[OpenSIPS-Users] Compiling modules

2024-06-11 Thread Callum Guy via Users
Hi All,

I'm compiling the load_balancer module with some pretty minor changes
however the resulting load_balancer.so is ~500k however the standard
release is 120k - a size increase of 4x.

My question is simple - why is my version so much bigger? Are there
"make" flags that are used for the official releases which I'm failing to
include? I build and run on Almalinux 9, is it a simple matter of the
libraries used on the build host?

Not a show stopper in any way but I wanted to ask the question in case I'm
about to deploy a less performant module etc.

Thanks,

Callum

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Re: [OpenSIPS-Users] Load Balancer Probing Question

2024-04-16 Thread Callum Guy via Users
If the backend servers are both the same instance then this seems to be the
correct behaviour?

I believe the probing is supposed to be a simple SIP response healthcheck
which applies to the destination globally (i.e. 1.2.3.4 is offline), the
groups are just a way of splitting up resources logically for load
balancing purposes.

On Tue, 16 Apr 2024 at 08:31, Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> What OpenSIPS version you have? And as I understand, as configuration, you
> do permanent probing to the destinations and the disabling happens because
> of this probing ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 12.04.2024 05:41, Alexander Perkins wrote:
>
> Hi.  I have an interesting issue.  We have two OpenSIPS servers with load
> balancer (with two different group IDs in the lb table) and we have probing
> set correctly and we are using the event, E_LOAD_BALANCER_STATUS, to
> capture changes to servers that were probed.  But we noticed that we have
> the same server URI listed in the lb table, but with two different group
> IDs, if one of the OpenSIPS servers probes that URI and it does not return,
> then lb disables both groups.  I'd expect it to only disable one group.
>
> My question is how can we tell the LB module to disable the IP, but also
> look for the groupID.  For example, I have a printout of lb_list below.
>
> "uri": "sip:1.2.3.4:5060", "id": 27, "group": 12, "enabled": "no",
> "auto-reenable": "on", "Resources": [ { "name": "vz12", "max": 600, "load":
> 0 } ], "attrs": "0"
>
> AND
> { "uri": "sip:1.2.3.4:5060", "id": 29, "group": 13, "enabled": "no",
> "auto-reenable": "on", "Resources": [ { "name": "vz13", "max": 600, "load":
> 0 } ], "attrs": "0" },
>
> Thank you,
> Alex
>
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Re: [OpenSIPS-Users] load_balancer usage counter

2024-02-05 Thread Callum Guy
Yes, that is correct.

My apologies in advance for the half formed issue - my implementation
with multiple OpenSIPs instances in front of the freeswitches isn't
ideal for the module and I'm just looking for the best solution for
such a scenario. I'm happy to share more details if that's useful but
the main goal is to ensure that channels are used "somewhat" evenly as
a percentage of the maximum sessions reported by each freeswitch
instance.

On Mon, 5 Feb 2024 at 10:58, Bogdan-Andrei Iancu  wrote:
>
> Hi Callum,
>
> Just to be 100% sure I got this right - the exact issue you report here
> is also (in a more detailed way) reported in 3297 HG ticket, right ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>https://www.siphub.com
>
> On 02.02.2024 16:53, Callum Guy wrote:
> > Hi Bogdan,
> >
> > Thanks for taking a look.
> >
> > I'm still working on this and was in the process of writing a second
> > issue to discuss my findings from looking at the module. In essence we
> > can see the profiles potentially being counted multiple times - both
> > in the max_load calculation (lb_update_max_loads) and then again in
> > the balancing code itself (get_dst_load).
> >
> > You are correct that I have multiple OpenSIPs in front of these
> > freeswitch instances. I operate separate registrars and SBCs and these
> > do not replicate data to each other. The main complication arises as I
> > am gradually releasing this onto a busy system - callers are currently
> > allocated to a specific freeswitch server using manually controlled
> > logic and am aiming to use the load balancer to automatically
> > distribute the calls, allowing instances to be easily added and
> > removed without reallocating callers to devices. As the vast majority
> > are not initially enrolled in the load balancer the calls are not
> > being counted and therefore the test accounts all have zero calls in
> > the profile and the call is being distributed to any instance rather
> > than the instance which has the most available channels
> > proportionally.
> >
> > I've just gone ahead and posted the issue
> > (https://github.com/OpenSIPS/opensips/issues/3297) as you've responded
> > - that might add confusion or clarity, I'm not sure yet!
> >
> > I think I need to take some more time to review the module and propose
> > a solution that will work for my scenario but I'm happy to share more
> > details of the setup if you are interested in helping me to find a
> > solution that would work for both me and the community.
> >
> > Many thanks,
> >
> > Callum
> >
> > On Fri, 2 Feb 2024 at 14:37, Bogdan-Andrei Iancu  
> > wrote:
> >> Hi Callum,
> >>
> >> I can confirm the module increments its internal load with each call, so
> >> you are good to go. Still, I do not understand why using the FS
> >> heartbeat here? are the FS servers receiving calls from other
> >> destination than OpenSIPS too ?
> >>
> >> Regards,
> >>
> >> Bogdan-Andrei Iancu
> >>
> >> OpenSIPS Founder and Developer
> >> https://www.opensips-solutions.com
> >> https://www.siphub.com
> >>
> >> On 30.01.2024 16:27, Callum Guy wrote:
> >>> Hi All,
> >>>
> >>> I'm implementing the load_balancer module on a very busy system where
> >>> thousands of calls may arrive in a matter of seconds. The module is
> >>> configured to receive heartbeats every 1 second from many freeswitch
> >>> servers, I have this set as a low value to try and keep OpenSIPs up to
> >>> date with the real call load, as close to real time as possible.
> >>>
> >>> All servers exist in a single group "channels" to keep the initial
> >>> implementation simple. When I kick off the session I use lb_start(1,
> >>> "channels", "rs") and if that destination fails I use lb_is_started()
> >>> and lb_next() to select the next destination.
> >>>
> >>> With the high call rate I'm concerned that the load values acquired 1
> >>> second ago will be used for the entirety of the following second which
> >>> would likely lead to a highly imbalanced load. If 1000 calls arrive in
> >>> that second I need them spread evenly over the freeswitch servers
> >>> which can only happen if OpenSIPs is counting each call as its
> >>> allocated.
>

[OpenSIPS-Users] load_balancer usage counter

2024-01-30 Thread Callum Guy
Hi All,

I'm implementing the load_balancer module on a very busy system where
thousands of calls may arrive in a matter of seconds. The module is
configured to receive heartbeats every 1 second from many freeswitch
servers, I have this set as a low value to try and keep OpenSIPs up to
date with the real call load, as close to real time as possible.

All servers exist in a single group "channels" to keep the initial
implementation simple. When I kick off the session I use lb_start(1,
"channels", "rs") and if that destination fails I use lb_is_started()
and lb_next() to select the next destination.

With the high call rate I'm concerned that the load values acquired 1
second ago will be used for the entirety of the following second which
would likely lead to a highly imbalanced load. If 1000 calls arrive in
that second I need them spread evenly over the freeswitch servers
which can only happen if OpenSIPs is counting each call as its
allocated.

My hope is that the module increments its counters each time a call is
allocated to a destination; however I have been unable to isolate the
line of code which performs this operation so I'm reaching out for
confirmation. I can see the module adding the dialogs to profiles and
that a separate lb_count_call() method is provided for counting calls
but I'm unclear on the exact usage although it certainly encourages me
that this counting is taking place somewhere.

Thanks for reading,

Callum

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Re: [OpenSIPS-Users] regular expression matching question

2024-01-02 Thread Callum Guy
Hi Faheem,

This is a simple case of regex support - I believe OpenSIPs uses Extended
Regular Expressions and not Perl Compatible Regular Expressions.

Swap out \d with the longer [0-9] format and your expression should match.

Happy new year,

Callum

On Tue, 2 Jan 2024 at 11:27, Faheem Muhammad  wrote:

> Hi everyone,
>
> Firstly, I wish you all a Happy New Year with the best wishes for each
> community member and the OpenSIPS project.
>
> I have a question related to regular expressions matching in an IF
> condition.
> There are two identical regular expressions. One RE is matching and one RE
> is not matching the pattern.
>
> *Sample $fU Range:  *+966123456700 to +96612345650
>
> # RE1 - Matching Regular expression
> if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
> xlog("re matched\n")
> }
>
> # RE2  - Non matching Regular expression
> if($fU ~="\+9661234567([0-4]\d|50)"){
> } # RE not matched
>
> Can anyone please suggest what I'm doing wrong?
>
> *OpenSIPS version*: 3.2.3
> OS: *Debian 12*
>
>
> Best regards,
> M. Faheem
>
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Re: [OpenSIPS-Users] Async rest_client timeout

2023-11-02 Thread Callum Guy
I wanted to follow up with confirmation that opensips is behaving
normally here and an issue with delayed HTTP is in fact occurring
outside of the server.

My apologies for any confusion.

Best regards,

Callum

On Tue, 31 Oct 2023 at 16:21, Callum Guy  wrote:
>
> Hi All,
>
> I'm seeing a small number of sessions reporting timeouts for async
> rest_client post requests. These occur at peak times for system load
> and present the following error:
>
> ERROR:rest_client:_resume_async_http_req: connected, but transfer timed out 
> (5s)
>
> The return code is -3 and HTTP response arrives as 0. My web server
> itself shows that the requests complete in <200ms. The
> _resume_async_http_req errors arrive ~400ms after the request was
> issued.
>
> I suspect that something is exhausted (i.e. too many async or curl
> sessions) however I am not aware of any configuration options that
> would help.
>
> The following settings are employed:
>
>  modparam("rest_client", "connection_timeout", 4)
>  modparam("rest_client", "curl_timeout", 5)
>  modparam("rest_client", "max_async_transfers", 250)
>  modparam("rest_client", "ssl_verifyhost", 0)
>
> Can anyone offer advice on where I should be looking to resolve this issue?
>
> version: opensips 3.2.10 (x86_64/linux)
> libcurl-7.76.1-23.el9_2.1.x86_64
>
> Many thanks,
>
> Callum

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[OpenSIPS-Users] Async rest_client timeout

2023-10-31 Thread Callum Guy
Hi All,

I'm seeing a small number of sessions reporting timeouts for async
rest_client post requests. These occur at peak times for system load
and present the following error:

ERROR:rest_client:_resume_async_http_req: connected, but transfer timed out (5s)

The return code is -3 and HTTP response arrives as 0. My web server
itself shows that the requests complete in <200ms. The
_resume_async_http_req errors arrive ~400ms after the request was
issued.

I suspect that something is exhausted (i.e. too many async or curl
sessions) however I am not aware of any configuration options that
would help.

The following settings are employed:

 modparam("rest_client", "connection_timeout", 4)
 modparam("rest_client", "curl_timeout", 5)
 modparam("rest_client", "max_async_transfers", 250)
 modparam("rest_client", "ssl_verifyhost", 0)

Can anyone offer advice on where I should be looking to resolve this issue?

version: opensips 3.2.10 (x86_64/linux)
libcurl-7.76.1-23.el9_2.1.x86_64

Many thanks,

Callum

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Re: [OpenSIPS-Users] Reset branch route

2023-10-10 Thread Callum Guy
Hi Bogdan,

Thanks for your response! I wasn't clear if named routes were supported for
branch/failure routes on 3.2 (I use them fine for reply routes) having
recently had issues with a failure route named 1_LB - perhaps that was due
to the leading integer?

Unfortunately I'm currently heavily invested in a production project so
there isn't much time for testing a non-critical patch. I will be upgrading
to 3.4 before the end of the year though and will certainly try to
implement then.

Similarly thank you for resolving #3216
<https://github.com/OpenSIPS/opensips/issues/3216> however I'll also have
to wait for the next major upgrade before disabling the probes!

I do wish I was more available for testing, my apologies.

Callum

On Tue, 10 Oct 2023 at 11:34, Bogdan-Andrei Iancu 
wrote:

> Hi Callum,
>
> In the old, old times, doing a t_on_branch(0) would have reset it. But
> when moving to named routes, this ability was somehow lost :(...The
> internal function still allows it, but the parameter fixup does not. The
> fix will be really small and quite, if you want to try it - it's just about
> adding the CMD_PARAM_OPT flag to the parameter in the internal definition
> of the function.
>
> Would you be able to run a test if I'll get you a patch ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 10/10/23 12:50 PM, Callum Guy wrote:
>
> Hi All,
>
> I can't find a function to reset a branch route. Instead I'm just creating
> a new empty branch route which I set in the failure route but this seems
> superfluous.
>
> Am I missing something or does it simply not exist?
>
> Callum
>
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[OpenSIPS-Users] Reset branch route

2023-10-10 Thread Callum Guy
Hi All,

I can't find a function to reset a branch route. Instead I'm just creating
a new empty branch route which I set in the failure route but this seems
superfluous.

Am I missing something or does it simply not exist?

Callum

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Re: [OpenSIPS-Users] Configuration file

2023-09-14 Thread Callum Guy
The idea is that you create your own based on your unique
infrastructure requirements.

Learn you must.

https://www.opensips.org/Documentation/Manual-3-3


On Thu, 14 Sept 2023 at 15:21, Prathibha B  wrote:
>
> I need the conf file for opensips 3.3
>
> Sent from Outlook for Android
> 
> From: Prathibha B 
> Sent: Thursday, September 14, 2023 6:23:35 PM
> To: users@lists.opensips.org 
> Subject: Configuration file
>
> Can anyone share the opensips conf file with rtpengine?
>
> Sent from Outlook for Android
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[OpenSIPS-Users] Dialog contact not updating until answer

2023-08-01 Thread Callum Guy
Hi All,

I've observed a behaviour (OpenSIPs 3.2.4) where the contact stored
against an active dialog is not populated until the call is answered.

Using opensips-cli dlg_list I see the following:

Ringing - https://gist.github.com/spacetourist/2502f6b76a95bb2f500fda5291e1c93b
Answered - https://gist.github.com/spacetourist/81b9e19de14a04aeff7f1f7b8965e2e5

During ringing we have:
- "state": 2
- CALLEES > "callee_contact": "",

Once answered we have:
- "state": 4
- CALLEES > "callee_contact": "sip:b2o80qss@77.95.114.132:45656;transport=ws",

This is problematic for me as I'm working on a system to prevent
duplicate registrations from an AoR (sip:6190...@sk-1.rtc.sip.net)
whilst preventing other instances from overruling and taking ownership
of the session by blocking registrations from separate contacts when a
call is ongoing.

The exact scenario where I'm currently seeing errors is when an active
contact (i.e. sip:b2o80qss@77.95.114.132:45656;transport=ws) issues a
re-REGISTER during call setup. I use get_dialogs_by_profile() in
OpenSIPs to track sessions against the AoR and loop through any active
dialogs associated whenever a call or registration comes in. Using
this approach has allowed me to block other devices from creating a
second registration however the lookup is currently failing during the
ringing part of the session as the contact cannot be checked.

Is there a reason why callee_contact cannot be updated immediately
when the session begins? Is this to accommodate branching or similar?
I'll review alternative approaches whilst awaiting a reply however it
would be great if there was a path for me to solve this issue with my
current approach!

Thanks,

Callum

P.S. if this makes more sense as a Github issue then please let me
know, I'm not sure where is best to ask!

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Re: [OpenSIPS-Users] NAT & RTPProxy

2023-04-24 Thread Callum Guy
Excellent, glad you're up and running. Note that you need to manually
implement any late sdp dialogue and handle any calls originating from the
application server using the reverse logic.

Best,

Callum

On Wed, 19 Apr 2023, 19:53 Gavin Murphy,  wrote:

> I have fixed most of the issues I had. I think that in the process of
> testing I mixed up an ie<->ei. Things are set up properly now and audio is
> flowing to the client on my mobile phone.
>
> Thanks for the help.
>
> Gavin
>
> On Wednesday, April 19, 2023 at 02:19:44 PM ADT, Gavin Murphy <
> iiaga...@aim.com> wrote:
>
>
> So I did remember that the rtpproxy_offer and rtpproxy_answer functions
> can take an IP address as a second parameter. That was able to address the
> SDP issue in the 183.
>
> However, I am still not getting audio flowing to the client. I have a
> suspicion that it's related to the SDP on the incoming INVITE having a
> connection address of 192.0.0.2, and rtpproxy being unable to route to that
> address. I though that I could call fix_nated_sdp(2) ahead of the call to
> rtpproxy_offer() in order to adjust the SDP before rtpproxy got hold of it,
> but that results in a mangled connection line with the client address and
> the private address getting concatenated in the resulting SDP (e.g. c: IN
> IP4 72.1.2.3.4192.168.2.251).
>
> Is there any way to address this? I was hoping that the client I'm using
> (ZoiPer) might support STUN/TURN/ICE, but it doesn't appear that it does.
>
> Thanks,
>
> Gavin
>
> On Wednesday, April 19, 2023 at 02:09:13 PM ADT, Gavin Murphy via Users <
> users@lists.opensips.org> wrote:
>
>
> Yes. I am assuming the rtpproxy_offer("ei") should be used when the INVITE
> comes in from the external network, and rtpproxy_answer("ie") should be
> used in the onreply_route to handle the 183 and 200.
>
> Gavin
>
>
>
> Sent via BlackBerry Hub+ Inbox for Android
>
>
>   Original Message
>
>
> From: callum@x-on.co.uk
> Sent: April 19, 2023 1:12 p.m.
> To: iiaga...@aim.com
> Cc: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] NAT & RTPProxy
>
>
> Just to confirm, for a call from the client to the application server
> you should be running rtp_offer("ei") in the main route for a non-LR
> INVITE and rtp_answer("ie") in the on reply route (when handling the
> 183 response).
>
> Is that what you're doing?
>
> On Wed, 19 Apr 2023 at 17:08, Gavin Murphy  wrote:
> >
> > Hi Callum,
> >
> >thanks for the additional info. I removed the -A option from the
> rtpproxy command line but am essentially getting the same result. The
> INVITE does get the address translated correctly, such that it's the
> rtpproxy's internal IP address when the INVITE is forwarded on to the
> application server.
> >
> > However when a 183 comes back from the application server, the IP is
> updated to the internal address, regardless of which i/e combination I use.
> I can't seem to get it to populate the external IP.
> >
> > Thoughts?
> >
> > Gavin
> >
> > On Wednesday, April 19, 2023 at 12:52:04 PM ADT, Callum Guy <
> callum@x-on.co.uk> wrote:
> >
> >
> > Hi Gavin,
> >
> > Its been a while since i used rtpproxy (favouring RTPEngine these
> > days) however I believe the main issue may be the listen option - this
> > should be "-l [internal]/[external]"
> >
> > -l 192.168.2.251/142.1.2.3
> >
> > Once implemented you do indeed need to set ei/ie/ee etc in the offer
> > options to specify which address you wish to present for each scenario
> > you are handling.
> >
> > This also applies to -A however you should only need that if running
> > those addresses behind NAT.
> >
> > Hope that helps to move you forward!
> >
> > Callum
> >
> > On Wed, 19 Apr 2023 at 15:01, Gavin Murphy  wrote:
> > >
> > > Hi Callum,
> > >
> > >yes, I do have rtpproxy set up in bridged mode. It is run using the
> following command (public IPs changed):
> > >
> > > rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.2.251/192.168.2.251 -A
> 142.1.2.3/142.1.2.3 -m 1 -M 10100
> > >
> > > The client source IP seen by opensips is 72.1.2.3, and the SDP has a
> connection IP of 192.0.0.2.
> > >
> > > In opensips' route configuration, when the incoming request is
> determined to be an INVITE, I call rtpproxy_offer(). I've tried passing no
> flags as well as every combination of "i" and "e".

Re: [OpenSIPS-Users] NAT & RTPProxy

2023-04-19 Thread Callum Guy
Just to confirm, for a call from the client to the application server
you should be running rtp_offer("ei") in the main route for a non-LR
INVITE and rtp_answer("ie") in the on reply route (when handling the
183 response).

Is that what you're doing?

