date:20110111

Re: [OpenSIPS-Users] How to implement a SIP Trunk in between two SIP servers.

2011-01-11 Thread Bogdan-Andrei Iancu

Hi Steven,

sorry, but I know nothing on CUCM :(

Regards,
Bogdan

steven chew wrote:

Hi Bogdan,

Thanks for your reply.

Your script is very useful for calling between two opensips servers 
which I have tested.

However, I don't know how to configure on CUCM 7.0 which I am using.

At the moment, CUCM 7.0 is using Web Config via the Web Browser. 

Can you let me know how to configure on CUCM 7.0?

I will appreciate very much if you give some instructions 
for  configuring SIP Trunk on CUCM7.0

Thanks
Kind regards,
Steven,

On 10 January 2011 19:33, Bogdan-Andrei Iancu > wrote:

Hi Steven,

To do that, you need to add in opensips some routing to 1)
recognize the numbers that needs to be sent to CUCM and 2)route
that calls to CUCM.

For script logic it sounds like : if you receive a new call
(initial INVITE) for your local domain, check the URI and divert.
If you look at the default config file, there is comment "#
requests for my domain" -> from that point further you have only
initial INVITEs for your local domain, so you can add after:

  # all numbers starting with 55 are to be sent to CUCM
  if ($rU =~ "^55[0-9]+$") {
# replace the domain part of RURI to point to CUCM
rewritehostport("CUCM_IP:CUCM_PORT");
# route the call out based on RURI
route(1);
  }

For the other way around, you have to put a similar logic in CUCM,
like to divert all calls starting with "12" to opensips - and
replace the domain on RURI with the IP/domain of opensips.

Regards,
Bogdan

steven chew wrote:

Hi Bogdan,

Thank you very much for your reply.

I have an Opensips Server and a Cisco Unified Communication
Manager (CUCM).

If I want to make calls from Opensips Server to CUCM and CUCM
to Opensips Server.

For example:
1) If I dial an extension number "5566" from a SIP Phone
"12345" under Opensips Server, it will try to call to a Cisco
IP Phone "5566" from CUCM through a SIP Trunk.
2) If I dial an extension number "12345" from a Cisco IP Phone
"5566" under CUCM, it will try to call to a SIP Phone "12345"
under Opensips Server through a SIP Trunk.

Can you give some instructions how to configure the above
scenario for dialing extension numbers?

Thanks
Steven,
On 6 January 2011 21:31, Bogdan-Andrei Iancu
mailto:bog...@voice-system.ro>
>> wrote:

   Hi Steven,

   If you use the opensips default script, your opensips will
accept
   calls from any other external SIP entities (call targeting
a local
   opensips subscriber).

   If you want to configure your opensips to accept foreign calls
   only form a specific IP address, you can use the permission
   module, with address table to implement IP-based
authentication.

   Best regards,
   Bogdan

   steven chew wrote:

   Hi everyone,

   I am a newbie with SIP-Trunk in OpenSips.
   I have a Cisco Communication Unified Manager and a OpenSips
   Server running in two different Virtual Machines.

   I would like to have a SIP trunk in between them "Cisco
   Communication Unified Manager and OpenSips Server".
   Therefore, I can make a call from OpenSips Server's SIP
   Clients to Cisco IP Phone.
   What should I need to add into opensips.cfg
configuration file?

   I hope you can give some simple examples how to do it.
   I look forward to hearing from your advise asap.

   Thanks
   Regards,
   -Steven.

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   -- Bogdan-Andrei Iancu
   OpenSIPS Event - expo, conf, social, bootcamp
   2 - 4 February 2011, ITExpo, Miami,  USA
   www.voice-system.ro 

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Re: [OpenSIPS-Users] insert customized SIP-Reason column in ACC

2011-01-11 Thread Jayesh Nambiar

Ho Bogdan,
Will the acc_db_request function accept PV in it. If at all I'd like to
fetch the reason phrases from the DB and send it back to the caller?

