Re: [OpenSIPS-Users] [Release] Planing 1.8.1 minor release

[Saúl Ibarra Corretgé]

Hi Rudy,

On Aug 13, 2012, at 8:19 PM, Rudy wrote:

> Bogdan,
> 
> Great work on all the improvements. There is a patch I submitted a
> while back for media proxy. I explained this to Vlad that there could
> be other scripts / modules effected. Basically, once you call
> strip_body() on a message, internally you are still able to access
> that body (in some cases an SDP). The get_body() function should not
> return anything if strip_body() was called. My patch fixes this
> problem only in mediaproxy by checking lumps for deleted body. A more
> elegant solution would be to fix opensips internally as described
> above (ie: get_body() should fail after strip_body() being called) .
> 
> https://sourceforge.net/tracker/?func=detail&aid=3530859&group_id=232389&atid=1086412
> 
> It would be great if we can get this fix, or a more proper one into 1.8.1 .
> 

I talked about this with Vlad at ClueCon and there doesn't seem to be a better 
approach given the limitations of the lump system. I think we can get this into 
1.8.1 after a final look.

--
Saúl Ibarra Corretgé
AG Projects




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Re: [OpenSIPS-Users] [Release] Planing 1.8.1 minor release

[Rudy]

Bogdan,

 Great work on all the improvements. There is a patch I submitted a
while back for media proxy. I explained this to Vlad that there could
be other scripts / modules effected. Basically, once you call
strip_body() on a message, internally you are still able to access
that body (in some cases an SDP). The get_body() function should not
return anything if strip_body() was called. My patch fixes this
problem only in mediaproxy by checking lumps for deleted body. A more
elegant solution would be to fix opensips internally as described
above (ie: get_body() should fail after strip_body() being called) .

https://sourceforge.net/tracker/?func=detail&aid=3530859&group_id=232389&atid=1086412

 It would be great if we can get this fix, or a more proper one into 1.8.1 .

Thanks in advance,
--Rudy
Dynamic Packet
Toll-Free: 888.929.VOIP ( 8647 )


On Mon, Aug 13, 2012 at 2:06 PM, Bogdan-Andrei Iancu
 wrote:
> Hello all,
>
> We plan to have 1.8.1 minor version released by Wednesday 15th of August.
> Once again, a minor release includes only bug fixes (on the 1.8 major
> branch).
>
> Thanks to a lot of bugs reports and patches, several minor and critical
> things were fixed. We still have couple of fixes on the pipe (working on
> them) and continuously evaluate the new reports.
>
> So please bring in front whatever reports or patches (regarding bugs) you
> may have, so that we can include them into 1.8.1 release.
>
> Once again, many thanks to all how contributed here!
>
> Best regards,
> Bogdan
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
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[OpenSIPS-Users] [Security Updates] Alerts service for free

[Bogdan-Andrei Iancu]

Hello all,

Following countless inquiries on how the Security Alerts Service runs 
and looks (what's the user experience and benefit), we decided to have 
this service running for free for one months.


What is the Security Alerts service ? Learn more on see 
http://www.opensips.org/Resources/AlertsMain


So, starting from tomorrow 14th of August, for one months, the alerts 
will be automatically send on the "users" mailing list, so people can 
evaluate the value and implementation of this services. You do not need 
to subscribe or anything else - just follow the users mailing list (or 
archive).


In respect to the paid users, we will prolong their subscription with 
one more month :).


Enjoy it :)

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


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[OpenSIPS-Users] [Release] Planing 1.8.1 minor release

[Bogdan-Andrei Iancu]

Hello all,

We plan to have 1.8.1 minor version released by Wednesday 15th of 
August. Once again, a minor release includes only bug fixes (on the 1.8 
major branch).


Thanks to a lot of bugs reports and patches, several minor and critical 
things were fixed. We still have couple of fixes on the pipe (working on 
them) and continuously evaluate the new reports.


So please bring in front whatever reports or patches (regarding bugs) 
you may have, so that we can include them into 1.8.1 release.


Once again, many thanks to all how contributed here!

Best regards,
Bogdan

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


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Re: [OpenSIPS-Users] Click to Dial example that comes with OpenSIPS

[Bogdan-Andrei Iancu]

Hi Duane,

That is really strange. Logically speaking, a change in SDP should have 
no impact on receiving (or not) a message - the SDP is just payload and 
is not even parsed by opensips (unless you ask it from script by using 
rtpproxy/mediaproxy/sipmsgops modules). Anyhow, the request should make 
it to the main route.