On Wed, 19 Apr 2023 at 17:08, Gavin Murphy  wrote:
>
> Hi Callum,
>
>thanks for the additional info. I removed the -A option from the rtpproxy 
> command line but am essentially getting the same result. The INVITE does get 
> the address translated correctly, such that it's the rtpproxy's internal IP 
> address when the INVITE is forwarded on to the application server.
>
> However when a 183 comes back from the application server, the IP is updated 
> to the internal address, regardless of which i/e combination I use. I can't 
> seem to get it to populate the external IP.
>
> Thoughts?
>
> Gavin
>
> On Wednesday, April 19, 2023 at 12:52:04 PM ADT, Callum Guy 
>  wrote:
>
>
> Hi Gavin,
>
> Its been a while since i used rtpproxy (favouring RTPEngine these
> days) however I believe the main issue may be the listen option - this
> should be "-l [internal]/[external]"
>
> -l 192.168.2.251/142.1.2.3
>
> Once implemented you do indeed need to set ei/ie/ee etc in the offer
> options to specify which address you wish to present for each scenario
> you are handling.
>
> This also applies to -A however you should only need that if running
> those addresses behind NAT.
>
> Hope that helps to move you forward!
>
> Callum
>
> On Wed, 19 Apr 2023 at 15:01, Gavin Murphy  wrote:
> >
> > Hi Callum,
> >
> >yes, I do have rtpproxy set up in bridged mode. It is run using the 
> > following command (public IPs changed):
> >
> > rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.2.251/192.168.2.251 -A 
> > 142.1.2.3/142.1.2.3 -m 1 -M 10100
> >
> > The client source IP seen by opensips is 72.1.2.3, and the SDP has a 
> > connection IP of 192.0.0.2.
> >
> > In opensips' route configuration, when the incoming request is determined 
> > to be an INVITE, I call rtpproxy_offer(). I've tried passing no flags as 
> > well as every combination of "i" and "e". In each case the SDP's connection 
> > IP that is passed to the internal server is rtpproxy's advertised address 
> > (142.1.2.3), as it seems to think that the 192.0.0.2 is an internal address 
> > that needs to be converted to the advertised address, even when the "ei" 
> > flags are passed in rtpproxy_offer(). I've also tried passing the "s", "c", 
> > and  "o" flags, with the same fundamental issue each time.
> >
> > On the SDP answer, I am calling rtpproxy_answer("ie"), and that seems to 
> > result in no update to the SDP. The same IP (from the application server) 
> > that comes in is the one that is sent back to the client. It should be the 
> > public IP address (142.1.2.3).
> >
> > Thoughts?
> >
> > Thanks,
> >
> > Gavin
> >
> >
> > On Wednesday, April 19, 2023 at 07:06:54 AM ADT, Callum Guy 
> >  wrote:
> >
> >
> > Hi Gavin,
> >
> > Using an RTP proxy is a good approach, you'll need to set it up in
> > bridge mode so that it is aware of the internal and external addresses
> > so that it can present the public IP to the client and private to the
> > application server.
> >
> > You probably don't need fix_nated_sdp as rtpproxy/rtpengine will do
> > that for you, you'll just need to tell it which address to use which
> > differs for each product.
> >
> > Good luck!
> >
> > Callum
> >
> > On Wed, 19 Apr 2023 at 06:39, Gavin Murphy via Users
> >  wrote:
> > >
> > > Hello,
> > >
> > >I'm trying to set up an instance of opensips to support a testing SIP 
> > > phone calling into my simulated network. The client is running from a 
> > > mobile phone. The connection from the client comes in from the public 
> > > network, but the client sees its own IP as private (192.0.0.2). My test 
> > > network is running on virtual machines on my laptop, and is behind a 
> > > NATed home router, so all of the VMs are in private IP space 
> > > (192.168.x.x). It looks something like this:
> > >
> > > client -> mobile network (NAT) -> home router (NAT) -> opensips -> 
> > > application server
> > >
> > > I am having trouble relaying the media from the network to the client. I 
> >

Re: [OpenSIPS-Users] NAT & RTPProxy

2023-04-19 Thread Callum Guy
Hi Gavin,

Its been a while since i used rtpproxy (favouring RTPEngine these
days) however I believe the main issue may be the listen option - this
should be "-l [internal]/[external]"

-l 192.168.2.251/142.1.2.3

Once implemented you do indeed need to set ei/ie/ee etc in the offer
options to specify which address you wish to present for each scenario
you are handling.

This also applies to -A however you should only need that if running
those addresses behind NAT.

Hope that helps to move you forward!

Callum

On Wed, 19 Apr 2023 at 15:01, Gavin Murphy  wrote:
>
> Hi Callum,
>
>yes, I do have rtpproxy set up in bridged mode. It is run using the 
> following command (public IPs changed):
>
> rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.2.251/192.168.2.251 -A 
> 142.1.2.3/142.1.2.3 -m 1 -M 10100
>
> The client source IP seen by opensips is 72.1.2.3, and the SDP has a 
> connection IP of 192.0.0.2.
>
> In opensips' route configuration, when the incoming request is determined to 
> be an INVITE, I call rtpproxy_offer(). I've tried passing no flags as well as 
> every combination of "i" and "e". In each case the SDP's connection IP that 
> is passed to the internal server is rtpproxy's advertised address 
> (142.1.2.3), as it seems to think that the 192.0.0.2 is an internal address 
> that needs to be converted to the advertised address, even when the "ei" 
> flags are passed in rtpproxy_offer(). I've also tried passing the "s", "c", 
> and  "o" flags, with the same fundamental issue each time.
>
> On the SDP answer, I am calling rtpproxy_answer("ie"), and that seems to 
> result in no update to the SDP. The same IP (from the application server) 
> that comes in is the one that is sent back to the client. It should be the 
> public IP address (142.1.2.3).
>
> Thoughts?
>
> Thanks,
>
> Gavin
>
>
> On Wednesday, April 19, 2023 at 07:06:54 AM ADT, Callum Guy 
>  wrote:
>
>
> Hi Gavin,
>
> Using an RTP proxy is a good approach, you'll need to set it up in
> bridge mode so that it is aware of the internal and external addresses
> so that it can present the public IP to the client and private to the
> application server.
>
> You probably don't need fix_nated_sdp as rtpproxy/rtpengine will do
> that for you, you'll just need to tell it which address to use which
> differs for each product.
>
> Good luck!
>
> Callum
>
> On Wed, 19 Apr 2023 at 06:39, Gavin Murphy via Users
>  wrote:
> >
> > Hello,
> >
> >I'm trying to set up an instance of opensips to support a testing SIP 
> > phone calling into my simulated network. The client is running from a 
> > mobile phone. The connection from the client comes in from the public 
> > network, but the client sees its own IP as private (192.0.0.2). My test 
> > network is running on virtual machines on my laptop, and is behind a NATed 
> > home router, so all of the VMs are in private IP space (192.168.x.x). It 
> > looks something like this:
> >
> > client -> mobile network (NAT) -> home router (NAT) -> opensips -> 
> > application server
> >
> > I am having trouble relaying the media from the network to the client. I 
> > have made various attempts of using the rtpproxy and the matmodule, but 
> > nothing has been successful so far. When using rtpproxy it writes the SDP 
> > going back to the client in the 183 with the internal IP. But if I just use 
> > fix_nated_sdp() the media doesn't go through the proxy server. If I try to 
> > use both (fix_nated_sdp() followed by rtpproxy_answer()), rtpproxy doesn't 
> > properly re-write the SDP (it ends up with a concatenation of the the 
> > private and the public advertised address.
> >
> > Anyone have any advice or experience with this kind of setup?
> >
> > Thanks,
> >
> > Gavin
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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>
>
>
>
>
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Re: [OpenSIPS-Users] NAT & RTPProxy

2023-04-19 Thread Callum Guy
Hi Gavin,

Using an RTP proxy is a good approach, you'll need to set it up in
bridge mode so that it is aware of the internal and external addresses
so that it can present the public IP to the client and private to the
application server.

You probably don't need fix_nated_sdp as rtpproxy/rtpengine will do
that for you, you'll just need to tell it which address to use which
differs for each product.

Good luck!

Callum

On Wed, 19 Apr 2023 at 06:39, Gavin Murphy via Users
 wrote:
>
> Hello,
>
> I'm trying to set up an instance of opensips to support a testing SIP 
> phone calling into my simulated network. The client is running from a mobile 
> phone. The connection from the client comes in from the public network, but 
> the client sees its own IP as private (192.0.0.2). My test network is running 
> on virtual machines on my laptop, and is behind a NATed home router, so all 
> of the VMs are in private IP space (192.168.x.x). It looks something like 
> this:
>
> client -> mobile network (NAT) -> home router (NAT) -> opensips -> 
> application server
>
> I am having trouble relaying the media from the network to the client. I have 
> made various attempts of using the rtpproxy and the matmodule, but nothing 
> has been successful so far. When using rtpproxy it writes the SDP going back 
> to the client in the 183 with the internal IP. But if I just use 
> fix_nated_sdp() the media doesn't go through the proxy server. If I try to 
> use both (fix_nated_sdp() followed by rtpproxy_answer()), rtpproxy doesn't 
> properly re-write the SDP (it ends up with a concatenation of the the private 
> and the public advertised address.
>
> Anyone have any advice or experience with this kind of setup?
>
> Thanks,
>
> Gavin
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Company Registration No. 2578478.

The information in this e-mail is 
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[OpenSIPS-Users] Contact param removal

2023-04-11 Thread Callum Guy
Hi All,

I'm looking for a clean way to de-register a WS location when I
receive a REGISTER with expires set as a contact param:

;expires=0

The issue is that a ul record is stored with the following contact and
the parameters are included in the matching process for registrar
module remove() method:

"AOR": "6190...@rtc.example.net",
"Contacts": [
{
"Contact":
"sip:v2fera4b@rjeh4sth4ien.invalid;transport=ws",
"ContactID": "3803995821952764351",
"Expires": 92,
"Q": "0.8",
"Callid": "6ndu8fnhpm267tnfpj36",
"Cseq": 3,
"User-agent": "SoftPhone Linux Chrome
112.0.0.0 (315e66c2)",
"Received": "sip:1.2.3.4:43010;transport=wss",
"State": "CS_NEW",
"Flags": 0,
"Cflags": "DST_WS SIP_PING_ENABLE TCP_NO_CONN",
"Socket": "wss:172.16.113.200:4433",
"Methods": 5439
}
]
}


To make this work I have demonstrated a method using regex to remove
the expires param and <> brackets, this is acceptable but I wanted to
reach out in case there was a cleaner way!

I only want to remove this individual contact address so a broader
match won't work in this situation.

Thanks,

Callum

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Company Registration No. 2578478.

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Re: [OpenSIPS-Users] tls_mgm domain database configuration

2023-04-07 Thread Callum Guy
Hi James,

It could certainly be clearer!

Here's an extract from my script and some example inserts for a client and
server record.

https://gist.github.com/spacetourist/788ea722901e81d355850842e2b17cda

INSERT INTO opensips_dev.tls_mgm (id, domain, match_ip_address,
match_sip_domain, type, method, verify_cert, require_cert, certificate,
private_key, crl_check_all, crl_dir, ca_list, ca_dir, cipher_list,
dh_params, ec_curve) VALUES (5, 'default', '*', '*', 1, 'TLSv1_2', 0, 0,
0x2D2D2D2D2D424547494E20434...2D2D2D2D0A,
0x2D2D2D2D2D424547494...B45592D2D2D2D2D0A, 0, null, null, null,
'EECDH+AESGCM,EDH+AESGCM,AES256+EECDH,AES256+EDH,!RSA,!SHA',
0x2D2D2D2D2D4245...D2D2D0A, 'secp384r1');

INSERT INTO opensips_dev.tls_mgm (id, domain, match_ip_address,
match_sip_domain, type, method, verify_cert, require_cert, certificate,
private_key, crl_check_all, crl_dir, ca_list, ca_dir, cipher_list,
dh_params, ec_curve) VALUES (15, 'default', '*', '*', 2, 'TLSv1_2', 0, 0,
0x2D2D2D2...D2D0A, 0x2D2D2D2D2...D2D2D0A, 0, null, null, null,
'EECDH+AESGCM,EDH+AESGCM,AES256+EECDH,AES256+EDH,!RSA,!SHA',
0x2D2D2D2D2D42454...2D0A, 'secp384r1');

Hope that helps get you moving!

Callum

On Fri, 7 Apr 2023 at 09:12, James Nicholls via Users <
users@lists.opensips.org> wrote:

> Hi all,
>
> I have an existing opensips 3.3.4 setup that uses modparam to set tls_mgm
> certificates with separate server_domain and client_domain entries. This
> works fine for registration and calling using TLS but I want to be able to
> update certificates with tls_reload so I'm trying to move them to the
> database instead.
>
> The tls_mgm table schema added by opensips-cli has a domain and type
> column. Does "type" mean client/server or is it something else? I have
> tried having separate entries for client/server certs, or combining them
> into one row, but I can't get it to work. Everything seems to result in "no
> TLS client domain found" as below.
>
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]:
> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]:
> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn
> 0x7f3c9f1b5e98
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]:
> DBG:core:tcpconn_destroy: delaying (0x7f3c9f1b5e98, flags 0018) ref = -1 ...
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]:
> ERROR:core:tcp_async_connect: tcp_conn_create failed, closing the socket
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]:
> ERROR:proto_tls:proto_tls_send: async TCP connect failed
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]: ERROR:tm:msg_send:
> send() to (PBX IP):5061 for proto tls/3 failed
> Apr 05 16:02:34 (hostname) /usr/sbin/opensips[22277]:
> ERROR:tm:t_forward_nonack: sending request failed
>
> Example row in the tls_mgm table:
>
>   domain: (SIP branded hostname)
> match_ip_address: (opensips IP):4003
> match_sip_domain: *
> type: 1
>   method: TLSv1_2-
>  verify_cert: 0
> require_cert: 0
>  certificate: -BEGIN CERTIFICATE- [...]
>  private_key: -BEGIN RSA PRIVATE KEY- [...]
>crl_check_all: 0
>  crl_dir: NULL
>  ca_list: /etc/pki/ca-trust/extracted/pem/tls-ca-bundle.pem
>   ca_dir: NULL
>  cipher_list: NULL
>dh_params: NULL
> ec_curve: NULL
>
> Is there any documentation for adding certificates to the tls_mgm table? I
> haven't found anything in the 3.3.x docs, the only examples use modparam.
> Hopefully I have got something really obvious wrong.
>
> Kind regards,
>
> James Nicholls
>
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Re: [OpenSIPS-Users] Check status of routes loading - drouting

2023-03-24 Thread Callum Guy
Hi Daniel,

I believe you're looking for this feature as included since 3.3

https://www.opensips.org/Documentation/Interface-StatusReport-3-3

Enjoy,

Callum

On Fri, 24 Mar 2023 at 14:19, Daniel Zanutti 
wrote:

> Hi
>
> Is there a way to check the status of initial loading of routes, on the
> drouting module?
>
> If routes are being loaded after a cold start, I want to do some alternate
> routing.
>
> Thanks
>
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Re: [OpenSIPS-Users] 3.2.11 vs 3.2.10 authorization

2023-03-14 Thread Callum Guy
Hi Matt,

You are not alone, I have just performed the same update and ran into the
same problem!

No config changes, just an opensips package update on a CentOS 7 server.

My only lead so far is the server referring to a stale nonce in its reply,
this may be a red herring as I reverted so quickly I don't have much data!

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS
4dd4fgrikg9v.invalid;received=99.100.77.236;rport=59260;branch=z9hG4bK2984131
To: ;tag=8dca.7f4e487b9989bf9b0212db1e9a1410a8
From: ;tag=ad8qlo9gaq
Call-ID: 2cjev1ph5gr0m2337tur
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="fr-1.rtc.example.net",
nonce="j9+Qf6N87TeziTRHnWacqtUl3wCf8nvD2I9jm21q+48A", *stale=true*
Content-Length: 0

I'll set up a test lab tomorrow and see if I can get to the bottom of it
and will continue to follow this thread.

Good luck,

Callum

On Tue, 14 Mar 2023 at 13:05, L S  wrote:

> Hi,
>
> We are trying to upgrade from Opensips 3.2.10 to 3.2.11, but we are
> running to an issue with registrations. It's the same server, same opensips
> cfg file, but 3.2.10 allows/authorizes the registrations, but 3.2.11
> returns 401 Unauthorized.
>
> The code that checks the credentials is:
>
> if (is_method("REGISTER|SUBSCRIBE")) {
>$avp(password)="xyz";
>if (!pv_www_authorize("")) {
>www_challenge("");
>exit;
>   };
> consume_credentials();
> }
>
> Again it's same code, the same physical server. What might be causing this?
>
> Thanks,
> Matt
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Re: [OpenSIPS-Users] rtpengine_offer on REINVITE

2023-03-13 Thread Callum Guy
Hi All,

Following up on this following another media server crash.

Since the previous email it was discovered that on occasion RTPEngine would
be delayed in responding to OpenSIPs and therefore a second RTP server was
tried triggering the failure scenario (unnecessary DTLS renegotiation
mid-call).

This time there was no detectable delay, a call was simply established
through OpenSIPs however a RE-INVITE arriving at the 1 minute mark engaged
a separate RTP server. Can anyone explain the mechanism by which a RTP
server which was selected during the dialog setup is attached to the dialog
data for subsequent requests, and identify any possible weakness where this
would not be recorded? In over 99.99% of situations it certainly selects
the RTP server previously engaged however there must be a weakness for this
to occur.

Any thoughts on debugging/mitigation would be appreciated! The affected
system is running 3.2.4 - happy to upgrade if there has been any work in
this area!

Thanks,

Callum

On Wed, 22 Feb 2023 at 21:40, Callum Guy  wrote:

> Hi All,
>
> I operate OpenSIPs in front of a bank of FreeSWITCH instances and I'm
> currently seeing an issue with FreeSWITCH crashing during a SRTP DTLS
> renegotiation triggered by a RE-INVITE.
>
> I've tracked this down to a WSS registrar which is issuing
> rtpengine_offer(...) for the INVITE and any subsequent RE-INVITE. This
> happens thousands of times a day without issue however on occasion OpenSIPs
> sends that RE-INVITE request to a different rtpengine server from the pool.
> This writes in the new RTP proxy IP and initiates a new ICE
> negotiation, the client fails to handle this and the DTLS negotiation
> breaks down - RTPEngine warns "*Received invalid STUN packet from
> 1.8.4.1:58184 <http://1.8.4.1:58184>: MESSAGE_INTEGRITY attribute missing*"
> and FreeSWITCH segfaults.
>
> Can anyone advise on what might be causing OpenSIPs to pick an unrelated
> RTP instance for the dialog? Any ideas for preventing that would be
> appreciated! I could conceivably save the socket after the initial offer
> against the dialog and use that for additional offers but hopefully that
> can be avoided. I use RTPEngine only as a proxy, perhaps I should simply
> prevent RE-INVITE's from sending additional offers so I only have one
> offer/answer/delete per call?
>
> Thanks,
>
> Callum
>

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[OpenSIPS-Users] rtpengine_offer on REINVITE

2023-02-22 Thread Callum Guy
Hi All,

I operate OpenSIPs in front of a bank of FreeSWITCH instances and I'm
currently seeing an issue with FreeSWITCH crashing during a SRTP DTLS
renegotiation triggered by a RE-INVITE.

I've tracked this down to a WSS registrar which is issuing
rtpengine_offer(...) for the INVITE and any subsequent RE-INVITE. This
happens thousands of times a day without issue however on occasion OpenSIPs
sends that RE-INVITE request to a different rtpengine server from the pool.
This writes in the new RTP proxy IP and initiates a new ICE
negotiation, the client fails to handle this and the DTLS negotiation
breaks down - RTPEngine warns "*Received invalid STUN packet from
1.8.4.1:58184 : MESSAGE_INTEGRITY attribute missing*"
and FreeSWITCH segfaults.

Can anyone advise on what might be causing OpenSIPs to pick an unrelated
RTP instance for the dialog? Any ideas for preventing that would be
appreciated! I could conceivably save the socket after the initial offer
against the dialog and use that for additional offers but hopefully that
can be avoided. I use RTPEngine only as a proxy, perhaps I should simply
prevent RE-INVITE's from sending additional offers so I only have one
offer/answer/delete per call?

Thanks,

Callum

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[OpenSIPS-Users] Memory Leak

2023-02-13 Thread Callum Guy
Hi All,

*(originally sent back in December with an image embedded which possibly
got the mail rejected!)*

I'm experiencing a memory leak on an OpenSIPs 3.2.10 instance (was 3.2.4,
upgrades to 3.2.9 and 3.2.10 did not resolve) which is acting as a WebRTC
proxy - this is running on AlmaLinux 9.0

The config is not much different from existing CentOS 7 servers which are
running 3.2.4 so I'm wondering if the issue is a difference in an
underlying library? Both use tls_openssl.so and the AlmaLinux install was
self packaged.

Here is the output of the memory dump:
https://gist.github.com/spacetourist/9a49ee7b55f2e5f0264f56adc10bf0fc

One item stands out as a strong candidate:
*opensips[287301]:   314792712 : 19101 x [ssl/record/ssl3_buffer.c:
os_malloc, line 119]*

The affected server is running AlmaLinux 9 with the following openssl
utilities:

openssl-pkcs11-0.4.11-7.el9.x86_64
xmlsec1-openssl-1.2.29-9.el9.x86_64
openssl-libs-3.0.1-23.el9_0.x86_64
openssl-3.0.1-23.el9_0.x86_64
openssl-devel-3.0.1-23.el9_0.x86_64

Our CentOS 7 server has:

openssl-libs-1.0.2k-19.el7.x86_64
openssl-devel-1.0.2k-19.el7.x86_64
openssl-1.0.2k-19.el7.x86_64

The symptom is memory creep of around 90MB/day for ~300k call sessions. Any
ideas would be appreciated, I'm on a weekly restart cycle to clear the
memory which I'd rather not implement as a cron!

Thanks,

Callum

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Re: [OpenSIPS-Users] Opensips 3.2 Ratecacher

2022-11-28 Thread Callum Guy
Added the worlds simplest PR for you!

https://github.com/OpenSIPS/opensips/pull/2958

On Mon, 28 Nov 2022 at 11:17, Callum Guy  wrote:

> Looks like a typo in the module!
>
>
> https://github.com/OpenSIPS/opensips/blob/master/modules/rate_cacher/rate_cacher.c#L127
>
> I suggest trying "cients_hash_size" until patched :)
>
> On Mon, 28 Nov 2022 at 08:19, Wadii ELMAJDI | Evenmedia <
> wa...@evenmedia.fr> wrote:
>
>> Hello,
>>
>> I was able to install ratecacher module and make it work.
>> But editing the clients hash size parameter generates an error.
>> “ERROR:core:set_mod_param_regex: parameter  not found
>> in module ”
>> however vendors_hash_size is correctly identified as a rate_cacher
>> parameter
>> Everything works when removing clients one
>> I am running opensips :
>>
>> version: opensips 3.2.9 (x86_64/linux)
>>
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>
>> poll method support: poll, epoll, sigio_rt, select.
>>
>> main.c compiled on  with gcc 10
>>
>> ### RATECACHER module
>>
>> loadmodule "rate_cacher.so"
>>
>> modparam("rate_cacher", "clients_db_url", "mysql://DB_STRING/opensips")
>>
>> modparam("rate_cacher", "clients_db_table", "rc_clients")
>>
>> modparam("rate_cacher", "rates_db_url", "mysql://DB_STRING/opensips")
>>
>> modparam("rate_cacher", "rates_db_table", "rc_ratesheets")
>>
>> modparam("rate_cacher", "vendors_db_url", "mysql://DB_STRING/opensips")
>>
>> modparam("rate_cacher", "vendors_db_table", "rc_vendors")
>>
>> modparam("rate_cacher", "vendors_hash_size", 1024)
>>
>> modparam("rate_cacher", "clients_hash_size", 1024)
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>>
>

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Re: [OpenSIPS-Users] Opensips 3.2 Ratecacher

2022-11-28 Thread Callum Guy
Looks like a typo in the module!

https://github.com/OpenSIPS/opensips/blob/master/modules/rate_cacher/rate_cacher.c#L127

I suggest trying "cients_hash_size" until patched :)

On Mon, 28 Nov 2022 at 08:19, Wadii ELMAJDI | Evenmedia 
wrote:

> Hello,
>
> I was able to install ratecacher module and make it work.
> But editing the clients hash size parameter generates an error.
> “ERROR:core:set_mod_param_regex: parameter  not found
> in module ”
> however vendors_hash_size is correctly identified as a rate_cacher
> parameter
> Everything works when removing clients one
> I am running opensips :
>
> version: opensips 3.2.9 (x86_64/linux)
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
>
> poll method support: poll, epoll, sigio_rt, select.
>
> main.c compiled on  with gcc 10
>
> ### RATECACHER module
>
> loadmodule "rate_cacher.so"
>
> modparam("rate_cacher", "clients_db_url", "mysql://DB_STRING/opensips")
>
> modparam("rate_cacher", "clients_db_table", "rc_clients")
>
> modparam("rate_cacher", "rates_db_url", "mysql://DB_STRING/opensips")
>
> modparam("rate_cacher", "rates_db_table", "rc_ratesheets")
>
> modparam("rate_cacher", "vendors_db_url", "mysql://DB_STRING/opensips")
>
> modparam("rate_cacher", "vendors_db_table", "rc_vendors")
>
> modparam("rate_cacher", "vendors_hash_size", 1024)
>
> modparam("rate_cacher", "clients_hash_size", 1024)
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Re: [OpenSIPS-Users] Check the User-Agent for registered user

2022-10-17 Thread Callum Guy
Hi Vlad,

Thank you for your recommendation, I can't use branch routes for my
implementation as I'm making the decision at registration time -
"User-Agent A" captures registrations from "User-Agent B", but not the
other way around.

