVITE")) {
>
>  $avp(timeout2) = 3; 
>
>   } else if (is_method("ACK")) {
>
>  $avp(timeout2) = 3540; 
>
>   }
>
>  
>
> Do this function effect on my calls or cause  any problem
>
>  
>
> Regards
>
>  
>
>  
>
>  
>
> Khaled Chehab
>
> Senior NGN Engineer
>
> [image: Description: icucall]****
>
> Operations Office - Lebanon
>
> Office: +961 1 515155 ext 300
>
> Mobile  : +961 3 045212
>
> E-mail: kche...@icucall.com
>
> MSN ID :khalidche...@hotmail.com 
>
> Skype: k_chehab ****
>
> Web Site: http://www.icucall.com
>
>  http://www.allohi.com
>
>  
>
>
>
> 
>
> ** **
>
> --
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com 
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
<>___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] ACK Timer

[Muhammad Shahzad]

This may work, only if you create dialog with 'B' flag, also 3 seconds look
very short, destination must be very quick to ACK the call.

I think there is a better way to achieve this, you only need to ensure
CANCEL is received at destination.

Thank you.


On Tue, Feb 12, 2013 at 1:45 PM, M.Khaled W Chehab wrote:

> Hi ,
>
> ** **
>
> I face a lot of scenarios where the customer send a cancel while the
> trunk send 200 OK and start the billing from its side ,so the client call
> will be canceled from his side and goes to the max call duration from my
> side and trunk side .
>
> Si I draw this function below 
>
> loadmodule "dialog.so"
>
> modparam("dialog", "timeout_avp", "$avp(timeout2)")
>
> ** **
>
> if (has_totag()) {
>
>   if ( is_method("INVITE")) {
>
>  $avp(timeout2) = 3; 
>
>   } else if (is_method("ACK")) {
>
>  $avp(timeout2) = 3540; 
>
>   }
>
> ** **
>
> Do this function effect on my calls or cause  any problem
>
> ** **
>
> Regards
>
> ** **
>
> ** **
>
> ** **
>
> Khaled Chehab
>
> Senior NGN Engineer
>
> [image: Description: icucall]
>
> Operations Office - Lebanon
>
> Office: +961 1 515155 ext 300
>
> Mobile  : +961 3 045212****
>
> E-mail: kche...@icucall.com
>
> MSN ID :khalidche...@hotmail.com 
>
> Skype: k_chehab 
>
> Web Site: http://www.icucall.com
>
>  http://www.allohi.com
>
> ** **
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
<>___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSips in other directory

[Muhammad Shahzad]

Just export PREFIX variable to your desired location, e.g.

export PREFIX='/tmp/opensips'; make all -j4

Thank you.


On Mon, Feb 11, 2013 at 3:04 PM, Dragomir Haralambiev wrote:

> Hello,
>
> By default OpenSips is installed in /usr/lib/opensips.
> How to install in other directory?
>
> Best regards,
> PlayMen
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] JSON Bug, empty strings

[Muhammad Shahzad]

Nope, i was wrong. It works fine.

$json(object) := "{'id':'1234567890','name':NULL}";
$json(object/element) = '';

xlog("L_WARN", "id: $json(object/id), name: $json(object/name),
element: $json(object/element)");

Results as,

Feb 10 19:39:47 ubuntu-1204 osip-service[13705]: id: 1234567890, name:
, element:

Also, here is null value test,

https://github.com/json-c/json-c/blob/master/tests/test_null.c


Thank you.


On Sun, Feb 10, 2013 at 8:05 PM, Muhammad Shahzad wrote:

> I think libjson_c treats empty string as null. Let me run some tests to
> confirm it.
>
> Thank you.
>
>
> On Sun, Feb 10, 2013 at 6:39 PM, Brett Nemeroff wrote:
>
>> Hey all,
>> I think I see a bug in the JSON module; but it might just be a "feature"
>> of JSON.
>>
>> Seems that when I assign an empty string to a JSON element, it's storing
>> it as NULL. So for example:
>>
>> $json(object/element) = '';
>>
>> If I print that, I get  in an xlog.
>>
>> Is this expected? I'd really like to be able to store empty strings.
>>
>> Thanks!
>> -Brett
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] JSON Bug, empty strings

[Muhammad Shahzad]

I think libjson_c treats empty string as null. Let me run some tests to
confirm it.