I rather suspect that something else is filtering your SIP traffic, 
dropping the INVITEs in the first format :-/ .


Regarding the second issue:

1) the angle brackets are not mandatory by RFC - they should be used 
only if you have URI params or a complex display name (which is not your 
case here)


2) assuming that the phone does not like TO - it should reply with a 400 
Bad request, not a 404 - a 404 represents a routing indication and 
routing is done based RURI not TO.


Regards,

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


On 08/09/2012 06:13 AM, Duane Larson wrote:

I changed the following in the ctd.sh script
Changed the default of
"`printf "v=0\r\no=click-to-dial 0 0 IN IP4 
0.0.0.0\r\ns=session\r\nc=IN IP4 0.0.0.0\r\nb=CT:1000\r\nt=0 
0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:0 
PCMU/8000\r\n"`

To
"`printf "v=0\r\no=click2dial 0 0 IN IP4 50.XX.XX.156\r\ns=click2dial 
call\r\nc=IN IP4 173.XX.XX.111\r\nt=0 0\r\nm=audio 12790 RTP/AVP 0 8 
18 3 4 97 98\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:18 
G729/8000\r\na=rtpmap:97 ilbc/8000\r\na=rtpmap:98 speex/8000\r\n"`

And now it is making it into the OpenSIPS/SBC's main route.  Not sure why.
I noticed another issue now.  My snom phone is receiving the INVITE 
but it is replying with a "404 Not Found" error.  (If I test with a 
Jitsi client I don't have the 404 issue)
This shouldn't happen since the TO header is the correct  SIP URI.  
The only thing that can be wrong is that the To: URI is not in <>
I think the TM MI function t_uac_dlg isn't placing the <> around the 
TO: header URI.  Reading the RFC I am not 100% sure if the <> are 
required.
U 2012/08/08 22:09:13.756976 192.168.88.1:5060 
 -> 192.168.88.13:3072 

INVITE sip:9016XX6XX4@192.168.88.13:3072 
 SIP/2.0.

Max-Forwards: 10.
Record-Route: .
Record-Route: .
Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.
Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.
*To: sip:9016xx6...@irck.com .*
From: >;tag=134448175329440.

CSeq: 1 INVITE.
Call-ID: 134448175329440.fifouacctd.
Content-Length: 226.
User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
Contact: >.

Content-Type: application/sdp.
.
v=0.
o=click2dial 0 0 IN IP4 50.XX.XX.156.
s=click2dial call.
c=IN IP4 173.XX.XX.111.
t=0 0.
m=audio 12790 RTP/AVP 0 8 18 3 4 97 98.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:97 ilbc/8000.
a=rtpmap:98 speex/8000.
#
U 2012/08/08 22:09:13.766974 192.168.88.13:3072 
 -> 192.168.88.1:5060 


SIP/2.0 404 Not found.
Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.
Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.
From: >;tag=134448175329440.

To: mailto:sip%3a9016726...@irock.com>>.
Call-ID: 134448175329440.fifouacctd.
CSeq: 1 INVITE.
User-Agent: snom821/8.7.3.10 .
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
PRACK, MESSAGE, INFO, UPDATE.

Allow-Events: talk, hold, refer, call-info.
Supported: timer, replaces, from-change.
Content-Length: 0.

On Fri, Jul 27, 2012 at 3:24 PM, > wrote:


Very sure. Normal calls are working with clients behind the
OpenSIPS/SBC.




On , Bogdan-Andrei Iancu mailto:bog...@opensips.org>> wrote:
>
>
>
>
>
>
> Duane,some stupid question : are you sure your opensips is
> listening on the given IP:port ? have you check with netstat ?
> also have you checked with netstat also if there is traffic queued
> on the sockets ?
>
>
>
> Regards,
>
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 07/27/2012 12:48 AM, Duane Larson wrote:
> Oh yeah.  My first email has the SIPTrace from the
> OpenSIPS/SBC.  So I am logged into the OpenSIPS/SBC and did the
> NGREP.  So I see it SIP invite (99.XX.XX.161 is the IP of the
> OpenSIPS/SBC).   I would even let you log into the OpenSIPS/SBC
> and see it for yourself.  Makes no sense.
>
>
>
> U 2012/07/19 18:20:13.486847 50.XX.XX.156:5060 -> 99.XX.XX.161:5060
>
> INVITE sip:9016X@192.168.88.13:3072;line=g2hfphrk SIP/2.0.
>
> Via: SIP/2.0/UDP 50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0.
>
> To: sip:9016xx...@irock.com .
>
> From: s

Re: [OpenSIPS-Users] httpd module not working

[duane . larson]

Thanks for the help. Setting pkg to (-M 8) worked.

