[OpenSIPS-Users] Prefered Media Proxy or RTP Proxy

[abisai matangira]

Hi

Which is the preferred or stable Media proxy for Opensips from below

RTP Proxy

Media Proxy



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Re: [OpenSIPS-Users] Port changes

[Pasan Meemaduma via Users]

Hi Schneur,
do you have force_rport() in your opensips.cfg ? hard to make any other 
comments without looking at your config.
 

On Monday, 12 February 2018, 20:36, Schneur Rosenberg 
 wrote:
 

 I have this interesting scenario, caller sends call to our OpenSIPS
who actsd as a loadbalancer which sends the call to a gateway for
termination and the gateway sets Session-Expires: 1800;refresher=uas,
the caller sends call the call through port 1090 and the rport in the
Via shows rport=1090 and so far everything is ok, but after 15 minutes
the gateway sends a reinvite,  OpenSIPS sends it to the client and the
client sends a OK and OpenSIPS properly sends the OK to the gateway,
the gateway sends a ACK to OpenSIPS, but here the problem starts
OpenSIPS sends the ACK to the port in the Contact header which is not
the same port as the rport, the clients router does not recognize the
packet and blocks it, the client sends multiple OK's to OpenSIPS who
ignores it because he has already moved on, as far as OpenSIPS is
concerned the ACK has already been sent to the next hop, and after
approx 30 seconds the client does not receive the ACK and sends a BYE
and the call terminates.

Why is OpenSIPS changing the port to the Contact port? and what can I
do to fix it?

    
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[OpenSIPS-Users] acc extra_fields param

[Pasan Meemaduma via Users]

 Hi List,
I'm working on getting my opensips 1.10 -> 2.3.3 :)
As db_extra param is not there anymore I tried to use extra_fields param in acc 
module and it appears it can hold string values only. Is it the case ?
I tried to use it to hold an integer value and it comes up as empty string ''. 

Is it a bug or purpose set only to accept string data only ? cause my db column 
has integer type for it. 

following is my config
modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; 
islocal->islocal; callid->\"exe-callid\"; direction->direction")



$acc_extra(islocal) = 1;

When call hangups it failed to insert the cdr record to acc table as islocal 
comes up as ''.
Ex:-ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 PQsendQuery 
Error: ERROR:  invalid input syntax for integer: ""#012LINE 1: 
...sip:xx@xx;transport=UDP','','376ab2...#012   
  ^#012 Query: insert into acc 
(method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration
 ) values 
('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-09
 
17:55:41','sip:xx@xx;transport=UDP','sip:xx@xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09
 17:55:30',4,4298)
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Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

[Bogdan-Andrei Iancu]

Yes, the chain is: carrier -> LB -> B2B -> Asterisk. And when Asterisk 
generates an REFER (for call transfer) , the REFER will be received by 
B2B which will handle it (it will generate a new call leg) without 
changing anything on the leg connecting the b2b to the carrier.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 02/12/2018 04:41 PM, Brian Southworth wrote:


Hi Bogan,

Thanks for the reply, so are you saying the load balancer will send 
the call over to the B2B and then to asterisk ?


Again sorry for my lack of knowledge there is still a lot I don’t 
understand or know.


Regards,

Brian Southworth

Communications Developer

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 12 February 2018 14:23
*To:* Brian Southworth ; OpenSIPS users 
mailling list 
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no 
socket found to match RR [1][XXX.XXX.XXX.XXX:5060]


Hi Brian,

In this case, I guess that the OPenSIPS B2B (handling the REFER) 
should sit between the OpenSIPS LB and Asterisk - again , this is the 
case only if the result of the transfer is a call to an Asterisk box 
too. If the call may be redirected back to a carrier, the OpenSIPS B2B 
should sit in front of the LB.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam

On 02/07/2018 03:38 PM, Brian Southworth wrote:

Opensips handles the refer sending it to the asterisk box

Regards,

Brian Southworth

Communications Developer



111 Wilmslow Road

Handforth

Wilmslow

SK9 3ER

T: 0 446677

W: www.clocom.uk __









__

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 11:36
*To:* Brian Southworth 
; OpenSIPS users mailling list
 
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict:
no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

So the target of the refer is to another Asterisk or may be also
back to the carrier ?



Bogdan-Andrei Iancu

  


OpenSIPS Founder and Developer

   http://www.opensips-solutions.com

OpenSIPS Summit 2018

   http://www.opensips.org/events/Summit-2018Amsterdam

On 02/07/2018 01:32 PM, Brian Southworth wrote:

Hi Bogdan,

The Cisco phone, generates the refer once you press the xfer
button when inside a call.

