[OpenSIPS-Users] NOTIFY, BYE ends with 404 Not here

[Artur Mega]

Hello community,
I see, that BYE (from carrier) comes to Opensips without Route field...
Which gives me error '404 Not here'.
Before doing INVITE to the carrier, I execute record_route(). What can be
the reason why carrier don't put Route to his BYE? I checked 2 different
carriers, in every case the problem is same

-- 
​С уважением, ​
Артур
​Regards, Arthur​
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Re: [OpenSIPS-Users] NOTIFY, BYE ends with 404 Not here

[Liviu Chircu]

Hi Artur,

I would first make sure the outgoing packets actually contain the added 
"Record-Route" header by doing a trace - scripting mistakes come in all 
shapes and sizes. If the carriers are really to blame, you also have the 
option of fixing the routing set for the BYEs using fix_route_dialog() [1]


[1]: http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#idp5801280

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 28.06.2017 12:01, Artur Mega wrote:

Hello community,
I see, that BYE (from carrier) comes to Opensips without Route 
field... Which gives me error '404 Not here'.
Before doing INVITE to the carrier, I execute record_route(). What can 
be the reason why carrier don't put Route to his BYE? I checked 2 
different carriers, in every case the problem is same


--
​С уважением, ​
Артур
​Regards, Arthur​


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[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

[Alex Megalokonomos]

Hello,

We have the following scenario: our office call center is an Alcatel
OmniPCX Office setup.

This handles most of our needs and also provides 4 SIP extensions.

These are provided by what appears to be a Kamailio SIP server v 3.2.2 (no
webrtc or websockets support)

What we would like to do is set up an OpenSIPS instance to handle WebRTC
and proxy everything to this Kamailio SIP server.

The idea is to allow a web client (using sip js or something similar) to
register / make / receive calls as one of the Kamailio extensions.


I think half of the configuration is this :
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1

which I've already completed and indeed, clients can register to opensips
and chat/make calls over websockets between them.

How do I go about proxying registrations/invites/etc to the kamailio server
instead?

best regards
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[OpenSIPS-Users] Query regarding db_text.

[Sasmita Panda]

Hi All,

   I am trying to use db_text module of opensips-1.11 . While creating
DB , its giving error .

[root@2e922d8237f4 opensips]# /usr/local/sbin/opensipsdbctl create opensips
INFO: creating DBTEXT tables at: opensips ...
Install presence related tables? (y/n): y
INFO: creating DBTEXT presence tables at: opensips ...
Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist
b2b registrant call_center? (y/n): y
INFO: creating DBTEXT extra tables at: opensips ...
cp: cannot stat '/usr/local//share/opensips//dbtext/opensips/call_center':
No such file or directory
ERROR: Creating extra tables failed!


 What should I do to solve the problem ? Its not creating all the
tables . Please help me .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

[Mundkowsky, Robert]

Curious, why would you want to use OpenSIPS and Kamailio?

There both SIP proxies.

Robert

From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Alex 
Megalokonomos
Sent: Wednesday, June 28, 2017 5:47 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

[https://mailtrack.io/trace/mail/3c783866054bc5242de3e7df06ab3a49285c0f0b.png?u=1422671]Hello,

We have the following scenario: our office call center is an Alcatel OmniPCX 
Office setup.

This handles most of our needs and also provides 4 SIP extensions.

These are provided by what appears to be a Kamailio SIP server v 3.2.2 (no 
webrtc or websockets support)

What we would like to do is set up an OpenSIPS instance to handle WebRTC and 
proxy everything to this Kamailio SIP server.

The idea is to allow a web client (using sip js or something similar) to 
register / make / receive calls as one of the Kamailio extensions.


I think half of the configuration is this : 
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1

which I've already completed and indeed, clients can register to opensips and 
chat/make calls over websockets between them.

How do I go about proxying registrations/invites/etc to the kamailio server 
instead?

best regards



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[OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug fixes!

[Liviu Chircu]

Hi, all!

We have planned an OpenSIPS minor release bump for 2.2 and 2.3 - due 
next week, July 5th.The headline of this release, however, is the 
long-awaited 2.3-compatible release for the OpenSIPS Control Panel!


