[OpenSIPS-Users] Phone Registration :(

[Voice TAC]

Hello,

I'm totally new with OpenSIPS and I manged to install it and adjust the
configuration script. However, I tried to register a phone using X-Lite and
ZOPIPER but I could not.

I tried to sniff using ngrep but I did not find any SIP messages. I used
the following command:

ngrep -pqtw byline port 5060 >sip_Messages_Debug.sip


===

In addition, I used the opensipsctl ul show command but I got:

ERROR: /tmp/opensips_fifo does not exist
ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name",
"/tmp/opensips_fifo")' in your config
ERROR: and also have loaded the mi_fifo module.

Any help please?
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[OpenSIPS-Users] OpenSIPS migth be changing CSeq numbers after SIP OPTIONS. How to avoid it?

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS users,

I'm facing a problem related to CSeq numbers in SIP messages and maybe it is 
being caused by SIP OPTIONS messages.
When a caller sends a SIP ACK to the callee, the SIP ACK has CSeq = 2. However, 
after being relayed by OpenSIPS, the CSeq number become 3.
It seems that OpenSIPS decided to change such value on its own. But, maybe it 
was caused by the previous SIP OPTIONS message receive before.
The diagram bellow shows what is happening.

Can the SIP OPTIONS message cause such issue? If yes, should I give up of using 
SIP OPTIONS? Can it be a bug in OpenSIPS?
If not, what is happening here??
If that CSeq remains like 3, the callee will simply discard the SIP ACK and 
send another SIP OK, will not discard?



CallerOpenSIPs  Callee
|  |
|
|  |
|
|   INVITE  ||
|   CSeq = 2  ||
|--->|INVITE|
| |CSeq = 2|
| |--->|
| | 
|
| | TRYING |
| | CSeq = 2   |
|  TRYING|<---|
|  CSeq = 2  | |
|<---| |
|  |
|
|  | RING |
|  | CSeq = 2   |
|  RING  |<---|
|  CSeq = 2  |  |
|<---|  |
| | SIP OK   |
| | CSeq = 2   |
|  SIP OK  |<---|
|  CSeq = 2  | |
|<---| |
| | SIP OK   |
| | CSeq = 2   |
|  SIP OK |<---|
|  CSeq = 2  | |
|<---| |
| | 
   |
| |SIP OPTIONS   |
| |CSeq = 3|
<< new CSeq
| |<---|
| | 
|
| | SIP 500  |
| | CSeq = 3   |
| |--->|
|   ACK  ||
|   CSeq = 2 | |
|--->|ACK|
| |CSeq = 3|
  << Cseq of ACK was changed by OpenSIPS.
| |--->|
|   ACK |  |
|   CSeq = 2 | |
|--->|ACK|
| |CSeq = 3|
| |--->|
| | 
|
| | SIP OK   |
| | CSeq = 2   |
|  SIP OK  |<---|
|  CSeq = 2  | |
|<---| |
| 

Re: [OpenSIPS-Users] Location table - how to use socket

[John Quick]

Hi Bogdan,

The main reason to want to use the standard location table for both servers is 
so we can keep the original database schema.
Keeping to the original schema makes upgrades smoother. Upgrade tools you 
provide expect to find one location table, called "location"

To overcome problems, we have now created one location table per site and 
therefore had to give the new tables a new name. So when we upgrade, there will 
be extra work.

Sorry, but I cannot see how it is a benefit for OpenSIPS to retrieve an 
unfiltered list of contacts and have no easy mechanism to filter it. It 
probably depends how the multiple OpenSIPS servers are working together - how 
they are cooperating. If they are load sharing and every server is identical to 
the next, then there may be a benefit although I cannot see immediately how you 
overcome the problem that each OpenSIPS server has a unique IP address and that 
UA's behind restricted cone NAT will not be contactable except from the IP 
address where they registered.

The architecture we have is dual Data Centre - for resilience, not load 
sharing; UA's may register once only at each DC; the DC's operate as 
Active-Active; each OpenSIPS server is on a different IP address (in fact on 
different subnets). The database is a distributed high availability solution 
but appears as a single entity to both data centres. Maybe we should be using a 
different database for each data centre, but a lot of the data are common (e.g. 
user credentials, trusted address list) so admin is much easier with just one 
database.

