Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available

[Bogdan-Andrei Iancu]

Hi Maxim,

During the 2.1 devel there was an attempt to have an auto load of the 
proto modules (modules providing proto implementations for SIP, like 
UDP, TCP, TLS, SCTP, WS, etc). This attempt failed due multiple reasons 
like some inconsistency in the syntax of config file (dependencies 
between "listen", "mpath" , "loadmodule") or like user experience.


Before releasing the first 2.1, the decision was made to explicitly load 
the proto modules as any other opensips module.


There is nothing to state in the release notes as it was never 
officially states the other way around. It was an experiment with no 
happy end.


Best regards,

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

On 24.08.2015 22:02, Maxim Sobolev wrote:
No, we have not loaded it yet. Is it now always required? As I said my 
point here is not so much how to fix it, but the fact that Liviu 
Chircu said that loading such module is just a workaround and the 
proper fix would be applied before the 2.1 x release goes out. If it 
was decided that loading module is now the "official" way to go, then 
it should be reflected in the relnotes IMHO.


-Maxim

On Sat, Aug 22, 2015 at 11:18 AM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Maxim,

Do you load the proto_udp module ?

Regards,
Bogdan


Sent from Samsung Mobile


 Original message 
From: Maxim Sobolev
Date:22/08/2015 18:48 (GMT+02:00)
To: OpenSIPS devel mailling list
Cc: n...@lists.opensips.org
,users@lists.opensips.org
, busin...@lists.opensips.org

Subject: Re: [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now
available

Hi Bogdan,

For some reason 2.1.x is still failing our voiptests travis run
with the following error when trying to run in the UDP-only mode:

Aug 22 15:32:00 [10854] ERROR:core:fix_all_socket_lists: listeners
found for protocol udp, but no module can handle it

Aug 22 15:32:00 [10854] ERROR:core:main: failed to initialize list
addresses

It was told on the mailing list before that it would be fixed
before the release:


http://opensips-open-sip-server.1449251.n2.nabble.com/Unable-to-start-opensips-2-1-1dev-in-UDP-only-mode-td7595563.html

So I guess that never happened. Could you guys look into it or at
least add some kind of errata or relnotes entry?

Thanks!

-Maxim


On Wed, Aug 19, 2015 at 11:03 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hello everyone,

Minor version 2.1.1 is now available on branch 2.1. This is a
release bringing multiple and valuable fixes, a result of the
continues work of testing and fixing the revolutionary 2.1
version.

Please update as soon as possible as it worth it ! Download
the tarball with sources from :
http://opensips.org/pub/opensips/2.1.1/

RPM and DEB packages will be shortly available on the official
repositories, after the nightly builts.

There are hundreds of reports, tens of fixes and maybe several
hundreds of commits - all these are the result of the entire
OpenSIPS community - people testing, reporting and fixes. And
I want to thanks to all these people, to these OpenSIPS'ers !

Enjoy 2.1.1 !!

-- 
Bogdan-Andrei Iancu

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


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Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474 
Tel (Toll-Free): +1-855-747-7779 
Fax: +1-866-857-6942 
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com 
Skype: SippySoft




--
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com 
Skype: SippySoft


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[OpenSIPS-Users] Public Meetings Schedule

[Răzvan Crainea]

Hello, all!

According to our plans, Public Meetings should be held every last 
Wednesday of the month, at 17:00 EET. However, due to vacation plans, we 
skipped the July's Public Meeting and we will also skip the August one 
(which should have been held tomorrow).
Therefore the next Public Meeting is scheduled for 30th September 2015. 
We haven't chosen a topic yet, but we will announce it during the next 
weeks.


Enjoy your summer,

--
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com


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Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Nabeel]

Currently, OpenSIPS is using the actual IP address in the record-route URI,
but I believe my SIP client needs the domain name in the record-route
instead.


For example, it should be:

Record-Route: 



NOT:

Record-Route: 



How can I make this change in the OpenSIPS config?

This should solve the problem because in a working setup (different SIP
server), the logs state *"Resolving host address 'sipdomain.com
'"* and the record route URI includes the domain
name, but in the OpenSIPS setup the logs state *"Resolving host address
'162.242.153.259'* and the record route URI contains the IP address.


On 24 August 2015 at 18:37, Nabeel  wrote:

> Hi,
>
> I see the cause now on the UAC side; I know it seems simple to just add
> some DNS records to the server IP,  but I'm still pondering on the best way
> to solve this and where exactly to add the SRV records because:
>
> 1) I already have the SRV records set up on the actual hostname / domain,
> hosted by a DNS service third party, which is easier for me to maintain.
> However the UAC seems to be ignoring this.
>
> 2) I have used the same UAC with another server and did not have to set up
> SRV on the actual server machine IP.
>
> I'm not sure if this has anything to do with the OpenSIPS config but I'll
> let you know if I solve it.
> On 24 Aug 2015 17:56, "Bogdan-Andrei Iancu"  wrote:
>
>> Hi ,
>>
>> So, is the problem solved (by your findings in the UAS side) ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 24.08.2015 18:25, Nabeel wrote:
>>
>> I just discovered that the SIP client logs show an error message only on
>> the recipient side, not on the caller's side.  I missed this previously
>> because the caller's side log does not show any error:
>>
>> java.lang.Exception: No DNS SRV or A results found for: 162.242.153.259
>>>  (IP address of OpenSIPS server).
>>
>>
>> I have the SRV records set on the actual hostname/domain, but it seems to
>> be looking for SRV at the actual IP address itself.
>>
>> On 21 August 2015 at 17:57, Nabeel  wrote:
>>
>>> The log doesn't show any errors when the Timeout occurs, it only shows
>>> this:
>>>
>>> opensips[1842]: ACC: call missed:
 timestamp=1440174643;method=INVITE;from_tag=z9hG4bK04147190;to_tag=;call_id=
 424618310389@10.137.181.237;code=408;reason=Request Timeout