Caller àopensipsàasteriskàCarrier

(cisco)

Regards,

Brian Southworth

T: 0 446677

W: www.clocom.uk __


















*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 09:38
*To:* Brian Southworth 
; OpenSIPS users mailling
list  
*Subject:* Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: no socket found to match RR
[1][XXX.XXX.XXX.XXX:5060]

Hi Brian,

Which party is generating the REFER ? the asterisk boxes from
behind the LB ? or the carrier side ?

and yes, see you in Amsterdam !!

Regards,



Bogdan-Andrei Iancu

  


OpenSIPS Founder and Developer

   http://www.opensips-solutions.com

OpenSIPS Summit 2018

   http://www.opensips.org/events/Summit-2018Amsterdam

On 02/05/2018 05:52 PM, Brian Southworth wrote:

I think I get it now thank you Bogdan.

So I would forward the traffic using the opensips proxy,
using the if (is_method(“refer”)) to the opensips box that
would be the B2BUA? To bridge the call ?.

Also look forward to Opensips summit in may 😊ill see you
all there got it booked Saturday 😊

Regards,

Brian Southworth



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Re: [OpenSIPS-Users] Database search inside opensips

[Schneur Rosenberg]

You can use avp_db_query


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On Mon, Feb 12, 2018 at 4:48 PM, Brian Southworth
 wrote:
> Hi Opensips,
>
>
>
> Sorry for all my emails lately.
>
>
>
> Is there any way I can run a database search in opensips config while a
> route is being processed.
>
>
>
> Example scenario: Company has over 15 clients each with their own user id
> (lets call this a callgroup ID), so the call group id needs to be changed
> based on the extension calling to match the correct call group.
>
> $rU->accountcode (this is just an example).
>
>
>
> I just thought being able to change this on the fly would be easier than
> writing loads of new call groups into the Even based routing.
>
>
>
> Regards,
>
>
>
> Brian Southworth
>
> Communications Developer
>
>
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> Users@lists.opensips.org
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>

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[OpenSIPS-Users] Port changes

[Schneur Rosenberg]

I have this interesting scenario, caller sends call to our OpenSIPS
who actsd as a loadbalancer which sends the call to a gateway for
termination and the gateway sets Session-Expires: 1800;refresher=uas,
the caller sends call the call through port 1090 and the rport in the
Via shows rport=1090 and so far everything is ok, but after 15 minutes
the gateway sends a reinvite,  OpenSIPS sends it to the client and the
client sends a OK and OpenSIPS properly sends the OK to the gateway,
the gateway sends a ACK to OpenSIPS, but here the problem starts
OpenSIPS sends the ACK to the port in the Contact header which is not
the same port as the rport, the clients router does not recognize the
packet and blocks it, the client sends multiple OK's to OpenSIPS who
ignores it because he has already moved on, as far as OpenSIPS is
concerned the ACK has already been sent to the next hop, and after
approx 30 seconds the client does not receive the ACK and sends a BYE
and the call terminates.

Why is OpenSIPS changing the port to the Contact port? and what can I
do to fix it?


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[OpenSIPS-Users] Database search inside opensips

[Brian Southworth]

Hi Opensips,

 
Sorry for all my emails lately.

 
Is there any way I can run a database search in opensips config while a route 
is being processed.

 
Example scenario: Company has over 15 clients each with their own user id (lets 
call this a callgroup ID), so the call group id needs to be changed based on 
the extension calling to match the correct call group.

$rU->accountcode (this is just an example).

 
I just thought being able to change this on the fly would be easier than 
writing loads of new call groups into the Even based routing.

 
Regards,

 
Brian Southworth

Communications Developer

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Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

[Brian Southworth]

Hi Bogan,

 
Thanks for the reply, so are you saying the load balancer will send the call 
over to the B2B and then to asterisk ?

Again sorry for my lack of knowledge there is still a lot I don’t understand or 
know.

 
Regards,

 
Brian Southworth

Communications Developer

 
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: 12 February 2018 14:23
To: Brian Southworth ; OpenSIPS users mailling list 

Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found 
to match RR [1][XXX.XXX.XXX.XXX:5060]

 
Hi Brian,

In this case, I guess that the OPenSIPS B2B (handling the REFER) should sit 
between the OpenSIPS LB and Asterisk - again , this is the case only if the 
result of the transfer is a call to an Asterisk box too. If the call may be 
redirected back to a carrier, the OpenSIPS B2B should sit in front of the LB.

