Re: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications

[Callum Guy]

Thinking outside the box here but... maybe update to 3.1.1?

https://www.opensips.org/Documentation/Migration-3-0-0-to-3-1-0

On Wed, 14 Apr 2021 at 17:35, HS  wrote:

> Bogdan-Andrei,
>
> Thanks a lot for sharing these links. I had actually looked at them
> previously, all are for Opensips 3.1 - any possibility of getting PN to
> work with Openips 3.0 please?
>
> Thanks.
>
> On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu 
> wrote:
>
>> See
>>
>>
>> https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/
>>
>>
>> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/
>>
>>
>> https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2021 online
>>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>>
>> On 4/9/21 12:39 PM, HS wrote:
>>
>> Hello all.
>>
>> We are using Opensips 3.0 (with rtpproxy) and been looking to implement
>> Push Notifications for iOS devices. Did come across quite a few resources,
>> but they seem either too old ~2013 time or about V3.1. Has anyone
>> implemented something to make calls to devices with the UAC in background
>> mode on V3.0 pls?
>>
>> Or suggestions on how to make this work on v3.0?
>>
>> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/
>>
>> Appreciate the help.
>>
>> Thanks.
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example

[xaled]

Hi Răzvan,

Thank you for the reply.
Indeed I was able to get the prepaid B2B scenario running with all the needed 
headers once I understood the mechanics of B2B module.

Thank you and the team for the great module that will allow me to implement 
some very interesting scenarios!

Greetings,
Xaled

-Original Message-
From: Users  On Behalf Of Razvan Crainea
Sent: Wednesday, April 14, 2021 3:25 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example

Hi, Xaled!

No, there should bot be an SDP, because you don't have an SDP (the media server 
is gone, and you don't know the callee's SDP). Instead, the module is doing a 
late negotiation to find out advertised caller's SDP, and present it to the new 
callee.

Best regards,

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

On 4/14/21 3:17 PM, xa...@web.de wrote:
> Hi,
> 
> could somebody please shed some light on this scenario?
> 
> Thanks,
> Xaled
> 
> -Original Message-
> From: Users  On Behalf Of 
> xa...@web.de
> Sent: Sunday, April 11, 2021 11:59 PM
> To: 'OpenSIPS users mailling list' 
> Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example
> 
> Hi,
> 
> I tested the 3.2  Prepaid example from B2B tutorial
> 
> https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2
> 
> In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - 
> this reINVITE in my tests comes without SDP. Should not there be an SDP, or 
> am I missing something in this scenario?
> 
> INVITE sip:TEST001@1.2.3.201:5060;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0
> To: ;tag=f79aacce
> From: 
> ;tag=B2B.81.186.1618177460
> CSeq: 3 INVITE
> Call-ID: 80a9a9cfbe9920f8@1.2.3.201
> Max-Forwards: 70
> Content-Length: 0
> Contact: 
> 
> Thanks,
> Xaled
> 
> 
> ___
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> 
> 
> ___
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Re: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications

[HS]

Bogdan-Andrei,

Thanks a lot for sharing these links. I had actually looked at them
previously, all are for Opensips 3.1 - any possibility of getting PN to
work with Openips 3.0 please?

Thanks.

On Wed, Apr 14, 2021 at 3:53 PM Bogdan-Andrei Iancu 
wrote:

> See
>
>
> https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/
>
>
> https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/
>
>
> https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2021 online
>   https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 4/9/21 12:39 PM, HS wrote:
>
> Hello all.
>
> We are using Opensips 3.0 (with rtpproxy) and been looking to implement
> Push Notifications for iOS devices. Did come across quite a few resources,
> but they seem either too old ~2013 time or about V3.1. Has anyone
> implemented something to make calls to devices with the UAC in background
> mode on V3.0 pls?
>
> Or suggestions on how to make this work on v3.0?
>
> http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/
>
> Appreciate the help.
>
> Thanks.
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example

[Răzvan Crainea]

Hi, Xaled!

No, there should bot be an SDP, because you don't have an SDP (the media 
server is gone, and you don't know the callee's SDP). Instead, the 
module is doing a late negotiation to find out advertised caller's SDP, 
and present it to the new callee.


Best regards,

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

On 4/14/21 3:17 PM, xa...@web.de wrote:

Hi,

could somebody please shed some light on this scenario?