Thank you.


On Sun, Feb 10, 2013 at 6:39 PM, Brett Nemeroff  wrote:

> Hey all,
> I think I see a bug in the JSON module; but it might just be a "feature"
> of JSON.
>
> Seems that when I assign an empty string to a JSON element, it's storing
> it as NULL. So for example:
>
> $json(object/element) = '';
>
> If I print that, I get  in an xlog.
>
> Is this expected? I'd really like to be able to store empty strings.
>
> Thanks!
> -Brett
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] local_route and relayed messages

[Muhammad Shahzad]

$mb should you whole message which $rb will give only the body of the
message (if one exists).

Thank you.


On Sun, Feb 10, 2013 at 6:21 PM, Brett Nemeroff  wrote:

> Hello All,
> If I use $mb in local_route it's capturing the received message. I'd like
> in local route to be able to capture the full message that is about to be
> relayed. Is this possible?
>
> Thanks,
> Brett
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 1.9.0 lb_count_call issue

[Muhammad Shahzad]

humm, then its most likely a bug. Please report it in tracker. :-(

Thank you.


On Sun, Feb 10, 2013 at 1:04 AM, Seth Schultz wrote:

>  Muhammad,
>
> Thank you for the reply.  I also tried a hard coded value with the same
> result.
>
> Thanks,
> Seth
>
>  On 2/9/2013 6:50 PM, Muhammad Shahzad wrote:
>
> I think both group and resource parameters can not be variables. Can you
> check with hard coded values?
>
> Thank you.
>
>
> On Sat, Feb 9, 2013 at 9:58 PM, Seth Schultz wrote:
>
>>  Hello,
>>
>> I am having trouble with the new function in 1.9.0 lb_count_call.  Please
>> look at the snippet below.
>>
>> failure_route[1]
>> {
>> ...
>>
>> xlog("L_WARN", "trying to count call ip=$dlg_val(route) port=$dp
>> group=$dlg_val(balance_pool) resource=$dlg_val(balance_resource)\n");
>>
>> if ($dlg_val(balance_pool) = "1")
>> {
>> if (lb_is_destination("$dlg_val(route)", "$dp", "1", "1"))
>> {
>> xlog("L_WARN", "counting call!\n");
>> if (lb_count_call("$dlg_val(route)", "$dp", "1",
>> "$dlg_val(balance_resource)"))
>> {
>> xlog("L_WARN", "call counted on sip:$du:$dp\n");
>> }
>> }
>> }
>>
>>  ...
>> }
>>
>> Here is the output I see in my log file:
>>
>> Feb  9 15:53:07 opensips-vm /sbin/opensips[16562]: trying to count call
>> ip=xxx.xxx.xxx.xxx port=5060 group=1 resource=pstn
>> Feb  9 15:53:07 opensips-vm /sbin/opensips[16562]: counting call!
>> Feb  9 15:53:07 opensips-vm /sbin/opensips[16562]:
>> ERROR:load_balancer:lb_count_call: no destination to match the given IP and
>> port (xxx.xxx.xxx.xxx:5060)
>>
>> Notice that it gets a positive result when calling lb_is_destination,
>> however, lb_count_call is throwing an error message indicating it can't
>> match a destination.
>>
>> On a side note, I am having trouble getting the "group" parameter to
>> accept variable input (i.e. $dlg_val(group)).
>>
>> Thanks,
>> Seth
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Muhammad Shahzad
> -------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] 1.9.0 lb_count_call issue

[Muhammad Shahzad]

I think both group and resource parameters can not be variables. Can you
check with hard coded values?