If you have any pending GitHub issues / mailing list bug threads 
concerning OpenSIPS 2.2+ or the Control Panel which are yet to be 
resolved, this would be a good time to bump them!


Best regards,

--
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com


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[OpenSIPS-Users] [Blog] Traffic balancing – the insertion into the SIP flow

[Bogdan-Andrei Iancu]

In the last post we talked about the different available flavors of 
traffic balancing 
with 
OpenSIPS. But picking the right balancing logic is just the first step. 
The next step, and an important one, is to decide how you want to have 
the balancer inserted into the SIP flow (between the end-points and the 
main servers)


https://blog.opensips.org/2017/06/28/traffic-balancing-the-insertion-into-the-sip-flow/

Enjoy this follow up on the balancing topic

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

OpenSIPS Bootcamp 2017, Houston, US
  http://opensips.org/training/OpenSIPS_Bootcamp_2017.html

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Re: [OpenSIPS-Users] [Blog] Traffic balancing – load, weights, round robin ??

[Bogdan-Andrei Iancu]

Hi Robert,

See the new blog post about how a balancer should be inserted into the 
SIP flow:


https://blog.opensips.org/2017/06/28/traffic-balancing-the-insertion-into-the-sip-flow/

Thanks for the idea of the post ;)

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

OpenSIPS Bootcamp 2017, Houston, US
  http://opensips.org/training/OpenSIPS_Bootcamp_2017.html

On 06/16/2017 07:52 PM, Mundkowsky, Robert wrote:

Thanks.

FYI, the main reason we decided to use openSIPS is that is a lot more 
documentation than other solutions.

Anyways, I have read lot of the books ("Building Telephony..2nd and 1.6"), some 
of the website (modules, how to, ); and the major missing things are:
1 - Need a "openSIPS CookBook of recipes" that has more route examples (e.g. 
how to setup user authentication, how to do HA, security (drop ghost calls, ...), setup 
to work with RTP proxies,  ...)
2 - module documentation needs better details on how to pass in and pass out 
values. For example, some modules do not expand variables, so it is really hard 
to pass in values.



Robert Mundkowsky

-Original Message-
From: Devel [mailto:devel-boun...@lists.opensips.org] On Behalf Of 
Bogdan-Andrei Iancu
Sent: Friday, June 16, 2017 10:47 AM
To: OpenSIPS users mailling list 
Cc: busin...@lists.opensips.org; n...@lists.opensips.org; OpenSIPS devel mailling 
list 
Subject: Re: [OpenSIPS-Devel] [OpenSIPS-Users] [Blog] Traffic balancing – load, 
weights, round robin ??

Hi Robert,

All the time there is space for more :). This blog post is the first from a set 
of docs trying to explain the routing with OpenSIPS.

Including some feedback from you, I can draft as following chapters:
  1) how to insert a balancer in your SIP traffic - like dialog statefull 
proxy, transaction statefull proxy, stateless proxy, etc
  2) routing modules in opensips, like dispatcher versus drouting versus 
load-balancer..


In your classification, I noticed many missing features for Dynamic Routing:
  - in memory matching with prefix-tree - O(prefix_len)
  - has gui in Control Panel
  - destination pinging, failover, re-enable
  - clustering capabilities for the state of the GW/destinations
  - rule fallback (on matching)
  ..

Again, any feedback is more the welcome as it will give us some ideas about the 
hot topics from the user perspective.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer

https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips-solutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7Ccb9609d3270b4091081108d4b4c6c079%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332213102517343&sdata=ejvSuDmud%2FPcsYLuGhCjfEX3Zug1WSCwBmWblGUTcus%3D&reserved=0

OpenSIPS Bootcamp 2017, Houston, US

https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fopensips.org%2Ftraining%2FOpenSIPS_Bootcamp_2017.html&data=02%7C01%7Crmundkowsky%40ets.org%7Ccb9609d3270b4091081108d4b4c6c079%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636332213102517343&sdata=30yqvccNVQZmoeGjEWkjWQC2n7CZ7KgzccZxoRtPBek%3D&reserved=0

On 06/15/2017 09:36 PM, Mundkowsky, Robert wrote:

This is helpful. But would like a little more details there. Such as:

1) openSIPS when it actions as a Load Balancer is always a SIP proxy during the 
complete dialog. In other words, the caller RTP is direct connected backend 
gateway, but openSIPS is always between them for the SIP traffic.
2) Some information for the traffic distributions modules is stored in
the database, but some info is only in memory (e.g. which gateway is
enabled, count of active dialogs)

This might be naïve details for most telecom folks, but helpful for naïve users 
like myself.