>>>
>>>
>>> This seems to occur sporadically; some calls connect without problem but
>>> others don't; so perhaps it is a genuine timeout... maybe it simply longer
>>> to connect on some calls?
>>>
>>>
>>> On 21 August 2015 at 17:46, Nabeel  wrote:
>>>
 Sorry to bring this up again, but I still get the 408 Request Timeout
 on some calls.

 Isn't there just a way to increase the request timeout limit?

 Here is the trace:

 http://pastebin.com/jvCPGYDu

 There is even an ACK in the trace after the request timeout message,
 but the call doesn't connect.

 On 7 August 2015 at 18:10, Bogdan-Andrei Iancu 
 wrote:

> Indeed,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 07.08.2015 20:08, Nabeel wrote:
>
> You mean like this, right?
>
> if (is_method("REGISTER"))
>
> {
> if (   0 ) setflag(TCP_PERSISTENT);
>
> setbflag(SIP_PING_FLAG);
>
> if (!save("location"))
> sl_reply_error();
>
> exit;
> }
>
>
>
> On 7 August 2015 at 17:52, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Nabeel,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 07.08.2015 19:39, Nabeel wrote:
>>
>> []
>> Bogdan,
>>
>> Regarding UDP, I realised that the UDP port could not be in LISTEN
>> state and this was probably preventing my server from fully opening that
>> port.  Running nmap on that port showed result "open|filtered", unlike 
>> with
>> TCP which showed fully open.  I am not running any firewalls on my 
>> server,
>> so this seems to be the default behaviour of my network.
>>
>> A bidirectional traffic through the NAT will keep the NAT pinhole
>> open, while a unidirectional one may not. This is the advantage of the 
>> SIP
>> pinging versus simple UDP pinging.
>>
>>
>> I would like to clarify one thing.  You mentioned adding
>> setbflag(SIP_PING_FLAG) before doing save(), but in my config file I 
>> don't
>> see save() anywhere, there is only this line: "if (!save("location"))".
>> Where exactly do I add this line?
>>
>> exactly.
>>
>> Regards,
>> Bogdan
>>
>>
>
>

>>>
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
_

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available

[Bogdan-Andrei Iancu]

Hi Maxim,

The inter-module dependencies is something new added in 2.1 and indeed, 
the check is misplaced in relation to the "-c/C" checks. It is not a 
regression, but rather an bug in the new code.
Thank you for reporting and opening the ticket, Liviu will have it fixed 
today.


Best regards,

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

On 25.08.2015 01:47, Maxim Sobolev wrote:
Another issue with 2.1.x, the -C option (check config) seems to have 
rotten. It was expected to check basic things about config that does 
not require setting up full run-time environment, but it is not even 
checking inter-module dependencies. I.e.:


[sobomax@van01 ~/projects/voiptests]$ ./dist/opensips/opensips -f 
opensips.cfg -C

Listening on
 udp: 127.0.0.1 [127.0.0.1]:5060
Aliases:
 udp: localhost.my.domain:5060
 udp: localhost:5060

Aug 24 13:51:54 [31495] NOTICE:core:main: config file ok, exiting...
[sobomax@van01 ~/projects/voiptests]$ ./dist/opensips/opensips -f 
opensips.cfg -D -E

Listening on
 udp: 127.0.0.1 [127.0.0.1]:5060
Aliases:
 udp: localhost.my.domain:5060
 udp: localhost:5060

Aug 24 13:52:02 [31820] WARNING:core:main: no fork mode
Aug 24 13:52:02 [31820] NOTICE:core:main: version: opensips 2.1.1 
(x86_64/freebsd)
Aug 24 13:52:02 [31820] WARNING:core:solve_module_dependencies: module 
rtpproxy depends on module dialog, but it was not loaded!
Aug 24 13:52:02 [31820] ERROR:core:init_modules: failed to solve 
module dependencies

Aug 24 13:52:02 [31820] ERROR:core:main: error while initializing modules

I've opened a ticket on that (#616).

On Mon, Aug 24, 2015 at 12:02 PM, Maxim Sobolev > wrote:


No, we have not loaded it yet. Is it now always required? As I
said my point here is not so much how to fix it, but the fact
that Liviu Chircu said that loading such module is just a
workaround and the proper fix would be applied before
the 2.1 x release goes out. If it was decided that loading module
is now the "official" way to go, then it should be reflected in
the relnotes IMHO.

-Maxim

On Sat, Aug 22, 2015 at 11:18 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Maxim,

Do you load the proto_udp module ?