John Quick
Smartvox Limited


-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: 07 June 2017 15:45
To: john.qu...@smartvox.co.uk; 'OpenSIPS users mailling list' 

Subject: Re: [OpenSIPS-Users] Location table - how to use socket

Hi John,

Sorry, my bad, $fs, not $si :)

There is something I mis here - all the downsides you list are actually 
benefits of a shared location table :). So, if you want each server to "read" 
only its own registered contact, why do you use shared location ?

Best regards,

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 06/06/2017 08:58 PM, John Quick wrote:
> Hi Bogdan,
>
> Surely $si is the IP source address of the request. It would show me where 
> the INVITE came from, not the socket for the current R-URI wouldn't it?
>
> The main problem with shared location table is reading back and using 
> contacts that are relevant on the local server. I don't want to read back 
> registrations that happened on the other server OR if I have to work with a 
> complete list of all contacts then I need simple way to filter and use them 
> sensibly. Possible cases were:
>   - User registered on this server
>   - User registered on the other server
>   - User registered on both servers
>   - User is not registered on either server The lookup() function 
> returns true for all of the first three cases. You then need to determine how 
> many branches are there and then check if the socket for the R-URI and for 
> the branch (if present) matches a local socket - allowing for socket to be 
> tls or udp or, as often happened,  - then remove the non-local contact 
> if present and relay the INVITE OR decide that the contact was non-local and 
> return 302. Consequently, this involved a lot of messy code and I never got 
> it working.
>
> Another issue was the limit for maximum contacts. I was using 
> save("location", "fc2") when it was a single site solution. When we changed 
> to dual site with a shared location table, it had to be changed to 
> save("location", "fc3") because the save function counts all contact records 
> for that user irrespective of socket. The relaxation required to allow 
> registration on both servers would also allow two registrations on one server 
> which I don't want to allow.  [btw, fc2 limits to one record per user and fc3 
> limits to 2 - curious].
>
> John Quick
> Smartvox Limited
>
>
> -Original Message-
> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> Sent: 06 June 2017 18:11
> To: john.qu...@smartvox.co.uk; 'OpenSIPS users mailling list' 
> 
> Subject: Re: [OpenSIPS-Users] Location table - how to use socket
>
> Hi John,
>
> Use $si to read/write the "send socket" associated with the RURI.
>
> Now, the discussion deviated and touched several topics, but trying to 
> understand the "There were too many problems with sharing one location table 
> for two different sites" - besides the socket aspect, is there anything else 
> in there ?
>
> Regards,
>
> 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 06/06/2017 07:19 

Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Daniel Zanutti]

That's great!

I'll take a look.

Thanks William

On Wed, Jun 7, 2017 at 12:19 PM, William Simon 
wrote:

> Web site certificates are fine. Do not purchase anything more than simple
> name validation because SIP clients won't care and you will waste your
> money. You can even use free certificates from LetsEncrypt.org
>
>
> On Jun 7, 2017, at 11:14 AM, Daniel Zanutti 
> wrote:
>
> Hi Alex
>
> I have tried with self-generated certificate and it is working fine.
>
> The problem is that this server will be "public" to any kind of sip
> client, so I need a Certificate Authority behind it.
>
> I couldn't find the right certificate, can anyone point the right one to
> buy?
>
> Looked at instantssl, they are all website certificates:
>
> https://www.instantssl.com/ssl-certificate.html?track=8183
>
> Thanks in advance!
>
> On Wed, Jun 7, 2017 at 11:20 AM, Alexander Jankowsky <
> e75a4...@exemail.com.au> wrote:
>
>>
>>
>> It is fairly straight forward to generate your own TLS certificates,
>>
>> if it is for your own Opensips server or for your own Voip phones.
>>
>>
>>
>> It is putting the certificates to use that may require a little trial and
>> error.
>>
>> You really might first want and need to test everything and make sure
>>
>> it is all sorted out and working properly with your own certificates.
>>
>>
>>
>> Do you really just only need a certificate to get things running
>>
>> for now or does it have to be from a recognised authority.
>>
>>
>>
>> Alex
>>
>>
>>
>> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
>> Zanutti
>> *Sent:* Wednesday, 7 June 2017 11:31 PM
>> *To:* OpenSIPS users mailling list
>> *Subject:* [OpenSIPS-Users] Which SSL Certificate for SIP+TLS
>>
>>
>>
>> I need to install an TLS certificate for secure SIP communication.
>>
>>
>>
>> Could you guys please point a valid certificate so I can buy it?
>>
>>
>>
>> Thanks
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> “The information transmitted is intended only for the person or entity to
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> dissemination, distribution, reproduction or any action taken in reliance
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>
>
>
>
> “The information transmitted is intended only for the person or entity to
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Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[William Simon]