--- Jayesh


> Hi Jayesh,
>
> one way to go is to use manual missed call accounting, instead of
> automatic one (with flags).  Use acc_db_request("403
> Account-Inactive","missed_calls") in your failure route, when you
> overwrite the reply.
>
> Regards,
> Bogdan
>
> Jayesh Nambiar wrote:
> > Hi All,
> > For all the failed calls, I respond with customized SIP-Codes and
> > SIP-Reasons to my callers using t_reply function in my script. The
> > SIP_code gets inserted correctly in the acc table but the SIP reasons
> > are not inserted as what I set. For example, if I send a
> > t_reply("403", "Account-Inactive"), the entry in sip_code column is
> > "403" but the entry in sip_reason column is still "Forbidden".
> > I know I can add an extra column and insert those values in a separate
> > column using AVPs, but I intend to use the default column in acc table.
> > Is it possible to insert the customized reason code generated from
> > script to be entered into the sip_reason column of the acc table?
> >
> > Thanks for any pointers or help in advance.
> >
> > --- Jayesh
> > 
> >
> > ___
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
>
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Re: [OpenSIPS-Users] RTPProxy timeout notifications

2011-01-11 Thread Denis Putyato

Hello Razvan!

Now it's working, thank you.

But I want to tell you that -W key in rtpproxy and Opensips doesn't work. When 
-W timer expires rtpproxy notifies about it Opensips but last one cannot drop 
call because there is no information about callee and caller contacts in 
dialog. 

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, January 03, 2011 7:25 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications

Hello Denis,

The problem you are dealing with is that you are using a TCP socket to 
receive timeout notifications from RTPProxy.
When a timeout notification is received through TCP, the nathelper 
module searches for the sender in the rtpproxies specified in 
"rtpproxy_sock" module parameter. It cannot find it in that list, so it 
ignores the notification.
Because you are using a UNIX socket to communicate with RTPProxy, then 
you should also use a UNIX socket to receive timeout notifications from it.
Note that I posted today a bug fix for this behavior. You can watch the 
thread at:
http://lists.rtpproxy.org/pipermail/devel/2011-January/thread.html

Regards,
Razvan

On 12/28/2010 04:52 PM, Denis Putyato wrote:
> Hello Bogdan
>
> RTP Proxy is working but timeout notification does not.
> There is error "/usr/local/opensips1.6.4/sbin/opensips[26496]:
>>> DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring "
> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> Sent: Tuesday, December 28, 2010 5:49 PM
> To: OpenSIPS users mailling list; Razvan Crainea
> Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications
>
> Hi Denis,
>
>
> Denis Putyato wrote:
>> Hello Bogdan!
>>
>> 1) There is no patch in official release of Opensips 1.6.4 which I can 
>> download from web site (source tar). There is a patch only in SVN version of 
>> Opensips 1.6.4
>>
> Hmm..that's a packaging bug :(I will take care of this.
>> 2) The patch which I can use from SVN version I can apply only to rtpproxy 
>> from git. If I use rtpproxy from web site I cannot apply patch to it (there 
>> are some errors during process of patch).
>>
> I will ask Razvan (the author of this work) to see if the patch can be
> ported to official rtpproxy release too (not as coding, but as
> functionality).
>> In my case I use rtpproxy from git with applied patch from SVN version of 
>> Opensips (when I start rtpproxy I use such command 
>> "/usr/local/rtpproxy/bin/rtpproxy -l 1.1.1.1 -s unix:/var/run/rtpproxy.sock 
>> -F -i -n tcp:1.1.1.1:2 -T 20 -W 60". As I understand without patch -W 
>> doesn`t work) and official release of Openspis 1.6.4 which I downloaded from 
>> web site (not from SVN)
>>
> And this works ?
>
> Regards,
> Bogdan
>
>> -Original Message-
>> From: users-boun...@lists.opensips.org 
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
>> Sent: Tuesday, December 28, 2010 1:31 PM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications
>>
>> Hi Denis,
>>
>> Silly question, but have you applied to the official RTPproxy the
>> patches that comes with the nathelper module ?
>>
>> Regards,
>> Bogdan
>>
>> Denis Putyato wrote:
>>
>>> Hello!
>>>
>>> During tests of new feature in rtpproxy I received such problem:
>>>
>>> “Dec 27 11:42:42 opensips
>>> /usr/local/opensips1.6.4/sbin/opensips[26496]:
>>> DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring”
>>>
>>> And log for rtpproxy
>>>
>>> “Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new
>>> session 6fe1-00bf-8e08-8065-0002a405c...@172.31.255.250, tag
>>> 6f008e65a4;1 requested, type strong
>>>
>>> Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: new
>>> session on a port 64922 created, tag 6f008e65a4;1
>>>
>>> Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command: setting
>>> timeout handler
>>>
>>> Dec 28 07:43:37 opensips rtpproxy[28223]: INFO:handle_command:
>>> pre-filling caller's address with 3.3.3.3:23066
>>>
>>> Dec 28 07:43:37 opensips
>>> /usr/local/opensips1.6.4/sbin/opensips[28196]:
>>> ERROR:nathelper:force_rtp_proxy: Unable to parse body
>>>
>>> Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command: lookup
>>> on ports 64922/4, session timer restarted
>>>
>>> Dec 28 07:43:38 opensips rtpproxy[28223]: INFO:handle_command:
>>> pre-filling callee's address with 2.2.2.2:18408
>>>
>>> Dec 28 07:43:39 opensips rtpproxy[28223]: INFO:handle_command: lookup
>>> on ports 64922/4, session timer restarted
>>>
>>> Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:process_rtp: session
>>> timeout
>>>
>>> Dec 28 07:44:07 opensips rtpproxy[28223]: INFO:remove_session: RTP
>>> stats: 1449 in from callee, 421 in from caller, 1870 relayed, 0 dropped
>>>
>>> Dec 28 07:44:07 opensips rt