Regards,
Bogdan


Sent from Samsung Mobile


 Original message 
From: Maxim Sobolev
Date:22/08/2015 18:48 (GMT+02:00)
To: OpenSIPS devel mailling list
Cc: n...@lists.opensips.org
,users@lists.opensips.org
, busin...@lists.opensips.org

Subject: Re: [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now
available

Hi Bogdan,

For some reason 2.1.x is still failing our voiptests travis
run with the following error when trying to run in the
UDP-only mode:

Aug 22 15:32:00 [10854] ERROR:core:fix_all_socket_lists:
listeners found for protocol udp, but no module can handle it

Aug 22 15:32:00 [10854] ERROR:core:main: failed to initialize
list addresses

It was told on the mailing list before that it would be fixed
before the release:


http://opensips-open-sip-server.1449251.n2.nabble.com/Unable-to-start-opensips-2-1-1dev-in-UDP-only-mode-td7595563.html

So I guess that never happened. Could you guys look into it or
at least add some kind of errata or relnotes entry?

Thanks!

-Maxim


On Wed, Aug 19, 2015 at 11:03 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hello everyone,

Minor version 2.1.1 is now available on branch 2.1. This
is a release bringing multiple and valuable fixes, a
result of the continues work of testing and fixing the
revolutionary 2.1 version.

Please update as soon as possible as it worth it !
Download the tarball with sources from :
http://opensips.org/pub/opensips/2.1.1/

RPM and DEB packages will be shortly available on the
official repositories, after the nightly builts.

There are hundreds of reports, tens of fixes and maybe
several hundreds of commits - all these are the result of
the entire OpenSIPS community - people testing, reporting
and fixes. And I want to thanks to all these people, to
these OpenSIPS'ers !

Enjoy 2.1.1 !!

-- 
Bogdan-Andrei Iancu

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


___
Devel mailing list
de...@lists.opensips.org 

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Bogdan-Andrei Iancu]

Hi,

According to the RFC, in RR header can be IP or FQDN (any kind of SIP 
URI). Even more, the best practice is to actually use IPs in RR to be 
100% sure that the following requests to hit exactly the same box (if 
using FQDN, subject to DNS resolving, a different IP may be lookup up 
later).


If you really want to put an IP there, use the record_route_preset() 
function:

http://www.opensips.org/html/docs/modules/1.11.x/rr.html#id293864

Regards,

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

On 25.08.2015 16:47, Nabeel wrote:
Currently, OpenSIPS is using the actual IP address in the record-route 
URI, but I believe my SIP client needs the domain name in the 
record-route instead.



For example, it should be:

Record-Route: http://sipdomain.com>;lr;nat=yes;did=29.3daff1f4>


NOT:

Record-Route: 



How can I make this change in the OpenSIPS config?

This should solve the problem because in a working setup (different 
SIP server), the logs state /"Resolving host address 'sipdomain.com 
'"/ and the record route URI includes the domain 
name, but in the OpenSIPS setup the logs state /"Resolving host 
address '162.242.153.259'/ and the record route URI contains the IP 
address.



On 24 August 2015 at 18:37, Nabeel > wrote:


Hi,

I see the cause now on the UAC side; I know it seems simple to
just add some DNS records to the server IP,  but I'm still
pondering on the best way to solve this and where exactly to add
the SRV records because:

1) I already have the SRV records set up on the actual hostname /
domain, hosted by a DNS service third party, which is easier for
me to maintain.  However the UAC seems to be ignoring this.

2) I have used the same UAC with another server and did not have
to set up SRV on the actual server machine IP.

I'm not sure if this has anything to do with the OpenSIPS config
but I'll let you know if I solve it.

On 24 Aug 2015 17:56, "Bogdan-Andrei Iancu" mailto:bog...@opensips.org>> wrote:

Hi ,

So, is the problem solved (by your findings in the UAS side) ?

Regards,

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

On 24.08.2015 18:25, Nabeel wrote:

I just discovered that the SIP client logs show an error
message only on the recipient side, not on the caller's
side.  I missed this previously because the caller's side log
does not show any error:

java.lang.Exception: No DNS SRV or A results found for:
162.242.153.259  (IP address of OpenSIPS server).


I have the SRV records set on the actual hostname/domain, but
it seems to be looking for SRV at the actual IP address itself.

On 21 August 2015 at 17:57, Nabeel mailto:nabeelshik...@gmail.com>> wrote:

The log doesn't show any errors when the Timeout occurs,
it only shows this:

opensips[1842]: ACC: call missed:

timestamp=1440174643;method=INVITE;from_tag=z9hG4bK04147190;to_tag=;call_id=424618310389@10.137.181.237
;code=408;reason=Request
Timeout 



This seems to occur sporadically; some calls connect
without problem but others don't; so perhaps it is a
genuine timeout... maybe it simply longer to connect on
some calls?


On 21 August 2015 at 17:46, Nabeel
mailto:nabeelshik...@gmail.com>> wrote:

Sorry to bring this up again, but I still get the 408
Request Timeout on some calls.

Isn't there just a way to increase the request
timeout limit?

Here is the trace:

http://pastebin.com/jvCPGYDu

There is even an ACK in the trace after the request
timeout message, but the call doesn't connect.