Thank you.


On Sat, Feb 9, 2013 at 9:58 PM, Seth Schultz wrote:

>  Hello,
>
> I am having trouble with the new function in 1.9.0 lb_count_call.  Please
> look at the snippet below.
>
> failure_route[1]
> {
> ...
>
> xlog("L_WARN", "trying to count call ip=$dlg_val(route) port=$dp
> group=$dlg_val(balance_pool) resource=$dlg_val(balance_resource)\n");
>
> if ($dlg_val(balance_pool) = "1")
> {
> if (lb_is_destination("$dlg_val(route)", "$dp", "1", "1"))
> {
> xlog("L_WARN", "counting call!\n");
> if (lb_count_call("$dlg_val(route)", "$dp", "1",
> "$dlg_val(balance_resource)"))
> {
> xlog("L_WARN", "call counted on sip:$du:$dp\n");
> }
> }
> }
>
>  ...
> }
>
> Here is the output I see in my log file:
>
> Feb  9 15:53:07 opensips-vm /sbin/opensips[16562]: trying to count call
> ip=xxx.xxx.xxx.xxx port=5060 group=1 resource=pstn
> Feb  9 15:53:07 opensips-vm /sbin/opensips[16562]: counting call!
> Feb  9 15:53:07 opensips-vm /sbin/opensips[16562]:
> ERROR:load_balancer:lb_count_call: no destination to match the given IP and
> port (xxx.xxx.xxx.xxx:5060)
>
> Notice that it gets a positive result when calling lb_is_destination,
> however, lb_count_call is throwing an error message indicating it can't
> match a destination.
>
> On a side note, I am having trouble getting the "group" parameter to
> accept variable input (i.e. $dlg_val(group)).
>
> Thanks,
> Seth
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [RELEASES] Drafting OpenSIPS 1.10.0 TODO list

[Muhammad Shahzad]

I would second Daniel Geopp for db_http improvements, its much needed. Many
people ask me from time to time to have an out of band call manager to
control call processing in opensips and i think db_http or some new module
with suggested functionality would really be a huge feature.

Thank you.


On Sat, Feb 9, 2013 at 10:01 PM, Dragomir Haralambiev wrote:

> Hello ,
>
> I'm reading every message from this mailling list.
> I see that there are many questions about rtpproxy and mediaproxy.
> I would like to recommend more detailed documentation with more
> examples about NAT support  in the new version.
>
> Best regards,
> PlayMen
>
> 2013/2/9 Bogdan-Andrei Iancu :
> > Hi all,
> >
> > According the the release policy
> > (http://www.opensips.org/Development/Development), I would like to call
> for
> > a brainstorming, ideas, discussion, etc regarding what should be the
> roadmap
> > for OpenSIPS 1.10 - more or less, what new goodies should be in 1.10
> release
> > (next major release).
> >
> > The page is already ready (http://www.opensips.org/Main/Ver1100) and
> > pre-populated with the pending items from 1.9 release plus some items
> from
> > my side .
> >
> >
> > I would like to stat the discussion here, on the mailing list first, to
> get
> > from all community ideas on what should be done in 1.10 - things to
> improve,
> > supporting new RFC/drafts, new functionalitites, etc.
> >
> > So, please do not be shy and make your points here ;).
> >
> > Also we plan a second round of discussion / selection via an IRC meeting
> > (probably in 2 weeks or so).
> >
> > Best regards,
> >
> > --
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.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
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] siptrace problem

[Muhammad Shahzad]

Use only one method, not both at the same time in opensips.cfg. For
sip_trace method, use it immediately after sanity checks in main route, e.g.

route {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};

if (msg:len > max_len) {
sl_send_reply("513","Message Too Big");
exit;
};

   sip_trace();
...

OK make sure you call it for each single SIP method you are interested in,
whenever you are processing it opensips.cfg, e.g.

   ...

if (is_method("INVITE")) {
 sip_trace();
 ...