It looks like attr_avp should get populated when using is_registered so
that will work well for me - I'm only expecting a single active
registration per AoR so this should be a good fit.

Many thanks for your help, I'll let you know if I run into any issues!

Callum

On Sat, 15 Oct 2022 at 09:54, Vlad Paiu  wrote:

> Hello,
>
> At registration time you could save the User Agent in the attributes per
> registration [1] and at INVITE time you can check that in branch_route and
> make your decision there.
>
> [1]
> https://opensips.org/html/docs/modules/3.2.x/registrar.html#param_attr_avp
>
>
> Regards,
> Vlad
> On 14.10.2022 17:02, Callum Guy wrote:
>
> Hi All,
>
> I'm working on a project that requires me to evaluate the user agent of a
> registered contact before making a decision on a current registration
> attempt.
>
> Is there a method to do this natively? If not, what is the best approach?
>
> I haven't found anything yet, the current options i'm evaluating are to:
>
> 1. Use mi_script module to allow me to pull the contact record
> via ul_show_contact
> 2. Use rest_get to pull this data via a web service
>
> Neither is ideal for my requirements but both would do the job, I'm hoping
> someone in the community has a cleaner solution before I start!
>
> Best regards,
>
> Callum
>
>
>
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X-on
is a trading name of Storacall 
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Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, 
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[OpenSIPS-Users] Check the User-Agent for registered user

2022-10-14 Thread Callum Guy
Hi All,

I'm working on a project that requires me to evaluate the user agent of a
registered contact before making a decision on a current registration
attempt.

Is there a method to do this natively? If not, what is the best approach?

I haven't found anything yet, the current options i'm evaluating are to:

1. Use mi_script module to allow me to pull the contact record
via ul_show_contact
2. Use rest_get to pull this data via a web service

Neither is ideal for my requirements but both would do the job, I'm hoping
someone in the community has a cleaner solution before I start!

Best regards,

Callum

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[OpenSIPS-Users] Event E_UL_CONTACT_DELETE delayed

2022-06-01 Thread Callum Guy
Hi Team,

I just wanted to report an incorrect behaviour for a comment from the dev
team.

I'm operating an RTC registrar on 3.2.4 and defining that each device (web
console) can have a maximum of one active contact, overridden by any new
registration for that AoR. The previously active devices are notified via
an internal API call whenever the E_UL_CONTACT_DELETE event is raised,
indicating that the session has been closed.

This works great most of the time however in certain situations such as
when a call has recently ended on one device the contact delete event is
often delayed by a minute or so. I have resolved this issue by changing
the sql_write_mode for usrloc to "write-through" rather than the more
performant "write-back" setting I typically rely on.

Is there any technical reason why we can't have the event generated at the
time that the old contact is flagged as deleted rather than the time that
this is persisted to the database, as appears to be the current behaviour?
I'm comfortable running in write-through mode for now but if there is a
scope to fix this I think raising the event in a timely manner seems the
correct thing to do.

Many thanks,

Callum

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[OpenSIPS-Users] Event on DE-REGISTRATION

2022-03-03 Thread Callum Guy
Hi All,

I have configured my registrar with max_contacts 1, allowing
subsequent registrations from that contact to overwrite.

I am looking to intercept the de-registration and send a message to the
losing contact. Ideally I would use the existing SIP connection to send it
before termination, the client devices are softphones under my control, but
I suppose the TCP connection to the previous device may have been severed
by the time the event is raised (I can test this when I get there but
advice is always welcome).

Event capture is straightforward using E_UL_CONTACT_DELETE however I am
wondering if there is any way to generate SIP towards the leaving contact
at this point? t_new_request() is not available in event_route and I wanted
to reach out to the community for ideas before building an external system
to do this (i.e. rest_post to a simple HTTP client).

Is there a neat way to do this within the config script or am I already on
the best-fit path here?

Many thanks,

Callum

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Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0

2022-02-25 Thread Callum Guy
Excellent, another string to the bow!

On Thu, 24 Feb 2022 at 10:28, HS  wrote:

> Callum.
>
> Thanks a lot for the help. I seem to have been able to compile the module
> and add it. Now working on port 443 on apache2 + Opensips. The fun never
> ends :D
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Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0

2022-02-23 Thread Callum Guy
Manually compiling modules is relatively straightforward if you've got the
build tools installed. I can't provide a copy as I'm using CentOS.

The process is simply to clone the repo on your target system (or
equivalent), checkout the release tag and then compile as shown below
(after adjusting to target the module you want). A single make command
followed by a copy of the .so file into your modules root.

🔥 make modules=modules/tls_wolfssl modules
🔥 sudo cp modules/tls_wolfssl/tls_wolfssl.so /usr/lib64/opensips/modules/

Good luck, if this is outside your scope then maybe someone else can find
you the module package you need!


On Wed, 23 Feb 2022 at 13:07, HS  wrote:

> Thanks a lot Callum.
>
> Haven't compiled a module ever (yet) and upgrade scares me :) Anyone have
> a copy that I can use please or any other suggestions?
>
> Or would the module have changed significantly - what if I use the 3.1
> build version?
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Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0

2022-02-23 Thread Callum Guy
Depending on your OS version, looks like 3.0 repo has been removed:

https://apt.opensips.org/dists/buster/

If so you'll either need to find a copy or compile it yourself (or upgrade!)

On Wed, 23 Feb 2022 at 09:49, HS  wrote:

> Apologies, I have already tried:
>
> apt-get install opensips-wss-module
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Re: [OpenSIPS-Users] WebRTC Implementation on Opensips 3.0

2022-02-23 Thread Callum Guy
You'll probably want to ensure you have the following packages installed,
presuming you are using the provided repo's:

opensips-wss-module
opensips-tls-module
opensips-tlsmgm-module


On Wed, 23 Feb 2022 at 09:08, HS  wrote:

> Dear all.
>
> Trying to following the tutorials, however, when I restart opensips after
> adding proto_wss. to the config file - I get an error message saying the
> module can't be loaded and it isn't in the modules folder.
>
> I recall having downloaded a module previously, but have no recollection
> of it. Any pointers please?
>
> Thanks again.
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[OpenSIPS-Users] iOS 15 WebRTC

2022-01-25 Thread Callum Guy
Hi All,

I'm wondering if anyone in the community has encountered the iOS 15 WebRTC
bug as reported here:

https://www.wowza.com/community/t/ios-15-webrtc-problem/93868/4
https://developer.apple.com/forums/thread/689293

The bug causes INVITES to fail to be digested by opensips with a log
message as shown below:

opensips[217976]: ERROR:proto_wss:ws_process: Made 4 read attempts but
message is not complete yet - closing connection

Worth noting is that REGISTER messages still arrive so as mentioned in the
article is does appear related to fragmentation of the messages being sent
through, perhaps failing to conclude this for reassembly but I can't make
sense of the network trace.

My client device is using SIP.js and I can confirm that it continues to
work on Android and other iOS versions. I'm currently running opensips
3.0.5 and would happily upgrade to the latest version if there was anything
worth trying or wolfssl has any potential to resolve this issue!

Thanks,

Callum

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Re: [OpenSIPS-Users] Opensips 3.2 install issue with WolfSSL in CentOS7

2021-11-29 Thread Callum Guy
You could try to install the required version?

https://centos.pkgs.org/7/okey-x86_64/automake-1.14-1.el7.x86_64.rpm.html

On Mon, 29 Nov 2021 at 11:52, Virendra Bhati via Users <
users@lists.opensips.org> wrote:

> Dear Team,
> I am not able to install Opensips 3.2. I am facing an issue with Automake
> requiring version 1.14.1but CentOS 7.x have 1.13.X
>
> Please suggest the best way to overcome this problem.
>
> Linking avpops.so
> make[2]: Leaving directory `/usr/src/bhati/opensips-3.2/modules/avpops'
>
>
> make[2]: Entering directory
> `/usr/src/bhati/opensips-3.2/modules/tls_wolfssl'
> make[2]: Leaving directory
> `/usr/src/bhati/opensips-3.2/modules/tls_wolfssl'
> make[2]: Entering directory
> `/usr/src/bhati/opensips-3.2/modules/tls_wolfssl'
> configure: error: cannot find install-sh, install.sh, or shtool in
> build-aux "."/build-aux
> make[2]: *** [lib/lib/libwolfssl.a] Error 1
> make[2]: Leaving directory
> `/usr/src/bhati/opensips-3.2/modules/tls_wolfssl'
> make[1]: *** [modules] Error 2
> make[1]: Leaving directory `/usr/src/bhati/opensips-3.2'
>
>
> --
> Regards
> Virendra Bhati
>
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Re: [OpenSIPS-Users] concatenate an integer and a string

2021-10-18 Thread Callum Guy
Try this:

$var(result) = "$avp(id_asterisk)|$avp(codecs)";

On Mon, 18 Oct 2021 at 10:10, Alain Bieuzent  wrote:

> Hi All,
>
>
>
> I’m’ trying to concatenate two avp result to one string
>
> The first avp(id_asterisk)  is an integer,
>
> the second $avp(codecs) is a string.
>
>
>
> When I try to concatenate with $var(result) = $avp(id_asterisk) + "|" +
> $avp(codecs);) I got this error message :
>
>
>
> ERROR:core:eval_elem: invalid numeric operands
>
> WARNING:core:do_assign: no value in right expression at
> /usr/local//etc/opensips/opensips.cfg:881
>
>
>
> Someone can help me?
>
>
>
> Regards,
>
> Alain
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Re: [OpenSIPS-Users] TLS_MGM module with DB(MYSQL)

2021-05-28 Thread Callum Guy
I'm a bit lost with regards to your setup/intentions but I think you might
be misusing those config values - if in doubt try the vanilla
implementation using an unmodified table and limited module parameters,
start small and build!

Of concern:

modparam("tls_mgm", "domain_col", "xx.i3clogic.com") < is that really
your column name??
modparam("tls_mgm", "match_ip_address_col", "xx.xx36.64:443") < is that
really your column name??

I'm wondering if you have interpreted those as default values?

https://opensips.org/html/docs/modules/3.0.x/tls_mgm.html#idp5640928

Initially just try with the value "default" for domain - thats a special
catch all value:

[image: image.png]


On Fri, 28 May 2021 at 12:59, Saurabh Chopra  wrote:

> Hi Opensips Team/ Callum,
>
> Thanks for the response, I successfully inserted the content of my
> files(certificate) in mysql DB. But when I try to run the script the
> opensips is crashing now with errors.
>
> May 28 11:45:59  /usr/local/sbin/opensips[1268]:
> CRITICAL:db_mysql:wrapper_single_mysql_real_query: driver error (1064): You
> have an error in your SQL syntax; check the manual that corresponds to your
> MySQL server version for the right syntax to use near
> '.36.64:443,match_sip_domain,type,method,0,0,certificate,private_key,crl_check_al'
> at line 1
> May 28 11:45:59  /usr/local/sbin/opensips[1268]: ERROR:core:db_do_query:
> error while submitting query - [select 
> id,.i3clogic.com,x.x.36.64:443,match_sip_domain,type,method,0,0,certificate,private_key,crl_check_all,crl_dir,ca_list,ca_dir,cipher_list,dh_params,ec_curve
> from tls_mgm ]
> May 28 11:45:59   /usr/local/sbin/opensips[1268]: ERROR:tls_mgm:load_info:
> DB query failed - retrieve valid connections
> May 28 11:45:59  /usr/local/sbin/opensips[1268]: ERROR:tls_mgm:load_info:
> Unable to load domains info from DB
> May 28 11:45:59 /usr/local/sbin/opensips[1268]: ERROR:core:init_mod:
> failed to initialize module tls_mgm
> May 28 11:45:59  /usr/local/sbin/opensips[1268]: ERROR:core:main: error
> while initializing modules
>
> Given these parameters in the config script.
>
> loadmodule "tls_mgm.so"
> modparam("tls_mgm", "db_url", "mysql://root:x@localhost/opensips")
> modparam("tls_mgm", "db_table", "tls_mgm")
> modparam("tls_mgm", "domain_col", "xx.i3clogic.com")
> modparam("tls_mgm", "match_ip_address_col", "xx.xx36.64:443")
> modparam("tls_mgm", "tls_method_col", "method")
> modparam("tls_mgm", "verify_cert_col", "0")
> modparam("tls_mgm", "require_cert_col", "0")
> modparam("tls_mgm", "certificate_col", "certificate")
> modparam("tls_mgm", "private_key_col", "private_key")
> modparam("tls_mgm", "crl_check_all_col", "crl_check_all")
> modparam("tls_mgm", "crl_dir_col", "crl_dir")
> modparam("tls_mgm", "ca_list_col", "ca_list")
>
> Kindly help in this matter.
>
> Best Regards
> Saurabh Chopra
> +918861979979
>
>
> On Wed, May 26, 2021 at 5:06 PM Callum Guy  wrote:
>
>> You need to insert the certificate contents into the table rather than a
>> file, to my understanding - you'd probably want to convert the certs to PEM
>> format prior to doing this.
>>
>>
>>
>>
>>
>> On Tue, 25 May 2021 at 14:20, Saurabh Chopra 
>> wrote:
>>
>>> Hi Opensips Team/Razvan,
>>>
>>> I am using the TLS_MGM module to validate my certificate file through a
>>> database (mysql) but getting the below error.
>>>
>>> May 25 12:41:28 ip-20-0-36-64 /usr/local/sbin/opensips[16297]:
>>> NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none
>>> May 25 12:41:28 ip-20-0-36-64 /usr/local/sbin/opensips[16297]: 
>>> *ERROR:tls_mgm:load_certificate_db:
>>> Unable to load certificate from buffer*
>>> May 25 12:41:28 ip-20-0-36-64 /usr/local/sbin/opensips[16297]:
>>> WARNING:tls_mgm:init_tls_domains: Failed to init TLS domain 'default',
>>> skipping...
>>>
>>> The data i have put in DB is depicted below:-
>>>
>>> insert into tls_mgm
>>> (id,domain,match_ip_address,match_sip_domain,type,method,verify_cert,require_cert,certificate,private_key,crl_check_all,crl_dir,ca_list,ca_dir,
>>> cipher_list,dh_params,ec_curve) values
>>> ('1','default','20.0.xx.xx:44

Re: [OpenSIPS-Users] TLS_MGM module with DB(MYSQL)

2021-05-26 Thread Callum Guy
You need to insert the certificate contents into the table rather than a
file, to my understanding - you'd probably want to convert the certs to PEM
format prior to doing this.





On Tue, 25 May 2021 at 14:20, Saurabh Chopra  wrote:

> Hi Opensips Team/Razvan,
>
> I am using the TLS_MGM module to validate my certificate file through a
> database (mysql) but getting the below error.
>
> May 25 12:41:28 ip-20-0-36-64 /usr/local/sbin/opensips[16297]:
> NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none
> May 25 12:41:28 ip-20-0-36-64 /usr/local/sbin/opensips[16297]: 
> *ERROR:tls_mgm:load_certificate_db:
> Unable to load certificate from buffer*
> May 25 12:41:28 ip-20-0-36-64 /usr/local/sbin/opensips[16297]:
> WARNING:tls_mgm:init_tls_domains: Failed to init TLS domain 'default',
> skipping...
>
> The data i have put in DB is depicted below:-
>
> insert into tls_mgm
> (id,domain,match_ip_address,match_sip_domain,type,method,verify_cert,require_cert,certificate,private_key,crl_check_all,crl_dir,ca_list,ca_dir,
> cipher_list,dh_params,ec_curve) values
> ('1','default','20.0.xx.xx:443','','1','SSLv23','1','0','2021.der','.der','0','','x2021.der','','ALL','','');
>
> *.der* files are the binary file for certificate file, private key file
> and ca_list .  However the same is working fine through the configuration
> script.
>
> Can anybody confirm what I am doing wrong ?
>
>
> Best Regards
> Saurabh Chopra
> +918861979979
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Re: [OpenSIPS-Users] TLS to UDP, record route

2021-05-17 Thread Callum Guy
subst_uri only works on the request uri, try again with subst()!


On Mon, 17 May 2021 at 08:58, Miha via Users 
wrote:

> Hello
>
> i need a little help how to chnage RR in responses to UDP GW (requestes
> goes via TLS to MS teams).
>
> So in reply i have like this:
>  RECORD-ROUTE:
> ,
> .
>
> But i should have like this:  RECORD-ROUTE:
> ,
> .
>
> I tried to do it like:  
> subst_uri('/mtsbc.test.com:5061;transport=tls/xxx.xxx.xxx.:5060/i');
> but it does not match.
>
>
> thank you
> miha
>
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Re: [OpenSIPS-Users] Opensips 3.0 + iOS Push Notifications

2021-04-14 Thread 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.
>>
>> ___
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>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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Re: [OpenSIPS-Users] TLS certificate reloading

2021-04-14 Thread 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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[OpenSIPS-Users] TLS certificate reloading

2021-04-14 Thread 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] Question regarding expire time of mid-register

2021-03-17 Thread Callum Guy
You should try setting the min expires value, the documentation states that
any values lower than this will be overridden with the minimum you define!

https://opensips.org/html/docs/modules/3.0.x/registrar.html#param_min_expires

On Wed, 17 Mar 2021 at 06:43, Jeffrey Zhao  wrote:

> Dear Team
>
> For legacy Cisco IP phones, the SIP registration expiration time must be
> set to less than 120 seconds, and there is no way to control the timer on
> phone end.
> It must be controlled on the SIP registrar end(Expires value set to 120 in
> registration reply SIP message).
>
> I am not sure whether there is a way in OpenSIPS script to update the
> 'Expires' header(base on user agent string, if Cisco Phone, then Expires:
> 120) in the onreply_route.
>
> Which parameter is writtable to achieve the target?
>
> Thank you so much!
>
>
> REGISTER sip:test.com SIP/2.0
> Via: SIP/2.0/UDP 10.167.6.60:5060;branch=z9hG4bK3593290796
> From: "Ooi, Liyong" ;tag=30622762
> To: "Ooi, Liyong" 
> Call-ID: 0_2497201195@10.167.6.60
> CSeq: 1 REGISTER
> Contact: 
> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
> SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
> Max-Forwards: 70
> User-Agent: Cisco CP-7975
> Expires: 900
> Allow-Events: talk,hold,conference,refer,check-sync
> Content-Length: 0
>
>
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Re: [OpenSIPS-Users] Handling mobile switch from wifi to data

2021-03-17 Thread Callum Guy
My advice would be to get stuck in, I had plenty of questions after reading
the release blog posts but once I started implementing it all made sense.
Anything specific concerning/confusing you at this stage?

Good luck!

On Wed, 17 Mar 2021 at 09:46, Mark Allen  wrote:

> Thanks Johan - I can see how that would be the case.
>
> I've read through the blog posts about setting up PN which are very
> helpful, though I've still got some questions. Hopefully, the video from
> the 2020 Summit will fill in the gaps.
>
> On Wed, 17 Mar 2021 at 09:35, Johan De Clercq  wrote:
>
>> sending push before you send the invite solves many many problems.
>>
>> Op wo 17 mrt. 2021 om 08:54 schreef Mark Allen :
>>
>>> OK - thanks for that. I'll give it a try
>>>
>>> On Tue, 16 Mar 2021 at 12:22, Johan De Clercq  wrote:
>>>
 Implement push.