On 7 August 2015 at 18:10, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Indeed,

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

On 07.08.2015 20:08, Nabeel wrote:

You mean like this, right?

if (is_method("REGISTER"))

{
if ( 0 ) setflag(TCP_PERSISTENT);

setbflag(SIP_PING_FLAG);

if (!save("location"))
sl_reply_error();

exit;
}



On 7 August 2015 at 17:52, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Nabeel,

Bogdan-Andrei Iancu
   

Re: [OpenSIPS-Users] Use of socket_info and local_contact columns in the active_watcher table

[Bogdan-Andrei Iancu]

Hi,

When the second SUBSCRIBE is received, the watcher info in DB will be 
updated accordingly  (or a new record will be added, depending if a 
re-SUBSCRIBE).

So, server 2 will use this new info in order to generate the NOTIFY's

Regards,

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

On 25.08.2015 05:20, surya wrote:

I was wondering what is the use of socket_info and local_contact columns in
the active_watcher table.

My concern is that if I have two opensips server using the shared db does
these columns pose any problem. For example the initial subscribe is
received on server 1 and the next one is received on server 2, will the
second server able to send the NOTIFY or it fails.

I googled for answers  but could not find any. However, there are few
similar questions here.

Any help will be appreciated.

Thanks.



--
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http://opensips-open-sip-server.1449251.n2.nabble.com/Use-of-socket-info-and-local-contact-columns-in-the-active-watcher-table-tp7598639.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Nabeel]

Please show me an example of where / how to use record_route_preset() to
add the FQDN.

On 25 August 2015 at 16:54, Bogdan-Andrei Iancu  wrote:

> Hi,
>
> According to the RFC, in RR header can be IP or FQDN (any kind of SIP
> URI). Even more, the best practice is to actually use IPs in RR to be 100%
> sure that the following requests to hit exactly the same box (if using
> FQDN, subject to DNS resolving, a different IP may be lookup up later).
>
> If you really want to put an IP there, use the record_route_preset()
> function:
> http://www.opensips.org/html/docs/modules/1.11.x/rr.html#id293864
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 25.08.2015 16:47, Nabeel wrote:
>
> Currently, OpenSIPS is using the actual IP address in the record-route
> URI, but I believe my SIP client needs the domain name in the record-route
> instead.
>
>
> For example, it should be:
>
> Record-Route: 
>
>
>
> NOT:
>
> Record-Route: 
>
>
>
> How can I make this change in the OpenSIPS config?
>
> This should solve the problem because in a working setup (different SIP
> server), the logs state *"Resolving host address 'sipdomain.com
> '"* and the record route URI includes the domain
> name, but in the OpenSIPS setup the logs state *"Resolving host address
> '162.242.153.259'* and the record route URI contains the IP address.
>
>
> On 24 August 2015 at 18:37, Nabeel  wrote:
>
>> Hi,
>>
>> I see the cause now on the UAC side; I know it seems simple to just add
>> some DNS records to the server IP,  but I'm still pondering on the best way
>> to solve this and where exactly to add the SRV records because:
>>
>> 1) I already have the SRV records set up on the actual hostname / domain,
>> hosted by a DNS service third party, which is easier for me to maintain.
>> However the UAC seems to be ignoring this.
>>
>> 2) I have used the same UAC with another server and did not have to set
>> up SRV on the actual server machine IP.
>>
>> I'm not sure if this has anything to do with the OpenSIPS config but I'll
>> let you know if I solve it.
>> On 24 Aug 2015 17:56, "Bogdan-Andrei Iancu"  wrote:
>>
>>> Hi ,
>>>
>>> So, is the problem solved (by your findings in the UAS side) ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 24.08.2015 18:25, Nabeel wrote:
>>>
>>> I just discovered that the SIP client logs show an error message only on
>>> the recipient side, not on the caller's side.  I missed this previously
>>> because the caller's side log does not show any error:
>>>
>>> java.lang.Exception: No DNS SRV or A results found for: 162.242.153.259
  (IP address of OpenSIPS server).
>>>
>>>
>>> I have the SRV records set on the actual hostname/domain, but it seems
>>> to be looking for SRV at the actual IP address itself.
>>>
>>> On 21 August 2015 at 17:57, Nabeel  wrote:
>>>
 The log doesn't show any errors when the Timeout occurs, it only shows
 this:

 opensips[1842]: ACC: call missed:
> timestamp=1440174643;method=INVITE;from_tag=z9hG4bK04147190;to_tag=;call_id=
> 424618310389@10.137.181.237;code=408;reason=Request Timeout
>


 This seems to occur sporadically; some calls connect without problem
 but others don't; so perhaps it is a genuine timeout... maybe it simply
 longer to connect on some calls?


 On 21 August 2015 at 17:46, Nabeel  wrote:

> Sorry to bring this up again, but I still get the 408 Request Timeout
> on some calls.
>
> Isn't there just a way to increase the request timeout limit?
>
> Here is the trace:
>
> http://pastebin.com/jvCPGYDu
>
> There is even an ACK in the trace after the request timeout message,
> but the call doesn't connect.
>
> On 7 August 2015 at 18:10, Bogdan-Andrei Iancu 
> wrote:
>
>> Indeed,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 07.08.2015 20:08, Nabeel wrote:
>>
>> You mean like this, right?
>>
>> if (is_method("REGISTER"))
>>
>> {
>> if (   0 ) setflag(TCP_PERSISTENT);
>>
>> setbflag(SIP_PING_FLAG);
>>
>> if (!save("location"))
>> sl_reply_error();
>>
>> exit;
>> }
>>
>>
>>
>> On 7 August 2015 at 17:52, Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi Nabeel,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 07.08.2015 19:39, Nabeel wrote:
>>>
>>> []
>>> Bogdan,
>>>
>>> Regarding UDP, I realised that the UDP port could not be in LISTEN
>>> state and this was probably preventing my server from fully opening that
>>> port.  Running nmap on that port showed result "open|filtered", unlike 
>>> with
>>> TCP which showed 