[OpenSIPS-Users] making media proxy use all the cores in a server

2011-01-11 Thread Jayesh Nambiar

Hi All,
I am running mediaproxy 2.4 and I at a little peak I observed media-relay
taking 70% CPU. This is a quad-core server with 8 cores and I don't see
multiple cores being used in that server.
Is there anything to make mediaproxy run in a multi-threaded environment. I
am running it on Debian.

Thanks for any pointers !!

--- Jayesh
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[OpenSIPS-Users] Pacth rtpproxy

2011-01-11 Thread Denis Putyato

Hello!

 

I try patch rtpproxy gotten from git. And there is such error during patching

 

patch < rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

 

rtpproxy_timeout_notification.patch is a patch for timeout notification which  
divide rtp timeout and session initiation timeout notification as said in

http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

 

This patch I got from SVN version of latest Opensips. 

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[OpenSIPS-Users] Dialog Module and Bogus Event 8 in state 2

2011-01-11 Thread Sven Schulz

Running opensips 1.6.3, dialog module seems to function correctly however I
keep getting these messages:

CRITICAL:dialog:log_next_state_dlg: bogus event 8 in state 2 for dlg
0x2b33956052c0 [2284:61359203] with clid
'11a0bd80-d2c160b0-1a-34020...@10.1.2.52' and tags
'd5edda48-9c10-424c-b200-8ec1eb8e532c-42504961' '292F5834-D13'

They only seem to happen when an INVITE is followed by a 183 RINGING
message. INVITES without a 183 wont get this error messege. Is this normal
or should I be concerned?


Sven Schulz
Penn State University
Telecommunications and Network Services
814.865.6116
sip:s...@psu.edu


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Re: [OpenSIPS-Users] b2b refer scenario

2011-01-11 Thread Anton Zagorskiy

Hi Anca.

As I see from the b2b documentation, there is no easy way to catch an
attended refer and make it as INVITE. The problem is in the refer-to field
that contains "Replaces" tag.


May be anyone has an idea how to do this?




Also, why there isn't documnetation for 1.6.4 on the site? There are 1.6.3
and trunk/devel in Cookbooks and 1.6.x and 1.7.x in the tutorials. What is
1.7.x?




WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru



> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Anca Vamanu
> Sent: Wednesday, January 05, 2011 3:53 PM
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] b2b refer scenario
> 
> Hi Anton,
> 
> 01/05/2011 02:09 PM, a.zagors...@oyster-telecom.ru wrote:
> > Hi Anca,
> > thanks for your reply, but could you explain a bit more?
> >
> > I don't understand how B2B will understand that he should execute
> > scenario on REFER request when b2b_init_request is calling on an
> > INVITE request?
> >
> > I'm talking about situation, where there are 2 UAs, they has
> > established session, and one of them sending the REFER request to
> > another.
> >
> Yes, this is exactly how it works, by calling the b2b_init_request on
> the initial Invite, so that B2BUA is in the middle of the call and can
> control it. If you look in the xml scenario file you will see that
> there
> is a rule for REFER there - this is how it know that when a REFER is
> received it has to do what it has to do.
> Try this out  and you will see ;) .
> 
> Regards,
> 
> --
> Anca Vamanu
> www.voice-system.ro
> 
> 
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Re: [OpenSIPS-Users] Pacth rtpproxy

2011-01-11 Thread Razvan Crainea


 Hello Denis,

You are right, OpenSIPS shouldn't even try to terminate the call because 
it isn't established yet. I just added a small fix to solve this 
problem. Please update your code from svn to use this fix.
The RTPProxy patch was done against commit 
"600c80493793bafd2d69427bc22fcb43faad98c5". You can either get the 
RTPProxy from git, change it's branch and then apply the patch, or you 
can download an already patched version from 
http://opensips.org/pub/rtpproxy/.


