date:20150824

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

2015-08-24 Thread surya

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

2015-08-24 Thread Nabeel

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
>
>


>>>
>>
>
>
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> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] Protocol/port mismatch warning (TLS - UDP)

2015-08-24 Thread Bogdan-Andrei Iancu


Matt,

Try to do a:
 subst_uri('/(sip:.*);transport=tls/\1/')
before t_relay().

It requires the URI module.

Regards,

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

On 24.08.2015 19:16, Matt Hamilton wrote:


Hi Bogdan,


UAC RURI doesn't have SIPS schema; it's regular SIP with transport=tls.


UAC(TLS)->  Opensips(force UDP)-> Asterisk


In spite of the warning message, the traffic flows fine, and Asterisk 
receives the packets unencrypted (as we want). To get rid of that 
warning, is is possible to do a protocol/port exchange from TLS(5061) 
to UDP(5060) in this specific case?


Thanks,
Matt



*From:* Bogdan-Andrei Iancu 
*Sent:* Monday, August 24, 2015 5:41 AM
*To:* OpenSIPS users mailling list; mistral9...@hotmail.com
*Subject:* Re: [OpenSIPS-Users] Protocol/port mismatch warning (TLS - 
UDP)

Hi Matt,

The RURI you receive from the UAC side, does it contain a SIPS schema 
or a "transport=tls" indication ?


The error you see translates into "there is a conflict between the 
requests protocol and the interface you force". So even if you force 
the UDP interface, maybe the RURI may force the TLS protocol, leading 
to an internal conflict.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 23.08.2015 19:53, Matt Hamilton wrote:



We use Opensips (1.7.1, in the process of upgrading to 2.1) as a 
dispatcher to multiple Asterisk servers. Currently we are trying to 
set up TLS.




UAC -> Opensips -> Asterisk


TLS is available and enabled on all three. The traffic between UAC 
and Opensips is on TLS and encrypted. On the other hand, it's not 
encrypted between Opensips and Asterisk since we use 
force_send_socket (UDP) to the Asterisk servers. This is the way we 
want it - encrypted between UAC and Opensips and not encrypted 
between Opensips and Asterisk. The communication is fine, but we get 
a warning "core:get_send_socket: protocol/port mismatch" in our logs.



My question is: Is forcing TLS traffic thru UDP is the only way to 
get Opensips - Asterisk traffic is unencrypted?



Thanks,

Matt



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

2015-08-24 Thread Bogdan-Andrei Iancu


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 > 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
OpenSIPS Founder and Developer
http://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









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Re: [OpenSIPS-Users] fifo t_uac_dlg syntax

2015-08-24 Thread Bogdan-Andrei Iancu


Hi Ralf,

Try this magic line (also thanks to Liviu for his shell expertise):

opensipsctl fifo t_uac_dlg NOTIFY "sip:3003@193.144.1.112" . . "\"From: 
sip:3001@192.180.1.107\r\nTo: sip:3003@193.144.1.112\r\nContent-Type: 
application/simservs+xml\r\n\"" '"encoding=\\"UTF-8\\"?>   
"'


Regards,

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

On 21.08.2015 20:57, Ralf_Neumann_hgw wrote:

Yes, it works until the character <=> because it does not contains the <">

The body should looks like...


-Ursprüngliche Nachricht-
Von: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 
Im Auftrag von Bogdan-Andrei Iancu
Gesendet: Freitag, 21. August 2015 16:36
An: OpenSIPS users mailling list; ralf_neumann_...@web.de
Betreff: Re: [OpenSIPS-Users] fifo t_uac_dlg syntax

Ralf,

You have a trailing <"> at the end of the command:

try
/usr/local/opensips_proxy/sbin/opensipsctl fifo t_uac_dlg NOTIFY "sip:3003@193.144.1.112" 
. . "\"From: sip:3001@192.180.1.107\r\nTo:
sip:3003@193.144.1.112\r\nContent-Type: application/simservs+xml\r\n"\"
'http://www.opensips-solutions.com

On 21.08.2015 16:36, Ralf Neumann wrote:

/usr/local/opensips_proxy/sbin/opensipsctl fifo t_uac_dlg NOTIFY "sip:3003@193.144.1.112" . . 
"\"From: sip:3001@192.180.1.107\r\nTo: sip:3003@193.144.1.112\r\nContent-Type: 
application/simservs+xml\r\n"\" '
An: "OpenSIPS users mailling list" ,
ralf_neumann_...@web.de
Betreff: Re: [OpenSIPS-Users] fifo t_uac_dlg syntax Hi Ralf,

Could you post the exact command (from shell) you send together with
the opensips error log please.