 Op di 16 mrt. 2021 om 13:15 schreef Mark Allen :

> We are using OpenSIPS 3.1 as a mid_registrar in AOR Throttling mode.
> When a mobile moves from data to wifi or vice versa, my understanding is
> that the registration on OpenSIPS becomes stale (because the phone has a
> new IP address). Does this mean that the mobile phone is not registered
> until registration expiry is reached on the mobile app and a new REGISTER
> is sent? If so, is there anything that can be done to mitigate this?
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>
 ___
 Users mailing list
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Re: [OpenSIPS-Users] Changing the expire parameter in the Contact field

2021-03-12 Thread Callum Guy
Hi Joseph,

I haven't fully digested your scenario however you may have some luck using
the nathelper function fix_nated_contact - presuming NAT is not an issue.

https://opensips.org/html/docs/modules/3.1.x/nathelper.html#func_fix_nated_contact

Otherwise you'll probably be able to achieve this using general
transformations, should be quite straightforward if you only have a single
Contact:

https://www.opensips.org/Documentation/Script-Tran-3-1#toc60

Hope that helps,

Callum


On Fri, 12 Mar 2021 at 03:55, Joseph Barrero  wrote:

> Is there a way to replace the expires parameter in a Contact header within
> a reply route?
>
> The issue I'm trying to solve is the following.
>
> I have two OpenSIPS servers, a primary and a backup.  The OpenSIPS server
> is performing mid-registrar duties for Freeswitch.  However, when I take
> the primary OpenSIPS server down for maintenance and bring it back online,
> Freeswitch still has the other registration active.  When the UA tries to
> register again on the primary OpenSIPS server, the mid-register module
> sends a request over to Freeswitch, and Freeswitch returns two contacts;
> one contact with the modparam("mid_registrar", "outgoing_expires", 600)
> value and the other contact with an expiration of whatever value it happens
> to be on (in this example it is 600 - time elapsed) from the registration
> created by the secondary OpenSIPS server.
>
> The problem is that the expiration parameter in the 200 Ok reply to the UA
> from OpenSIPS on a successful registration has the second expiration value
> instead of the shorter expiration value set in modparam("mid_registrar",
> "max_expires", 60).
>
> The result, is that the UA goes from 60-second registrations to whatever
> the second expiration value had at the time (a value between 0 and 599).
> When the second expiration is higher than 60 seconds, the UA's registration
> is expired on OpenSIPS for up to 599 seconds.
>
> What I'd like to do is simply change the expire parameter in the contact
> header to 60 seconds on the reply route.
>
> Any help will be greatly appreciated.
>
> Thank you,
> Joe Barrero
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Re: [OpenSIPS-Users] Handling missing BYEs

2021-03-10 Thread Callum Guy
Might be a case of adding in a record_route() call in the appropriate
place, hard to say without a trace :)

On Wed, 10 Mar 2021 at 10:42, Mark Allen  wrote:

> Hi Callum - thanks for that!
>
> Yes - it's generating the BYE at the Linux end but not sending it to the
> remote OpenSIPS IP address but rather to an address on the local LAN -
> hence the problem. Thanks for your help.
>
> Cheers,
>
> Mark
>
>
>
> On Wed, 10 Mar 2021 at 09:26, Callum Guy  wrote:
>
>> Hi Mark,
>>
>> It sounds like you may be having issues with the proxy not keeping itself
>> in path for certain call scenarios.
>>
>> Are you able to provide a SIP trace and/or opensips config? Also if
>> you're running Blink on a Linux system, can you get a SIP trace there to
>> see if the BYE is being generated and sent somewhere else?
>>
>> Callum
>>
>> On Tue, 9 Mar 2021 at 16:32, Mark Allen  wrote:
>>
>>> I'm seeing some odd behaviour which also leads into a broader question
>>>
>>> I have a NATed Blink app running on Linux on my home LAN. It connects to
>>> an OpenSIPS 3.1 server in on our office LAN which is a mid-registrar for an
>>> Asterisk server. I'm running sngrep on the OpenSIPS box to watch the
>>> traffic.
>>>
>>> If I call from the Blink app to another extension it all connects and
>>> audio works correctly. If I hangup in Blink, a BYE is sent via OpenSIPS to
>>> Asterisk - all good so far.
>>> If I call from another extension to the Blink app it all connects and
>>> audio works correctly. However, if I hangup in the Blink app, no BYE is
>>> sent to OpenSIPS.
>>>
>>> In most situations, this is merely inconvenient because, with the loss
>>> of RTP traffic, Asterisk generates a BYE after about 35 seconds to tidy
>>> everything up. However, if I'm doing an attended transfer, the BYE is
>>> needed to exit the call so that the transfer completes successfully. At the
>>> moment, if I hangup in the Blink app, there's a wait of 35 seconds until
>>> Asterisk creates the BYE before the call transfer is completed.
>>>
>>> While I'm mostly using Blink, I've seen similar failures to send BYEs
>>> from other apps. Does OpenSIPS offer anything that could help with this?
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
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>> Riduna Park, Melton IP12 1QT.*
>>
>> X-on is a trading name of Storacall Technology Ltd a limited company
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>> X-on immediately on +44(0)333 332  and delete the
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>> 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 attachments
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Re: [OpenSIPS-Users] Handling missing BYEs

2021-03-10 Thread Callum Guy
Hi Mark,

It sounds like you may be having issues with the proxy not keeping itself
in path for certain call scenarios.

Are you able to provide a SIP trace and/or opensips config? Also if you're
running Blink on a Linux system, can you get a SIP trace there to see if
the BYE is being generated and sent somewhere else?

Callum

On Tue, 9 Mar 2021 at 16:32, Mark Allen  wrote:

> I'm seeing some odd behaviour which also leads into a broader question
>
> I have a NATed Blink app running on Linux on my home LAN. It connects to
> an OpenSIPS 3.1 server in on our office LAN which is a mid-registrar for an
> Asterisk server. I'm running sngrep on the OpenSIPS box to watch the
> traffic.
>
> If I call from the Blink app to another extension it all connects and
> audio works correctly. If I hangup in Blink, a BYE is sent via OpenSIPS to
> Asterisk - all good so far.
> If I call from another extension to the Blink app it all connects and
> audio works correctly. However, if I hangup in the Blink app, no BYE is
> sent to OpenSIPS.
>
> In most situations, this is merely inconvenient because, with the loss of
> RTP traffic, Asterisk generates a BYE after about 35 seconds to tidy
> everything up. However, if I'm doing an attended transfer, the BYE is
> needed to exit the call so that the transfer completes successfully. At the
> moment, if I hangup in the Blink app, there's a wait of 35 seconds until
> Asterisk creates the BYE before the call transfer is completed.
>
> While I'm mostly using Blink, I've seen similar failures to send BYEs from
> other apps. Does OpenSIPS offer anything that could help with this?
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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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.

*From April 1st 2021 our office address will change to: Units 22-24 Riduna 
Park, Melton IP12 1QT.*

X-on
is a trading name of Storacall Technology Ltd 
a limited company registered in
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Re: [OpenSIPS-Users] Push Notification without registration timers

2021-02-15 Thread Callum Guy
Quick update, since sending my post I have made a small amount of
progress using event_routing.

My prototype is currently doing the following:

- INVITE arrives targeting unregistered device
- Execute launch(rest_post()) to initiate the notification in another process
- Execute async( wait_for_event("E_UL_AOR_INSERT", "aor=UA@DOM", 10),
RESUME_ON_REG);
- (request exits)
- RESUME_ON_REG called when registration arrives
- send provisional reply
- lookup() (ignoring any advice that the user isn't ready without another PN!)
- t_relay() to forward the INVITE to the new registration

This feels like a workable solution although there is some scope for a
race condition as I'd like to book the event waiter before sending the
POST request however I'm not sure that's possible?

I'm hopeful that I can progress and solve the remaining issues shortly..

Happy SIPing all.

On Mon, 15 Feb 2021 at 15:42, Callum Guy  wrote:
>
> Hi All,
>
> Running 3.1 release.
>
> I'm trying to implement a proxy which *only* supports PN-enabled
> devices however I'm running into some implementation issues for my use
> case.
>
> My project is targeting a very low usage subscriber base (users may be
> idle for months) so I'd like to disable the timer based registration
> refresh and simply collect the PNID at time of registration for later
> use. I don't want to lose devices (which may be turned off) through
> contact purging when they don't respond to PN refresh requests. Is
> this possible? If not I'm happy to manually implement an external
> system for tracking PNID to subscriber relationships if that is
> necessary/simpler. Crucially I want to preserve a list of all the
> PNID's associated with a subscriber (i.e. a user with multiple
> devices) for a long time.
>
> The main problem I am running into is transaction parking without
> using the default module implementation. If the location table does
> not contain contact information for these devices I can't use the
> default park/resume features provided by the registrar module even
> though I can load PRID's from another source. At the point of dial I
> want to park the INVITE and dispatch all the notifications however I
> get the impression that these features are under the hood. Is this
> possible at the script level without the magic behind the registrar
> lookup() method or pn_process_purr()?
>
> Furthermore this project runs on a private network so I'm not sure its
> strictly necessary to generate and track PURR values for active
> sessions, is there any reason I can't just store the PRID against the
> dialog and use that to generate PN mid-dialog? Every call passing
> through this proxy will be for PN-enabled devices so this seems like a
> reasonable implementation even if it doesn't conform to RFC 8599?
>
> Thanks,
>
> Callum

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X-on
is a trading name of Storacall Technology Ltd a limited company 
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[OpenSIPS-Users] Push Notification without registration timers

2021-02-15 Thread Callum Guy
Hi All,

Running 3.1 release.

I'm trying to implement a proxy which *only* supports PN-enabled
devices however I'm running into some implementation issues for my use
case.

My project is targeting a very low usage subscriber base (users may be
idle for months) so I'd like to disable the timer based registration
refresh and simply collect the PNID at time of registration for later
use. I don't want to lose devices (which may be turned off) through
contact purging when they don't respond to PN refresh requests. Is
this possible? If not I'm happy to manually implement an external
system for tracking PNID to subscriber relationships if that is
necessary/simpler. Crucially I want to preserve a list of all the
PNID's associated with a subscriber (i.e. a user with multiple
devices) for a long time.

The main problem I am running into is transaction parking without
using the default module implementation. If the location table does
not contain contact information for these devices I can't use the
default park/resume features provided by the registrar module even
though I can load PRID's from another source. At the point of dial I
want to park the INVITE and dispatch all the notifications however I
get the impression that these features are under the hood. Is this
possible at the script level without the magic behind the registrar
lookup() method or pn_process_purr()?

Furthermore this project runs on a private network so I'm not sure its
strictly necessary to generate and track PURR values for active
sessions, is there any reason I can't just store the PRID against the
dialog and use that to generate PN mid-dialog? Every call passing
through this proxy will be for PN-enabled devices so this seems like a
reasonable implementation even if it doesn't conform to RFC 8599?

Thanks,

Callum

-- 





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   **  |  Coronavirus 
*


THE 
ITSPA AWARDS 2020  AND Best ITSP - 
Mid Market, Best Software and Best Vertical Solution are trade marks of the 
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X-on
is a trading name of Storacall Technology Ltd a limited company 
registered in
England and Wales.

Registered Office : Avaland House, 110 
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Herts, HP3 9SD. Company Registration 
No. 2578478.

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Re: [OpenSIPS-Users] [BLOG] Exploring SSL/TLS libraries for OpenSIPS 3.2

2021-02-11 Thread Callum Guy
Thanks for that Vlad, I always learn something from these article
releases and this was no exception 🐺

On Thu, 11 Feb 2021 at 12:35, Vlad Patrascu  wrote:
>
> Hello everyone,
>
> Check out this new blog post [1] that recounts our process of searching
> for and choosing a SSL/TLS library to use in OpenSIPS 3.2, as an
> alternative to the problematic OpenSSL.
>
> [1]
> https://blog.opensips.org/2021/02/11/exploring-ssl-tls-libraries-for-opensips-3-2/
>
> Cheers,
>
> --
> Vlad Patrascu
> OpenSIPS Developer
> http://www.opensips-solutions.com
>
>
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   **  |  Coronavirus 
*


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.

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.

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X-on immediately on +44(0)333 332  and delete the
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Re: [OpenSIPS-Users] EXEC not working

2021-02-04 Thread Callum Guy
Maybe permissions or similar - have you tested the example from the docs?

exec("ls -l", , "$var(out)", "$var(err)", "$avp(env)");
xlog("The output is $var(out)\n");
xlog("Received the following error\n$var(err)");


On Thu, 4 Feb 2021 at 10:36, Dragomir Haralambiev  wrote:
>
> Hello,
>
> I try to run external program like this
>
> exec("/test/opensips.pl",,,"$var(err)");
> xlog("Received the following error $var(err)");
>
> Here is log record:
> Received the following error 
>
> The program is not running.
>
> Where is the reason?
>
> Best regards:
> Dragomir
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   **  |  Coronavirus 
*


THE 
ITSPA AWARDS 2020  AND Best ITSP - 
Mid Market, Best Software and Best Vertical Solution are trade marks of the 
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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.

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Re: [OpenSIPS-Users] Limit Call per Second

2021-01-25 Thread Callum Guy
An alternative option would be to leverage cachedb_local and opensips-cli
to implement your list of accounts and rate limits. It has the advantage of
using the internal opensips cache service and is probably your most high
performance option, with the CLI you can automate data refreshes using
basic cron scripts or anything you fancy!

https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc4
https://www.opensips.org/Documentation/Interface-CoreMI-3-0#toc15
https://www.opensips.org/Documentation/Tutorials-KeyValueInterface

On Sun, 24 Jan 2021 at 14:56, Ovidiu Sas  wrote:

> There are no db queries, unless explicitly enabled for distributed
> rate limiting, in which case memcache or redis can be used.
>
> -ovidiu
>
> On Sun, Jan 24, 2021 at 1:57 AM Schneur Rosenberg
>  wrote:
> >
> > Maybe use memcache, it's global and it won't create a bottleneck like a
> database query would.
> >
> > Scott (Schneur)
> >
> > On Thu, Jan 21, 2021, 05:43 Ovidiu Sas  wrote:
> >>
> >> Take a look at the ratelimit module:
> >> https://opensips.org/docs/modules/3.1.x/ratelimit.html
> >>
> >> -ovidiu
> >>
> >> On Wed, Jan 20, 2021 at 10:01 PM Alexander Perkins
> >>  wrote:
> >> >
> >> > Hi All.  Is there a way to limit the calls per second by pulling the
> information from a database? I was looking at the call_control module and
> that seems to be a global value. but I would need something that I can
> control at the call level (I will be handling different accounts that have
> different limits).
> >> >
> >> > Thanks, All.  Any help is appreciated.
> >> >
> >> > Alex
> >> > ___
> >> > Users mailing list
> >> > Users@lists.opensips.org
> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >>
> >>
> >> --
> >> VoIP Embedded, Inc.
> >> http://www.voipembedded.com
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://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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> http://www.voipembedded.com
>
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Re: [OpenSIPS-Users] Managing drouting data

2020-10-30 Thread Callum Guy
I believe the DB followed by a dr_reload is your only option...

On Fri, 30 Oct 2020 at 15:49, Mark Farmer  wrote:

> Hi everyone
>
> I am looking for a way to manage routes etc in drouting without using
> opensips-cp.
> I was hoping to find a MI function to add/remove routes but there only
> seems to be dr_number_routing which just returns an existing route.
>
> Has anyone done this somehow? Interacting with the database directly seems
> less than ideal.
>
> Many thanks
> Mark.
>
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Re: [OpenSIPS-Users] rtpengine_delete not returning immediately

2020-10-27 Thread Callum Guy
Have you double checked it's not a firewall issue on the SIP proxy?

The transport is typically UDP so there is a fair chance it's blocked.

On Tue, 27 Oct 2020 at 19:05, John Quick  wrote:

> Using OpenSIPS v2.4.8 and the latest release of rtpengine. The same was
> also
> happening with some older releases. I upgraded to see if it would help.
>
> When OpenSIPS receives BYE, the script calls rtpengine_delete() but I can
> see from xlog messages before and after the call that the function is
> taking
> up to 5 seconds to return.
> This is also reflected in the logs on the server where rtpengine is hosted:
>
> Oct 27 18:06:12 vps684757 rtpengine: INFO: [d2s4qf6thm2tmtg40khg]: Received
> command 'delete' from 83.120.14.234:56677
> Oct 27 18:06:12 vps684757 rtpengine: INFO: [d2s4qf6thm2tmtg40khg]:
> Scheduling deletion of call branch '5cclfhu59e' (via-branch '') in 12
> seconds
> Oct 27 18:06:12 vps684757 rtpengine: INFO: [d2s4qf6thm2tmtg40khg]: Replying
> to 'delete' from 83.120.14.234:56677 (elapsed time 0.000192 sec)
> Oct 27 18:06:13 vps684757 rtpengine: INFO: Detected command from
> 83.120.14.234:56677 as a duplicate
> Oct 27 18:06:14 vps684757 rtpengine: INFO: Detected command from
> 83.120.14.234:56677 as a duplicate
> Oct 27 18:06:15 vps684757 rtpengine: INFO: Detected command from
> 83.120.14.234:56677 as a duplicate
>
> It looks as if OpenSIPS is failing to detect/recognise the reply from
> rtpengine. This only happens for the delete command - I have no problem
> with
> offer and answer.
>
> I do not generally see any error messages in the opensips logs, but once it
> did report timeout waiting for a response from rtpengine.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
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Re: [OpenSIPS-Users] Max async transfers - opensips-cli feature request

2020-08-28 Thread Callum Guy
Hi Liviu,

Sounds great to me, my particular scenario was due to failing
elsewhere in our systems so we just need (eventual) timeouts - this
would be an ideal fix from my perspective.

Hope all is well - looking forward to the virtual Summit!

Callum

On Fri, 28 Aug 2020 at 12:25, Liviu Chircu  wrote:
>
> On 20.08.2020 17:46, Callum Guy wrote:
> > I presume this is fallout from a recent network issue so I plan to
> > restart all instances during a quiet period which I'm sure will
> > resolve it.
> >
> > Is there any change that visibility of the async stack could be added
> > to opensips-cli - if I can't see whats in there then I can't put
> > mitigations in place to prevent it. The option to clean all requests
> > older than X seconds would be very helpful right now :)
> >
> > Thanks,
> >
> > Callum
>
> Hey, Callum!
>
> Maybe an equivalent feature would be to introduce a timeout for
> async(rest_xxx()) operations?  The latter feature request is already
> being kept track of in issue #1838 [1], and will definitely make it into
> the next release.
>
> Best regards,
>
> [1]: https://github.com/OpenSIPS/opensips/issues/1838
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
> OpenSIPS Summit 2020 Distributed
>www.opensips.org/events/Summit-2020Distributed
>
>
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[OpenSIPS-Users] Max async transfers - opensips-cli feature request

2020-08-20 Thread Callum Guy
Hi All,

Noticed these in my logs suggesting that something has jammed in the
async tables:

2020-08-20T15:42:24.145805+01:00 FR-P-SIPSBC-1 opensips[240129]:
WARNING:rest_client:get_multi: max async transfers! (250)
2020-08-20T15:42:24.146290+01:00 FR-P-SIPSBC-1 opensips[240129]:
WARNING:rest_client:start_async_http_req: failed to get a multi
handle, doing a blocking transfer

I presume this is fallout from a recent network issue so I plan to
restart all instances during a quiet period which I'm sure will
resolve it.

Is there any change that visibility of the async stack could be added
to opensips-cli - if I can't see whats in there then I can't put
mitigations in place to prevent it. The option to clean all requests
older than X seconds would be very helpful right now :)

Thanks,

Callum

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Re: [OpenSIPS-Users] OpenSIPS 3.0.2 - 'IF' statement doesn't evaluating correctly

2020-08-20 Thread Callum Guy
Excellent, I'm not clear on exactly how the type conversion system
works on OpenSIPs but I'm glad this resolved the confusion!

From my few visits to the source I recall that most of the script
variables are custom types which have both a string and integer
property but I'm not clear how it chooses which one to use for
conditional statements etc

On Thu, 20 Aug 2020 at 07:35, Igor Pavlov  wrote:
>
> Thanks a lot for {s.int} ! I forgot that my dialog value is string. 
> Transforming to int helped.
>
>
> ср, 19 авг. 2020 г. в 02:33, Callum Guy :
>>
>> Have you matched the dialog before running this check? Just wondering
>> if one of those values is stale, do the durations match up with
>> reality for the example calls?
>>
>> Also maybe rule out type issues with $(dlg_val(dialog_min_time){s.int})
>>
>> On Tue, 18 Aug 2020 at 17:35, Igor Pavlov  wrote:
>> >
>> > Hi all, I have found the strange behavior of evaluating boolean value in 
>> > 'if' statement. Here is the part of my routing script, it handles the BYE 
>> > msg (I removed real logic and left only xlog).
>> >
>> > if ($DLG_lifetime < $dlg_val(dialog_min_time)) {
>> > xlog("L_DBG","[$ci] Dialog lifetime less then dialog_min_time ; 
>> > duration: $DLG_lifetime ; $dlg_val(dialog_min_time)");
>> > } else {
>> > xlog("L_DBG","[$ci] Dialog lifetime greater then dialog_min_time ; 
>> > duration: $DLG_lifetime ; $dlg_val(dialog_min_time)");
>> > }
>> >
>> > The '$dlg_val(dialog_min_time)' setup during INVITE handling, after 
>> > create_dialog().
>> >
>> > Under load I see that '$DLG_lifetime < $dlg_val(dialog_min_time)' is not 
>> > evaluating correctly. Here is some logs:
>> >
>> > opensips[1589]: [56276459-0-1637911800@1.1.1.48] BYE from 2.2.2.143:5060, 
>> > dialog_min_time: 30, duration: 161, status: 5
>> > opensips[1589]: [56276459-0-1637911800@1.1.1.48] Dialog lifetime less then 
>> > dialog_min_time ; duration: 161 ; 30
>> >
>> > Here is DLG_lifetime = 161 and $dlg_val(dialog_min_time) = 30 (161 < 30 
>> > ???)
>> >
>> > Another example:
>> >
>> > opensips[1590]: [58514636-0-1638386270@1.1.1.48] BYE from 1.1.1.50:5060, 
>> > dialog_min_time: 15, duration: 1212, status: 5
>> > opensips[1590]: [58514636-0-1638386270@1.1.1.48] Dialog lifetime less then 
>> > dialog_min_time ; duration: 1212 ; 15
>> >
>> > Here is DLG_lifetime = 1212 and $dlg_val(dialog_min_time) = 15 (1212 < 15 
>> > ???)
>> >
>> > My OpenSIPS version is:
>> >
>> > version: opensips 3.0.2 (x86_64/linux)
>> > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
>> > Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
>> > MAX_URI_SIZE 1024, BUF_SIZE 65535
>> > poll method support: poll, epoll, sigio_rt, select.
>> > git revision: 3a8f6f137
>> > main.c compiled on 22:11:53 Jul 20 2020 with gcc 8
>> >
>> > --
>> >
>> > Best regards,
>> > Igor Pavlov
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
>>
>>
>>
>>
>>
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>>
>>
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>> registered in
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>>
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>>
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>> X-on immediately on +44(0)333 332  an

Re: [OpenSIPS-Users] OpenSIPS 3.0.2 - 'IF' statement doesn't evaluating correctly

2020-08-18 Thread Callum Guy
Have you matched the dialog before running this check? Just wondering
if one of those values is stale, do the durations match up with
reality for the example calls?