Regards,



Bogdan-Andrei Iancu


 

OpenSIPS Founder and Developer


  http://www.opensips-solutions.com  


OpenSIPS Summit 2018


  http://www.opensips.org/events/Summit-2018Amsterdam 
 

On 02/07/2018 03:38 PM, Brian Southworth wrote:

Opensips handles the refer sending it to the asterisk box

 
Regards,

 
Brian Southworth

Communications Developer




111 Wilmslow Road

Handforth

Wilmslow

SK9 3ER

 
T: 0 446677

W: www.clocom.uk  


 
 

 

 
 
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org 
 ] 
Sent: 07 February 2018 11:36
To: Brian Southworth  
 ; OpenSIPS users mailling list 
  
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found 
to match RR [1][XXX.XXX.XXX.XXX:5060]

 
So the target of the refer is to another Asterisk or may be also back to the 
carrier ?





Bogdan-Andrei Iancu


 

OpenSIPS Founder and Developer


  http://www.opensips-solutions.com  


OpenSIPS Summit 2018


  http://www.opensips.org/events/Summit-2018Amsterdam 
 

On 02/07/2018 01:32 PM, Brian Southworth wrote:

Hi Bogdan,

 
The Cisco phone, generates the refer once you press the xfer button when inside 
a call.

Caller àopensipsà asteriskàCarrier 

(cisco)

Regards,

 
Brian Southworth

 
T: 0 446677

W: www.clocom.uk  



 

 

 
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org 
 ] 
Sent: 07 February 2018 09:38
To: Brian Southworth  
 ; OpenSIPS users mailling list 
  
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found 
to match RR [1][XXX.XXX.XXX.XXX:5060]

 
Hi Brian,

Which party is generating the REFER ? the asterisk boxes from behind the LB ? 
or the carrier side ?

and yes, see you in Amsterdam !!

Regards,





Bogdan-Andrei Iancu


 

OpenSIPS Founder and Developer


  http://www.opensips-solutions.com  


OpenSIPS Summit 2018


  http://www.opensips.org/events/Summit-2018Amsterdam 
 

On 02/05/2018 05:52 PM, Brian Southworth wrote:

I think I get it now thank you Bogdan.

So I would forward the traffic using the opensips proxy, using the if 
(is_method(“refer”)) to the opensips box that would be the B2BUA? To bridge the 
call ?.

 
Also look forward to Opensips summit in may 😊 ill see you all there got it 
booked Saturday 😊

 
Regards,

 
Brian Southworth

 
 
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Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

[Bogdan-Andrei Iancu]

Hi Brian,

In this case, I guess that the OPenSIPS B2B (handling the REFER) should 
sit between the OpenSIPS LB and Asterisk - again , this is the case only 
if the result of the transfer is a call to an Asterisk box too. If the 
call may be redirected back to a carrier, the OpenSIPS B2B should sit in 
front of the LB.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
OpenSIPS Summit 2018
  http://www.opensips.org/events/Summit-2018Amsterdam

On 02/07/2018 03:38 PM, Brian Southworth wrote:


Opensips handles the refer sending it to the asterisk box

Regards,

Brian Southworth

Communications Developer


111 Wilmslow Road

Handforth

Wilmslow

SK9 3ER

T: 0 446677

W: www.clocom.uk __









__

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 11:36
*To:* Brian Southworth ; OpenSIPS users 
mailling list 
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no 
socket found to match RR [1][XXX.XXX.XXX.XXX:5060]


So the target of the refer is to another Asterisk or may be also back 
to the carrier ?



Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam

On 02/07/2018 01:32 PM, Brian Southworth wrote:

Hi Bogdan,

The Cisco phone, generates the refer once you press the xfer
button when inside a call.

Caller àopensipsàasteriskàCarrier

(cisco)

Regards,

Brian Southworth

T: 0 446677

W: www.clocom.uk __


















__

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* 07 February 2018 09:38
*To:* Brian Southworth 
; OpenSIPS users mailling list
 
*Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict:
no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]

Hi Brian,

Which party is generating the REFER ? the asterisk boxes from
behind the LB ? or the carrier side ?

and yes, see you in Amsterdam !!

Regards,


Bogdan-Andrei Iancu

  


OpenSIPS Founder and Developer

   http://www.opensips-solutions.com

OpenSIPS Summit 2018

   http://www.opensips.org/events/Summit-2018Amsterdam

On 02/05/2018 05:52 PM, Brian Southworth wrote:

I think I get it now thank you Bogdan.

So I would forward the traffic using the opensips proxy, using
the if (is_method(“refer”)) to the opensips box that would be
the B2BUA? To bridge the call ?.

Also look forward to Opensips summit in may 😊ill see you all
there got it booked Saturday 😊

Regards,

Brian Southworth


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