Re: [OpenSIPS-Users] OpenSIPS2.4 HA and Dialog Replication

[Steve Kumar]

Thanks, it worked.
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Re: [OpenSIPS-Users] OpenSIPS2.4 HA and Dialog Replication

[Ben Newlin]

You need to allow binding to non-local sockets on each machine.

# echo 1 > /proc/sys/net/ipv4/ip_nonlocal_bind

Ben Newlin

From: Users  on behalf of Steve Kumar 

Reply-To: OpenSIPS users mailling list 
Date: Wednesday, August 21, 2019 at 8:12 PM
To: "users@lists.opensips.org" 
Subject: [OpenSIPS-Users] OpenSIPS2.4 HA and Dialog Replication

Hi,
Need some help from smart folks.

I have setup a HA between 2 nodes. So this is the current setup  and these are 
the problems I am getting ->


So I have installed keepalived and only node has floating IP at one time.
I have enabled clusterer module and both nodes can see each other and ping each 
other.
And I have also setup the dialog replication

Problems ->
Where I define my listeners in opensips script I define first Virtual IP listen 
and then local IP. But other opensips node doesn't have VIP so it's opensips 
fails when I start the service, you know understand what I am saying.
This are the listeners on first active node ->
# Proto Bin Listeners
listen = bin:10.10.10.236:

# Virtual / HA IP
listen=udp:10.10.10.111:5060 as X.X.X.X:5060
listen=tcp:10.10.10.111:5060 as X.X.X.X:5060

# Administrative IP
listen=udp:10.10.10.236:5060 as X.X.X.X.:5060
listen=tcp:10.10.10.236:5060 as X.X.X.X:5060

You can see this node has VIP -> 10.10.10.111

But other backup node in HA doesn't have VIP so opensips service was failing to 
start, so what I did I removed that VIP listener from other node and I started 
the openSIPS service.But now I am getting problem in dialog replication because 
other node is not accepting dialogs because active node is sending dialog from 
VIP 10.10.10.111 and other node is not accepting. I am getting this error.

ERROR:dialog:dlg_replicated_created dialog doesn't match caller's listening 
socket udp:10.10.10.111:5060
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[OpenSIPS-Users] OpenSIPS2.4 HA and Dialog Replication

[Steve Kumar]

Hi,
Need some help from smart folks.

I have setup a HA between 2 nodes. So this is the current setup  and these
are the problems I am getting ->


So I have installed keepalived and only node has floating IP at one time.
I have enabled clusterer module and both nodes can see each other and ping
each other.
And I have also setup the dialog replication

Problems ->
Where I define my listeners in opensips script I define first Virtual IP
listen and then local IP. But other opensips node doesn't have VIP so it's
opensips fails when I start the service, you know understand what I am
saying.
This are the listeners on first active node ->
# Proto Bin Listeners
listen = bin:10.10.10.236:

# Virtual / HA IP
listen=udp:10.10.10.111:5060 as X.X.X.X:5060
listen=tcp:10.10.10.111:5060 as X.X.X.X:5060

# Administrative IP
listen=udp:10.10.10.236:5060 as X.X.X.X.:5060
listen=tcp:10.10.10.236:5060 as X.X.X.X:5060

You can see this node has VIP -> 10.10.10.111

But other backup node in HA doesn't have VIP so opensips service was
failing to start, so what I did I removed that VIP listener from other node
and I started the openSIPS service.But now I am getting problem in dialog
replication because other node is not accepting dialogs because active node
is sending dialog from VIP 10.10.10.111 and other node is not accepting. I
am getting this error.

ERROR:dialog:dlg_replicated_created dialog doesn't match caller's listening
socket udp:10.10.10.111:5060
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Re: [OpenSIPS-Users] opensips_cp 3.0

[johan]

Hi Bogdan,

netstat -tulpn on opensipsbox:



tcp    0  0 0.0.0.0: 0.0.0.0:*   LISTEN  
8881/opensips


...

opensips.cfg

 HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", )
 MI_HTTP module
loadmodule "json.so"
loadmodule "mi_http.so"


boxes.global.inc.php

// each server is a box

$box_id=0;

//  MI connector (via JSON backend):   json:host:port/json
$boxes[$box_id]['mi']['conn']="json:127.0.0.1:/JSON";


Is there something that I overlook ?