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Eric Tamme]

http://www.opensips.org/html/docs/modules/devel/rr.html#id293868

On 08/25/2015 10:20 AM, Nabeel wrote:
Please show me an example of where / how to use record_route_preset() 
to add the FQDN.
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Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Nabeel]

Unfortunately, I am not so experienced with OpenSIPS and need a more
specific example.

On 25 August 2015 at 17:23, Eric Tamme  wrote:

>
> http://www.opensips.org/html/docs/modules/devel/rr.html#id293868
>
> On 08/25/2015 10:20 AM, Nabeel wrote:
>
> Please show me an example of where / how to use record_route_preset() to
> add the FQDN.
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available

[Maxim Sobolev]

Hi Bogdan,

For some reason 2.1.x is still failing our voiptests travis run with the
following error when trying to run in the UDP-only mode:

Aug 22 15:32:00 [10854] ERROR:core:fix_all_socket_lists: listeners
found for protocol udp, but no module can handle it

Aug 22 15:32:00 [10854] ERROR:core:main: failed to initialize list addresses

It was told on the mailing list before that it would be fixed before the
release:

http://opensips-open-sip-server.1449251.n2.nabble.com/Unable-to-start-opensips-2-1-1dev-in-UDP-only-mode-td7595563.html

So I guess that never happened. Could you guys look into it or at least add
some kind of errata or relnotes entry?

Thanks!

-Maxim


On Wed, Aug 19, 2015 at 11:03 AM, Bogdan-Andrei Iancu 
wrote:

> Hello everyone,
>
> Minor version 2.1.1 is now available on branch 2.1. This is a release
> bringing multiple and valuable fixes, a result of the continues work of
> testing and fixing the revolutionary 2.1 version.
>
> Please update as soon as possible as it worth it ! Download the tarball
> with sources from :
> http://opensips.org/pub/opensips/2.1.1/
>
> RPM and DEB packages will be shortly available on the official
> repositories, after the nightly builts.
>
> There are hundreds of reports, tens of fixes and maybe several hundreds of
> commits - all these are the result of the entire OpenSIPS community -
> people testing, reporting and fixes. And I want to thanks to all these
> people, to these OpenSIPS'ers !
>
> Enjoy 2.1.1 !!
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> ___
> Devel mailing list
> de...@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>



-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available

[Maxim Sobolev]

No, we have not loaded it yet. Is it now always required? As I said my
point here is not so much how to fix it, but the fact that Liviu Chircu
said that loading such module is just a workaround and the proper fix would
be applied before the 2.1 x release goes out. If it was decided that
loading module is now the "official" way to go, then it should be reflected
in the relnotes IMHO.

-Maxim

On Sat, Aug 22, 2015 at 11:18 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Maxim,
>
> Do you load the proto_udp module ?
>
> Regards,
> Bogdan
>
>
> Sent from Samsung Mobile
>
>
>  Original message 
> From: Maxim Sobolev
> Date:22/08/2015 18:48 (GMT+02:00)
> To: OpenSIPS devel mailling list
> Cc: n...@lists.opensips.org,users@lists.opensips.org,
> busin...@lists.opensips.org
> Subject: Re: [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available
>
> Hi Bogdan,
>
> For some reason 2.1.x is still failing our voiptests travis run with the
> following error when trying to run in the UDP-only mode:
>
> Aug 22 15:32:00 [10854] ERROR:core:fix_all_socket_lists: listeners found for 
> protocol udp, but no module can handle it
>
> Aug 22 15:32:00 [10854] ERROR:core:main: failed to initialize list addresses
>
> It was told on the mailing list before that it would be fixed before the
> release:
>
>
> http://opensips-open-sip-server.1449251.n2.nabble.com/Unable-to-start-opensips-2-1-1dev-in-UDP-only-mode-td7595563.html
>
> So I guess that never happened. Could you guys look into it or at least
> add some kind of errata or relnotes entry?
>
> Thanks!
>
> -Maxim
>
>
> On Wed, Aug 19, 2015 at 11:03 AM, Bogdan-Andrei Iancu  > wrote:
>
>> Hello everyone,
>>
>> Minor version 2.1.1 is now available on branch 2.1. This is a release
>> bringing multiple and valuable fixes, a result of the continues work of
>> testing and fixing the revolutionary 2.1 version.
>>
>> Please update as soon as possible as it worth it ! Download the tarball
>> with sources from :
>> http://opensips.org/pub/opensips/2.1.1/
>>
>> RPM and DEB packages will be shortly available on the official
>> repositories, after the nightly builts.
>>
>> There are hundreds of reports, tens of fixes and maybe several hundreds
>> of commits - all these are the result of the entire OpenSIPS community -
>> people testing, reporting and fixes. And I want to thanks to all these
>> people, to these OpenSIPS'ers !
>>
>> Enjoy 2.1.1 !!
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> ___
>> Devel mailing list
>> de...@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>>
>
>
>
> --
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sa...@sippysoft.com
> Skype: SippySoft
>