Web site certificates are fine. Do not purchase anything more than simple name 
validation because SIP clients won't care and you will waste your money. You 
can even use free certificates from LetsEncrypt.org


On Jun 7, 2017, at 11:14 AM, Daniel Zanutti 
> wrote:

Hi Alex

I have tried with self-generated certificate and it is working fine.

The problem is that this server will be "public" to any kind of sip client, so 
I need a Certificate Authority behind it.

I couldn't find the right certificate, can anyone point the right one to buy?

Looked at instantssl, they are all website certificates:

https://www.instantssl.com/ssl-certificate.html?track=8183

Thanks in advance!

On Wed, Jun 7, 2017 at 11:20 AM, Alexander Jankowsky 
> wrote:

It is fairly straight forward to generate your own TLS certificates,
if it is for your own Opensips server or for your own Voip phones.

It is putting the certificates to use that may require a little trial and error.
You really might first want and need to test everything and make sure
it is all sorted out and working properly with your own certificates.

Do you really just only need a certificate to get things running
for now or does it have to be from a recognised authority.

Alex

From: Users 
[mailto:users-boun...@lists.opensips.org]
 On Behalf Of Daniel Zanutti
Sent: Wednesday, 7 June 2017 11:31 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

I need to install an TLS certificate for secure SIP communication.

Could you guys please point a valid certificate so I can buy it?

Thanks

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privileged material. If you are not the intended recipient of this message you 
are hereby notified that any use, review, retransmission, dissemination, 
distribution, reproduction or any action taken in reliance upon this message is 
prohibited. If you received this in error, please contact the sender and delete 
the material from any computer.”
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“The information transmitted is intended only for the person or entity to which 
it is addressed and may contain proprietary, business-confidential and/or 
privileged material. If you are not the intended recipient of this message you 
are hereby notified that any use, review, retransmission, dissemination, 
distribution, reproduction or any action taken in reliance upon this message is 
prohibited. If you received this in error, please contact the sender and delete 
the material from any computer.”
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Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Daniel Zanutti]

Hi Alex

I have tried with self-generated certificate and it is working fine.

The problem is that this server will be "public" to any kind of sip client,
so I need a Certificate Authority behind it.

I couldn't find the right certificate, can anyone point the right one to
buy?

Looked at instantssl, they are all website certificates:

https://www.instantssl.com/ssl-certificate.html?track=8183

Thanks in advance!

On Wed, Jun 7, 2017 at 11:20 AM, Alexander Jankowsky <
e75a4...@exemail.com.au> wrote:

>
>
> It is fairly straight forward to generate your own TLS certificates,
>
> if it is for your own Opensips server or for your own Voip phones.
>
>
>
> It is putting the certificates to use that may require a little trial and
> error.
>
> You really might first want and need to test everything and make sure
>
> it is all sorted out and working properly with your own certificates.
>
>
>
> Do you really just only need a certificate to get things running
>
> for now or does it have to be from a recognised authority.
>
>
>
> Alex
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
> Zanutti
> *Sent:* Wednesday, 7 June 2017 11:31 PM
> *To:* OpenSIPS users mailling list
> *Subject:* [OpenSIPS-Users] Which SSL Certificate for SIP+TLS
>
>
>
> I need to install an TLS certificate for secure SIP communication.
>
>
>
> Could you guys please point a valid certificate so I can buy it?
>
>
>
> Thanks
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] Building a redirect server

[Royee Tichauer via Users]

Hi,

We are thinking of going through a path of building a redirect registrar
server using opensips. In addition we are planning to add an additional SBC
layer also using opensips. My question is can we use opensips in such a way
that when an authenticated INVITE comes in to the SBC it is sent over to
the redirect registrar server which if authorized will respond with a 3xx
that will indicate the SBC that it should redirect this INVITE to the
user's assigned PBX. I would of course expect the SBC to absorb the 3xx
response and have this all seem less for the client. Our objective here to
get the authentication server out of the call path so future SIP traffic
for a dialog does not go through this server.