Also maybe rule out type issues with $(dlg_val(dialog_min_time){s.int})

On Tue, 18 Aug 2020 at 17:35, Igor Pavlov  wrote:
>
> Hi all, I have found the strange behavior of evaluating boolean value in 'if' 
> statement. Here is the part of my routing script, it handles the BYE msg (I 
> removed real logic and left only xlog).
>
> if ($DLG_lifetime < $dlg_val(dialog_min_time)) {
> xlog("L_DBG","[$ci] Dialog lifetime less then dialog_min_time ; duration: 
> $DLG_lifetime ; $dlg_val(dialog_min_time)");
> } else {
> xlog("L_DBG","[$ci] Dialog lifetime greater then dialog_min_time ; 
> duration: $DLG_lifetime ; $dlg_val(dialog_min_time)");
> }
>
> The '$dlg_val(dialog_min_time)' setup during INVITE handling, after 
> create_dialog().
>
> Under load I see that '$DLG_lifetime < $dlg_val(dialog_min_time)' is not 
> evaluating correctly. Here is some logs:
>
> opensips[1589]: [56276459-0-1637911800@1.1.1.48] BYE from 2.2.2.143:5060, 
> dialog_min_time: 30, duration: 161, status: 5
> opensips[1589]: [56276459-0-1637911800@1.1.1.48] Dialog lifetime less then 
> dialog_min_time ; duration: 161 ; 30
>
> Here is DLG_lifetime = 161 and $dlg_val(dialog_min_time) = 30 (161 < 30 ???)
>
> Another example:
>
> opensips[1590]: [58514636-0-1638386270@1.1.1.48] BYE from 1.1.1.50:5060, 
> dialog_min_time: 15, duration: 1212, status: 5
> opensips[1590]: [58514636-0-1638386270@1.1.1.48] Dialog lifetime less then 
> dialog_min_time ; duration: 1212 ; 15
>
> Here is DLG_lifetime = 1212 and $dlg_val(dialog_min_time) = 15 (1212 < 15 ???)
>
> My OpenSIPS version is:
>
> version: opensips 3.0.2 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, 
> F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll, sigio_rt, select.
> git revision: 3a8f6f137
> main.c compiled on 22:11:53 Jul 20 2020 with gcc 8
>
> --
>
> Best regards,
> Igor Pavlov
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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THE 
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is a trading name of Storacall Technology Ltd a limited company 
registered in
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Registered Office : Avaland House, 110 
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Herts, HP3 9SD. Company Registration 
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Re: [OpenSIPS-Users] Intercept 477 on TCP closed

2020-08-17 Thread Callum Guy
Thanks Johan, exactly what the doctor ordered!

Most appreciated


On Mon, 17 Aug 2020 at 14:26, Johan De Clercq  wrote:
>
> use t_relay wih 0x2 option.
>
> On Mon, Aug 17, 2020, 15:16 Callum Guy  wrote:
>>
>> Hi All,
>>
>> Using OpenSIPs 3.0.3
>>
>> I'm dealing with a client device with a faulty network, they are using a 
>> softphone WebRTC client and the TCP connections disappear sporadically.
>>
>> When the media server issues a RE-INVITE session timer OpenSIPs discovers 
>> the closed TCP connection and returns 477 to the media server. In this case 
>> the media server is FreeSWITCH which promptly ignores the non-standard 
>> session timer response and the call hangs. I want to close the call 
>> immediately in this situation as there is no way I can see to reestablish 
>> the connection. The call is typically bridged on the media server so the 
>> other call leg is left dangling indefinitely so I need to find a way to kill 
>> the session.
>>
>> Reading RFC4028 the only response codes that should trigger a BYE are 
>> 408/481 so my aim is to intercept the 477 and alter it. Unfortunately I have 
>> been unable to intercept it within my OpenSIPs config as the messages don't 
>> hit failure_route or similar. Am I missing a trick? Is there somewhere where 
>> these can be intercepted or should I be looking into a solution on the media 
>> server instead?
>>
>> Many thanks,
>>
>> Callum
>>
>>
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[OpenSIPS-Users] Intercept 477 on TCP closed

2020-08-17 Thread Callum Guy
Hi All,

Using OpenSIPs 3.0.3

I'm dealing with a client device with a faulty network, they are using a
softphone WebRTC client and the TCP connections disappear sporadically.

When the media server issues a RE-INVITE session timer OpenSIPs discovers
the closed TCP connection and returns 477 to the media server. In this case
the media server is FreeSWITCH which promptly ignores the
non-standard session timer response and the call hangs. I want to close the
call immediately in this situation as there is no way I can see to
reestablish the connection. The call is typically bridged on the media
server so the other call leg is left dangling indefinitely so I need to
find a way to kill the session.

Reading RFC4028  the only
response codes that should trigger a BYE are 408/481 so my aim is to
intercept the 477 and alter it. Unfortunately I have been unable to
intercept it within my OpenSIPs config as the messages don't hit
failure_route or similar. Am I missing a trick? Is there somewhere where
these can be intercepted or should I be looking into a solution on the
media server instead?

Many thanks,

Callum

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Re: [OpenSIPS-Users] Single OpenSIPS instance as a full proxy for several softswitches

2020-08-13 Thread Callum Guy
Yes this is very much achievable and a common topology.

You'll need to look into media proxy software (RTPEngine/rtpproxy/etc)
as well but this can run on the same device if sufficient resources
are available.

OpenSIPs is a great choice!


On Wed, 12 Aug 2020 at 15:00, Adam Obuchowski  wrote:
>
> I would like to deploy several asterisk boxes on  a docker network (swarm or 
> kubernetes).  Looking for something that may act as a proxy (for sip and 
> media), basing on a  domain.
> For example, if i configure my device to register with
> a...@asterisk1.mydomain.com,the sip proxy should route all the dialog to 
> asterisk instance 1, if i setup same device to a...@asterisk2.mydomain.com , 
> the proxy should route the dialog to instance 2 and so on.
> Instances may work in docker network with DNS names.
> The proxy should work same way for everything, registar, invites, BLF and 
> also proxy media.
> In other words only proxy should be exposed to the world, while asterisk 
> instances would remain only on docker network. Kind of fake multitenancy.  
> Would this be achievable with OpenSIPS ?
>
> Kind regards,
>
> --
> Adam Obuchowski
>
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Re: [OpenSIPS-Users] Sylk Mobile

2020-08-11 Thread Callum Guy
That'd be great, hope this comes together!

On Tue, 11 Aug 2020 at 14:40, Maxim Sobolev  wrote:

> Interesting work Adrian! Any chance you can be interested in coming over
> to our SIP Chronicles videocast to talk about it and perhaps do a live
> demo? 😀
>
> https://www.youtube.com/playlist?list=PL-U7hOT8zFXoSMgHLfVj_CX4MvFjD2gcj
>
> Let me know, we don't have anyone booked yet for this coming Saturday.
>
> Thanks!
>
> -Max
>
> -Max
>
> On Thu., Aug. 6, 2020, 11:07 p.m. Adrian Georgescu, 
> wrote:
>
>> Hi,
>>
>> We just published Sylk Mobile, a react-native mobile client for Android
>> and iOS.
>>
>> The server side is running OpenSIPS + Janus.
>>
>>
>> https://lists.ag-projects.com/pipermail/sipbeyondvoip/2020-August/003469.html
>>
>> Enjoy!
>> Adrian
>>
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Re: [OpenSIPS-Users] Opensips & media server

2020-07-21 Thread Callum Guy
It is not questions I seek, but answers to the great mystery.

On Tue, 21 Jul 2020 at 07:57, Alexey Kazantsev via Users <
users@lists.opensips.org> wrote:

> Hello,
> So what is the question?
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Re: [OpenSIPS-Users] rtpengine documentation

2020-05-18 Thread Callum Guy
Hi Rob,

I'm interested to follow your thread to hear more about this, I have
found that some flags are valid yet undocumented during initial setup
of some RTC compatable proxies.

Two in particular: DTLS-passive and SDES-disable both of which appear
to influence behaviour of RTPEngine in accordance with the RTPEngine
documentation.

YMMV however it might help you to try this format as you proceed with testing.

Good luck!

Callum


On Sat, 16 May 2020 at 20:38, Robert Dyck  wrote:
>
> I am wanting to convert my config/script to use rtpengine instead of rtpproxy.
> I think it would better deal with webrtc. After looking at some examples I
> found, I see a couple of parameters that are not mentioned in the opensips
> documentation. First there is the offer/answer option ice=force-relay and
> secondly DTLS=passive.
>
> Are these options obsolete/deprecated/intentionally omitted?
>
> On the subject of DTLS I noticed that when I use ice=force in offer and answer
> rtpengine sends new DTLS fingerprints to the parties. I appears to operate as
> back-to-back DTLS agent. I know this because both UAs sent SHA-256
> fingerprints but they received SHA-1 fingerprints. This may have worked but
> one UA will only accept SHA-256 and it drops the call.
>
> The documentation does not mention that the ice= option can influence DTLS.
>
> Regards, Rob
>
>
>
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Re: [OpenSIPS-Users] OpenSIPS 3.1 release - the final countdown

2020-05-14 Thread Callum Guy
OpenSIPs 3.1 and Nasa DM-2 (https://www.nasa.gov/specials/dm2/) on the same
day? Awesome!

On Thu, 14 May 2020 at 10:02, Bogdan-Andrei Iancu 
wrote:

> Hi all,
>
> We planned an ambitious roadmap [1] for OpenSIPS 3.1, but we were even
> more ambitious by trying to complete it.  It was a long way,
> brainstorming, designing, implementing or extended an awesome feature
> set that will make OpenSIPS an appealing platform for Class 5 services.
>
> But we are almost done and the date is set:
>  27th of May, the date of the 3.1 beta release
>
> To check the status, visit anytime the roadmap page and scroll at the
> end for the status table - yes, we do update it in realtime ;)
>
> [1] https://www.opensips.org/Development/Opensips-3-1-Planning
>
> Best regards,
>
> --
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>
>
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Re: [OpenSIPS-Users] Can't seem to get python module to work

2020-05-13 Thread Callum Guy
Do you have the module loaded? Have you
installed opensips-python-module.x86_64? Confirm in your module path that
it exists!

loadmodule "python.so"

On Wed, 13 May 2020 at 03:18, Gordon Yeong  wrote:

> Anyone know what's going on?
>
> Gordon
>
>
> On Tue, 12 May 2020 at 17:10, Gordon Yeong  wrote:
>
>> hi guys
>>  I managed to get opensips to work with the perl module.
>> I now need to port what I have to python 3.
>>
>>
>> I can't seem to start opensips 3.0 because I have the following error:
>>
>>
>> May 12 16:59:51 [47532] ERROR:core:set_mod_param_regex: no module
>> matching python found
>> May 12 16:59:51 [47532] Traceback (last included file at the bottom):
>> May 12 16:59:51 [47532]  0.
>> /opt/opensips/etc/opensips/opensips-Enum-Python.cfg
>> May 12 16:59:51 [47532] CRITICAL:core:yyerror: parse error in
>> /opt/opensips/etc/opensips/opensips-Enum-Python.cfg:137:20-21: Parameter
>>  not found in module  - can't set
>> May 12 16:59:51 [47532]
>> May 12 16:59:51 [47532] loadmodule "enum.so"
>> May 12 16:59:51 [47532] modparam("python", "script_name", "/tmp/test.py")
>> May 12 16:59:51 [47532] ^~
>> May 12 16:59:51 [47532]
>> May 12 16:59:51 [47532] ### Routing Logic 
>> May 12 16:59:51 [47532] CRITICAL:core:yyerror: parse error in
>> /opt/opensips/etc/opensips/opensips-Enum-Python.cfg:158:36-37: unknown
>> command , missing loadmodule?
>> May 12 16:59:51 [47532] CRITICAL:core:yyerror: parse error in
>> /opt/opensips/etc/opensips/opensips-Enum-Python.cfg:159:54-55: unknown
>> command , missing loadmodule?
>> May 12 16:59:51 [47532] ERROR:core:parse_opensips_cfg: bad config file (3
>> errors)
>> May 12 16:59:51 [47532] ERROR:core:main: failed to parse config file
>> /opt/opensips/etc/opensips/opensips-Enum-Python.cfg
>> May 12 16:59:51 [47532] NOTICE:core:main: Exiting
>>
>> 1) the first issue is "ERROR:core:set_mod_param_regex: no module matching
>> python found".
>> I find this really weird because I referred to
>> https://opensips.org/docs/modules/3.0.x/python.html and it says in 1.2.1.
>> that no loading is required.
>>
>> 2) Additionally, I am using the  line "modparam("python", "script_name",
>> "/tmp/test.py")" to load my test python functions as per the same
>> documentation in section 1.3.1.
>> Yet, it errors out, "CRITICAL:core:yyerror: parse error in
>> /opt/opensips/etc/opensips/opensips-Enum-Python.cfg:137:20-21: Parameter
>>  not found in module  - can't set".
>>
>> Any ideas?
>>
>> Thank you
>>
>>
>> Regards,
>> Gordon Yeong
>> perl, python3, elixir, rails, php = joy
>>
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THE 
ITSPA AWARDS 2020  AND Best ITSP - 
Mid Market, Best Software and Best Vertical Solution are trade marks of the 
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is a trading name of Storacall Technology Ltd a limited company 
registered in
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Registered Office : Avaland House, 110 
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Herts, HP3 9SD. Company Registration 
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Re: [OpenSIPS-Users] Load testing

2020-05-12 Thread Callum Guy
Sippy cup has some good media generation capabilities, still using sipp
under the hood.

https://mojolingo.github.io/sippy_cup/



On Mon, 11 May 2020 at 18:45, Tomi Hakkarainen  wrote:

> I agree
>
> BR, Tomi
>
> On 11. May 2020, at 19.48, johan  wrote:
>
> hmmm sipp with your own rtp files.
> On 11/05/2020 18:19, miha- via Users wrote:
>
> Hi
>
> What is best tool for load testing that can generate also RTP?
>
> Tnx
>
> miha
>
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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.


X-on
is a trading name of Storacall Technology Ltd a limited company 
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Registered Office : Avaland House, 110 
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[OpenSIPS-Users] Crash on BYE (free after free)

2020-05-11 Thread Callum Guy
Hi All,

I've had a couple of incidents this week where OpenSIPs crashed during BYE
processing, in both cases the SIP logs show BYE arriving from both sides of
a call within a couple of milliseconds - these overlap and result in 481's
being both sent to our media server and being received from our media
server as shown in the image below (UA <> OpenSIPs <> Media server):

[image: image.png]

It is during the processing of the final 481 that OpenSIPs hits the bug.
The error shown in the logs is:

opensips[234594]: CRITICAL:core:fm_free: freeing already freed shm pointer
(0x7f56fefaaa18), first free: (null): (null)(0) - aborting!

User logs show that the 481 was being processed for an established dialog
and executes an onreply route followed by a failure route. The failure
route issues t_check_trans and rtpengine_delete (which may have already
been removed during BYE processing). Is there likely to be anything that I
can do at the script level if this is a race condition?

This server is using the official CentOS 3.0.1 release, happy to upgrade if
there have been any relevant fixes since that version.

Thanks,

Callum

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[OpenSIPS-Users] TLS SNI Checks

2020-05-07 Thread Callum Guy
Hi All,

Some of our clients are brave enough to access our OpenSIPs WebRTC
gateway using Microsoft Edge.

We've had some teething issues which have been diagnosed as a failed
SNI check due to the character casing, our certificate presents common
and alt names in lowercase (i.e. rtc.opensips.org) however Edge was
checking using a mix of uppercase and lowercase (i.e.
RTC.opensips.org). The server responds with a fatal alert message
"Description: Unrecognized Name (112)" indicating the mismatch.

The fix in our case is just to amend the client configuration however
most other browsers have been demonstrated to fix this by default. I'm
wondering if this is worth patching server side such that the
certificate names and client provided SNI's are held only in lowercase
or a case insensitive match is performed?

Thanks,

Callum

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registered in
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Re: [OpenSIPS-Users] disable_503_translation

2020-05-04 Thread Callum Guy
Hi Bogdan and others,

I've updated my runtime config to include the memory debug option (-a
Q_MALLOC_DBG) - its working great and I'm collecting data to try and
identify a potential leak.

Following this change I've noted a difference in the way processes are
being utilised. I'm running 20 UDP listeners and during normal
operation without the debugger I can see memory use growing on a
couple of processes which I've presumed means that the processes are
used in a top down order such that memory use remains relatively
static for most processes but grows on the processes which are being
most utilised. When the debugger is on however the load appears to be
evenly distributed amongst all processes so memory growth is even for
all 20 processes. As I'm hunting for a very minor leak it may be a few
weeks until I have enough information to identify memory growth. This
is actually very useful but I wanted to ask if this is an expected
behaviour when the debugger is enabled?

My base startup memory use per process is ~12.4Mb and I've copied in
some stats below to illustrate what I mean.

Without debugger, after a couple of days execution (notice process 23/21):

"pkmem:4-real_used_size": 12477048,
"pkmem:5-real_used_size": 12485184,
"pkmem:6-real_used_size": 12497672,
"pkmem:7-real_used_size": 12585320,
"pkmem:8-real_used_size": 12493296,
"pkmem:9-real_used_size": 12520400,
"pkmem:10-real_used_size": 12522488,
"pkmem:11-real_used_size": 12689832,
"pkmem:12-real_used_size": 12530056,
"pkmem:13-real_used_size": 12526160,
"pkmem:14-real_used_size": 12505896,
"pkmem:15-real_used_size": 12606136,
"pkmem:16-real_used_size": 12783888,
"pkmem:17-real_used_size": 12651568,
"pkmem:18-real_used_size": 12961944,
"pkmem:19-real_used_size": 12744784,
"pkmem:20-real_used_size": 12544880,
"pkmem:21-real_used_size": 13155584,
"pkmem:22-real_used_size": 12844256,
"pkmem:23-real_used_size": 13963040,

With debugger after 10 days of execution:

"pkmem:4-real_used_size": 12793624,
"pkmem:5-real_used_size": 12793872,
"pkmem:6-real_used_size": 12793400,
"pkmem:7-real_used_size": 12793552,
"pkmem:8-real_used_size": 12793560,
"pkmem:9-real_used_size": 12794872,
"pkmem:10-real_used_size": 12793952,
"pkmem:11-real_used_size": 12793488,
"pkmem:12-real_used_size": 12794328,
"pkmem:13-real_used_size": 12794016,
"pkmem:14-real_used_size": 12793664,
"pkmem:15-real_used_size": 12793832,
"pkmem:16-real_used_size": 12793464,
"pkmem:17-real_used_size": 12794240,
"pkmem:18-real_used_size": 12793792,
"pkmem:19-real_used_size": 12794072,
"pkmem:20-real_used_size": 12801088,
"pkmem:21-real_used_size": 12794264,
"pkmem:22-real_used_size": 12796232,
"pkmem:23-real_used_size": 12797184,

Thanks for reading, I am curious to understand more about what I'm seeing here.




On Tue, 28 Apr 2020 at 12:18, Bogdan-Andrei Iancu  wrote:
>
> Just be sure you are not logging all the debug ops, but only the dump -
> in this case you should be safe .
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>
> On 4/28/20 1:21 PM, Callum Guy wrote:
> > OK will do, I've been hesitant to configure the debugger on a
> > production system in case it caused any performance/stability issues -
> > is that a reason for concern?
> >
> > For now I'll see if I can configure a test setup and see if I can
> > duplicate the problem there.
> >
> > Thanks for your time.
> >
> >
> > On Tue, 28 Apr 2020 at 10:01, Bogdan-Andrei Iancu  
> > wrote:
> >> Hi Callum,
> >>
> >> Maybe a better approach will be try to identify the potential leak as
> >> described here ->
> >> https://opensips.org/Documentation/TroubleShooting-OutOfMem
> >>
> >> Best regards,
> >>
> >> Bogdan-Andrei Iancu
> >>
> >> OpenSIPS Founder and Developer
> >> https://www.opensips-solutions.com
> >>
> >> On 4/28/20 1:38 AM, Callum Guy wrote:
> >>> Hi Bogdan,
> >>>
> >>> I'm still searching for my memory leak, just downgraded from 3.0.2 to
> >>> 3.0.1 and should be able to confirm if that has made a difference in a
> >>> day or two.
> >>>
> >>>

Re: [OpenSIPS-Users] disable_503_translation

2020-04-28 Thread Callum Guy
OK will do, I've been hesitant to configure the debugger on a
production system in case it caused any performance/stability issues -
is that a reason for concern?

For now I'll see if I can configure a test setup and see if I can
duplicate the problem there.

Thanks for your time.