-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft
___
Users mailing list
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available

[Maxim Sobolev]

Another issue with 2.1.x, the -C option (check config) seems to have
rotten. It was expected to check basic things about config that does not
require setting up full run-time environment, but it is not even checking
inter-module dependencies. I.e.:

[sobomax@van01 ~/projects/voiptests]$ ./dist/opensips/opensips -f
opensips.cfg -C
Listening on
 udp: 127.0.0.1 [127.0.0.1]:5060
Aliases:
 udp: localhost.my.domain:5060
 udp: localhost:5060

Aug 24 13:51:54 [31495] NOTICE:core:main: config file ok, exiting...
[sobomax@van01 ~/projects/voiptests]$ ./dist/opensips/opensips -f
opensips.cfg -D -E
Listening on
 udp: 127.0.0.1 [127.0.0.1]:5060
Aliases:
 udp: localhost.my.domain:5060
 udp: localhost:5060

Aug 24 13:52:02 [31820] WARNING:core:main: no fork mode
Aug 24 13:52:02 [31820] NOTICE:core:main: version: opensips 2.1.1
(x86_64/freebsd)
Aug 24 13:52:02 [31820] WARNING:core:solve_module_dependencies: module
rtpproxy depends on module dialog, but it was not loaded!
Aug 24 13:52:02 [31820] ERROR:core:init_modules: failed to solve module
dependencies
Aug 24 13:52:02 [31820] ERROR:core:main: error while initializing modules

I've opened a ticket on that (#616).

On Mon, Aug 24, 2015 at 12:02 PM, Maxim Sobolev 
wrote:

> No, we have not loaded it yet. Is it now always required? As I said my
> point here is not so much how to fix it, but the fact that Liviu Chircu
> said that loading such module is just a workaround and the proper fix would
> be applied before the 2.1 x release goes out. If it was decided that
> loading module is now the "official" way to go, then it should be reflected
> in the relnotes IMHO.
>
> -Maxim
>
> On Sat, Aug 22, 2015 at 11:18 AM, Bogdan-Andrei Iancu  > wrote:
>
>> Hi Maxim,
>>
>> Do you load the proto_udp module ?
>>
>> Regards,
>> Bogdan
>>
>>
>> Sent from Samsung Mobile
>>
>>
>>  Original message 
>> From: Maxim Sobolev
>> Date:22/08/2015 18:48 (GMT+02:00)
>> To: OpenSIPS devel mailling list
>> Cc: n...@lists.opensips.org,users@lists.opensips.org,
>> busin...@lists.opensips.org
>> Subject: Re: [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available
>>
>> Hi Bogdan,
>>
>> For some reason 2.1.x is still failing our voiptests travis run with the
>> following error when trying to run in the UDP-only mode:
>>
>> Aug 22 15:32:00 [10854] ERROR:core:fix_all_socket_lists: listeners found for 
>> protocol udp, but no module can handle it
>>
>> Aug 22 15:32:00 [10854] ERROR:core:main: failed to initialize list addresses
>>
>> It was told on the mailing list before that it would be fixed before the
>> release:
>>
>>
>> http://opensips-open-sip-server.1449251.n2.nabble.com/Unable-to-start-opensips-2-1-1dev-in-UDP-only-mode-td7595563.html
>>
>> So I guess that never happened. Could you guys look into it or at least
>> add some kind of errata or relnotes entry?
>>
>> Thanks!
>>
>> -Maxim
>>
>>
>> On Wed, Aug 19, 2015 at 11:03 AM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>
>>> Hello everyone,
>>>
>>> Minor version 2.1.1 is now available on branch 2.1. This is a release
>>> bringing multiple and valuable fixes, a result of the continues work of
>>> testing and fixing the revolutionary 2.1 version.
>>>
>>> Please update as soon as possible as it worth it ! Download the tarball
>>> with sources from :
>>> http://opensips.org/pub/opensips/2.1.1/
>>>
>>> RPM and DEB packages will be shortly available on the official
>>> repositories, after the nightly builts.
>>>
>>> There are hundreds of reports, tens of fixes and maybe several hundreds
>>> of commits - all these are the result of the entire OpenSIPS community -
>>> people testing, reporting and fixes. And I want to thanks to all these
>>> people, to these OpenSIPS'ers !
>>>
>>> Enjoy 2.1.1 !!
>>>
>>> --
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>>
>>> ___
>>> Devel mailing list
>>> de...@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>>>
>>
>>
>>
>> --
>> Maksym Sobolyev
>> Sippy Software, Inc.
>> Internet Telephony (VoIP) Experts
>> Tel (Canada): +1-778-783-0474
>> Tel (Toll-Free): +1-855-747-7779
>> Fax: +1-866-857-6942
>> Web: http://www.sippysoft.com
>> MSN: sa...@sippysoft.com
>> Skype: SippySoft
>>
>
>
>
> --
> Maksym Sobolyev
> Sippy Software, Inc.
> Internet Telephony (VoIP) Experts
> Tel (Canada): +1-778-783-0474
> Tel (Toll-Free): +1-855-747-7779
> Fax: +1-866-857-6942
> Web: http://www.sippysoft.com
> MSN: sa...@sippysoft.com
> Skype: SippySoft
>



-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Bogdan-Andrei Iancu]

The example is really straight forward, I would say :

record_route_preset("sipdomain.com  ");

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

On 25.08.2015 19:20, Nabeel wrote:
Please show me an example of where / how to use record_route_preset() 
to add the FQDN.