Regards,

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

On 21.08.2015 09:59, Ralf Neumann wrote:

not caused by the syntax. The command accept the following syntax:
'
An: "OpenSIPS users mailling list" 
Betreff: Re: [OpenSIPS-Users] fifo t_uac_dlg syntax are u sure about
the syntax errors not caused by your shell?

On Aug 20, 2015 11:10 AM, "Ralf_Neumann_hgw"
 wrote:Hi Bogdan,

the NOTIFY is used to switch on/off different services
(warmline/hotline) on the SIP client.
I get syntax errors if i use the characters like <;>;";?

What is your recommendation?

Ralf

-Ursprüngliche Nachricht-
Von:
users-boun...@lists.opensips.org[users-boun...@lists.opensips.org]
[mailto:users-boun...@lists.opensips.org[users-bounces@lists.opensips
.org]] Im Auftrag von Bogdan-Andrei Iancu
Gesendet: Donnerstag, 20. August 2015 16:45
An: OpenSIPS users mailling list;
ralf_neumann_...@web.de[ralf_neumann_...@web.de]
Betreff: Re: [OpenSIPS-Users] fifo t_uac_dlg syntax

Hi Ralf,

Body can be anything. The question is what you want to use the NOTIFY for ?

Regards,

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

On 20.08.2015 16:01, Ralf Neumann wrote:

The command "opensipsctl fifo t_uac_dlg ..." send the NOTIFY to the
client

as expected. I need support regarding the implementation of the body
in the NOTIFY packet based on xml. Is here anybody with knowledge of
the needed syntax or workaround.

opensipsctl fifo t_uac_dlg NOTIFY 
"sip:3003@193.144.1.112[sip%3A3003@193.144.1.112]" . .
"\"From: sip:3001@192.180.1.107[sip%3A3001@192.180.1.107]\r\nTo:
sip:3003@193.144.1.112[sip%3A3003@193.144.1.112]\r\nContent-Type:
application/simservs+xml\r\n"\" thebody

Body of SIP NOTIFY:
  
 

Thanks
Ralf






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

2015-08-24 Thread Nabeel

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

>>>
>>>
>>
>
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[OpenSIPS-Users] OpenSIPS Workshop 2015 in Chicago - Post Facts

2015-08-24 Thread Vlad Paiu


Hi everyone !

Following the requests of people who were not able to attend, we put 
together a nice collection of presentations and videos - all are shared 
here:


http://www.opensips.org/Community/Workshop-2015Chicago

Enjoy !
Thanks again to our sponsors, to the people who helped organize the 
event and, of course, to our beautiful audience !


Best regards,

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread dpa

Ooops and I was very happy some minutes ago:)))

Ok, will wait an options.

 

Thank you!

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, August 24, 2015 12:58 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hold on ! none of the options are available now . I suggested them as
options for a fix ;)

Regards,



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

On 24.08.2015 12:55, dpa wrote:

Second point seems interesting. I will test it.

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, August 24, 2015 12:49 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

Currently you cannot do this, but I see 2 options:

1) allow a carrier definition to contain other carriers (max 3 levels)
2) have a more complex syntax in destination set definition, with brackets,
like: 
#14,(#34=50,#35=50),#2881

Regards,




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

On 24.08.2015 10:42, dpa wrote:

Hello Bogdan!

 

Yes, you are right, I want use only carriers in routing.

 

For example.

I have for providers

#14, #34, #35, #2881.

 

Let`s  say that the goal of some prefix is

Prefix "9087": #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$

Prefix "8765": #34-1$, #35-2$, #14-3$, #2881 - 3$

 

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

 

First Opensips make serial routing and then parallel.

 

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms
routing table) and using it during forming routing table. 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 4:35 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

So what you want to do is to be able to serial and parallel (combined)
having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use carriers in the
definition of another carrier (when doing the parallel group), right ?

Regards,





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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted "do a combination between serial and parallel selection" but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for "sharing"
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

 

Thank you.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,






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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can "sharing" different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set "14,#X" (and do not use the W flag
anymore)

Regards,







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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list "#14,#34=50,#35=50,#2881"

Example 2.

I have such gw_list "#14,#34,#35,#2881"

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread Bogdan-Andrei Iancu

Hold on ! none of the options are available now . I suggested them as 
options for a fix ;)


Regards,

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

On 24.08.2015 12:55, dpa wrote:


Second point seems interesting. I will test it.