On , Ovidiu Sas  wrote:

The httpd module needs to reserve a buffer for building http



responses. If there are a lot of modules loaded, the pkg memory might



become fragmented and there will be not enough space to allocate a big



buffer.



Ways to workaround this issue:



- increase the pkg memory (use -M switch in the opensips command)



- play with the size of the buffer (buf_size parameter)





Regards,



Ovidiu Sas







--



VoIP Embedded, Inc.



http://www.voipembedded.com





On Sun, Aug 12, 2012 at 10:21 PM, duane.lar...@gmail.com> wrote:


> I ran the sample OpenSIPS config that comes with all installs and I was  
able


> to start OpenSIPS with the httpd and mi_http modules. Not sure why its  
not


> starting with my config. I will send you the debug to you directly  
after I



> send this email.



>



> On , Ovidiu Sas o...@voipembedded.com> wrote:



>> Can you enable debug probes and print out the output.



>>



>>



>> Also, you can try to run a very simple config and load only httpd and



>>



>>



>> mi_http modules.



>>



>>



>>



>>



>>



>> Regards,



>>



>>



>> Ovidiu Sas



>>



>>



>>



>>



>>



>> On Sun, Aug 12, 2012 at 9:45 PM, duane.lar...@gmail.com> wrote:



>>



>>


>> > I wanted to try out the mi-http module but I am not able to get  
OpenSIPS



>> > to



>>



>>


>> > start up when I have enabled the httpd module. When I start OpenSIPS  
I



>> > am



>>



>>



>> > seeing the following error



>>



>>



>> >



>>



>>



>> > Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]:



>>



>>



>> > NOTICE:presence:child_init: init_child [-2] pid [1537]



>>



>>



>> > Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]:



>>



>>



>> > WARNING:core:fm_malloc: Not enough free memory, will atempt



>> > defragmenation



>>



>>



>> > Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]:



>>



>>



>> > ERROR:httpd:httpd_proc: oom



>>



>>



>> >



>>



>>



>> >



>>



>>



>> > Before compiling the modules I installed both libmicrohttpd-dev and



>>



>>



>> > libmicrohttpd5 on Debian 6. Any idea why its giving me this error?





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Re: [OpenSIPS-Users] httpd module not working

[Ovidiu Sas]

The httpd module needs to reserve a buffer for building http
responses.  If there are a lot of modules loaded, the pkg memory might
become fragmented and there will be not enough space to allocate a big
buffer.
Ways to workaround this issue:
 - increase the pkg memory (use -M switch in the opensips command)
 - play with the size of the buffer (buf_size parameter)

Regards,
Ovidiu Sas


-- 
VoIP Embedded, Inc.
http://www.voipembedded.com

On Sun, Aug 12, 2012 at 10:21 PM,   wrote:
> I ran the sample OpenSIPS config that comes with all installs and I was able
> to start OpenSIPS with the httpd and mi_http modules. Not sure why its not
> starting with my config. I will send you the debug to you directly after I
> send this email.
>
> On , Ovidiu Sas  wrote:
>> Can you enable debug probes and print out the output.
>>
>>
>> Also, you can try to run a very simple config and load only httpd and
>>
>>
>> mi_http modules.
>>
>>
>>
>>
>>
>> Regards,
>>
>>
>> Ovidiu Sas
>>
>>
>>
>>
>>
>> On Sun, Aug 12, 2012 at 9:45 PM,  duane.lar...@gmail.com> wrote:
>>
>>
>> > I wanted to try out the mi-http module but I am not able to get OpenSIPS
>> > to
>>
>>
>> > start up when I have enabled the httpd module. When I start OpenSIPS I
>> > am
>>
>>
>> > seeing the following error
>>
>>
>> >
>>
>>
>> > Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]:
>>
>>
>> > NOTICE:presence:child_init: init_child [-2] pid [1537]
>>
>>
>> > Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]:
>>
>>
>> > WARNING:core:fm_malloc: Not enough free memory, will atempt
>> > defragmenation
>>
>>
>> > Aug 12 20:40:03 SIPProxy02 /usr/local/sbin/opensips[1537]:
>>
>>
>> > ERROR:httpd:httpd_proc: oom
>>
>>
>> >
>>
>>
>> >
>>
>>
>> > Before compiling the modules I installed both libmicrohttpd-dev and
>>
>>
>> > libmicrohttpd5 on Debian 6. Any idea why its giving me this error?