On Tue, 28 Apr 2020 at 10:01, Bogdan-Andrei Iancu  wrote:
>
> Hi Callum,
>
> Maybe a better approach will be try to identify the potential leak as
> described here ->
> https://opensips.org/Documentation/TroubleShooting-OutOfMem
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>
> On 4/28/20 1:38 AM, Callum Guy wrote:
> > Hi Bogdan,
> >
> > I'm still searching for my memory leak, just downgraded from 3.0.2 to
> > 3.0.1 and should be able to confirm if that has made a difference in a
> > day or two.
> >
> > During the hunt I've been reviewing the changelog commits - wondered
> > if you could check params var created at the link below:
> >
> > https://github.com/OpenSIPS/opensips/blob/559921962c69575bcdc90c1f03102e6d5c56d776/modules/dialog/dlg_req_within.c#L489
> >
> > My understanding of memory management in C isn't great but it looks to
> > be allocating without a free?
> >
> > Many thanks,
> >
> > Callum
> >
> >
> > On Fri, 24 Apr 2020 at 17:16, Callum Guy  wrote:
> >> Thanks Bogdan, I realise it was a long shot! There are only minor 
> >> differences between this and another system i am running so I'm going 
> >> through eliminating the config differences.
> >>
> >> Version wise the stable one is 3.0.1 and the leaky one is 3.0.2
> >>
> >> Slow progress but I will get there eventually!
> >>
> >> On Fri, 24 Apr 2020, 16:50 Bogdan-Andrei Iancu,  
> >> wrote:
> >>> Hi Callum,
> >>>
> >>> I 99.999% that that setting cannot generate a leak :)
> >>>
> >>> Best regards,
> >>>
> >>> Bogdan-Andrei Iancu
> >>>
> >>> OpenSIPS Founder and Developer
> >>>https://www.opensips-solutions.com
> >>>
> >>> On 4/24/20 6:31 PM, Callum Guy wrote:
> >>>
> >>> Hi All,
> >>>
> >>> I've been hunting a minor memory leak in my config and wanted to check in 
> >>> with the devs in case it is related to ues of the parameter:
> >>>
> >>> disable_503_translation=yes
> >>>
> >>> Here is the implementation link to save you a few seconds:
> >>>
> >>> https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425
> >>>
> >>> Is there any chance that this is failing to free the tiny amount it's 
> >>> allocating on each use?
> >>>
> >>> All the best,
> >>>
> >>> Callum
> >>>
> >>>
> >>> 0333 332   |  x-on.co.uk  |  |  Coronavirus
> >>>
> >>> 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.
> >>>
> >>> 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 attachments
> >>> for viruses. X-on makes no representation or warranty as to the absence 
> >>> of viruses in this email or any attachments.
> >>>
> >>>
> >>> ___
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> >>>

[OpenSIPS-Users] disable_503_translation

2020-04-24 Thread Callum Guy
Hi All,

I've been hunting a minor memory leak in my config and wanted to check in
with the devs in case it is related to ues of the parameter:

disable_503_translation=yes

Here is the implementation link to save you a few seconds:

https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425

Is there any chance that this is failing to free the tiny amount it's
allocating on each use?

All the best,

Callum

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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.


X-on
is a trading name of Storacall Technology Ltd a limited company 
registered in
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Registered Office : Avaland House, 110 
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Herts, HP3 9SD. Company Registration 
No. 2578478.

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Re: [OpenSIPS-Users] need some help to parse the data I got after doing lookup in location table .

2020-04-24 Thread Callum Guy
Hi Sasmita,

I would advise that you capture this information in a branch flag during
registration, these will be stored in location and retrieved when
performing a matching lookup().

So:

if ($pr == "ws" || $pr == "wss") {
setbflag(SRC_WS);
}
save("location_table")

When you do the lookup this bflag will be loaded and can be utilised to
route as required.

Good luck



On Fri, 24 Apr 2020 at 06:57, Sasmita Panda  wrote:

> Hi ,
>
>
> I have a 2 user , 1 is register from browser and another is register from
> a cisco phone . The protocol through which both get register is different .
> browser get register through WSS and cisco phone get register through UDP
> .
>
> Now while an INVITE comes for any of the user after doing lookup , while
> giving call to browser I wanted to involve Rtpengine in for media
> encryption but while giving call to cisco phone no need of Rtpengine
> involvement  .
>
> My question is , how I will differentiate both user  ? My knowledge says I
> have to parse the contact I got from lookup , parse that contact and find
> which user register through which protocol and accordingly differentiate
> them .
> If so , then how I will parse the contact ? Please do help me .
>
> Thank you in advance .
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
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Re: [OpenSIPS-Users] Compare $fU to drouting prefix

2020-04-18 Thread Callum Guy
That'll do it..

On Fri, 17 Apr 2020 at 16:00, Mark Farmer  wrote:

> OK, fixed it.
>
> Turned out to be this breaking it by overwriting $acc_extra(customer_id)
> with a blank value.
>
> ...
> else $acc_extra(Call_Flow) = "Internal";
> $acc_extra(customer_id) = $var(rule_attrs);
> ...
>
> Changed it to:
>
> ...
> else if (isflagset(TPTY_PBX) || isflagset(PBX_TPTY)) {
> $acc_extra(Call_Flow) = "Internal";
> $acc_extra(customer_id) = $var(rule_attrs);
> ...
>
> And all works nicely :)
>
> Thanks for the help!
> Mark.
>
>
> On Fri, 17 Apr 2020 at 14:08, Mark Farmer  wrote:
>
>> Thats what I thought :)
>>
>> This is getting quite odd now.
>> I am adding 2 extra fields, Call_Flow & customer_id. The odd thing is
>> that Call_Flow is working perfectly in all cases.
>> customer_id works fine for the latter 2 scenarios but in the first
>> scenario the customer_id data is never added to the database.
>>
>> I have this configured for the acc module:
>> modparam("acc", "extra_fields", "db: from_usr; to_usr; customer_id;
>> Call_Flow")
>>
>> The variable is set earlier on:
>> ...
>> do_routing("3",,,"$var(custID)");
>> ...
>>
>> And I am doing all of the accounting in a dedicated route:
>>
>> route[ACCEXTRA] {
>> do_accounting("db","cdr");
>> xlog("CUSTOM_LOG: Adding extra accounting: from_usr: $fU
>> customer_id: $var(rule_attrs) $var(custID)"); *# variable is
>> visible here*
>> $acc_extra(from_usr) = $fU;
>> $acc_extra(to_usr) = $tU;
>> if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) {
>> xlog("CUSTOM_LOG: Customer ID = $var(custID)"); *#
>> variable is visible here*
>> $acc_extra(Call_Flow) = "Outbound";
>> $acc_extra(customer_id) = $var(custID);
>> xlog("CUSTOM_LOG: $$acc_extra(customer_id) =
>> $acc_extra(customer_id)"); *# variable is visible here*
>> } else if (isflagset(PSTN_TPTY) || isflagset(PSTN_PBX)) {
>> $acc_extra(Call_Flow) = "Inbound";
>> $acc_extra(customer_id) = $var(rule_attrs);
>> } else $acc_extra(Call_Flow) = "Internal";
>> $acc_extra(customer_id) = $var(rule_attrs);
>> }
>>
>>
>> On Fri, 17 Apr 2020 at 13:50, Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Mark,
>>>
>>> You can populate the $acc_extra() from whatever other variable or string
>>> operations. Most probably your issue is in other place, in regards to the
>>> acc logic.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>>
>>> On 4/17/20 1:40 PM, Mark Farmer wrote:
>>>
>>> Thanks Bogdan, that's mostly working now.
>>>
>>> My issue now is with passing that identifier into acc_extra() as a
>>> variable which does not seem to be working.
>>> Using xlog() I can see that the variable is populated right before
>>> calling acc_extra()
>>>
>>> ...
>>> if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) {
>>> xlog("CUSTOM_LOG: Customer ID = $var(custID)");
>>> $acc_extra(customer_id) = $var(custID);
>>> ...
>>> do_accounting("db","cdr");
>>> }
>>>
>>> Does acc_extra() not accept variables as input?
>>>
>>> Thanks again!
>>> Mark.
>>>
>>>
>>>
>>>
>>> On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu 
>>> wrote:
>>>
 Hey Mark,

 It is not nice, but you can do:

 $var(tmp) = $rU;
 $rU = $fU
 do_routing();
 $rU = $var(tmp);

 Regards,

 Bogdan-Andrei Iancu

 OpenSIPS Founder and Developer
   https://www.opensips-solutions.com

 On 4/17/20 11:13 AM, Mark Farmer wrote:

 Hi Bogdan, I will try to explain better.

 In rule_attrs I have a customer identifier which is used by acc to add
 the identifier into the CDR database.
 This works fine for calls from PSTN which are routed to another SIP
 gateway but calls from that gateway routed to PSTN can come from multiple
 customers and there is no way to identify which. So I'd like to match the
 incoming $fU to the rule that would match $rU in the from PSTN scenario in
 order to retrieve the rule_attrs (the customer identifier) from that rule.

 Does that make sense?

 Many thanks and regards
 Mark.




 On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu 
 wrote:

> Hi Mark,
>
> What kind of matching you want to do between $fU and the dr prefixes ?
> You want to do the same as what drouting() does with $rU ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>
> On 4/16/20 6:14 PM, Mark Farmer wrote:
>
> Hi everyone
>
> I am looking for a way to compare $fU in INVITE to the matching
> drouting() prefix of another group and retrieve the rule_attrs from that
> rule.
>
> At the moment I am thinking I'll have to

Re: [OpenSIPS-Users] Errors

2020-03-31 Thread Callum Guy
You're not the only one my friend, I've seen plenty of discussions on the
topic in the mailing list so browse the archives for details.

i.e.
https://opensips.org/pipermail/users/2018-September/039895.html

Here's a quote from Bogdan later in that thread:

"If there is no load (worker processes are idle, no busy with anything
else), the reported delay may be generated only by the interprocess
communication (passing the job from the triggering process to the
executing process via internal pipes).

What are the values you typically observe ? maybe the warning is jst
about a slow context switching on your server (btw, how many cores ?)."

For what it's worth I've been seeing these in production for a long time
and they seem harmless. It was on my list to speak to people about at the
Summit but looks like I'll have to wait until September! We're running quad
core VM's.



On Tue, 31 Mar 2020 at 18:44, Saint Michael  wrote:

> Does anybody have any idea what causes this error?
> I am pulling my hair.
>
> ps[1883]: Mar 31 17:10:49 [1903] WARNING:core:utimer_ticker: utimer task
>  already scheduled for 190 ms (now 890 ms), it may overlap..
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1903]
> WARNING:core:utimer_ticker: utimer task  already scheduled for
> 190 ms (now 990 ms), it may overlap..
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1903]
> WARNING:core:utimer_ticker: utimer task  already scheduled for
> 190 ms (now 1090 ms), it may overlap..
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1961]
> WARNING:core:handle_timer_job: utimer job  has a 97 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1955]
> WARNING:core:handle_timer_job: utimer job  has a 97 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1961]
> WARNING:core:handle_timer_job: timer job  has a 17 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1930]
> WARNING:core:handle_timer_job: utimer job  has a 97 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1980]
> WARNING:core:handle_timer_job: timer job  has a 17 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1971]
> WARNING:core:handle_timer_job: timer job  has a 17 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1951]
> WARNING:core:handle_timer_job: utimer job  has a 97 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1928]
> WARNING:core:handle_timer_job: timer job  has a 17 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1914]
> WARNING:core:handle_timer_job: timer job  has a 17 us delay
> in execution
> Mar 31 17:10:49 jorge opensips[1883]: Mar 31 17:10:49 [1961]
> WARNING:core:handle_timer_job: timer job  has a 17 us
> delay in execution
> ___
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Re: [OpenSIPS-Users] AVP availability in CANCEL

2020-03-16 Thread Callum Guy
Hi Ben,

Just following up to confirm that you were correct and t_check_trans()
makes all the difference - without it the isflagset check doesn't
match.

Thank you!

  if (is_method("BYE|CANCEL")) {
 match_dialog();
 t_check_trans();
 if (is_dlg_flag_set(10) || isflagset(RTPENGINE_ENGAGED)) {
  rtpengine_delete();
 }
   }


On Mon, 16 Mar 2020 at 13:33, Callum Guy  wrote:
>
> Hi Ben,
>
> Thank you for the information, I've checked the tm module docs and 
> t_check_trans() doesn't highlight this behaviour - it just sounds like a 
> method to confirm the presence of the transaction. To try and confirm this 
> I've been looking at the tm module code and it matches the documentation 
> description - some of the nested methods are locating the transaction within 
> a hash table however there is no clear indicator to me that this would be 
> somehow loading the AVPs and/or flags into the current request process. I am 
> not a C expert and could definitely be missing something however but it would 
> be great if one the dev team could confirm if this is indeed the correct 
> approach. For now I'll implement some logging to try and confirm the 
> behaviour.
>
> Your feature request for dialog method flags sounds useful, I suspect once my 
> current CANCEL issue is resolved that there may be a few remaining call 
> scenarios where rtpengine isn't being cleared due to the dialog having been 
> closed, if that happens to me I'll see if I can find time to implement your 
> idea.
>
> Thanks,
>
> Callum
>
>
> On Mon, 16 Mar 2020 at 13:02, Ben Newlin  wrote:
>>
>> Callum,
>>
>>
>>
>> Both AVPs and transaction flags are tied to the transaction, so you must 
>> call t_check_trans so the transaction will be matched. I don’t see that 
>> happening in your code. I don’t believe match_dialog will do that for you.
>>
>>
>>
>> Also, I think the issues you are having with dialog variables as you 
>> mentioned in another response may be related to an feature request we have 
>> open [1]. Basically, once the BYE is received the dialog is set to ENDED 
>> state and the dialog variable retrieval ignores dialogs in this state. Our 
>> use case is a bit different than yours, so it may not be the same.
>>
>>
>>
>> [1] - https://github.com/OpenSIPS/opensips/issues/1637
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> From: Users  on behalf of Callum Guy 
>> 
>> Reply-To: OpenSIPS users mailling list 
>> Date: Monday, March 16, 2020 at 7:23 AM
>> To: OpenSIPS users mailling list 
>> Subject: [OpenSIPS-Users] AVP availability in CANCEL
>>
>>
>>
>> Hi All,
>>
>> I have a simple question regarding availability of AVP variables in CANCEL. 
>> I'm not sure when OpenSIPs will load the AVP's for a transaction so am 
>> looking for information here. The situation is that I want to flag sessions 
>> using a media proxy and close the sessions when a CANCEL arrives before the 
>> call is answered.
>>
>>
>>
>> This is how I enable and track rtpengine:
>>
>>
>>
>> route[RTPENGINE_OFFER_IE] {
>>   set_dlg_flag(10);
>>   $avp(media_proxy) := "on";
>>   rtpengine_offer("trust-address replace-session-connection 
>> in-iface=internal out-iface=external ICE=remove");
>> }
>>
>> The current logic sets both a dialog variable and an AVP when rtpengine is 
>> offered. When BYE/CANCEL messages arrive the script does this:
>>
>> if (is_method("BYE|CANCEL")) {
>>   match_dialog();
>>   if (is_dlg_flag_set(10) || $avp(media_proxy) == "on") {
>> rtpengine_delete();
>>   }
>> }
>>
>>
>>
>> I was hoping that the dialog would be matched by call ID and tags however 
>> this does not appear to work so I must be missing some info. The AVP was 
>> added later to try and make it work for the CANCEL requests but also did not 
>> work.
>>
>>
>>
>> I'm planning on replacing the AVP with a transaction flag tonight, is that 
>> likely to resolve the issue? Apologies if these are basic questions, 
>> availability of different types of flags and AVP's really aren't clear to me 
>> so any advice would be appreciated!
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Callum
>>
>>
>>
>>
>>
>> 0333 332   |  www.x-on.co.uk  |
>>
>> X-on is a trading name of Storacall Technology Ltd a limited company 
>> registered in England and 

Re: [OpenSIPS-Users] AVP availability in CANCEL

2020-03-16 Thread Callum Guy
Hi Ben,

Thank you for the information, I've checked the tm module docs
and t_check_trans() doesn't highlight this behaviour - it just sounds like
a method to confirm the presence of the transaction. To try and confirm
this I've been looking at the tm module code and it matches the
documentation description - some of the nested methods are locating the
transaction within a hash table however there is no clear indicator to me
that this would be somehow loading the AVPs and/or flags into the current
request process. I am not a C expert and could definitely be missing
something however but it would be great if one the dev team could confirm
if this is indeed the correct approach. For now I'll implement some logging
to try and confirm the behaviour.

Your feature request for dialog method flags sounds useful, I suspect once
my current CANCEL issue is resolved that there may be a few remaining call
scenarios where rtpengine isn't being cleared due to the dialog having been
closed, if that happens to me I'll see if I can find time to implement your
idea.

Thanks,

Callum


On Mon, 16 Mar 2020 at 13:02, Ben Newlin  wrote:

> Callum,
>
>
>
> Both AVPs and transaction flags are tied to the transaction, so you must
> call t_check_trans so the transaction will be matched. I don’t see that
> happening in your code. I don’t believe match_dialog will do that for you.
>
>
>
> Also, I think the issues you are having with dialog variables as you
> mentioned in another response may be related to an feature request we have
> open [1]. Basically, once the BYE is received the dialog is set to ENDED
> state and the dialog variable retrieval ignores dialogs in this state. Our
> use case is a bit different than yours, so it may not be the same.
>
>
>
> [1] - https://github.com/OpenSIPS/opensips/issues/1637
>
>
>
> Ben Newlin
>
>
>
> *From: *Users  on behalf of Callum Guy <
> callum@x-on.co.uk>
> *Reply-To: *OpenSIPS users mailling list 
> *Date: *Monday, March 16, 2020 at 7:23 AM
> *To: *OpenSIPS users mailling list 
> *Subject: *[OpenSIPS-Users] AVP availability in CANCEL
>
>
>
> Hi All,
>
> I have a simple question regarding availability of AVP variables in
> CANCEL. I'm not sure when OpenSIPs will load the AVP's for a transaction so
> am looking for information here. The situation is that I want to flag
> sessions using a media proxy and close the sessions when a CANCEL arrives
> before the call is answered.
>
>
>
> This is how I enable and track rtpengine:
>
>
>
> route[RTPENGINE_OFFER_IE] {
>   set_dlg_flag(10);
>   $avp(media_proxy) := "on";
>   rtpengine_offer("trust-address replace-session-connection
> in-iface=internal out-iface=external ICE=remove");
> }
>
> The current logic sets both a dialog variable and an AVP when rtpengine is
> offered. When BYE/CANCEL messages arrive the script does this:
>
> if (is_method("BYE|CANCEL")) {
>   match_dialog();
>   if (is_dlg_flag_set(10) || $avp(media_proxy) == "on") {
> rtpengine_delete();
>   }
> }
>
>
>
> I was hoping that the dialog would be matched by call ID and tags however
> this does not appear to work so I must be missing some info. The AVP was
> added later to try and make it work for the CANCEL requests but also did
> not work.
>
>
>
> I'm planning on replacing the AVP with a transaction flag tonight, is that
> likely to resolve the issue? Apologies if these are basic questions,
> availability of different types of flags and AVP's really aren't clear to
> me so any advice would be appreciated!
>
>
>
> Thanks,
>
>
>
> Callum
>
>
>
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>
>
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[OpenSIPS-Users] AVP availability in CANCEL

2020-03-16 Thread Callum Guy
Hi All,

I have a simple question regarding availability of AVP variables in CANCEL.
I'm not sure when OpenSIPs will load the AVP's for a transaction so am
looking for information here. The situation is that I want to flag sessions
using a media proxy and close the sessions when a CANCEL arrives before the
call is answered.

This is how I enable and track rtpengine:

route[RTPENGINE_OFFER_IE] {
  set_dlg_flag(10);
  $avp(media_proxy) := "on";
  rtpengine_offer("trust-address replace-session-connection
in-iface=internal out-iface=external ICE=remove");
}

The current logic sets both a dialog variable and an AVP when rtpengine is
offered. When BYE/CANCEL messages arrive the script does this:

if (is_method("BYE|CANCEL")) {
  match_dialog();
  if (is_dlg_flag_set(10) || $avp(media_proxy) == "on") {
rtpengine_delete();
  }
}

I was hoping that the dialog would be matched by call ID and tags however
this does not appear to work so I must be missing some info. The AVP was
added later to try and make it work for the CANCEL requests but also did
not work.

I'm planning on replacing the AVP with a transaction flag tonight, is that
likely to resolve the issue? Apologies if these are basic questions,
availability of different types of flags and AVP's really aren't clear to
me so any advice would be appreciated!

Thanks,

Callum

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Re: [OpenSIPS-Users] opensips + rtpengine

2020-03-16 Thread Callum Guy
I encountered a similar issue recently, I was using dialog variables
to flag sessions where RTPEngine is engaged so rtpengine_delete only
fired on applicable BYE/CANCEL requests. For reasons I have not yet
understood the dialog variable was not always available so the
sessions were left open and subject to an hours wait for the rtpengine
timeouts. I've tried to improve the logic with AVP's however I'm still
seeing unclosed sessions so am planning just to always fire it off on
a BYE/CANCEL and let RTPEngine drop requests when it is not engaged.
For your script I would recommend that you take the delete function
out of loose routing and implement it whenever you have BYE/CANCEL
requests arriving. I would also highly recommend that you enable the
rtpengine CLI so you are able to control session limits on a running
instance.

Alain, thanks for your anecdotal stats - I've been wondering how far
we'll be able to push an instance of RTPEngine. We have some powerful
24 core machines and are using the packet forwarding module and are
hoping to approach 1 sessions per instance. If your figures scale
up it sounds like this is very achievable!