Any thoughts/ suggestions/ past experience with this kind of setup would be
helpful.

Thanks a lot,
Royee
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Re: [OpenSIPS-Users] Location table - how to use socket

[Bogdan-Andrei Iancu]

Hi John,

Sorry, my bad, $fs, not $si :)

There is something I mis here - all the downsides you list are actually 
benefits of a shared location table :). So, if you want each server to 
"read" only its own registered contact, why do you use shared location ?


Best regards,

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 06/06/2017 08:58 PM, John Quick wrote:

Hi Bogdan,

Surely $si is the IP source address of the request. It would show me where the 
INVITE came from, not the socket for the current R-URI wouldn't it?

The main problem with shared location table is reading back and using contacts 
that are relevant on the local server. I don't want to read back registrations 
that happened on the other server OR if I have to work with a complete list of 
all contacts then I need simple way to filter and use them sensibly. Possible 
cases were:
  - User registered on this server
  - User registered on the other server
  - User registered on both servers
  - User is not registered on either server
The lookup() function returns true for all of the first three cases. You then need to 
determine how many branches are there and then check if the socket for the R-URI and 
for the branch (if present) matches a local socket - allowing for socket to be tls or 
udp or, as often happened,  - then remove the non-local contact if 
present and relay the INVITE OR decide that the contact was non-local and return 302. 
Consequently, this involved a lot of messy code and I never got it working.

Another issue was the limit for maximum contacts. I was using save("location", "fc2") when it was a 
single site solution. When we changed to dual site with a shared location table, it had to be changed to 
save("location", "fc3") because the save function counts all contact records for that user 
irrespective of socket. The relaxation required to allow registration on both servers would also allow two 
registrations on one server which I don't want to allow.  [btw, fc2 limits to one record per user and fc3 limits to 2 - 
curious].

John Quick
Smartvox Limited


-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 06 June 2017 18:11
To: john.qu...@smartvox.co.uk; 'OpenSIPS users mailling list' 

Subject: Re: [OpenSIPS-Users] Location table - how to use socket

Hi John,

Use $si to read/write the "send socket" associated with the RURI.

Now, the discussion deviated and touched several topics, but trying to understand the 
"There were too many problems with sharing one location table for two different 
sites" - besides the socket aspect, is there anything else in there ?

Regards,

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 06/06/2017 07:19 PM, John Quick wrote:

Hi Bogdan,

The problem could be from setting db_mode to 3.  As you said earlier, this is 
always required when using a shared location table.
My DB table would have two records - one where the socket is for server A and 
the other with the socket for server B.
It is possible that when the lookup() function loads matching contacts into 
memory, the first contact has a local socket and the second one has non-local.
My log statement would only print the socket info for the first branch. Perhaps 
it should print the info for the main R-URI but I was not sure how to get to 
that item of data.
The order that is used to load contacts into memory would be somewhat random - 
sometimes it might be the local contact first and sometimes not. I could use Q 
to force a consistent load order, but it would still be wrong on one site and 
right on the other. I would prefer that there was an option to only load 
contacts where the socket was a local one. That would be *much* more useful 
than loading the contact and discarding its socket data. Could you add this to 
the road map please?

Frankly, this is all a bit academic now. I have changed to use a location table 
per site and have also changed db_mode back to 1. There were too many problems 
with sharing one location table for two different sites.

So this discussion is only relevant if you and the other developers want to 
rationalise the usrloc module so it can be used in the scenario where two 
different OpenSIPS servers (both active) can share a single location table. 
Perhaps you just need to add a few notes to the documentation to warn users 
that this is not a recommended scenario. I am writing a new knowledgebase 
article that covers this topic. I will attempt to explain the problems there in 
more detail.

John Quick
Smartvox Limited


-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 06 June 2017 16:51
To: john.qu...@smartvox.co.uk; 

Re: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Alexander Jankowsky]

 

It is fairly straight forward to generate your own TLS certificates,

if it is for your own Opensips server or for your own Voip phones.

 

It is putting the certificates to use that may require a little trial and error.

You really might first want and need to test everything and make sure

it is all sorted out and working properly with your own certificates.

 

Do you really just only need a certificate to get things running

for now or does it have to be from a recognised authority.