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Re: [OpenSIPS-Users] Opensips and rtpproxy

[dpa]

I understand, thank you.

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, August 13, 2012 5:06 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy

 

Hello!

You can use for several sets, but only a single socket for all of them.

Regards,




Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/13/2012 03:56 PM, dpa wrote:

Hello!

 

And I cannot use one unix socket (timeout notification) for several rtpproxy 
set? 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, August 13, 2012 4:46 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy

 

Hello!

No, currently OpenSIPS doesn't support multiple notification sockets.

Best regards,




Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/13/2012 01:25 PM, dpa wrote:



Hello!

 

There is one question about Opensips and rtpproxy interaction.

Documentation says that I can use more than one rtpproxy set. And to choose 
which rtpproxy to use I should write to opensips.cfg something like 

“modparam("nathelper", "rtpproxy_sock", "1 == /sock/rtpproxy.sock")

modparam("nathelper", "rtpproxy_sock", "2 == /sock/rtpproxy1.sock")”

 

And what about “rtpp_notify_socket” parameter? Can I write it to opensips.cfg 
the same way or multiple rtpproxy can use similar unix socket for timeout 
notification?

 

Opensips 1.6.4-2.

 

Thank you for any help







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Re: [OpenSIPS-Users] opensips-cp MDB2 Error: unknown error

[Alex Ionescu]

Hi Ardogan,

OpenSIPS CP was initially built to work with MySQL. At some point PEAR 
was used in order to be able to use other DB too. Postgres has some 
particularities and there is a chance that a query was build for MySQL 
and Postgres does not support it. What you can do is to go to 
"/opensips-cp/web/tools/system/cdrviewer/template/cdrviewer.main.php" 
and insert this right before line 133:

echo "---".$sql."---"
Execute the query in pgsql and send back both the query and postgres error.

Regards,
Alex


On 08/13/2012 02:47 PM, Ardogan Tekinsoy wrote:


I have configured opensip-cp with postgresql. DB connection works and 
I see the results on Domains or Users/User Management pages.
But I get an error on CDRViewer page when I select default Start Date 
- End Date range. Page doesn't lists the results and this error 
message is seen on the page.


Failed to issue query, error message : MDB2 Error: unknown error

I have records in my cdrs table. Is it possible to trace the details 
of this error in a log file so I can give you more information as well.




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Re: [OpenSIPS-Users] Opensips and rtpproxy

[Răzvan Crainea]

  
  
Hello!

You can use for several sets, but only a single socket for all
of them.

Regards,

  
  Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

  On 08/13/2012 03:56 PM, dpa wrote:


  
  
  
  
  
  
Hello!
 
And I cannot use one unix socket (timeout
notification) for several rtpproxy set? 
 

  
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf
  Of Razvan Crainea
Sent: Monday, August 13, 2012 4:46 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and
rtpproxy
  

 

  Hello!
  
  No, currently OpenSIPS doesn't support multiple
notification sockets.
  
  Best regards,
  

  Razvan Crainea
  OpenSIPS Core Developer
  http://www.opensips-solutions.com
  On 08/13/2012 01:25 PM, dpa wrote:


  
  Hello!
   
  There is one question
  about Opensips and rtpproxy interaction.
  Documentation says
  that I can use more than one rtpproxy set. And to choose
  which rtpproxy to use I should write to opensips.cfg
  something like 
  “modparam("nathelper",
  "rtpproxy_sock", "1 == /sock/rtpproxy.sock")
  modparam("nathelper",
  "rtpproxy_sock", "2 == /sock/rtpproxy1.sock")”
   
  And what about
  “rtpp_notify_socket” parameter? Can I write it to
  opensips.cfg the same way or multiple rtpproxy can use
  similar unix socket for timeout notification?
   