Thanks,
Xaled

-Original Message-
From: Users  On Behalf Of xa...@web.de
Sent: Sunday, April 11, 2021 11:59 PM
To: 'OpenSIPS users mailling list' 
Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example

Hi,

I tested the 3.2  Prepaid example from B2B tutorial

https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2

In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this 
reINVITE in my tests comes without SDP. Should not there be an SDP, or am I 
missing something in this scenario?

INVITE sip:TEST001@1.2.3.201:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0
To: ;tag=f79aacce
From: ;tag=B2B.81.186.1618177460
CSeq: 3 INVITE
Call-ID: 80a9a9cfbe9920f8@1.2.3.201
Max-Forwards: 70
Content-Length: 0
Contact: 

Thanks,
Xaled


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Re: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example

[xaled]

Hi,

could somebody please shed some light on this scenario?

Thanks,
Xaled 

-Original Message-
From: Users  On Behalf Of xa...@web.de
Sent: Sunday, April 11, 2021 11:59 PM
To: 'OpenSIPS users mailling list' 
Subject: [OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example

Hi,

I tested the 3.2  Prepaid example from B2B tutorial 

https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2

In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this 
reINVITE in my tests comes without SDP. Should not there be an SDP, or am I 
missing something in this scenario?

INVITE sip:TEST001@1.2.3.201:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0
To: ;tag=f79aacce
From: ;tag=B2B.81.186.1618177460
CSeq: 3 INVITE
Call-ID: 80a9a9cfbe9920f8@1.2.3.201
Max-Forwards: 70
Content-Length: 0
Contact: 

Thanks,
Xaled


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Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

[xaled]

Hi Bogdan,

I understood and got it working with extra_hdrs and extra_hdr_bodies. 

Thank you,
Xaled  

-Original Message-
From: xa...@web.de  
Sent: Wednesday, April 14, 2021 1:54 PM
To: 'Bogdan-Andrei Iancu' ; 'users@lists.opensips.org' 

Subject: RE: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

Hi Bogdan,

I got it, the actual question is how can I add new headers (not only Route, but 
any header) with values to the outgoing B2B INVITE request. Please see my other 
mail.

Thanks,
Xaled

-Original Message-
From: Bogdan-Andrei Iancu 
Sent: Wednesday, April 14, 2021 12:30 PM
To: xa...@web.de; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

Hi Xaled,

The record routing is incompatible with the b2b, by SIP definition. The record 
routing is done only by a SIP proxy, not by a SIP b2b2.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/11/21 9:32 PM, xa...@web.de wrote:
> Hi,
>
> would it be possible to respect loose routing on the caller side of 
> B2B module and set route header on the callee side accordingly
>
> Something like this:
>
> Incoming INVITE:
>
> INVITE sip:+12345678@1.2.3.4;user=phone SIP/2.0
> Via: SIP/2.0/TCP
> 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
> Route: ,
> 
>
> Wanted outgoing INVITE:
>
> INVITE sip:+87654321@1.1.1.1;user=phone SIP/2.0
> Via: SIP/2.0/TCP
> test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
> Route: 
>
> I tried setting loose_route in both the original route and in the resulting 
> b2b route but without success.
>
> route[b2b_test_logic] {
>
> <-->loose_route();
>
>  if ($rm != "BYE") {
>  # for requests other than BYE, no special actions needs to be done,
>  # just pass the request to the peer
>  b2b_pass_request();
>  exit;
>  }
> ...
>
> Thanks,
> Xaled
>
>
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Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

[Bogdan-Andrei Iancu]

Use the "local_route" for that

https://www.opensips.org/Documentation/Script-Routes-3-1#toc6

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
  https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/14/21 2:54 PM, xa...@web.de wrote:

Hi Bogdan,

I got it, the actual question is how can I add new headers (not only Route, but 
any header) with values to the outgoing B2B INVITE request. Please see my other 
mail.

Thanks,
Xaled

-Original Message-
From: Bogdan-Andrei Iancu 
Sent: Wednesday, April 14, 2021 12:30 PM
To: xa...@web.de; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

Hi Xaled,

The record routing is incompatible with the b2b, by SIP definition. The record 
routing is done only by a SIP proxy, not by a SIP b2b2.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/11/21 9:32 PM, xa...@web.de wrote:

Hi,

would it be possible to respect loose routing on the caller side of
B2B module and set route header on the callee side accordingly

Something like this:

Incoming INVITE:

INVITE sip:+12345678@1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/TCP
4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route: ,


Wanted outgoing INVITE:

INVITE sip:+87654321@1.1.1.1;user=phone SIP/2.0
Via: SIP/2.0/TCP
test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route: 

I tried setting loose_route in both the original route and in the resulting b2b 
route but without success.

route[b2b_test_logic] {

<-->loose_route();

  if ($rm != "BYE") {
  # for requests other than BYE, no special actions needs to be done,
  # just pass the request to the peer
  b2b_pass_request();
  exit;
  }
...