} else if (is_method("REGISTER")) {
 sip_trace();
 ...
}

   ...


This should work, if it does not then its most likely a bug. I remember
somebody reported such bug in older opensips version (perhaps 1.7.x) and it
was fixed by the dev team. So it should work..

Thank you.


On Sat, Feb 9, 2013 at 7:34 AM, Dragomir Haralambiev wrote:

> I try with both , results the same.
>
> 2013/2/9 Muhammad Shahzad :
> > How you are doing sip trace? Are you calling sip_trace method or setting
> sip
> > trace dialog flag in opensips.cfg?
> >
> > http://www.opensips.org/html/docs/modules/1.8.x/siptrace.html#id248315
> >
> > Thank you.
> >
> >
> > On Fri, Feb 8, 2013 at 11:31 PM, Dragomir Haralambiev <
> goup2...@gmail.com>
> > wrote:
> >>
> >> Hello,
> >>
> >> I have problem with siptrace in OpenSips 1.8.
> >>
> >> OpenSips dumps only CANCEL which sends. It doesn't dump CANCEL which
> >> receive.
> >>
> >> Here is an example for dump:
> >>
> >> http://87.121.151.141/voipdump.htm
> >>
> >> Best regards,:
> >> PlayMen
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> >
> > --
> > Muhammad Shahzad
> > ---
> > CISCO Rich Media Communication Specialist (CRMCS)
> > CISCO Certified Network Associate (CCNA)
> > Cell: +49 176 99 83 10 85
> > MSN: shari_78...@hotmail.com
> > Email: shaherya...@googlemail.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
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Media Proxy /Opensips

[Muhammad Shahzad]

I am not sure why you are getting trouble in having more then 100 calls on
opensips service with media proxy. I need opensips and media proxy logs to
check whats happening. Perhaps the error is somewhere else.

Anyways, here are the steps to setup second media proxy,

1. Install media proxy on second server.
2. In /etc/mediaproxy/config.ini set your dispatcher ip address (where
opensips + first media proxy is running) under Relay section, e.g.

[Relay]
dispatchers = 

3. Shutdown media-dispatcher on second machine (its useless anyway) and
restart media-relay (on second machine).

4. Check syslogs on second machine, if everything is ok then you will see
message like,

debug: Connected to dispatcher at x.x.x.x:p

Where x.x.x.x and p are ip and port of your opensips server.

5. You only need to secure dispatcher (running on opensips). The relay uses
its on encrypted / secure communication protocol to connect to dispatcher,
so no need to worry about its security. You only make sure open RTP port
range you specify in config.ini for media relay.

Thank you.


On Sat, Feb 9, 2013 at 12:56 AM, M. KHaled W. Chehab wrote:

>
> Sorry i I have more than 600 to 1000 call What about limit.conf should i
> edit it.
> 2-please can you show me a sample on how to add the second mediaproxy to
> opensips.conf  as i will have the same media proxy config with differnt ip
> address on the second server
> 3-how to protect the  mediaproxy relay to donot be used from others,is
> that can be done by  config  or just by  firewall
>
> Thanks in advance
>
> Sent from my android device.
>
>
>
> -----Original Message-
> From: Muhammad Shahzad 
> To: "M.Khaled W Chehab" 
> Cc: Muhammad Shahzad ,
> users@lists.opensips.org, users-boun...@lists.opensips.org
> Sent: Sat, 09 Feb 2013 12:47 AM
> Subject: Re: Media Proxy /Opensips
>
> Are you sure, 500 Internal Server Error is caused by media proxy? Can you
> show some media proxy logs indicating this problem?
>
> Secondly load balancing media proxy is simple, just install media relay on
> another machine and point it media dispatcher running on opensips machine.
>
> Thank you.
>
>
> On Fri, Feb 8, 2013 at 11:27 PM, M.Khaled W Chehab wrote:
>
>> Hi,
>>
>> ** **
>>
>> I cant  find  more than 100 connected call on my server even I have aleot
>> of attempts most of calls ends with 500 internal error 
>>
>> My server is 16 GB with 24 core processor ,
>>
>> I didn’t edit /etc/security/limits.conf 
>>
>> ** **
>>
>> **1-  **What config I can do to have more than 600 connected calls
>> on the system 
>>
>> **2-  **If I install a mediaproxy in another server ,how can I
>> configure opensips  to make a loadbalance between the  mediaproxy that
>> installed on the same server and the new one ?
>>
>> ** **
>>
>> As I understand from the config that max_connections = 10 is not related
>> to the max number of calls unless  you can control it by limiting the port
>> range and can how to let it  print the error only in a file 
>>
>> ** **
>>
>> Sorry for my nested question but I want to be sure that everything is
>> going fine  
>>
>>  # 
>>
>> #
>>
>> ** **
>>
>> #*   softcore0
>>
>> #roothardcore    10****
>>
>> #*   hardrss 1
>>
>> #@studenthardnproc   20
>>
>> #@facultysoftnproc   20
>>
>> #@facultyhardnproc   50
>>
>> #ftp hardnproc   0
>>
>> #ftp -   chroot  /ftp
>>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] siptrace problem