  Opensips 1.6.4-2.
   
  Thank you for any help
  
  
  
  
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Re: [OpenSIPS-Users] Opensips and rtpproxy

[dpa]

Hello!

 

And I cannot use one unix socket (timeout notification) for several rtpproxy 
set? 

 

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Monday, August 13, 2012 4:46 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpproxy

 

Hello!

No, currently OpenSIPS doesn't support multiple notification sockets.

Best regards,



Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 08/13/2012 01:25 PM, dpa wrote:



Hello!

 

There is one question about Opensips and rtpproxy interaction.

Documentation says that I can use more than one rtpproxy set. And to choose 
which rtpproxy to use I should write to opensips.cfg something like 

“modparam("nathelper", "rtpproxy_sock", "1 == /sock/rtpproxy.sock")

modparam("nathelper", "rtpproxy_sock", "2 == /sock/rtpproxy1.sock")”

 

And what about “rtpp_notify_socket” parameter? Can I write it to opensips.cfg 
the same way or multiple rtpproxy can use similar unix socket for timeout 
notification?

 

Opensips 1.6.4-2.

 

Thank you for any help






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Re: [OpenSIPS-Users] Opensips and rtpproxy

[Răzvan Crainea]

  
  
Hello!

No, currently OpenSIPS doesn't support multiple notification
sockets.

Best regards,
  
  Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

  On 08/13/2012 01:25 PM, dpa wrote:


  
  
  
  
  
Hello!
 
There is one question
about Opensips and rtpproxy interaction.
Documentation says that
I can use more than one rtpproxy set. And to choose which
rtpproxy to use I should write to opensips.cfg something
like 
“modparam("nathelper",
"rtpproxy_sock", "1 == /sock/rtpproxy.sock")
modparam("nathelper",
"rtpproxy_sock", "2 == /sock/rtpproxy1.sock")”
 
And what about
“rtpp_notify_socket” parameter? Can I write it to
opensips.cfg the same way or multiple rtpproxy can use
similar unix socket for timeout notification?
 
Opensips 1.6.4-2.
 
Thank you for any help
  
  
  
  
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[OpenSIPS-Users] opensips-cp MDB2 Error: unknown error

[Ardogan Tekinsoy]

I have configured opensip-cp with postgresql. DB connection works and I see
the results on Domains or Users/User Management pages.
But I get an error on CDRViewer page when I select default Start Date - End
Date range. Page doesn't lists the results and this error message is seen
on the page.

Failed to issue query, error message : MDB2 Error: unknown error

I have records in my cdrs table. Is it possible to trace the details of
this error in a log file so I can give you more information as well.
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Re: [OpenSIPS-Users] SIP School Certification

[Kevin Mathy]

Hi Muhammad,

Thanks for your answers. I already know about OpenSIPS trainings, Asterisk
and Cisco ones, but I'd like to find something more "general", and not
provided by a manufacturer or about a particular product.

In France there are some trainings provided by engineers schools, but no
one covering as much as SipSchool's one (according that everything written
in Sipschool outline is really introduced...)

So, I'll continue searching about Sipschool, and wait for someone else's
answer.

Thanks,

Regards,
*
**Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr

Pour toute demande de support, merci de contacter le *03.51.08.42.07*, ou
bien d'adresser un e-mail à *supp...@hexanet.fr*




2012/8/13 Muhammad Shahzad 

> I never heard of this school. I think OpenSIPs guys from time to time
> conduct training and workshops on SIP, that might be useful for you.
>
> In general SIP is a huge field, you should probably get certified in some
> particular SIP product, like OpenSIPs, or Asterisk, or CISCO.
>
> http://www.opensips.org/Training/Certification
> http://www.digium.com/en/training/certifications/
> https://learningnetwork.cisco.com/community/certifications/ccvp
>
>
> Thank you.
>
>
> On Mon, Aug 13, 2012 at 11:42 AM, Kevin Mathy  wrote:
>
>> Hi list,
>>
>> I'm looking for a SIP training, and I've found SIP School Certification
>> (SSCA) at http://www.thesipschool.com .
>>
>> What do you think about it ? According that I already have some knowledge
>> about SIP, and that I'd like to improve it.
>>
>> Does anyone have already pass this training ? Does the SSCA really have a
>> "value" in SIP world ?
>>
>> Then, if you know some other trainings, could you tell me what they are ?
>>
>> Thanks a lot,
>> Regards,
>>
>> *Kevin MATHY*
>> *HEXANET*
>> *
>> --
>> *
>> Téléphone : 03.26.79.30.05
>> Web : www.hexanet.fr
>>
>> Pour toute demande de support, merci de contacter le *03.51.08.42.07*,
>> ou bien d'adresser un e-mail à *supp...@hexanet.fr*
>>
>>
>>
>> ___
>> 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: +92 334 422 40 88
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Opensips and rtpproxy