Thanks,
Xaled


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Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

[xaled]

Hi Bogdan,

I got it, the actual question is how can I add new headers (not only Route, but 
any header) with values to the outgoing B2B INVITE request. Please see my other 
mail.

Thanks,
Xaled

-Original Message-
From: Bogdan-Andrei Iancu  
Sent: Wednesday, April 14, 2021 12:30 PM
To: xa...@web.de; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

Hi Xaled,

The record routing is incompatible with the b2b, by SIP definition. The record 
routing is done only by a SIP proxy, not by a SIP b2b2.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/11/21 9:32 PM, xa...@web.de wrote:
> Hi,
>
> would it be possible to respect loose routing on the caller side of 
> B2B module and set route header on the callee side accordingly
>
> Something like this:
>
> Incoming INVITE:
>
> INVITE sip:+12345678@1.2.3.4;user=phone SIP/2.0
> Via: SIP/2.0/TCP 
> 4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
> Route: , 
> 
>
> Wanted outgoing INVITE:
>
> INVITE sip:+87654321@1.1.1.1;user=phone SIP/2.0
> Via: SIP/2.0/TCP 
> test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
> Route: 
>
> I tried setting loose_route in both the original route and in the resulting 
> b2b route but without success.
>
> route[b2b_test_logic] {
>
> <-->loose_route();
>
>  if ($rm != "BYE") {
>  # for requests other than BYE, no special actions needs to be done,
>  # just pass the request to the peer
>  b2b_pass_request();
>  exit;
>  }
> ...
>
> Thanks,
> Xaled
>
>
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Re: [OpenSIPS-Users] The next OpenSIPS Bootcamp online training

[Bogdan-Andrei Iancu]

Heads-up ALL, the deadline for the early-bird discount is almost here. 
IF you care about your pocket, register now ;-)


https://opensips.org/training/OpenSIPS_eBootcamp_2021/#mu-pricing

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
  https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 3/31/21 2:30 PM, Bogdan-Andrei Iancu wrote:




17th May-28th May 2021,


online, worldwide



*OpenSIPS Bootcamp*

Join the core developers and founder of the OpenSIPS project for an 
*OpenSIPS Bootcamp* 
 online session 
- an in-cloud training, a ten days, 4 hours per day (40 hours) 
intensive and practical training, covering installation, configuration 
and administration on OpenSIPS.


All the knowledge transferred to the students will be strongly backed 
up by practice sessions where you will get hands-on experience in 
handling OpenSIPS. The training is structured to be offer 50% / 50% 
between the theoretical and practical sessions.


Check Syllabus 



*Early Birds open*

The Early Bird 10% discount is available for registrations before 
/*16th of April 2021*/, so do not mis the opportunity. The number of 
seats is limited, so be sure and book a seat now. Keep in mind that a 
10% group discount is also available - grab your work mate and start 
learning more OpenSIPS together .

.

Register Now 



*Certified training saves time and money*

OpenSIPS mistakes are easily avoided if you get proper training! 
Companies that use OpenSIPS waste time and money when they don't have 
a trained engineer on staff. Searching on Google, waiting on IRC, even 
the latency in mailing list replies takes it's toll over time. Take 
this rare opportunity to train your employees with the project members 
themselves.



Any questions? do not hesitate to contact us 
!



You received this email as part of your relationship with the OpenSIPS 
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Re: [OpenSIPS-Users] TLS certificate reloading

[Callum Guy]

Excellent, thanks for beginning this to my attention - will track.

On Wed, 14 Apr 2021 at 11:52, Alexey Vasilyev 
wrote:

> Hi, yes there is an issue.
>
> Please follow here: https://github.com/OpenSIPS/opensips/issues/2433
>
>
>
>
> -
> ---
> Alexey Vasilyev
> --
> Sent from:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>
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**  |  
Practice Index Reviews *


THE ITSPA 
AWARDS 2020  AND Best ITSP - Mid 
Market, Best Software and Best Vertical Solution are trade marks of the 
Internet Telephony Services Providers' Association, used under licence.

*Our new office address: 22 Riduna Park, Melton IP12 1QT.*

X-on
is a 
trading name of Storacall Technology Ltd a limited company registered in
England and Wales.