On Sun, 15 Mar 2020 at 12:05, Alain Bieuzent  wrote:
>
> Hi,
>
>
>
> Can you share value of  delete-delay, port-min and port-max of your rtpengine 
> configuration.
>
>
>
> Have you also check if you handle rtpengine_delete on failed calls (in case 
> sip cause code 4XX, 5XX and 6XX).
>
>
>
> At @job, we handle max 6000 calls on a 6 cores servers without any issue.
>
>
>
> Regards
>
>
>
>
>
>
>
> De : Users  au nom de volga629 via Users 
> 
> Répondre à : volga629 , OpenSIPS users mailling list 
> 
> Date : vendredi 13 mars 2020 à 18:39
> À : 
> Objet : [OpenSIPS-Users] opensips + rtpengine
>
>
>
> Hello Everyone,
>
> Might be  somebody can point me to right place.
>
> Under load Rtpengine on server with 12 core can't pass 400 channels/sessions.
>
> Mar 13 18:14:53 CentOS-77-64-minimal rtpengine[14588]: WARNING: 
> [1b17077c-654e-11ea-bd31-87b1c8fc-849]: Protocol error in packet from 
> 136.243.43.23:47763: Ran out of ports [d3:sdp289:
>
> WARNING: [1be05a46-654e-11ea-b136-573b6201-849]: Protocol error in packet 
> from 136.243.43.23:55847: Unknown call-id [d3:sdp250:
>
> It like it not closing calls properly, but I am running  rtpengine_delete()  
> in loose _route on BYE or CANCEL.
>
>
>
> Here are more details
>
>
> https://github.com/sipwise/rtpengine/issues/946
>
>
>
> # Handle requests within SIP dialogs
> route[handle_sequential] {
> if (has_totag()) {
> if (loose_route()) {
> # BYE rtpengine_delete()
> if (is_method("BYE|CANCEL")) {
> xlog("LOOSE_ROUTE:DBG: [$rm] trying delete 
> rtpengine\n");
> rtpengine_delete();
> xlog("Average MOS of the entire call is 
> $rtpstat(MOS-average)\r\n");
> xlog("Average MOS of caller is 
> $(rtpstat(MOS-average)[$ft])\r\n");
> xlog("Average MOS of callee is 
> $(rtpstat(MOS-average)[$tt])\r\n");
> xlog("Min MOS of caller is $(rtpstat(MOS-min)[$ft]) 
> reported at $(rtpstat(MOS-min-at)[$ft])\r\n");
> }
> t_relay();
> exit;
>
> volga629
>
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Re: [OpenSIPS-Users] Public IP in REGISTER

2020-03-04 Thread Callum Guy
Hi Jehan,

Sounds like you want to be using fix_nated_contact() - when the INVITE
arrives you can try the following:

# Check if contact is RFC1918
if (nat_uac_test(1)) {
# Replace the contact IP with the received address from the network
fix_nated_contact();
}

If you look at the registrations via CLI or database (if in use) you'll see
that OpenSIPs is tracking both a contact and received value to deal with
this - that will be where your NAT pings are getting the address from.

Hopefully that helps to get you on the correct path!

Callum

On Tue, 3 Mar 2020 at 21:06, Jehanzaib Younis 
wrote:

> Hi gang,
>
> I have quite strange issue. When in some cases when my customer's have
> 10.202.128.xx range they are behind NAT. when i see opensipsctl ul show
> number@sip
> i see contact header as Contact:: sip:num...@10.202.128.xx:5060;transport=udp
> Q=
> which means when i try to relay the invite goes to that 10.202.128.xx:5060
> which is wrong. It should go to its public IP:port
>
> but weird thing is, i see the OPTION is sent to its Public IP (OPTIONS
> sip:180.xx.xx.xx:1502). I also see the 200 OK which is perfect.
>
> I tried to use fix_nated_register(); but it does not change anything.
>
> Can anyone help please?
>
> Thank you
>
>
> Regards,
> Jehan
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Re: [OpenSIPS-Users] A word of caution on migrating to 3.1

2020-02-28 Thread Callum Guy
That is the same as the example above!

On Fri, 28 Feb 2020 at 10:11, johan  wrote:

> no, what I mean is this: if lookup_location(...) needs to be if
> (lookup_location())
> On 28.02.20 11:06, Grant Bagdasarian wrote:
>
> Hi Johan,
>
> I’ve been testing with a nightly build (not the latest) these past few
> days, and the body of an if statement doesn’t require any brackets if there
> is only a single statement. Haven’t tested with multiple statements, but I
> assumed the syntax of other languages.
>
> Can’t comment on the rest.
>
> Regards,
>
> Grant
>
> On Fri, 28 Feb 2020 at 10:56 Johan De Clercq  wrote:
>
>>
>> Hello,
>>
>> Migration from 3.0 to 3.1 is not so straightforward:
>> For script:
>> -everything after if needs to be in brackets.
>> -setflag ans etbflag : their parameters need to be in double quotes.
>>
>> For db: ther is a version change there also.
>>
>> Best regards,
>> Outlook voor iOS  downloaden
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>>
>
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Re: [OpenSIPS-Users] Usrloc Cluster

2020-02-19 Thread Callum Guy
Thanks @liviuchircu - sadly GitHub isn't letting me comment at the moment
(bug) so coming back here.

Checking my active masters I am indeed seeing:
`INFO:usrloc:replicate_ucontact_update: All destinations in cluster: 25 are
down or probing`

Browsing the repo I can see that patch 40b7422 is not included in any
tagged releases at this time so I will amend the `ping_timeout` as
suggested.

Thanks for the invaluable insight.

On Wed, 19 Feb 2020 at 14:29, Liviu Chircu  wrote:

> Hi, Callum!
>
> On 18.02.2020 15:56, Callum Guy wrote:
>
>
> INFO:usrloc:receive_ucontact_insert: failed to fetch local urecord -
> creating new one (ci: '0_751733367@10.0.0.13')
> INFO:usrloc:receive_ucontact_update: failed to fetch local urecord -
> create new record and contact (ci: '0_3576574827@10.0.0.26')
> INFO:usrloc:receive_ucontact_update: failed to fetch local urecord -
> create new record and contact (ci: '0_4169036315@10.0.0.12')
>
> Is this a simple matter of sessions timing out on the backup and being
> removed before the next registration on the primary instance?
>
> Exactly.  The UA re-REGISTERs too close to the expiration point, creating
> a race condition
> between the two nodes.  By the time the backup processes the contact
> refresh packet, its
> contact is long gone, so it has nothing to match it against.  Hence this
> harmless INFO message.
>
>
> Secondly I am graphing registration counts based on the following CLI
> request:
>
> opensips-cli -x mi ul_dump brief=1 | grep AOR | wc -l
>
> Several times a day I see a small drop in the backup registrations
> (lighter red line) before a resync operation as shown here:
>
>
> Can anyone help to explain what's going on here? There aren't any log
> messages at this time however I do see the backup node reporting a primary
> node ping loss ~1 hour before the dip:
>
> 2020-02-18T11:38:37.786021+00:00 opensips[91789]:
> INFO:clusterer:do_action_trans_2: Ping reply not received, node [13] is
> down
> 2020-02-18T11:38:38.795838+00:00 opensips[91804]:
> INFO:clusterer:handle_internal_msg: Node [13] is UP
>
> This is 100% identical to what Alexey reported in #1976 [1].  Please try
> to incorporate at least one of
> my proposed solutions -- this should make it much more harder for the link
> between the nodes to do down.
>
> Best regards,
>
> [1]: https://github.com/OpenSIPS/opensips/issues/1976
>
> --
> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
>
> OpenSIPS Summit, Amsterdam, May 2020
>   www.opensips.org/events
> OpenSIPS Bootcamp, Miami, March 2020
>   www.opensips.org/training
>
>

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[OpenSIPS-Users] Usrloc Cluster

2020-02-18 Thread Callum Guy
Hi All,

I'm running a full sharing cluster for hot standby purposes and have been
noticing that the backup node will periodically report the messages below.

INFO:usrloc:receive_ucontact_insert: failed to fetch local urecord -
creating new one (ci: '0_751733367@10.0.0.13')
INFO:usrloc:receive_ucontact_update: failed to fetch local urecord - create
new record and contact (ci: '0_3576574827@10.0.0.26')
INFO:usrloc:receive_ucontact_update: failed to fetch local urecord - create
new record and contact (ci: '0_4169036315@10.0.0.12')

Is this a simple matter of sessions timing out on the backup and being
removed before the next registration on the primary instance?

Secondly I am graphing registration counts based on the following CLI
request:

opensips-cli -x mi ul_dump brief=1 | grep AOR | wc -l

Several times a day I see a small drop in the backup registrations
(lighter red line) before a resync operation as shown here:

[image: image.png]

Can anyone help to explain what's going on here? There aren't any log
messages at this time however I do see the backup node reporting a primary
node ping loss ~1 hour before the dip:

2020-02-18T11:38:37.786021+00:00 opensips[91789]:
INFO:clusterer:do_action_trans_2: Ping reply not received, node [13] is down
2020-02-18T11:38:38.795838+00:00 opensips[91804]:
INFO:clusterer:handle_internal_msg: Node [13] is UP

There shouldn't have been any loss of connectivity at that time so I'm at a
loss to explain the communication failure.

I'm running 3.0.0 on these systems so if relevant patches have been applied
on 3.0.2 please let me know and I'll plan the upgrade.

Callum

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Re: [OpenSIPS-Users] is_request(), is_reply() functions in OpenSips

2020-02-17 Thread Callum Guy
Hi Grant,

There could very well be a better way so hopefully someone else will chime
in if they have a better solution however I would leverage a simple
$var(example_mode)
= 'reply'; - $var's operate on a per process basis so you would simply set
it before the route(GEN) call and check its value within the block. This
seems like a lightweight and error prone implementation, just be aware that
a var will persist on a given process so you'll just need to set it every
time before calling the route to be sure you're not using stale data.

Again, hope that helps and I'll be interested if there is a better way - I
haven't seen that route parameter syntax before so if that's valid I would
also be happy with that approach.

Good luck,

Callum


On Mon, 17 Feb 2020 at 22:01, Grant Bagdasarian 
wrote:

> Hi Callum,
>
> Thanks for the reply. I’m aware of the different types of routes and how
> to arm reply and branch routes for the parent requests.
>
> I’m currently trying to port a Kamailio script I’ve written a while back
> to OpenSips. One of the routes is fairly generic and called  from both
> request routes as onreply routes, but depends on the functions is_request
> and is_reply, but these are Kamailio specific.
>
> The only solution I can think of now is passing the type as a route
> parameter when called from request or reply routes.
>
> route(GENERIC_ROUTE, “request”)
> Or
> route(GENERIC_ROUTE, “reply”)
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Re: [OpenSIPS-Users] is_request(), is_reply() functions in OpenSips

2020-02-17 Thread Callum Guy
Hi,

All replies arrive at a separate route block, configurable during
processing of the parent request.

You can define many reply processing routes and allocate requests to these
at your discretion using t_on_reply("example_a"); or
t_on_reply("example_b"); etc

Docs are here:

https://www.opensips.org/Documentation/Script-Routes-3-0

It's all pretty straightforward once you've taken it for a spin!

Does that help?

Callum

On Mon, 17 Feb 2020 at 19:51, Grant Bagdasarian 
wrote:

> Hello,
>
> My experience is mainly in Kamailio, and I was wondering how to check
> in OpenSips if a message is a request or response? Kamailio exposes
> the functions is_request() and is_reply(), but I couldn't find an
> equivalent in OpenSips.
>
> Thanks and regards,
>
> Grant
>
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Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server

2020-02-13 Thread Callum Guy
I would imagine Liviu is correct, the answer will be quite clear

Not sure if you have seen my earlier comments but wanted to reiterate that
your initial email shows a resource limit error "Starting opensips (via
systemctl):  Job for opensips.service failed because a configured resource
limit was exceeded."

Try looking at the service file systemd is using, try something like the
following (P.S. your init.d file is probably not relevant):

[16:36]  FR-P-SIPSBC-3 🙂 $ sudo systemctl cat opensips
# /etc/systemd/system/opensips.service
[Unit]
Description=OpenSIPS is a very fast and flexible SIP (RFC3261) server
Documentation=man:opensips
After=network.target mysqld.service postgresql.service rtpproxy.service

[Service]
Type=forking
User=opensips
Group=opensips
RuntimeDirectory=opensips
RuntimeDirectoryMode=775
Environment=
EnvironmentFile=-/etc/sysconfig/opensips
PermissionsStartOnly=yes
PIDFile=%t/opensips/opensips.pid
ExecStart=/usr/sbin/opensips -P %t/opensips/opensips.pid -f
/etc/opensips/opensips.cfg -m $S_MEMORY -M $P_MEMORY $OPTIONS
ExecStop=/usr/bin/pkill --pidfile %t/opensips/opensips.pid
Restart=always
TimeoutStopSec=30s
LimitNOFILE=262144

[Install]
WantedBy=multi-user.target

Try changing the P_MEMORY/S_MEMORY values in the listed Env file and if you
tweak this file run systemctl daemon-reload before trying to start it.

On Thu, 13 Feb 2020 at 16:10, Liviu Chircu  wrote:

> On 13.02.2020 11:19, Tekin, Arda wrote:
>
> It prints too many logs when I run below command manually
>
> opensips –DD –E
>
> Well, those logs are _exactly_ where you should be starting at!  Redirect
> them
> to a file, maybe with "opensips -D -E &>/tmp/out &", then do "less
> /tmp/out" and
> take a look for any helpful errors.
>
> I assume you just want to start with the default
> /etc/opensips/opensips.cfg that
> comes with the "opensips" CentOS package, right?
>
> Regards,
>
> --
> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
>
> OpenSIPS Summit, Amsterdam, May 2020
>   www.opensips.org/events
> OpenSIPS Bootcamp, Miami, March 2020
>   www.opensips.org/training
>
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Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server

2020-02-11 Thread Callum Guy
Sorry, just read that again. Looks like opensipsctl found the systemd
service just fine except your config somehow exhausted resources. This may
be open file limits or memory allocation.

Try running the binary manually:

opensips -D

It should provide sufficient debug.


On Wed, 12 Feb 2020, 00:15 Callum Guy,  wrote:

> You're on Centos 7, systems my friend.
>
> Try systemctl start opensips
>
> Otherwise put the service file in systemd tree, an example exists on
> GitHub, and run systemctl daemon-reload before trying to start again.
>
> If it still doesn't work you'd need to check that you have the opensips
> binary in the correct path.
>
> Hope that helps you on your way!
>
>
> On Tue, 11 Feb 2020, 23:23 David Villasmil, <
> david.villasmil.w...@gmail.com> wrote:
>
>> And the output of
>>
>> journalctl -xe
>>
>> ?
>>
>> On Tue, 11 Feb 2020 at 17:20, Tekin, Arda  wrote:
>>
>>> How can I run OpenSIPS 3.0?
>>>
>>>
>>>
>>> Previously we are running `opensipsctl start` command.
>>>
>>>
>>>
>>> How to set and use db connection configuration before starting OpenSIPS?
>>>
>>>
>>>
>>> I have compiled latest source code on CentOS 7 successfully. Copied init
>>> script
>>>
>>> cp /root/src/opensips-3.0/packaging/redhat_fedora/opensips.init
>>> /etc/rc.d/init.d/.
>>>
>>>
>>>
>>> Edited the init file
>>>
>>> prog=opensips
>>>
>>> opensips=/usr/local/sbin/$prog
>>>
>>> cfgdir="/usr/local/etc/$prog"
>>>
>>> pidfile="/var/run/$prog.pid"
>>>
>>> lockfile="/var/lock/subsys/$prog"
>>>
>>> configfile="$cfgdir/$prog.cfg"
>>>
>>> m4configfile="$cfgdir/$prog.m4"
>>>
>>> m4archivedir="$cfgdir/archive"
>>>
>>> OPTIONS=""
>>>
>>> S_MEMORY=32
>>>
>>> P_MEMORY=32
>>>
>>> RETVAL=0
>>>
>>>
>>>
>>>
>>>
>>> When I run init script I get this error.
>>>
>>> Starting opensips (via systemctl):  Job for opensips.service failed
>>> because a configured resource limit was exceeded. See "systemctl status
>>> opensips.service" and "journalctl -xe" for details.
>>>
>>>[FAILED]
>>>
>>>
>>>
>>> Where is doc page explaining how OpenSIPS 3 starts?
>>>
>>>
>>>
>>> Kind Regards,
>>>
>>> Arda
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>

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Re: [OpenSIPS-Users] How to start OpenSIPS 3.0 server

2020-02-11 Thread Callum Guy
You're on Centos 7, systems my friend.

Try systemctl start opensips

Otherwise put the service file in systemd tree, an example exists on
GitHub, and run systemctl daemon-reload before trying to start again.

If it still doesn't work you'd need to check that you have the opensips
binary in the correct path.

Hope that helps you on your way!


On Tue, 11 Feb 2020, 23:23 David Villasmil, 
wrote:

> And the output of
>
> journalctl -xe
>
> ?
>
> On Tue, 11 Feb 2020 at 17:20, Tekin, Arda  wrote:
>
>> How can I run OpenSIPS 3.0?
>>
>>
>>
>> Previously we are running `opensipsctl start` command.
>>
>>
>>
>> How to set and use db connection configuration before starting OpenSIPS?
>>
>>
>>
>> I have compiled latest source code on CentOS 7 successfully. Copied init
>> script
>>
>> cp /root/src/opensips-3.0/packaging/redhat_fedora/opensips.init
>> /etc/rc.d/init.d/.
>>
>>
>>
>> Edited the init file
>>
>> prog=opensips
>>
>> opensips=/usr/local/sbin/$prog
>>
>> cfgdir="/usr/local/etc/$prog"
>>
>> pidfile="/var/run/$prog.pid"
>>
>> lockfile="/var/lock/subsys/$prog"
>>
>> configfile="$cfgdir/$prog.cfg"
>>
>> m4configfile="$cfgdir/$prog.m4"
>>
>> m4archivedir="$cfgdir/archive"
>>
>> OPTIONS=""
>>
>> S_MEMORY=32
>>
>> P_MEMORY=32
>>
>> RETVAL=0
>>
>>
>>
>>
>>
>> When I run init script I get this error.
>>
>> Starting opensips (via systemctl):  Job for opensips.service failed
>> because a configured resource limit was exceeded. See "systemctl status
>> opensips.service" and "journalctl -xe" for details.
>>
>>[FAILED]
>>
>>
>>
>> Where is doc page explaining how OpenSIPS 3 starts?
>>
>>
>>
>> Kind Regards,
>>
>> Arda
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
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Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.7 and 3.0.2 minor releases

2020-01-24 Thread Callum Guy
Thanks Razvan, you bring me great joy! :)

On Fri, 24 Jan 2020 at 13:48, Răzvan Crainea  wrote:

> Hi, Callum!
>
> Yes, all fixes will be backported to the 3.0 branch, and it will be part
> of the 3.0.2 release. You should check the 3.0 branch to see all the
> commits that will take part of a future release.
> The tag will be created as soon as the release is made, that's why
> you're not seeing anything about 3.0.2 yet. But if it is in the 3.0
> branch, then it will be part of the release.
>
> Best regards,
> Răzvan
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 1/24/20 3:27 PM, Callum Guy wrote:
> > Hi Razvan,
> >
> > Just wondering if the below commit can/will be included in the 3.0.2
> > release? I don't see a 3.0.2 branch or tag on Github so I'm not sure how
> > to check!
> >
> >
> https://github.com/OpenSIPS/opensips/commit/7b9239d63f412a1194e10c97611489d5facfdf74
> >
> > Thanks,
> >
> > Callum
> >
> > On Thu, 23 Jan 2020 at 09:58, Răzvan Crainea  > <mailto:raz...@opensips.org>> wrote:
> >
> > Hello to you all!
> >
> > I am pleased to announce you that we will make two more stable minor
> > releases next week. I'm talking about OpenSIPS 2.4.7 and 3.0.2.
> >
> > These new releases will contain several bug fixes (more than 150
> > commits), the headline being the notorious OpenSSL fix for the TLS
> > connections.
> >
> > The new releases are scheduled for Tuesday, 28.01.2020.
> >
> > Please let us know on GitHub of any pending issues or tickets that
> are
> > affecting the current versions so that we can get them fixed until
> the
> > next release.
> >
> > Thank you all for your contributions to the OpenSIPS community!
> >
> > Best wishes,
> > --
> > Răzvan Crainea
> > OpenSIPS Core Developer
> > http://www.opensips-solutions.com
> >
> > ___
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> > Users@lists.opensips.org <mailto:Users@lists.opensips.org>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
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Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.7 and 3.0.2 minor releases

2020-01-24 Thread Callum Guy
Hi Razvan,

Just wondering if the below commit can/will be included in the 3.0.2
release? I don't see a 3.0.2 branch or tag on Github so I'm not sure how to
check!

https://github.com/OpenSIPS/opensips/commit/7b9239d63f412a1194e10c97611489d5facfdf74

Thanks,

Callum

On Thu, 23 Jan 2020 at 09:58, Răzvan Crainea  wrote:

> Hello to you all!
>
> I am pleased to announce you that we will make two more stable minor
> releases next week. I'm talking about OpenSIPS 2.4.7 and 3.0.2.
>
> These new releases will contain several bug fixes (more than 150
> commits), the headline being the notorious OpenSSL fix for the TLS
> connections.
>
> The new releases are scheduled for Tuesday, 28.01.2020.
>
> Please let us know on GitHub of any pending issues or tickets that are
> affecting the current versions so that we can get them fixed until the
> next release.
>
> Thank you all for your contributions to the OpenSIPS community!
>
> Best wishes,
> --
> Răzvan Crainea
> OpenSIPS Core Developer
>http://www.opensips-solutions.com
>
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Re: [OpenSIPS-Users] Include ECDHE cipher suites in TLS

2020-01-23 Thread Callum Guy
Hi Ali,

Glad the suggestions were helpful.

The certificate is just a regular SSL cert, in PEM format just with a
different file extension. Depending on your implementation you might want
to look into public verifiable certificates (if you're public facing) -
available for free if you want to check out a service like
https://letsencrypt.org/.