I haven't played with "Carrier Route","Dispatcher", and "Dynamic Routing", but 
similar notes for those would be helpful too.

The following video was very helpful in describing the different traffic 
distributions modules openSIPS supports:

https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.y
outube.com%2Fwatch%3Fv%3DyTLPs1-X0SM&data=02%7C01%7Crmundkowsky%40ets.
org%7Ccb9609d3270b4091081108d4b4c6c079%7C0ba6e9b760b34fae92f37e6ddd9e9
b65%7C0%7C0%7C636332213102517343&sdata=mDlSlzbtLQ%2FL3uDq67VkOa%2B8FPG
20nNx4AZ9duHVi1s%3D&reserved=0

Not sure if my notes are helpful, but they are below.

Routing modules:
Carrier Route
- old module and not maintained, but seems to work
- features: routing, load balancing, blacklisting
- not installed by default
- no web GUI

- performs longest prefix matching
- you use longest preefix to pull "least cost" route out of database
table
- strips prefix, adds prefix/suffix
- probabilities to load balancer


Load Balancer
- light weight
- balancing based on load
- can config via GUI
- single database table

- features:
- no prefix/suffix changing
- with version 1.8, you can keep counters in database

- docs:
Tutorial
https://na01.safelinks.protection.outlook.com/?url=h

[OpenSIPS-Users] SIP URI User Parameters

[Ben Newlin]

Hi,

We have run into an issue with OpenSIPs’ handling of user parameters in SIP 
URIs with Dynamic Routing module. When a parameter is added to a SIP URI user 
part, any subsequent modification of the URI by DR module results in the 
parameter being moved to be a URI parameter.

For example, starting with $ru of “sip:+1551...@gw1.com”, if we modify it 
this way:

$rU = $rU + “;npdi”;

then we get a new $ru of “sip:+1551212;n...@gw1.com”.

We send this call out and if it returns an error we want to use the next 
available gateway.

The Request URI in the failure route is still “sip:+1551212;n...@gw1.com”.

Note: this is the case even when the “;npdi” parameter was added in a branch 
route, which I didn’t expect. I thought changes made in a branch route were 
isolated to that branch.

Now from the failure route when we call use_next_gw the npdi parameter is moved 
and the URI is now “sip:+1551...@gw2.com;npdi”. This is not correct.

Is there some other way to properly manipulate SIP URI user parameters or is 
this a bug?


Thanks,

Ben Newlin
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Re: [OpenSIPS-Users] NOTIFY, BYE ends with 404 Not here

[Artur Mega]

Hi, thanks for reply!
Actually, the reason wasn't in `fix_route_dialog()`. But like you said, I
re-checked if Record-Route presents in messages, they weren't (I use
combination of 2 proxies, Record-Route wasn't present on second proxy). So,
now everything works. Thanks!

2017-06-28 14:08 GMT+05:00 Liviu Chircu :

> Hi Artur,
>
> I would first make sure the outgoing packets actually contain the added
> "Record-Route" header by doing a trace - scripting mistakes come in all
> shapes and sizes. If the carriers are really to blame, you also have the
> option of fixing the routing set for the BYEs using fix_route_dialog() [1]
>
> [1]: http://www.opensips.org/html/docs/modules/2.4.x/dialog.
> html#idp5801280
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 28.06.2017 12:01, Artur Mega wrote:
>
> Hello community,
> I see, that BYE (from carrier) comes to Opensips without Route field...
> Which gives me error '404 Not here'.
> Before doing INVITE to the carrier, I execute record_route(). What can be
> the reason why carrier don't put Route to his BYE? I checked 2 different
> carriers, in every case the problem is same
>
> --
> ​С уважением, ​
> Артур
> ​Regards, Arthur​
>
>
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>
>


-- 
​С уважением, ​
Артур
​Regards, Arthur​
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