On 21.08.19 17:59, Bogdan-Andrei Iancu wrote:

Hi,

I suspect your MI connector is not properly configured in CP - check 
if it matches the one offered by MI_HTTP in opensips (also opensips 
must run)


Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Summit 2019
   https://www.opensips.org/events/Summit-2019Amsterdam/
On 08/21/2019 06:34 PM, johan wrote:


Hi,


I installed opensips-cp 3.1.

Both uac_registrant and smpp work (thanks for that).

However, when I need to reload something there is no reply coming 
back (see below screenshot from dr_reload).


Is there something that I need to change in my opensips.cfg to make 
this working ?



Best regards,





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Re: [OpenSIPS-Users] OPENSIPS / Unicode (UTF-8) support for proto_smpp

[Bogdan-Andrei Iancu]

Hi,

I think what you need is UTF-16, not UTF-8.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 08/20/2019 11:14 AM, MAJID ENJOO wrote:


Hello,

We installed Opensips 3.0-3 on Fedora 29 and configure it as SMPP 
gateway and it working well only in Latin characters and it cannot 
support UTF-8. I would appreciate if you can help us to add/enable 
UTF-8 on Opensips 3.0-3 Proto SMPP to able handle multi language 
support and special characters.


On proto_smpp send/receive in English characters and numbers working 
well. But when we try to send/receive in deferent language characters 
does not works. Please see following test results. We need to 
activate/enable UTF-8 on proto_smpp to send/receive multi language 
characters.


Inbound

>> Sent in French>> Café >>Received>> Caf

>> Sent in Persian>> سلام >>Received>> 3D’E

>> Sent in Russian>> Привет >>Received>> @825B

>> Sent non alphabet character>> $ >>Received>> 2

>> Sent non alphabet character>> @ >>Received>> 2

outbound

>>Sent in French>> Café >>Received>> Caf

>>Sent in Persian>> سلام >>Received nothing>> “nothing/empty”

>>Sent in Russian>> Привет >>Received nothing>> “nothing/empty”

>>Sent non alphabet character>> $ >>Received>> ¤

>> Sent non alphabet character >> @>>Received>> ¡

Thanks,

Majid



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Re: [OpenSIPS-Users] opensips_cp 3.0

[Bogdan-Andrei Iancu]

Hi,

I suspect your MI connector is not properly configured in CP - check if 
it matches the one offered by MI_HTTP in opensips (also opensips must run)


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 08/21/2019 06:34 PM, johan wrote:


Hi,


I installed opensips-cp 3.1.

Both uac_registrant and smpp work (thanks for that).

However, when I need to reload something there is no reply coming back 
(see below screenshot from dr_reload).


Is there something that I need to change in my opensips.cfg to make 
this working ?



Best regards,





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Re: [OpenSIPS-Users] running opensips behind AWS NetworkLoadBalancer

[David Huang via Users]

Will it work with AWS Global Accelerator (anycast) ?

Kind regards,

David Huang  BCM (CS, IT)
Director






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> On 14 Aug 2019, at 11:27 am, Pasan Meemaduma via Users 
>  wrote:
> 
> Hi Guys,
> 
> Does anyone able to get opensips running behind AWS NLB? 
> 
> I tried opensips 2.4 with fedaration topology and things doesn't work well.
> 
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[OpenSIPS-Users] opensips_cp 3.0

[johan]

Hi,


I installed opensips-cp 3.1.

Both uac_registrant and smpp work (thanks for that).

However, when I need to reload something there is no reply coming back 
(see below screenshot from dr_reload).


Is there something that I need to change in my opensips.cfg to make this 
working ?



Best regards,


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[OpenSIPS-Users] OPENSIPS / Unicode (UTF-8) support for proto_smpp

[MAJID ENJOO]

Hello,

We installed Opensips 3.0-3 on Fedora 29 and configure it as SMPP gateway and 
it working well only in Latin characters and it cannot support UTF-8. I would 
appreciate if you can help us to add/enable UTF-8 on Opensips 3.0-3 Proto SMPP 
to able handle multi language support and special characters.