On 25 August 2015 at 16:54, Bogdan-Andrei Iancu > wrote:


Hi,

According to the RFC, in RR header can be IP or FQDN (any kind of
SIP URI). Even more, the best practice is to actually use IPs in
RR to be 100% sure that the following requests to hit exactly the
same box (if using FQDN, subject to DNS resolving, a different IP
may be lookup up later).

If you really want to put an IP there, use the
record_route_preset() function:
http://www.opensips.org/html/docs/modules/1.11.x/rr.html#id293864

Regards,

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

On 25.08.2015 16:47, Nabeel wrote:

Currently, OpenSIPS is using the actual IP address in the
record-route URI, but I believe my SIP client needs the domain
name in the record-route instead.


For example, it should be:

Record-Route: http://sipdomain.com>;lr;nat=yes;did=29.3daff1f4>


NOT:

Record-Route: 



How can I make this change in the OpenSIPS config?

This should solve the problem because in a working setup
(different SIP server), the logs state /"Resolving host address
'sipdomain.com '"/ and the record route URI
includes the domain name, but in the OpenSIPS setup the logs
state /"Resolving host address '162.242.153.259'/ and the record
route URI contains the IP address.


On 24 August 2015 at 18:37, Nabeel mailto:nabeelshik...@gmail.com>> wrote:

Hi,

I see the cause now on the UAC side; I know it seems simple
to just add some DNS records to the server IP,  but I'm still
pondering on the best way to solve this and where exactly to
add the SRV records because:

1) I already have the SRV records set up on the actual
hostname / domain, hosted by a DNS service third party, which
is easier for me to maintain.  However the UAC seems to be
ignoring this.

2) I have used the same UAC with another server and did not
have to set up SRV on the actual server machine IP.

I'm not sure if this has anything to do with the OpenSIPS
config but I'll let you know if I solve it.

On 24 Aug 2015 17:56, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org>> wrote:

Hi ,

So, is the problem solved (by your findings in the UAS
side) ?

Regards,

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

On 24.08.2015 18:25, Nabeel wrote:

I just discovered that the SIP client logs show an error
message only on the recipient side, not on the caller's
side.  I missed this previously because the caller's
side log does not show any error:

java.lang.Exception: No DNS SRV or A results found
for: 162.242.153.259  (IP address of OpenSIPS server).


I have the SRV records set on the actual
hostname/domain, but it seems to be looking for SRV at
the actual IP address itself.

On 21 August 2015 at 17:57, Nabeel
mailto:nabeelshik...@gmail.com>> wrote:

The log doesn't show any errors when the Timeout
occurs, it only shows this:

opensips[1842]: ACC: call missed:

timestamp=1440174643;method=INVITE;from_tag=z9hG4bK04147190;to_tag=;call_id=424618310389@10.137.181.237
;code=408;reason=Request
Timeout 



This seems to occur sporadically; some calls connect
without problem but others don't; so perhaps it is a
genuine timeout... maybe it simply longer to connect
on some calls?


On 21 August 2015 at 17:46, Nabeel
mailto:nabeelshik...@gmail.com>> wrote:

Sorry to bring this up again, but I still get
the 408 Request Timeout on some calls.

Isn't there just a way to increase the request
timeout limit?

Here is the trace:

http://pastebin.com/jvCPGYDu

There is even an ACK in the trace after the
request timeout message, but the call doesn't
connect.

On 7 August 2015 at 18:10, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

   

Re: [OpenSIPS-Users] 408 Request Timeout with UDP

[Eric Tamme]

http://www.opensips.org/html/docs/modules/devel/rr.html#id293868

On 08/25/2015 10:20 AM, Nabeel wrote:
Please show me an example of where / how to use record_route_preset() 
to add the FQDN.


On 25 August 2015 at 16:54, Bogdan-Andrei Iancu > wrote:


Hi,

According to the RFC, in RR header can be IP or FQDN (any kind of
SIP URI). Even more, the best practice is to actually use IPs in
RR to be 100% sure that the following requests to hit exactly the
same box (if using FQDN, subject to DNS resolving, a different IP
may be lookup up later).

If you really want to put an IP there, use the
record_route_preset() function:
http://www.opensips.org/html/docs/modules/1.11.x/rr.html#id293864

Regards,

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

On 25.08.2015 16:47, Nabeel wrote:

Currently, OpenSIPS is using the actual IP address in the
record-route URI, but I believe my SIP client needs the domain
name in the record-route instead.


For example, it should be:

Record-Route: http://sipdomain.com>;lr;nat=yes;did=29.3daff1f4>


NOT:

Record-Route: 



How can I make this change in the OpenSIPS config?

This should solve the problem because in a working setup
(different SIP server), the logs state /"Resolving host address
'sipdomain.com '"/ and the record route URI
includes the domain name, but in the OpenSIPS setup the logs
state /"Resolving host address '162.242.153.259'/ and the record
route URI contains the IP address.