[dpa]

Hello!

 

There is one question about Opensips and rtpproxy interaction.

Documentation says that I can use more than one rtpproxy set. And to choose
which rtpproxy to use I should write to opensips.cfg something like 

"modparam("nathelper", "rtpproxy_sock", "1 == /sock/rtpproxy.sock")

modparam("nathelper", "rtpproxy_sock", "2 == /sock/rtpproxy1.sock")"

 

And what about "rtpp_notify_socket" parameter? Can I write it to
opensips.cfg the same way or multiple rtpproxy can use similar unix socket
for timeout notification?

 

Opensips 1.6.4-2.

 

Thank you for any help

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Re: [OpenSIPS-Users] SIP School Certification

[Muhammad Shahzad]

I never heard of this school. I think OpenSIPs guys from time to time
conduct training and workshops on SIP, that might be useful for you.

In general SIP is a huge field, you should probably get certified in some
particular SIP product, like OpenSIPs, or Asterisk, or CISCO.

http://www.opensips.org/Training/Certification
http://www.digium.com/en/training/certifications/
https://learningnetwork.cisco.com/community/certifications/ccvp


Thank you.


On Mon, Aug 13, 2012 at 11:42 AM, Kevin Mathy  wrote:

> Hi list,
>
> I'm looking for a SIP training, and I've found SIP School Certification
> (SSCA) at http://www.thesipschool.com .
>
> What do you think about it ? According that I already have some knowledge
> about SIP, and that I'd like to improve it.
>
> Does anyone have already pass this training ? Does the SSCA really have a
> "value" in SIP world ?
>
> Then, if you know some other trainings, could you tell me what they are ?
>
> Thanks a lot,
> Regards,
>
> *Kevin MATHY*
> *HEXANET*
> *
> --
> *
> Téléphone : 03.26.79.30.05
> Web : www.hexanet.fr
>
> Pour toute demande de support, merci de contacter le *03.51.08.42.07*, ou
> bien d'adresser un e-mail à *supp...@hexanet.fr*
>
>
>
> ___
> 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: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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[OpenSIPS-Users] SIP School Certification

[Kevin Mathy]

Hi list,

I'm looking for a SIP training, and I've found SIP School Certification
(SSCA) at http://www.thesipschool.com .

What do you think about it ? According that I already have some knowledge
about SIP, and that I'd like to improve it.

Does anyone have already pass this training ? Does the SSCA really have a
"value" in SIP world ?

Then, if you know some other trainings, could you tell me what they are ?

Thanks a lot,
Regards,

*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr

Pour toute demande de support, merci de contacter le *03.51.08.42.07*, ou
bien d'adresser un e-mail à *supp...@hexanet.fr*
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[OpenSIPS-Users] openserSIPContactTable in Snmpstats Module

[Gomtesh Jain]

Hi ,
  I am using openserSIPContactTable  to monitor my registered contacts in
OpenSips 1.6.
  Opensips is running in  db_mod =2.
  But I notice openserSIPContactTable is not getting updated properly . I
noticed in location table there are 600 contacts but
openserSIPContactTable  reflects only 150 .
Is it but in Opensips or some configuration is required ?

Thanx,
Gomtesh
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[OpenSIPS-Users] opensips-cp MDB2 Error: unknown error

[Ardogan Tekinsoy]

I have configured opensip-cp with postgresql. DB connection works and I see
the results on Domains or Users/User Management pages.
But I get an error on CDRViewer page when I select default Start Date - End
Date range. Page doesn't lists the results and this error message is seen
on the page.

Failed to issue query, error message : MDB2 Error: unknown error

I have records in my cdrs table. Is it possible to trace the details of
this error in a log file so I can give you more information as well.
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