On proto_smpp send/receive in English characters and numbers working well. But 
when we try to send/receive in deferent language characters does not works. 
Please see following test results. We need to activate/enable UTF-8 on 
proto_smpp to send/receive multi language characters.

Inbound
>> Sent in French>> Café >>Received>> Caf
>> Sent in Persian>> سلام >>Received>> 3D’E
>> Sent in Russian>> Привет >>Received>> @825B
>> Sent non alphabet character>> $ >>Received>> 2
>> Sent non alphabet character>> @ >>Received>> 2

outbound
>>Sent in French>> Café >>Received>> Caf
>>Sent in Persian>> سلام >>Received nothing>> “nothing/empty”
>>Sent in Russian>> Привет >>Received nothing>> “nothing/empty”
>>Sent non alphabet character>> $ >>Received>> ¤
>> Sent non alphabet character >> @>>Received>> ¡


Thanks,
Majid
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Re: [OpenSIPS-Users] Opensips integration with asterisk

[John Tuxies]

i have not had any progress with the integration.
My setup consists of:
-Opensips (2.4.2) with Control Panel and IP:192.168.1.250. Created through
the Control Panel users 2500-2509 and registered them to softphones and
they can talk to each other.
-2 Freepbx boxes, no configuration at the time, with IPs: 192.168.1.100 &
192.168.1.101 respectively
i am looking how to send the alter the configuration so that users in 2500
can access voicemail in the freepbxes and the rest of the services.
Also looking to register my ITSPs that require username/passwd/domain. Also
they forward numbers in DID mode and i have to assign them to particular
extensions. eg
DID_A to 2500
DID_B to 2501
DID_C to 2502
DID_D to 2503



i would appreciate all the help available. Once successful, i will document
everything and make it available to anyone looking to create something
similar.




On Mon, Aug 19, 2019 at 1:11 PM Alexey Kazantsev via Users <
users@lists.opensips.org> wrote:

> Hi John!
>
> >i am trying for some time now to integrate Opensips with Asterisk,
> > but without success. I have seen the links to the Opensips blog for
> > Asterisk integration, but it is outdated both for Opensips and Opensips.
>
> Well, what confused you in this tutorial? It seems to be what you need:
> https://www.opensips.org/Documentation/Tutorials#toc18
>
> >What i am trying to achieve is a box running Opensips with control panel
> >and another box with Asterisk. The reason for that is to enhance the users
> >with services such as IVR, Voicemail, email to voicemail, faxing,etc
>
> >Up to now i managed to create users in Opensips and register on that.
> >Also they are able to make calls between them.
>
> You've written that you don't see such calls on the Asterisk.
> It means that you don't route such calls from OpenSIPS to Asterisk.
> Check this.
>
> >The numbering plan is 30XX and the port on the system is 5060.
> >Then i have another box with Asterisk that has the port as 55060
> >and the numbering plan is 30XX
>
> Well.. OK, let it be so.
>
> >and every time a user is created in Opensip's CP
> >then i create the same user in Asterisk,
> >eg Opensips 3000(port 5060) and Asterisk 3000(port 55060).
>
> But why?! You don't need this. Create SIP accounts either in OpenSIPS,
> or in Asterisk.
>
> >Then on the Asterisk box i made the following:
> >Created a trunk to Opensips
> >
> >[Opensips]
> >type=peer
> >host=192.168.1.113
> >context=from-opensips
> >insecure=port,invite
> >disallow=all
> >allow=alaw, g729, g722, ulaw
> >deny= 0.0.0.0/0.0.0.0
> >permit= 192.168.1.113/255.255.255.255
> >
> >
> >The problem is that i cannot see the call in Asterisk's terminal when 2
> users call each other.
>
> As I already wrote, it means that the call does not leave OpenSIPS.
> I guess you don't route it to Asterisk with smth like this:
>
>...
>t_relay(x.x.x.x);   # Asterisk's IP
>...
>
> >Also , i have a couple of ITSPs in Asterisk that require username/passwd
> and thet have a FQDN.
>
> If you'd like to use OpenSIPS as the front-end, you'd better connect to
> ITSPs also from OpenSIPS.
> In case of SIP-registration,
> use UAC_REGISTRANT
> https://opensips.org/html/docs/modules/3.0.x/uac_registrant.html module.
> In case of SIP trunks just be able to receive SIP traffic from them and
> control access to your OpenSIPS
> public IP via iptables or PERMISSIONS module
> https://opensips.org/html/docs/modules/3.0.x/permissions.html.
>
> >While in Asterisk registered the user can access the first ITSP with the
> following prefixes 0 and 1 respectively.
> >Is there any way to allow the Opensips registered users dial 0 or 1 as
> prefix and place outgoing calls through ITSP 0 or 1, please?
>
> OpenSIPS is _extremely_ flexible.
> This could be achieved in many ways, starting from hardcoding in the
> script (in case of static configuration
> without need of changing it often) and ending with appropriate modules
> using, such as
> DROUTING https://opensips.org/html/docs/modules/3.0.x/drouting.html
>
>
> ---
> BR, Alexey
> http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] Change SDP when re-directed