Registered Office : Avaland House, 110 London Road, 
Apsley, Hemel Hempstead,
Herts, HP3 9SD. Company Registration No. 2578478.

The information in this e-mail is confidential and for use by the 
addressee(s)
only. If you are not the intended recipient, please notify 
X-on immediately on +44(0)333 332  and delete the
message from your 
computer. If you are not a named addressee you must not use,
disclose, 
disseminate, distribute, copy, print or reply to this email. Views
or 
opinions expressed by an individual
within this email may not necessarily
reflect the views of X-on or its associated companies. Although X-on 
routinely
screens for viruses, addressees should scan this email and any 
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Re: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications

[Bogdan-Andrei Iancu]

See

https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/

https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/

https://blog.opensips.org/2017/03/10/advanced-sip-scenarios-with-event-based-routing/

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
  https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/9/21 12:39 PM, HS wrote:

Hello all.

We are using Opensips 3.0 (with rtpproxy) and been looking to 
implement Push Notifications for iOS devices. Did come across quite a 
few resources, but they seem either too old ~2013 time or about V3.1. 
Has anyone implemented something to make calls to devices with the UAC 
in background mode on V3.0 pls?


Or suggestions on how to make this work on v3.0?
http://web.archive.org/web/20131018015829/http://techvoiper.com/opensips-and-apple-push-notification-service-integration/

Appreciate the help.

Thanks.

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Re: [OpenSIPS-Users] TLS certificate reloading

[Alexey Vasilyev]

Hi, yes there is an issue.

Please follow here: https://github.com/OpenSIPS/opensips/issues/2433




-
---
Alexey Vasilyev
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[OpenSIPS-Users] TLS certificate reloading

[Callum Guy]

Hi All,

I recently encountered an issue where our certificates were renewed,
following which I issued: *opensips-cli -x mi tls_reload*

The CLI action indicated success however on closer inspection of the
handshake we could see the previous certificate was continuing to be
presented. Previously I have had success with the reload operation. In this
situation only a full restart resolved the issue and loaded the
updated certificates.

We are storing certificates in the database and only use domain default
with a type 1 and type 2 record, both using the same certificate.

The following global params are set:

tcp_connection_lifetime=720
tcp_connect_timeout=200

We also use extremely long module timeout settings to deal with an
outrageously slow peer (slow on first connect):

modparam("tls_mgm", "tls_send_timeout", 2000)
modparam("tls_mgm", "tls_handshake_timeout", 2000)

Is tls_reload expected to work under all conditions or is there something
else we need to do (i.e. tear down existing connections)? All log messages
indicated success and as we are using lets encrypt certs the subject/issuer
remained the same so only a packet capture revealed the actual serial
number of the cert - as an aside it would be useful to have the SN reported
in the reload operation.

Thanks,

Callum

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Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module

[Bogdan-Andrei Iancu]

Hi Xaled,

The record routing is incompatible with the b2b, by SIP definition. The 
record routing is done only by a SIP proxy, not by a SIP b2b2.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
  https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 4/11/21 9:32 PM, xa...@web.de wrote:

Hi,

would it be possible to respect loose routing on the caller side of B2B module 
and set route header on the callee side accordingly

Something like this:

Incoming INVITE:

INVITE sip:+12345678@1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/TCP 
4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route: , 


Wanted outgoing INVITE:

INVITE sip:+87654321@1.1.1.1;user=phone SIP/2.0
Via: SIP/2.0/TCP 
test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route: 

I tried setting loose_route in both the original route and in the resulting b2b 
route but without success.

route[b2b_test_logic] {

<-->loose_route();

 if ($rm != "BYE") {
 # for requests other than BYE, no special actions needs to be done,
 # just pass the request to the peer
 b2b_pass_request();
 exit;
 }
...

Thanks,
Xaled


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Re: [OpenSIPS-Users] Issue with opensips-cli .

[Saurabh Chopra]

Hi Opensips Team/ Razvan,

As per Sasmita's previous mail, we are still stuck in a situation where,

1. If we are using "FIFO" as a communication_type we are not getting data
from a remote machine instead getting data from a local host.
2. If we are using "HTTP" as a communication_type then we are unable to
switch to a remote machine.

For both scenarios logs have been attached in *pastebin*.