Regards,
Razvan

On 1/11/2011 2:19 PM, Denis Putyato wrote:


Hello!

I try patch rtpproxy gotten from git. And there is such error during 
patching


patch < rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

rtpproxy_timeout_notification.patch is a patch for timeout 
notification which  divide rtp timeout and session initiation timeout 
notification as said in


http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

This patch I got from SVN version of latest Opensips.


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Re: [OpenSIPS-Users] b2b refer scenario

2011-01-11 Thread Anca Vamanu


On 01/11/2011 04:56 PM, Anton Zagorskiy wrote:

Hi Anca.

As I see from the b2b documentation, there is no easy way to catch an
attended refer and make it as INVITE.

Why do you want to do that?


  The problem is in the refer-to field
that contains "Replaces" tag.


May be anyone has an idea how to do this?




Also, why there isn't documnetation for 1.6.4 on the site? There are 1.6.3
and trunk/devel in Cookbooks and 1.6.x and 1.7.x in the tutorials. What is
1.7.x?




WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru




-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Wednesday, January 05, 2011 3:53 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] b2b refer scenario

Hi Anton,

01/05/2011 02:09 PM, a.zagors...@oyster-telecom.ru wrote:

Hi Anca,
thanks for your reply, but could you explain a bit more?

I don't understand how B2B will understand that he should execute
scenario on REFER request when b2b_init_request is calling on an
INVITE request?

I'm talking about situation, where there are 2 UAs, they has
established session, and one of them sending the REFER request to
another.


Yes, this is exactly how it works, by calling the b2b_init_request on
the initial Invite, so that B2BUA is in the middle of the call and can
control it. If you look in the xml scenario file you will see that
there
is a rule for REFER there - this is how it know that when a REFER is
received it has to do what it has to do.
Try this out  and you will see ;) .

Regards,

--
Anca Vamanu
www.voice-system.ro


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--
Anca Vamanu
www.voice-system.ro


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Re: [OpenSIPS-Users] How to implement a SIP Trunk in between twoSIP servers.

2011-01-11 Thread Leon Li

Hi Steven,

 

To configure the trunk in CUCM, go to Device > Trunk, add a new "SIP
trunk".

 

The configuration fields are pretty straight forward. Important ones are

* Destination Address, i.e. opensips IP

* Port, if not 5060

* CSS for inbound and outbound calls. (this decide what number
you can send calls to and receive calls from opensips)

* Any number transformation if you have

 

This is the basic. If you have questions about particular fields, please
mail in details.

 

Regards,

Leon

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of steven chew
Sent: Tuesday, 11 January 2011 11:50 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] How to implement a SIP Trunk in between
twoSIP servers.

 

Hi Bogdan,

 

Thanks for your reply.

 

 

Your script is very useful for calling between two opensips servers
which I have tested.

However, I don't know how to configure on CUCM 7.0 which I am using.

At the moment, CUCM 7.0 is using Web Config via the Web Browser. 

Can you let me know how to configure on CUCM 7.0?

I will appreciate very much if you give some instructions for
configuring SIP Trunk on CUCM7.0

 

 

Thanks
Kind regards,

Steven,

On 10 January 2011 19:33, Bogdan-Andrei Iancu 
wrote:

Hi Steven,

To do that, you need to add in opensips some routing to 1) recognize the
numbers that needs to be sent to CUCM and 2)route that calls to CUCM.

For script logic it sounds like : if you receive a new call (initial
INVITE) for your local domain, check the URI and divert. If you look at
the default config file, there is comment "# requests for my domain" ->
from that point further you have only initial INVITEs for your local
domain, so you can add after:

  # all numbers starting with 55 are to be sent to CUCM
  if ($rU =~ "^55[0-9]+$") {
# replace the domain part of RURI to point to CUCM
rewritehostport("CUCM_IP:CUCM_PORT");
# route the call out based on RURI
route(1);
  }


For the other way around, you have to put a similar logic in CUCM, like
to divert all calls starting with "12" to opensips - and replace the
domain on RURI with the IP/domain of opensips.