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Monday, August 24, 2015 12:49 PM
*To:* dpa; 'OpenSIPS users mailling list'
*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Hi Denis,

Currently you cannot do this, but I see 2 options:

1) allow a carrier definition to contain other carriers (max 3 levels)
2) have a more complex syntax in destination set definition, with 
brackets, like:

#14,(#34=50,#35=50),#2881

Regards,

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

On 24.08.2015 10:42, dpa wrote:

Hello Bogdan!

Yes, you are right, I want use only carriers in routing.

For example.

I have for providers

#14, #34, #35, #2881.

Let`s  say that the goal of some prefix is

Prefix “9087”: #14 – 1$, #34-2$, #35 – 2$, #2881 – 3$

Prefix “8765”: #34-1$, #35-2$, #14-3$, #2881 – 3$

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

First Opensips make serial routing and then parallel.

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which
forms routing table) and using it during forming routing table.

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Friday, August 21, 2015 4:35 PM
*To:* dpa; 'OpenSIPS users mailling list'
*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Denis,

So what you want to do is to be able to serial and parallel
(combined) having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use
carriers in the definition of another carrier (when doing the
parallel group), right ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted “do a combination between serial and parallel
selection” but method you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for
“sharing” doesn`t convenient, because I must have all possible
combination of carriers and using (remember) it for making
routing table (contents of routing table formed by another
system, and I do not want complicate it).

Thank you.

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Friday, August 21, 2015 3:52 PM
*To:* dpa; 'OpenSIPS users mailling list'
*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Denis,

I had the impression that I haven't managed to answer to your
question. Maybe because I do not fully understand the
question. Could you detail why the solution I gave does not
fit you?

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

I understand and I thought about it before. But it doesn`t
fit to me, because I have many prefix which can “sharing”
different carriers.

Anyway, thank you for help.

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Friday, August 21, 2015 3:29 PM
*To:* OpenSIPS users mailling list; denis7...@mail.ru

*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Hi Denis,

If you use weights alg and the GW has no weight, a 0
weight will be assumed, so it will never match (unless
after a failover).

Using weights, the selection is over all the GW in the set.

If you want do a combination between serial and parallel
selection , you need to use carriers:

Make carrier X with wights flags and set for it
destinations 34=50,35=50
and in the rule have the destination set "14,#X" (and do
not use the W flag anymore)

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 21.08.2015 12:16, dpa wrote:

Hello!

There is one question about dr_routing module.

Example 1.

I have such gw_list “#14,#34=50,#35=50,#2881”

Example 2.

I have such gw_list “#14,#34,#35,#2881”

Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread dpa

Second point seems interesting. I will test it.

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Monday, August 24, 2015 12:49 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

Hi Denis,

Currently you cannot do this, but I see 2 options:

1) allow a carrier definition to contain other carriers (max 3 levels)
2) have a more complex syntax in destination set definition, with brackets,
like: 
#14,(#34=50,#35=50),#2881

Regards,

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

On 24.08.2015 10:42, dpa wrote:

Hello Bogdan!

Yes, you are right, I want use only carriers in routing.

For example.

I have for providers

#14, #34, #35, #2881.

Let`s  say that the goal of some prefix is

Prefix "9087": #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$

Prefix "8765": #34-1$, #35-2$, #14-3$, #2881 - 3$

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

First Opensips make serial routing and then parallel.

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms
routing table) and using it during forming routing table. 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 4:35 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

Denis,

So what you want to do is to be able to serial and parallel (combined)
having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use carriers in the
definition of another carrier (when doing the parallel group), right ?

Regards,

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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted "do a combination between serial and parallel selection" but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for "sharing"
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

Thank you.

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,

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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can "sharing" different carriers.

Anyway, thank you for help.

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set "14,#X" (and do not use the W flag
anymore)

Regards,

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

On 21.08.2015 12:16, dpa wrote:

Hello!

There is one question about dr_routing module.

Example 1.

I have such gw_list "#14,#34=50,#35=50,#2881"

Example 2.

I have such gw_list "#14,#34,#35,#2881"

I am using d_routing() with W flag.

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

Thank you for any help.

___
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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

Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread Bogdan-Andrei Iancu


Hi Denis,

Currently you cannot do this, but I see 2 options:

1) allow a carrier definition to contain other carriers (max 3 levels)
2) have a more complex syntax in destination set definition, with 
brackets, like:

#14,(#34=50,#35=50),#2881

Regards,

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

On 24.08.2015 10:42, dpa wrote:


Hello Bogdan!