 

Alex

 

From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Daniel 
Zanutti
Sent: Wednesday, 7 June 2017 11:31 PM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Which SSL Certificate for SIP+TLS

 

I need to install an TLS certificate for secure SIP communication.

 

Could you guys please point a valid certificate so I can buy it? 

 

Thanks

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Re: [OpenSIPS-Users] Call Pickup feature in Opensips 1.11

[maatohewetbi]

Is it any way to upload whole script to test Call pickup with 2.3? I've
installed this version and pasted pickup config from module event_routing
documentation. Call Pickup doesn't work as it should. When I call one leg,
and want to handle this call by third user I get only 480 message after
dialing "33". When I delete this line it works ok but on third user phone
(Grandstream) I get incomming call on second line whil first is still
calling. When I answer it looks ok. But I feel it doesn't work as it should.



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[OpenSIPS-Users] Which SSL Certificate for SIP+TLS

[Daniel Zanutti]

I need to install an TLS certificate for secure SIP communication.

Could you guys please point a valid certificate so I can buy it?

Thanks
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Re: [OpenSIPS-Users] acc extra_fields

[Bogdan-Andrei Iancu]

Try  Jessie + 2.3(stable) + nightly build

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 06/07/2017 04:18 PM, Edwin wrote:

I did, but there is no update available yet. I install from repro: deb
http://apt.opensips.org jessie 2.3-releases

opensips   2.3.0-1
opensips-b2bua-module:amd642.3.0-1
opensips-mysql-module:amd642.3.0-1




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Re: [OpenSIPS-Users] acc extra_fields

[Edwin]

I did, but there is no update available yet. I install from repro: deb
http://apt.opensips.org jessie 2.3-releases

opensips   2.3.0-1
opensips-b2bua-module:amd642.3.0-1
opensips-mysql-module:amd642.3.0-1




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Re: [OpenSIPS-Users] ratelimit (cps/timer_interval)

[Max Mühlbronner]

Cool, thanks.



BR


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Von: Users  im Auftrag von Răzvan Crainea 

Gesendet: Mittwoch, 7. Juni 2017 14:08:44
An: users@lists.opensips.org
Betreff: Re: [OpenSIPS-Users] ratelimit (cps/timer_interval)

Hello, Max!

If you want to limit to 50 CPS, you should use:
if (!rl_check("$avp(carriername)", "50", "RED"))

I will update the docs to reflect this.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/07/2017 01:07 PM, Max Mühlbronner wrote:

Hi,


just a small question regarding the ratelimit module. I am confused by the 
documentation:


rl_check():


"Note that the limit should be specified as per-second, not per-timer_interval."



But at the timer_interval modparam it says:


"All amounts of messages have to be divided by this timer to get a messages per 
second value. "




So which one is true, e.g. if i want to limit to 50 cps (and the default 
timer_interval is 10 seconds) would i just do something like:


if (!rl_check("$avp(carriername)", "50", "RED")) {
...


Or


if (!rl_check("$avp(carriername)", "500", "RED")) {





Thanks.


BR


Max Mühlbronner
--
42com Telecommunication GmbH
Straße der Pariser Kommune 12-16
10243 Berlin

Fon: +49-(0)30-2434299-28
Fax: +49-(0)30-2434299-99
E-Mail: m...@42com.com
Web: www.42com.com

Firmenangaben/Company information:
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Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

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Re: [OpenSIPS-Users] acc extra_fields

[Bogdan-Andrei Iancu]

Have you tried to update your 2.3 version and give it a try again ?

Regards,

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 06/07/2017 03:47 PM, Edwin wrote:

I will do that. Besides, I think the problem was already described and
resolved in bug 1123

https://github.com/OpenSIPS/opensips/issues/1123



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Re: [OpenSIPS-Users] acc extra_fields

[Edwin]

I will do that. Besides, I think the problem was already described and
resolved in bug 1123

https://github.com/OpenSIPS/opensips/issues/1123



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Re: [OpenSIPS-Users] OpenSIPS Dependencies Installation

[Alexander Jankowsky]

 

You might try the Opensips already precompiled Debian package on Debian Linux.

I first began by compiling Opensips, first up to start with and ran into some 
problems.

Then I tried the precompiled debian package and soon got that running alright.

It is a very good way to get started and have a good look at what is going on.