[Muhammad Shahzad]

How you are doing sip trace? Are you calling sip_trace method or setting
sip trace dialog flag in opensips.cfg?

http://www.opensips.org/html/docs/modules/1.8.x/siptrace.html#id248315

Thank you.


On Fri, Feb 8, 2013 at 11:31 PM, Dragomir Haralambiev wrote:

> Hello,
>
> I have problem with siptrace in OpenSips 1.8.
>
> OpenSips dumps only CANCEL which sends. It doesn't dump CANCEL which
> receive.
>
> Here is an example for dump:
>
> http://87.121.151.141/voipdump.htm
>
> Best regards,:
> PlayMen
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Media Proxy /Opensips

[Muhammad Shahzad]

Are you sure, 500 Internal Server Error is caused by media proxy? Can you
show some media proxy logs indicating this problem?

Secondly load balancing media proxy is simple, just install media relay on
another machine and point it media dispatcher running on opensips machine.

Thank you.


On Fri, Feb 8, 2013 at 11:27 PM, M.Khaled W Chehab wrote:

> Hi,
>
> ** **
>
> I cant  find  more than 100 connected call on my server even I have aleot
> of attempts most of calls ends with 500 internal error 
>
> My server is 16 GB with 24 core processor ,
>
> I didn’t edit /etc/security/limits.conf 
>
> ** **
>
> **1-  **What config I can do to have more than 600 connected calls on
> the system 
>
> **2-  **If I install a mediaproxy in another server ,how can I
> configure opensips  to make a loadbalance between the  mediaproxy that
> installed on the same server and the new one ?
>
> ** **
>
> As I understand from the config that max_connections = 10 is not related
> to the max number of calls unless  you can control it by limiting the port
> range and can how to let it  print the error only in a file 
>
> ** **
>
> Sorry for my nested question but I want to be sure that everything is
> going fine  
>
>  # 
>
> #
>
> ** **
>
> #*   softcore0
>
> #roothardcore10
>
> #*   hardrss 1
>
> #@studenthardnproc   20
>
> #@facultysoftnproc   20
>
> #@facultyhardnproc   50
>
> #ftp hardnproc   0
>
> #ftp -   chroot  /ftp
>
> #@student-   maxlogins   4
>
> ** **
>
> # End of file
>
> Best regards
>
> ** **
>
> Khaled Chehab
>
> Senior NGN Engineer
>
> [image: Description: icucall]
>
> Operations Office - Lebanon
>
> Office: +961 1 515155 ext 300
>
> Mobile  : +961 3 045212
>
> E-mail: kche...@icucall.com
>
> MSN ID :khalidche...@hotmail.com 
>
> Skype: k_chehab 
>
> Web Site: http://www.icucall.com
>
>  http://www.allohi.com
>
> ** **
>



-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
<>___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users