Kindly advise and provide us with an early resolution. We will be
waiting for your prompt response

Best Regards
Saurabh Chopra
+918861979979


On Tue, Mar 30, 2021 at 11:31 AM Sasmita Panda  wrote:

> Hi ,
>
> I have changed my instance.cfg like bellow .
>
> [10.0.0.1]
> prompt_name: opensips-cli@10.0.0.1
> communication_type: http
> url: http://10.0.0.1:/json
>
> With this config above I am getting some error . I am not able to switch
> to the remote instance as well .
>
> https://pastebin.com/1fDuHbBk
>
>
> if I am giving below config .
>
>  [10.0.0.1]
> prompt_name: opensips-cli@10.0.0.1
> communication_type: fifo
> fifo_file: /tmp/opensips_fifo
>
> I am able to switch to the remote instance . After switching when I am
> running any command its giving the result for localhost not  for the remote
> instance ;.
>
>
> What should I do ? Am I missing any dependency library ? Please do help .
>
>
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Fri, Mar 26, 2021 at 5:58 PM Sasmita Panda  wrote:
>
>> same result .
>>
>> my instance.cfg contains :
>> cat instance.cfg
>> [10.0.1.1]
>> prompt_name: opensips-cli@10.0.1.1
>> fifo_file: /tmp/opensips_fifo
>>
>> https://pastebin.com/FQBipyZJ
>> This is the result .
>>
>>
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>>
>> On Fri, Mar 26, 2021 at 5:38 PM Răzvan Crainea 
>> wrote:
>>
>>> You are not switching the instance correctly: should have used `instance
>>> switch 10.0.1.1`
>>>
>>> Best regards,
>>>
>>> Răzvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>>
>>> On 3/26/21 1:54 PM, Sasmita Panda wrote:
>>> > I am not getting any error for this . I have pasted the logs in
>>> pastebin .
>>> >
>>> > https://pastebin.com/4tzXzrrG 
>>> >
>>> > What needs to be done in the remote machine 10.0.1.1 . I have allowed
>>> >  tcp port . I saw default http url is like http://10.0.1.1:/mi
>>> >  .
>>> > Do suggest if I am missing anything .
>>> >
>>> >
>>> > */Thanks & Regards/*
>>> > /Sasmita Panda/
>>> > /Senior Network Testing and Software Engineer/
>>> > /3CLogic , ph:07827611765/
>>> >
>>> >
>>> > On Fri, Mar 26, 2021 at 5:00 PM Răzvan Crainea >> > > wrote:
>>> >
>>> > Hi, Sasmita!
>>> >
>>> > Can you post the logs of an execution for the 10.0.1.1 instance run
>>> > with
>>> > debugging (-d parameter)? Also, please post them on pastebin[1] or
>>> in a
>>> > different place, not in the mail itself.
>>> >
>>> > [1] https://pastebin.com/ 
>>> >
>>> > Best regards,
>>> >
>>> > Răzvan Crainea
>>> > OpenSIPS Core Developer
>>> > http://www.opensips-solutions.com <
>>> http://www.opensips-solutions.com>
>>> >
>>> > On 3/26/21 11:16 AM, Sasmita Panda wrote:
>>> >  > Hi All ,
>>> >  >
>>> >  > I am using opensips-cli  ( 0.1.0 ) and opensips-3.1 .
>>> >  >
>>> >  > I am able to get all the data of the local host through cli but
>>> > when I
>>> >  > am connecting to a remote host even though it's getting
>>> connected
>>> > but no
>>> >  > data is visible for the remote instance  . It only showing the
>>> > data of
>>> >  > the instance on which I am running the cli .
>>> >  >
>>> >  >
>>> >  >   my opensips-cli.cfg
>>> >  > *[default]
>>> >  > database_modules: All
>>> >  > prompt_intro: Welcome to OpenSIPS
>>> >  >
>>> >  > database_admin_url: mysql://root:cccl0g1c@localhost
>>> >  > #database_url:  mysql://opensips:opensipsrw@localhost
>>> >  > database_schema_path: /opt/opensips-3.1/scripts
>>> >  >
>>> >  > [10.0.1.1]
>>> >  > prompt_name: opensips-cli@10.0.1.1 >> opensips-cli@10.0.1.1>
>>> > >
>>> >  > prompt_intro: OpenSIPS CLI for 10.0.1.1 instance
>>> >  > communication_type: fifo
>>> >  > url: http://10.0.1.1:/mi 
>>> > >*
>>> >  >
>>> >  > Am I doing anything wrong ? Please do help me .
>>> >  >
>>> >  >
>>> >  > */Thanks & Regards/*
>>> >  > /Sasmita Panda/
>>> >  > /Senior Network Testing and Software Engineer/
>>> >  > /3CLogic , ph:07827611765/
>>> >  >
>>> >  > ___
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