 

There are also complete already running virtual machine image files around.

If you are still in a virtual machine you might try a prebuilt and running 
system.

Then you can have a good look around at what is going on in a working system.

I am not really all that familiar with virtual machine environments in practice.

 

I very much prefer a clean install onto a spare machine and a lot of note 
taking.

That way you can re-do things, fully clean reinstall and work from your own 
notes.

It only takes a little more time but I do find it more practical on a dedicated 
machine.

 

Once you have any working system then it is easier to plan and work from that.

 

Alex

 

 

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Re: [OpenSIPS-Users] ratelimit (cps/timer_interval)

[Răzvan Crainea]

Hello, Max!

If you want to limit to 50 CPS, you should use:
if (!rl_check("$avp(carriername)", "50", "RED"))

I will update the docs to reflect this.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 06/07/2017 01:07 PM, Max Mühlbronner wrote:


Hi,


just a small question regarding the ratelimit module. I am confused by 
the documentation:



rl_check():


/"Note that the limit should be specified as per-second, not 
per-timer_interval."/




But at the timer_interval modparam it says:

/
/

/"All amounts of messages have to be divided by this timer to get a 
messages per second value. "/





So which one is true, e.g. if i want to limit to 50 cps (and the 
default timer_interval is 10 seconds) would i just do something like:



if (!rl_check("$avp(carriername)", "50", "RED")) {
...


Or


if (!rl_check("$avp(carriername)", "500", "RED")) {





Thanks.


BR


Max Mühlbronner
--
42com Telecommunication GmbH*
*Straße der Pariser Kommune 12-16
10243 Berlin

Fon: +49-(0)30-2434299-28
Fax: +49-(0)30-2434299-99
E-Mail: m...@42com.com
Web: _www.42com.com _


Firmenangaben/Company information:
Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
Umsatzsteuer-ID/VAT-ID: DE223812306
Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. 
Diese sind möglicherweise vertraulich und ausschließlich für den 
Adressaten bestimmt. Sollten Sie diese elektronische Nachricht 
irrtümlicherweise erhalten haben, so informieren Sie uns bitte 
unverzüglich telefonisch oder per E-Mail.


This message is intended only for the use of the individual or entity 
to which it is addressed. If you have received this message by 
mistake, please notify us immediately.




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Re: [OpenSIPS-Users] acc extra_fields

[Bogdan-Andrei Iancu]

Hi Edwin,

First, can you send me a script snapshot (off list) that produces the 
CRITICAL message ? (and the "opensips -V")


Now, how you should do it is:

# put the $acc_extra(channel) content in "col_channel" DB column
modparam("acc", "extra_fields", "db: channel->col_channel")

if ( do_routing() ) {
# populate the ACC variable with the DR output
$acc_extra(channel) = $avp(channel);
..
}


Have you read 
http://www.opensips.org/Documentation/Tutorials-Advanced-Accounting ?


The old ACC approach (in using AVP) suffered from several problems due 
the nature of the AVPs (like multi-values, but cannot use NULL). Also, 
for many script writer was not clear when the actual value (from the 
AVP) is read by the ACC module (when the value is actually evaluated). 
So, we decided for a more rigorous and straight approach in terms of 
setting the ACC data.


Nevertheless, feedback is welcome ;)

Best regards,

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 06/06/2017 09:04 PM, Edwin wrote:

Hi Bogdan,

I use $avp's in i.a. 'avp_db_query' and 'store_dlg_value', and created extra
fields for channel, a_number, b_number, subscriber_id. The channel is
defined in drouting (modparam("drouting", "gw_id_avp", '$avp(channel)')).

First I tried to swap all the $avp for $acc_extra, but 'auth_db' didn't
appreciate this : "auth_db:parse_aaa_avps: malformed or non AVP
$acc_extra(subscriber_id) AVP definition".

Thereafter I tried to fill the $acc_extra with $avp values
($acc_extra(subscriber_id) = $avp(subscriber_id) before and after
do_routing. This resulted in an critical error:
"CRITICAL:core:pv_value_destroy: #012>>> pkg_free() on non-pkg ptr 0x807491
- aborting!#012#012It seems you have hit a programming bug.#012"

Then I wondered were to fill the $acc_extra(channel) variable, after
do_routing?