Regards,
Bogdan

steven chew wrote:

Hi Bogdan,

Thank you very much for your reply.

I have an Opensips Server and a Cisco Unified Communication
Manager (CUCM).

If I want to make calls from Opensips Server to CUCM and CUCM to
Opensips Server.

For example:
1) If I dial an extension number "5566" from a SIP Phone "12345"
under Opensips Server, it will try to call to a Cisco IP Phone "5566"
from CUCM through a SIP Trunk.
2) If I dial an extension number "12345" from a Cisco IP Phone
"5566" under CUCM, it will try to call to a SIP Phone "12345" under
Opensips Server through a SIP Trunk.

Can you give some instructions how to configure the above
scenario for dialing extension numbers?

Thanks
Steven, 

On 6 January 2011 21:31, Bogdan-Andrei Iancu
mailto:bog...@voice-system.ro>> wrote:

   Hi Steven,

   If you use the opensips default script, your opensips will
accept
   calls from any other external SIP entities (call targeting a
local
   opensips subscriber).

   If you want to configure your opensips to accept foreign
calls
   only form a specific IP address, you can use the permission
   module, with address table to implement IP-based
authentication.

   Best regards,
   Bogdan

   steven chew wrote:

   Hi everyone,

   I am a newbie with SIP-Trunk in OpenSips.
   I have a Cisco Communication Unified Manager and a
OpenSips
   Server running in two different Virtual Machines.

   I would like to have a SIP trunk in between them "Cisco
   Communication Unified Manager and OpenSips Server".
   Therefore, I can make a call from OpenSips Server's SIP
   Clients to Cisco IP Phone.
   What should I need to add into opensips.cfg configuration
file?

   I hope you can give some simple examples how to do it.
   I look forward to hearing from your advise asap.

   Thanks
   Regards,
   -Steven.




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Re: [OpenSIPS-Users] Pacth rtpproxy

2011-01-11 Thread Denis Putyato

Hello Razvan,

 

“OpenSIPS shouldn't even try to terminate the call because it isn't established 
yet”

As I understand I just do not need to use –W key when starting rtpproxy, it 
does not work at all?

 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Tuesday, January 11, 2011 6:49 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Pacth rtpproxy

 

Hello Denis,

You are right, OpenSIPS shouldn't even try to terminate the call because it 
isn't established yet. I just added a small fix to solve this problem. Please 
update your code from svn to use this fix.
The RTPProxy patch was done against commit 
"600c80493793bafd2d69427bc22fcb43faad98c5". You can either get the RTPProxy 
from git, change it's branch and then apply the patch, or you can download an 
already patched version from http://opensips.org/pub/rtpproxy/.

Regards,
Razvan 

On 1/11/2011 2:19 PM, Denis Putyato wrote: 

Hello!

 

I try patch rtpproxy gotten from git. And there is such error during patching

 

patch < rtpproxy_timeout_notification.patch

patching file main.c

Hunk #1 succeeded at 70 (offset 2 lines).

Hunk #2 FAILED at 120.

Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).

Hunk #4 succeeded at 211 with fuzz 2 (offset 4 lines).

Hunk #5 succeeded at 276 (offset 4 lines).

Hunk #6 succeeded at 742 with fuzz 2 (offset -26 lines).

Hunk #7 succeeded at 758 with fuzz 2 (offset -26 lines).

1 out of 7 hunks FAILED -- saving rejects to file main.c.rej

patching file rtpp_command.c

Hunk #1 FAILED at 795.

Hunk #2 FAILED at 888.

2 out of 2 hunks FAILED -- saving rejects to file rtpp_command.c.rej

patching file rtpp_defines.h

Hunk #1 FAILED at 95.

1 out of 1 hunk FAILED -- saving rejects to file rtpp_defines.h.rej

patching file rtpp_notify.c

 

rtpproxy_timeout_notification.patch is a patch for timeout notification which  
divide rtp timeout and session initiation timeout notification as said in

http://www.opensips.org/html/docs/modules/devel/nathelper.html#id249142

 

This patch I got from SVN version of latest Opensips. 

 
 
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