Yes, you are right, I want use only carriers in routing.

For example.

I have for providers

#14, #34, #35, #2881.

Let`s  say that the goal of some prefix is

Prefix “9087”: #14 – 1$, #34-2$, #35 – 2$, #2881 – 3$

Prefix “8765”: #34-1$, #35-2$, #14-3$, #2881 – 3$

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

First Opensips make serial routing and then parallel.

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms 
routing table) and using it during forming routing table.


*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Friday, August 21, 2015 4:35 PM
*To:* dpa; 'OpenSIPS users mailling list'
*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Denis,

So what you want to do is to be able to serial and parallel (combined) 
having carriers as elements in the list (and not GWs).


With the solution I gave you cannot do it as you cannot use carriers 
in the definition of another carrier (when doing the parallel group), 
right ?


Regards,

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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted “do a combination between serial and parallel selection”
but method you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for
“sharing” doesn`t convenient, because I must have all possible
combination of carriers and using (remember) it for making routing
table (contents of routing table formed by another system, and I
do not want complicate it).

Thank you.

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Friday, August 21, 2015 3:52 PM
*To:* dpa; 'OpenSIPS users mailling list'
*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Denis,

I had the impression that I haven't managed to answer to your
question. Maybe because I do not fully understand the question.
Could you detail why the solution I gave does not fit you?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

I understand and I thought about it before. But it doesn`t fit
to me, because I have many prefix which can “sharing”
different carriers.

Anyway, thank you for help.

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Friday, August 21, 2015 3:29 PM
*To:* OpenSIPS users mailling list; denis7...@mail.ru

*Subject:* Re: [OpenSIPS-Users] opensips1.11 dr_routing

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight
will be assumed, so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set.

If you want do a combination between serial and parallel
selection , you need to use carriers:

Make carrier X with wights flags and set for it destinations
34=50,35=50
and in the rule have the destination set "14,#X" (and do not
use the W flag anymore)

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com

On 21.08.2015 12:16, dpa wrote:

Hello!

There is one question about dr_routing module.

Example 1.

I have such gw_list “#14,#34=50,#35=50,#2881”

Example 2.

I have such gw_list “#14,#34,#35,#2881”

I am using d_routing() with W flag.

In Example 1 a call first had been sent to sharing between
#34 and #35 and only then to #14.

In Example 2 a call first had been sent to #14, then #34,
then #35 and #2881.

Is there any way to tell Opensips in Example 1 first sends
calls to #14 and only then to sharing between #34 and #35?

Thank you for any help.






___

Users mailing list

Users@lists.opensips.org  

http://lists.opensips.org/cgi-bin/mailman/listinfo/users



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

Re: [OpenSIPS-Users] Protocol/port mismatch warning (TLS - UDP)

2015-08-24 Thread Bogdan-Andrei Iancu


Hi Matt,

The RURI you receive from the UAC side, does it contain a SIPS schema or 
a "transport=tls" indication ?


The error you see translates into "there is a conflict between the 
requests protocol and the interface you force". So even if you force the 
UDP interface, maybe the RURI may force the TLS protocol, leading to an 
internal conflict.


Regards,

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

On 23.08.2015 19:53, Matt Hamilton wrote:



We use Opensips (1.7.1, in the process of upgrading to 2.1) as a 
dispatcher to multiple Asterisk servers.  Currently we are trying to 
set up TLS.




UAC -> Opensips -> Asterisk


TLS is available and enabled on all three. The traffic between UAC and 
Opensips is on TLS and encrypted. On the other hand, it's not 
encrypted between Opensips and Asterisk since we use force_send_socket 
(UDP) to the Asterisk servers. This is the way we want it - encrypted 
between UAC and Opensips and not encrypted between Opensips and 
Asterisk. The communication is fine, but we get a warning 
"core:get_send_socket: protocol/port mismatch" in our logs.



My question is: Is forcing TLS traffic thru UDP is the only way to get 
Opensips - Asterisk traffic is unencrypted?



Thanks,

Matt



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Re: [OpenSIPS-Users] Opensips TLS

2015-08-24 Thread Bogdan-Andrei Iancu


Hi Matt,

Yes, you can do that. OpenSIPS can do protocol exchange so it can switch 
from TLS to UDP. I would recommand to use 1.11 as 1.7 is outdates and 
not maintain.