OK, I have to read some more documentation and do more testing, but my first
impression is 'just keep it like it was, or at least also permit $avp's in
the extra fields'.



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[OpenSIPS-Users] ratelimit (cps/timer_interval)

[Max Mühlbronner]

Hi,


just a small question regarding the ratelimit module. I am confused by the 
documentation:


rl_check():


"Note that the limit should be specified as per-second, not per-timer_interval."



But at the timer_interval modparam it says:


"All amounts of messages have to be divided by this timer to get a messages per 
second value. "




So which one is true, e.g. if i want to limit to 50 cps (and the default 
timer_interval is 10 seconds) would i just do something like:


if (!rl_check("$avp(carriername)", "50", "RED")) {
...


Or


if (!rl_check("$avp(carriername)", "500", "RED")) {





Thanks.


BR


Max Mühlbronner
--
42com Telecommunication GmbH
Straße der Pariser Kommune 12-16
10243 Berlin

Fon: +49-(0)30-2434299-28
Fax: +49-(0)30-2434299-99
E-Mail: m...@42com.com
Web: www.42com.com

Firmenangaben/Company information:
Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
Umsatzsteuer-ID/VAT-ID: DE223812306
Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese sind 
möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. 
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informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail.

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it is addressed. If you have received this message by mistake, please notify us 
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Re: [OpenSIPS-Users] FW: Re: 401 Unauthorized after Authentication Digest

[David Peláez]

Hi John.

I configured "secure=INVITE" but the same behaivor continue. Also the
extensions on Asterisk server are pjsip and the trunk is chan_sip, could it
be the problem why the calls aren't reching the SIPphone? Or some problem
between the ports the servers are listen to?
I just have one peer defined which is the one I am sending the calls.

And now I have seen this error on Asterisk server:

[2017-06-06 10:58:20] NOTICE[3601] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"501" ' failed for '192.168.1.12:5060'
(callid: 880692485-17367...@bjc.bgi.B.C) - No matching endpoint found
[2017-06-06 10:58:20] NOTICE[3601] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"501" ' failed for '192.168.1.12:5060'
(callid: 880692485-17367...@bjc.bgi.B.C) - No matching endpoint found
[2017-06-06 10:58:20] NOTICE[3601] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"501" ' failed for '192.168.1.12:5060'
(callid: 880692485-17367...@bjc.bgi.B.C) - Failed to authenticate

What does it means?

Best regards
David


2017-06-02 12:20 GMT+02:00 John Quick :

> Hi David,
>
> In asterisk, "insecure=INVITE" should be sufficient to disable
> authentication, although I have only tried it using chan_sip, not pjsip.
> Is it possible you have another sip peer defined where the address for
> "host=" is the same? It is very difficult to know which one Asterisk will
> use for incoming calls when there are two with the same address for host.
> If you have parameters for username and secret in your sip peer, try
> commenting them out and see if that helps.
>
> I would not advise disabling authentication of SIP phones. In fact you
> should make sure you always use strong passwords.
> All makes of SIP phone will support username/password authentication and
> it is vital to keep it active if you don't want your phone system to be
> hacked.
> However, you should add this line to opensips.cfg after the SIP phone
> authentication section (www_authorize) and before you send the call to
> Asterisk (t_relay):
>
> consume_credentials();
>
> This will remove the headers that OpenSIPS and the SIP phone exchanged for
> authentication. If you don't remove those headers, Asterisk is likely to
> get confused and may request authorisation.
>
> The consume_credentials function is documented here:
> http://www.opensips.org/html/docs/modules/2.2.x/auth.html#idp5543680
>
> John Quick
> Smartvox Limited
>
>
> From: David Peláez [mailto:dvl...@gmail.com]
> Sent: 02 June 2017 10:56
> To: john.qu...@smartvox.co.uk
> Cc: users@lists.opensips.org
> Subject: Re: FW: Re: [OpenSIPS-Users] 401 Unauthorized after
> Authentication Digest
>
> Thanks a lot for your replay. I already change the option
> "insecure=INVITE" as you suggested but I am still having the same problem.
> Find attached the peer configuration maybe I am missing something else.
> About opensips authenticating calls from SIPphones how do I disabled that
> behavior? because my opensips sends an 407 Proxy Authentication to the Sip
> phone before sending the INVITE to asterisk server.
> Best regards
> David
>
>
>
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