On 24 August 2015 at 18:37, Nabeel mailto:nabeelshik...@gmail.com>> wrote:

Hi,

I see the cause now on the UAC side; I know it seems simple
to just add some DNS records to the server IP,  but I'm still
pondering on the best way to solve this and where exactly to
add the SRV records because:

1) I already have the SRV records set up on the actual
hostname / domain, hosted by a DNS service third party, which
is easier for me to maintain.  However the UAC seems to be
ignoring this.

2) I have used the same UAC with another server and did not
have to set up SRV on the actual server machine IP.

I'm not sure if this has anything to do with the OpenSIPS
config but I'll let you know if I solve it.

On 24 Aug 2015 17:56, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org>> wrote:

Hi ,

So, is the problem solved (by your findings in the UAS
side) ?

Regards,

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

On 24.08.2015 18:25, Nabeel wrote:

I just discovered that the SIP client logs show an error
message only on the recipient side, not on the caller's
side.  I missed this previously because the caller's
side log does not show any error:

java.lang.Exception: No DNS SRV or A results found
for: 162.242.153.259  (IP address of OpenSIPS server).


I have the SRV records set on the actual
hostname/domain, but it seems to be looking for SRV at
the actual IP address itself.

On 21 August 2015 at 17:57, Nabeel
mailto:nabeelshik...@gmail.com>> wrote:

The log doesn't show any errors when the Timeout
occurs, it only shows this:

opensips[1842]: ACC: call missed:

timestamp=1440174643;method=INVITE;from_tag=z9hG4bK04147190;to_tag=;call_id=424618310389@10.137.181.237
;code=408;reason=Request
Timeout 



This seems to occur sporadically; some calls connect
without problem but others don't; so perhaps it is a
genuine timeout... maybe it simply longer to connect
on some calls?


On 21 August 2015 at 17:46, Nabeel
mailto:nabeelshik...@gmail.com>> wrote:

Sorry to bring this up again, but I still get
the 408 Request Timeout on some calls.

Isn't there just a way to increase the request
timeout limit?

Here is the trace:

http://pastebin.com/jvCPGYDu

There is even an ACK in the trace after the
request timeout message, but the call doesn't
connect.

On 7 August 2015 at 18:10, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Indeed,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
  

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now available

[Eric Tamme]

Loading protocol modules is required,  It is not temporary.  As long as 
it is covered in the migration documentation I think that is reasonable.


I have updated the migration documentation to make it clearer that you 
must load proto_udp if you want to use UDP listeners.


see: http://www.opensips.org/Documentation/Migration-1-11-0-to-2-1-0

-Eric

On 08/24/2015 01:02 PM, Maxim Sobolev wrote:
No, we have not loaded it yet. Is it now always required? As I said my 
point here is not so much how to fix it, but the fact that Liviu 
Chircu said that loading such module is just a workaround and the 
proper fix would be applied before the 2.1 x release goes out. If it 
was decided that loading module is now the "official" way to go, then 
it should be reflected in the relnotes IMHO.


-Maxim

On Sat, Aug 22, 2015 at 11:18 AM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Maxim,

Do you load the proto_udp module ?

Regards,
Bogdan


Sent from Samsung Mobile


 Original message 
From: Maxim Sobolev
Date:22/08/2015 18:48 (GMT+02:00)
To: OpenSIPS devel mailling list
Cc: n...@lists.opensips.org
,users@lists.opensips.org
, busin...@lists.opensips.org

Subject: Re: [OpenSIPS-Devel] [RELEASE] OpenSIPS 2.1.1 is now
available

Hi Bogdan,

For some reason 2.1.x is still failing our voiptests travis run
with the following error when trying to run in the UDP-only mode:

Aug 22 15:32:00 [10854] ERROR:core:fix_all_socket_lists: listeners
found for protocol udp, but no module can handle it

Aug 22 15:32:00 [10854] ERROR:core:main: failed to initialize list
addresses

It was told on the mailing list before that it would be fixed
before the release:


http://opensips-open-sip-server.1449251.n2.nabble.com/Unable-to-start-opensips-2-1-1dev-in-UDP-only-mode-td7595563.html

So I guess that never happened. Could you guys look into it or at
least add some kind of errata or relnotes entry?

Thanks!

-Maxim


On Wed, Aug 19, 2015 at 11:03 AM, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hello everyone,

Minor version 2.1.1 is now available on branch 2.1. This is a
release bringing multiple and valuable fixes, a result of the
continues work of testing and fixing the revolutionary 2.1
version.

Please update as soon as possible as it worth it ! Download
the tarball with sources from :
http://opensips.org/pub/opensips/2.1.1/

RPM and DEB packages will be shortly available on the official
repositories, after the nightly builts.

There are hundreds of reports, tens of fixes and maybe several
hundreds of commits - all these are the result of the entire
OpenSIPS community - people testing, reporting and fixes. And
I want to thanks to all these people, to these OpenSIPS'ers !

Enjoy 2.1.1 !!

-- 
Bogdan-Andrei Iancu

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


___
Devel mailing list
de...@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/devel




-- 
Maksym Sobolyev

Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474 
Tel (Toll-Free): +1-855-747-7779 
Fax: +1-866-857-6942 
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com 
Skype: SippySoft




--
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
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