[Bogdan-Andrei Iancu]

Hi Mark,

I should try to call fix_nated_sdp (for changing a particular sdp line) 
only once.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 08/16/2019 01:35 PM, Mark Farmer wrote:

Good morning

I'm having a problem with SDP info when a call results in 486, 603 
etc. OpenSIPS is running in mhomed mode with 2 interfaces, the phones 
are behind NAT & working nicely but when the call is routed to 
Asterisk, the SDP needs to change such that the internal IP's are used.


rtpproxy_offer etc is used for the initial call so I tried using 2 
calls to fix_nated_sdp to change the c & o parameters but that seems 
to append the new IP to the existing one followed by rewritehostport 
& force_send_socket


How should I correctly handle this?

Many thanks
Mark.



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Re: [OpenSIPS-Users] is_contact_registered - not working as expected

[Bogdan-Andrei Iancu]

Hi Peter,

To doublecheck, do this:

* before calling the is_contact_registered(), use xlog() to print the 
contact you are testing (use $ct).


* before running the test, use the ul_show_contact MI function [1] to 
print the in-memory contacts for the desired AOR.



[1] 
https://opensips.org/html/docs/modules/2.4.x/usrloc.html#mi_ul_show_contact


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 08/16/2019 06:38 PM, Peter Pulham wrote:

Hi,

I am using mid_registrar to save registrations to the location table.

If I use is_registered - everything works as expected. So AOR matching 
is working.


However, when I use is_contact_registered - it returns false.

I have checked the contact header in the request against the contact 
value of the AOR in the db and they match (the request has a display 
name but I am presuming only the uri is checked.


The endpoint is behind NAT, but the contacts match in the db and 
request. Can some explain what is actually checked when using 
is_contact_registered?


Many thanks


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Re: [OpenSIPS-Users] extension data with SIPREC module

[Răzvan Crainea]

But you can still use participants and add some other data to it. If you 
can't do that, unfortunately there's no other way to do it.


Best regards,
Răzvan

On 8/20/19 9:27 PM, Mickael Hubert wrote:

Hi thanks for your answer
But I already use participants (from and to).
I can't change participants data :(
Can I add other part into siprec xml ?

Thanks

Le mar. 20 août 2019 18:07, Răzvan Crainea > a écrit :


Hi, Mickael!

You can create your own participant's xml using the caller and callee
fields. Check Example 1.6 in the siprec documentation[1] to see how you
can build this xml, that will appear in the participant's node.

[1] https://opensips.org/html/docs/modules/3.0.x/siprec.html#idp5579200

Best regards,
Răzvan

On 8/20/19 3:54 PM, Mickael Hubert wrote:
 > Hi all,
 > I use SIPREC module, that works like a charm with SRS drachtio
server ;)
 >
 > In my initial invite (sip only, no siprec), I have User-To-User
header,
 > and I want to copy it into SIPREC xml part, like the participants.
 >
 > Ex:
 > 
 > 00FA081875333AA;encoding=hex
 > 
 >
 > Do you have a way for me to add this sip header in siprec xml part ?
 >
 > Thanks in advance
 >
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-- 
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OpenSIPS Core Developer
http://www.opensips-solutions.com

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