And yes, the TLS module had a lot of fixes in the last years, not to 
mentioned the TCP stack (TLS relies on it!).



Regards,

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

On 22.08.2015 02:34, Matt Hamilton wrote:



We are using Opensips as a dispatcher to Asterisk servers. The call 
flow for incoming calls to UAC is below (outbound is reversed).



telco -> opensips1 -> asterisk -> opensips1 -> UAC (SIP phones)


We are at the planning stages of implementing TLS. Asterisk (1.8.x), 
Opensips (1.7.1 TLS) and the phones are TLS-capable. Is it possible to 
have TLS just between Opensips and the phones, and not touch the 
traffic between Asterisk and Opensips? If TLS on Asterisk is not 
enabled, will traffic flow between Opensips and Asterisk 
(unencrypted)? Both Opensips server and Asterisk servers are at the 
same location, so it's not really necessary to secure that leg.



Also, do you recommend upgrading Opensips to 2.1 first and then 
enabling TLS? I know our version (1.7.1) is pretty old, but it's been 
very stable for us without TLS. I'm wondering if the TLS module has 
improved (performance, etc.) since then? (We will do the upgrade this 
year - just trying to time it).



Thanks,
Matt



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Re: [OpenSIPS-Users] opensips1.11 dr_routing

2015-08-24 Thread dpa

Hello Bogdan!

 

Yes, you are right, I want use only carriers in routing.

 

For example.

I have for providers

#14, #34, #35, #2881.

 

Let`s  say that the goal of some prefix is

Prefix "9087": #14 - 1$, #34-2$, #35 - 2$, #2881 - 3$

Prefix "8765": #34-1$, #35-2$, #14-3$, #2881 - 3$

 

In ideal I want to see in routing table:

9087 #14,#34=50,#35=50,#2881

8765 #34,#35,#14=50,#2881=50

 

First Opensips make serial routing and then parallel.

 

In method you gave I must create carriers:

34=50, 35=50

34=33, 35=33, 2881=33 an so on.

And all these carriers I must remember in another system (which forms
routing table) and using it during forming routing table. 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 4:35 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

So what you want to do is to be able to serial and parallel (combined)
having carriers as elements in the list (and not GWs).

With the solution I gave you cannot do it as you cannot use carriers in the
definition of another carrier (when doing the parallel group), right ?

Regards,



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

On 21.08.2015 16:14, dpa wrote:

No, Bogdan, everything fine. You wright understand my question.

I wanted "do a combination between serial and parallel selection" but method
you suggest doesn`t fit to me.

The main reason I have many providers and using carrier for "sharing"
doesn`t convenient, because I must have all possible combination of carriers
and using (remember) it for making routing table (contents of routing table
formed by another system, and I do not want complicate it).

 

Thank you.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:52 PM
To: dpa; 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Denis,

I had the impression that I haven't managed to answer to your question.
Maybe because I do not fully understand the question. Could you detail why
the solution I gave does not fit you?

Regards,




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

On 21.08.2015 15:35, dpa wrote:

Hello Bogdan!

 

I understand and I thought about it before. But it doesn`t fit to me,
because I have many prefix which can "sharing" different carriers.

Anyway, thank you for help.

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Friday, August 21, 2015 3:29 PM
To: OpenSIPS users mailling list; denis7...@mail.ru
Subject: Re: [OpenSIPS-Users] opensips1.11 dr_routing

 

Hi Denis,

If you use weights alg and the GW has no weight, a 0 weight will be assumed,
so it will never match (unless after a failover).

Using weights, the selection is over all the GW in the set. 

If you want do a combination between serial and parallel selection , you
need to use carriers:

Make carrier X with wights flags and set for it destinations 34=50,35=50
and in the rule have the destination set "14,#X" (and do not use the W flag
anymore)

Regards,





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

On 21.08.2015 12:16, dpa wrote:



Hello!

 

There is one question about dr_routing module.

 

Example 1.

I have such gw_list "#14,#34=50,#35=50,#2881"

Example 2.

I have such gw_list "#14,#34,#35,#2881"

 

I am using d_routing() with W flag.

 

In Example 1 a call first had been sent to sharing between #34 and #35 and
only then to #14.

In Example 2 a call first had been sent to #14, then #34, then #35 and
#2881.

 

Is there any way to tell Opensips in Example 1 first sends calls to #14 and
only then to sharing between #34 and #35?

 

Thank you for any help.

 








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

 

 

 

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