Callum


On Wed, 22 Jan 2020 at 22:09, Ali Alawi  wrote:

> Dear Callum,
> Thanks a lot. it help me to establish a TLS connection with ECDH suite.
> but I used my own certificate.pem rather than the one you mentioned
> sip.crt. Actually, I couldn't figure out from where I can get this kind of
> .crt files.
>
> On Mon, Jan 20, 2020 at 11:49 AM Callum Guy  wrote:
>
>> Hi Ali,
>>
>> You'll need to setup your cipher list and DH file. You can generate a DH
>> param file like this: *openssl dhparam -out dhparam.pem 4096*
>>
>> If you want to review locally available cipher suites you can run: *openssl
>> ciphers -v*
>>
>> The OpenSIPs documentation clarifies the module configuration options
>> however the following setup will provide a set of strong ciphers and maybe
>> you can pick from this to add to your existing config to get things working.
>>
>> modparam("tls_mgm", "dh_params", "/etc/pki/tls/certs/dhparam.pem")
>> modparam("tls_mgm", "ec_curve", "secp384r1")
>> modparam("tls_mgm", "ciphers_list",
>> "EECDH+AESGCM,EDH+AESGCM,AES256+EECDH,AES256+EDH")
>> modparam("tls_mgm", "verify_cert", "1")
>> modparam("tls_mgm", "require_cert", "1")
>> modparam("tls_mgm", "tls_method", "TLSv1_2")
>> modparam("tls_mgm", "certificate", "/etc/pki/tls/certs/sip.crt")
>> modparam("tls_mgm", "private_key", "/etc/pki/tls/private/sip.key")
>> modparam("tls_mgm", "ca_list", "/etc/pki/tls/certs/ca-bundle.crt")
>> modparam("tls_mgm", "ca_dir", "/etc/pki/tls/certs/")
>>
>> Good luck,
>>
>> Callum
>>
>> On Sat, 18 Jan 2020 at 20:32, Ali Alawi  wrote:
>>
>>> Hello every one.
>>> I am trying to test TLS in OpenSIPS 2.4, the testing is going fine but
>>> it only support certain cipher suite methods such as (
>>>
>>> AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA
>>> )
>>> For some reason, I need to use ECDHE cipher suites but it is unsupported
>>> here.
>>> How can I include ECDHE in my TLS test?
>>> BTW, I am using OpenSSL 1.0.2g
>>>
>>> ALi
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
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>>   <https://twitter.com/xonuk> *
>>
>> 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 attachments
>> for viruses. X-on makes no representation or warranty as to the absence
>> of viruses in this email or any attachments.
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Re: [OpenSIPS-Users] Memory leak - solved

2020-01-20 Thread Callum Guy
Hi Liviu,

Thanks for coming back to me.

I am continuing to have memory problems on these servers however I have
stemmed the flow as per my previous emails here. I am still in an
uncomfortable position where the systems need to be restarted weekly so I'd
like to get this resolved as soon as possible.

To provide a very quick rundown of the history, this is a new registrar as
part of our migration to v3. The instances run in clustered pairs in
hot-standby (full-sharing) using a shared database. Initially I didn't
notice the issue as I had excessive memory allocated to each process (8GB!)
however as more UACs were migrated we eventually exhausted this. The first
step was to simply respond to handset NOTIFY pings at the start of the
script which prevented all routes executing and bypassing the leak. After
this I noticed memory usage continuing to grow and applied the patch
described above which helped to further reduce the leak indicating that the
change was relevant however there is something else there.

As a production instance I am currently looking into deploying a
development server to allow me to dig deeper into the problem however it
would be great if you could take a look at my config - that would be very
much welcomed. Would you like me to send you a complete config to
li...@opensips.org directly? If you'd prefer for me to trim it then I can
remove what I think are irrelevant sections however I'm concerned that I
might trim something that would be helpful in your review, just let me know
what you would prefer and any other details that could be useful.

Thanks in advance,

Callum


On Mon, 20 Jan 2020 at 09:50, Liviu Chircu  wrote:

> Hi Callum,
>
> Sorry for the late follow-up: did you make any progress with your leak?
> If not, could you prepare a minimal opensips.cfg that exposes the
> problem?  A quick
> code review did not show any obvious leaks, so I suspect there is something
> about your specific script that I am overlooking.
>
> Best regards,
>
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
> OpenSIPS Summit, Amsterdam, May 2020
>www.opensips.org/events/Summit-2020Amsterdam
> OpenSIPS Bootcamp, Miami, March 2020
>www.opensips.org/training
>
> On 09.12.2019 13:13, Callum Guy wrote:
> > Hi All,
> >
> > I wanted to follow up on a recent issue I experienced to understand if
> > it was due to user error or a bug that needs to be patched.
> >
> > The issue was traced back to a simple function call in the permissions
> module:
> >
> > check_source_address(0, $avp(address_desc))
> >
> > Nearly every request processed would have been an unlisted source
> > address and a negative response would have been expected. As an in
> > memory hash lookup for a small address list (<50 records) this seemed
> > like a very safe operation to perform.
> >
> > The AVP is uninitialised at the point of invocation - I am guessing
> > that this is key to the problem. To resolve the problem I have simply
> > removed the AVP and the method call is now:
> >
> > check_source_address(0)
> >
> > I would like to learn whether using an AVP for this operation was
> > incorrect or whether there was another reason for the leak. I've had a
> > go at reviewing the source for permissions and pvar however I quickly
> > got lost trying to find where the AVP initialisation would have been
> > invoked. Any advice would be appreciated.
> >
> > Many thanks,
> >
> > Callum
> >
>
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: [OpenSIPS-Users] Include ECDHE cipher suites in TLS

2020-01-20 Thread Callum Guy
Hi Ali,

You'll need to setup your cipher list and DH file. You can generate a DH
param file like this: *openssl dhparam -out dhparam.pem 4096*

If you want to review locally available cipher suites you can run: *openssl
ciphers -v*

The OpenSIPs documentation clarifies the module configuration options
however the following setup will provide a set of strong ciphers and maybe
you can pick from this to add to your existing config to get things working.

modparam("tls_mgm", "dh_params", "/etc/pki/tls/certs/dhparam.pem")
modparam("tls_mgm", "ec_curve", "secp384r1")
modparam("tls_mgm", "ciphers_list",
"EECDH+AESGCM,EDH+AESGCM,AES256+EECDH,AES256+EDH")
modparam("tls_mgm", "verify_cert", "1")
modparam("tls_mgm", "require_cert", "1")
modparam("tls_mgm", "tls_method", "TLSv1_2")
modparam("tls_mgm", "certificate", "/etc/pki/tls/certs/sip.crt")
modparam("tls_mgm", "private_key", "/etc/pki/tls/private/sip.key")
modparam("tls_mgm", "ca_list", "/etc/pki/tls/certs/ca-bundle.crt")
modparam("tls_mgm", "ca_dir", "/etc/pki/tls/certs/")

Good luck,

Callum

On Sat, 18 Jan 2020 at 20:32, Ali Alawi  wrote:

> Hello every one.
> I am trying to test TLS in OpenSIPS 2.4, the testing is going fine but it
> only support certain cipher suite methods such as (
>
> AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA
> )
> For some reason, I need to use ECDHE cipher suites but it is unsupported
> here.
> How can I include ECDHE in my TLS test?
> BTW, I am using OpenSSL 1.0.2g
>
> ALi
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] NAT for in-dialog requests

2020-01-16 Thread Callum Guy
Hi Bogdan,

Apologies for the late reply, I've been away on paternity leave but your
response is very much appreciated as I resume work on the project.

Thank you for the clarification, I didn't expect a one size fits all
approach to NAT but wanted to confirm that I wasn't missing a trick.

Callum

On Tue, 7 Jan 2020 at 12:41, Bogdan-Andrei Iancu 
wrote:

> Hi Callum,
>
> I would say you are on the right tracks.
>
> For detecting different NAT situation, there is no other way than to
> play with the flags of the `nat_uac_test()`.
>
> For the in-dialog NAT handling, during the initial request you need to
> learn the NAT status of the parties involved (for the caller, based on
> the nat_uac_test(), for the callee, based on the data from lookup()).
> Once you learned the caller/callee 's NAT status, you should preserve
> this information across the whole dialog, so, when handling in-dialog
> requests, you already know the NAT status of each party.
>
> And yes, when you receive traffic (requests or replies) for a party
> behind the NAT, you need to do the Contact fixing (and SDP if the case),
> in order to get rid of the private IPs.
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS Summit, Amsterdam, May 2020
>https://www.opensips.org/events/Summit-2020Amsterdam/
> OpenSIPS Bootcamp, Miami, March 2020
>https://opensips.org/training/OpenSIPS_Bootcamp_2020/
>
> On 12/13/19 2:12 PM, Callum Guy wrote:
> > Hi All,
> >
> > I am operating a registrar which proxies calls to an internal network
> > of media servers.
> >
> > Most of my subscribers are operating using RFC1918 addresses behind
> > NAT. We detect this configuration through nat_uac_test() and patch the
> > SIP using fix_nated_contact(). By rewriting the requests the media
> > servers will send in-dialog requests with request URI set to the NAT
> > address.
> >
> > This works fine for the majority of my use cases however I am now
> > dealing with a UAC which has chosen to use public addresses on the
> > handsets but continues to run behind NAT. The effect here is that my
> > choice of flags for nat_uac_test() do not match the scenario and the
> > rewrites are not happening.
> >
> > I can resolve this issue with some additional flags and logic however
> > I wonder if there is a "more correct" way to do this. In an ideal
> > world I would lookup the registration during INVITE processing, notice
> > that there is a different received address and use this for all future
> > communications for the rest of the dialog.
> >
> > For example the initial requests out to these handsets are performing
> > a lookup() operation which will retrieve the original contact for use
> > as request URI and received address for use as destination URI
> > allowing the request to be properly formed and forwarded. The issue
> > arises when the dialog is started, the UAC responds to the well formed
> > INVITE with 200 OK and unless I patch the contact with the received IP
> > the subsequent ACK will end up routing back to the UAC contact rather
> > than the NAT device. What I am hoping is that there may be a correct
> > way for me to track the received IP against the dialog such that it
> > can be used in subsequent in-dialog requests, allowing the request URI
> > to correctly represent the UAC contact whilst still delivering
> > requests to the NAT address. I can probably achieve this by recording
> > the information using dialog variables however I imagine this is a
> > common scenario so I wouldn't want to add this logic in manually if
> > there was a proper way to do this natively in OpenSIPs.
> >
> > Hopefully this isn't one of those questions that gets asked too
> > regularly however if it is please point me in the direction of an
> > article that might help, I've re-read the nathelper, nat_trasveral,
> > registrar and usrloc documentation today and haven't found what I'm
> > looking for. Any pointers would be appreciated before I embark on a
> > homegrown solution.
> >
> > Thanks,
> >
> > Callum
> >
>
>

-- 





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<https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel> 
  <https://twitter.com/xonuk> *


X-on
is a trading name of Storacall 
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Registered Office : Avaland House, 110 London Road, Apsley, Hemel 
Hempstead,
Herts, HP3 9SD. Company Registration No. 2578478.


[OpenSIPS-Users] NAT for in-dialog requests

2019-12-13 Thread Callum Guy
Hi All,

I am operating a registrar which proxies calls to an internal network
of media servers.

Most of my subscribers are operating using RFC1918 addresses behind
NAT. We detect this configuration through nat_uac_test() and patch the
SIP using fix_nated_contact(). By rewriting the requests the media
servers will send in-dialog requests with request URI set to the NAT
address.

This works fine for the majority of my use cases however I am now
dealing with a UAC which has chosen to use public addresses on the
handsets but continues to run behind NAT. The effect here is that my
choice of flags for nat_uac_test() do not match the scenario and the
rewrites are not happening.

I can resolve this issue with some additional flags and logic however
I wonder if there is a "more correct" way to do this. In an ideal
world I would lookup the registration during INVITE processing, notice
that there is a different received address and use this for all future
communications for the rest of the dialog.

For example the initial requests out to these handsets are performing
a lookup() operation which will retrieve the original contact for use
as request URI and received address for use as destination URI
allowing the request to be properly formed and forwarded. The issue
arises when the dialog is started, the UAC responds to the well formed
INVITE with 200 OK and unless I patch the contact with the received IP
the subsequent ACK will end up routing back to the UAC contact rather
than the NAT device. What I am hoping is that there may be a correct
way for me to track the received IP against the dialog such that it
can be used in subsequent in-dialog requests, allowing the request URI
to correctly represent the UAC contact whilst still delivering
requests to the NAT address. I can probably achieve this by recording
the information using dialog variables however I imagine this is a
common scenario so I wouldn't want to add this logic in manually if
there was a proper way to do this natively in OpenSIPs.

Hopefully this isn't one of those questions that gets asked too
regularly however if it is please point me in the direction of an
article that might help, I've re-read the nathelper, nat_trasveral,
registrar and usrloc documentation today and haven't found what I'm
looking for. Any pointers would be appreciated before I embark on a
homegrown solution.

Thanks,

Callum

-- 





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   *


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
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[OpenSIPS-Users] Memory leak - solved

2019-12-09 Thread Callum Guy
Hi All,

I wanted to follow up on a recent issue I experienced to understand if
it was due to user error or a bug that needs to be patched.

The issue was traced back to a simple function call in the permissions module:

check_source_address(0, $avp(address_desc))

Nearly every request processed would have been an unlisted source
address and a negative response would have been expected. As an in
memory hash lookup for a small address list (<50 records) this seemed
like a very safe operation to perform.

The AVP is uninitialised at the point of invocation - I am guessing
that this is key to the problem. To resolve the problem I have simply
removed the AVP and the method call is now:

check_source_address(0)

I would like to learn whether using an AVP for this operation was
incorrect or whether there was another reason for the leak. I've had a
go at reviewing the source for permissions and pvar however I quickly
got lost trying to find where the AVP initialisation would have been
invoked. Any advice would be appreciated.

Many thanks,

Callum

-- 





*0333 332   |  www.x-on.co.uk   |   ** 
    
   *


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
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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
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Re: [OpenSIPS-Users] Memory Leak - RTP Engine?

2019-12-06 Thread Callum Guy
Hi All,

Just wanted to come back on this thread to confirm that the memory
leak issue is now under control although not directly identified.

After many variations of module configuration parameters were
attempted to no avail I went back to basics and simply identified our
most frequently processed packets which were inbound SIP NOTIFY
handset keep alive pings. Identification and processing for these was
moved to the top of the core route and as soon as this was deployed
the memory usage increases dropped from 200MB/hour to maybe 2MB/hour -
enough for this to no longer be a major issue.

Following on from that I have now identified the actions that would
have been performed on these NOTIFY packets prior to the change as one
of these must be responsible for the memory allocation. The methods
are listed below:

check_source_address(0, $avp(address_desc))
t_on_reply("example")
record_route();
fix_nated_contact();
add_rr_param(";nat=yes");

As you can tell none of these were ever actually required for PING
processing however all would have been called ahead of the previous
ping processing block. I'm out of my depth here however if I was to
consider one of these as a most likely candidate for the issue it
would be check_source_address. For the device pings in question these
would never match an address and would therefore presumably return
NULL context? If so would this mean that an AVP could be allocated a
memory location? I've simply removed the avp assignment at this point
however I would like to know if there is any chance that hitting this
repeatedly could cause an issue?

Anyway, onwards and upwards - just happy that I don't need to restart
it every night now :)

Callum

P.S. apologies for my employers banner.. I believe my last message was
caught in your spam bin for being too large


On Tue, 3 Dec 2019 at 17:31, Johan De Clercq  wrote:
>
> I think you can. Check the documentation of rtpengine on github. And if you 
> can, please use the latest commit.
>
> On Tue, 3 Dec 2019, 18:02 Callum Guy,  wrote:
>>
>> Hi All,
>>
>> I'm working through this memory issue and have some additional data. The 
>> server crashed this afternoon due to memory exhaustion on a UDP listener 
>> process, as far as I know at this time. Note that the processes all have 4GB 
>> assigned so this is a gradual and constant growth issue.
>>
>> At this point I have my suspicions that it could be related to the RTPEngine 
>> module - the first memory allocation error messages are all related to what 
>> I believe the be RTCP session reports coming in from the RTP engine servers. 
>> These came through about every second for an hour before OpenSIPs finally 
>> gave up the ghost and restarted.
>>
>> Is this normal behaviour? We have only recently moved to RTPEngine from 
>> RTPProxy so I am new to this software. Does anyone know if it is possible to 
>> prevent RTPEngine from sending this data to OpenSIPs - its not something 
>> that we require and I'd like to check if this is at all related to the 
>> memory growth. Any other ideas would also be appreciated!
>>
>> Many thanks,
>>
>> Callum
>>
>> 2019-12-03T11:48:14.653225+00:00 TH-P-SIPREG-1 opensips[2521]: 
>> ERROR:core:fm_malloc: not enough free pkg memory (214628248 bytes left, need 
>> 536), please increase the "-M" command line parameter!
>> 2019-12-03T11:48:14.653851+00:00 TH-P-SIPREG-1 opensips[2521]: 
>> INFO:core:fm_malloc: attempting defragmentation...
>> 2019-12-03T11:48:15.651495+00:00 TH-P-SIPREG-1 opensips[2521]: 
>> INFO:core:fm_malloc: unable to alloc a big enough fragment!
>> 2019-12-03T11:48:15.652218+00:00 TH-P-SIPREG-1 opensips[2521]: 
>> ERROR:rtpengine:rtpe_function_call: failed to decode bencoded reply from 
>> proxy: d7:createdi1575373650e10:created_usi505483e11:last 
>> signali1575373670e4:SSRCd10:3831331386d11:average MOSd3:MOSi44e15:round-trip 
>> timei6754e6:jitteri0e11:packet lossi0e7:samplesi9ee10:lowest 
>> MOSd3:MOSi44e15:round-trip timei6175e6:jitteri0e11:packet lossi0e11:reported 
>> ati1575373656ee11:highest MOSd3:MOSi44e15:round-trip 
>> timei6175e6:jitteri0e11:packet lossi0e11:reported ati1575373656ee15:MOS 
>> progressiond8:intervali3e7:entriesld3:MOSi44e15:round-trip 
>> timei6175e6:jitteri0e11:packet lossi0e11:reported 
>> ati1575373656eed3:MOSi44e15:round-trip timei6307e6:jitteri0e11:packet 
>> lossi0e11:reported ati1575373660eed3:MOSi44e15:round-trip 
>> timei6884e6:jitteri0e11:packet lossi0e11:reported 
>> ati1575373664eed3:MOSi44e15:round-trip timei7016e6:jitteri0e11:packet 
>> lossi0e11:reported ati1575373668eed3:MOSi44e15:round-trip 
>> timei6368e6:jitteri0e11:packet lossi0e11:reported 
>>

Re: [OpenSIPS-Users] Opensips 3 & rtpengine

2019-12-04 Thread Callum Guy
Methods are almost interchangeable though - check out the docs:

https://opensips.org/html/docs/modules/3.0.x/rtpengine.html


On Wed, 4 Dec 2019 at 21:29, David Villasmil 
wrote:

> are you setting up rtpENGINE or rtpPROXY?
> They're not the same...
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Wed, Dec 4, 2019 at 9:14 PM info  wrote:
>
>> Hello,
>>
>>
>>
>> I setup an opensips 3 and I want to load rtpproxy.so like this :
>>
>>
>>
>> loadmodule "rtpproxy.so"
>>
>> modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:2223")
>>
>>
>>
>> I install and setup the master of rtpengine
>>
>>
>>
>> /usr/sbin/rtpengine -v
>>
>> Version: 8.2.0.0+0~mr8.2.0.0 git-master-c36a445
>>
>>
>>
>> Rtpengine is running and listen on port 2223 (udp)
>>
>>
>>
>> netstat -anp | grep 2223
>>
>> udp0  0 127.0.0.1:2223  0.0.0.0:*
>> 3484/rtpengine
>>
>>
>>
>>
>>
>> when I start opensips I get in log :
>>
>>
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3571]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3571]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3578]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3578]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3577]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3577]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3572]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3572]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3575]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3575]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3579]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3579]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3567]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3567]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3586]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3586]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3581]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3581]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3584]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3584]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3568]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3568]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3576]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3576]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Dec  4 22:11:50 opensips1 /usr/local/sbin/opensips[3587]:
>> WARNING:rtpproxy:rtpp_test: unsupported version of RTP proxy
>>  found: 20040107 supported,0 present
>>
>> Dec  4 22:11:50 opensips1

Re: [OpenSIPS-Users] Memory Leak

2019-12-04 Thread Callum Guy
Thanks Johan, I've been through the rtpengine docs and source and can't see
any way to disable the RTCP call stat reports. Having said that I'm no
longer confident that RTPEngine is the problem - I'm losing memory
constantly at a rate of about 130MB/hour at all times, even overnight when
we are not seeing many calls. This seems to rule out the RTP reports or
anything call load related.

I now wonder if the issue is related to registrations and NAT pings (~3500
registrations). This is the only constant activity on the system overnight
(other than OpenSIPs clustering?). NAT pings are enabled for all contacts
and many of the handsets are also pinging the server, according to client
config.

At this time I am not sure where to focus my attention - I've collected
some data from my server (version, relevant module configs, memory dump) in
the hope that someone might be able to spot a problem with my setup - you
can find this at the following link:

https://gist.githubusercontent.com/spacetourist/2b880c71eda54bcabf691282d6389209/raw/25c42b90ffa4e592cd761237dffca6f85aabbf4f/gistfile1.txt

Please let me know if anything else would be useful. All ideas are very
much appreciated, thanks.

Callum


On Tue, 3 Dec 2019 at 17:31, Johan De Clercq  wrote:

> I think you can. Check the documentation of rtpengine on github. And if
> you can, please use the latest commit.
>
> On Tue, 3 Dec 2019, 18:02 Callum Guy,  wrote:
>
>> Hi All,
>>
>> I'm working through this memory issue and have some additional data. The
>> server crashed this afternoon due to memory exhaustion on a UDP listener
>> process, as far as I know at this time. Note that the processes all have
>> 4GB assigned so this is a gradual and constant growth issue.
>>
>> At this point I have my suspicions that it could be related to the
>> RTPEngine module - the first memory allocation error messages are all
>> related to what I believe the be RTCP session reports coming in from the
>> RTP engine servers. These came through about every second for an hour
>> before OpenSIPs finally gave up the ghost and restarted.
>>
>> Is this normal behaviour? We have only recently moved to RTPEngine from
>> RTPProxy so I am new to this software. Does anyone know if it is possible
>> to prevent RTPEngine from sending this data to OpenSIPs - its not something
>> that we require and I'd like to check if this is at all related to the
>> memory growth. Any other ideas would also be appreciated!
>>
>> Many thanks,
>>
>> Callum
>>
>> 2019-12-03T11:48:14.653225+00:00 TH-P-SIPREG-1 opensips[2521]:
>> ERROR:core:fm_malloc: not enough free pkg memory (214628248 bytes left,
>> need 536), please increase the "-M" command line parameter!
>> 2019-12-03T11:48:14.653851+00:00 TH-P-SIPREG-1 opensips[2521]:
>> INFO:core:fm_malloc: attempting defragmentation...
>> 2019-12-03T11:48:15.651495+00:00 TH-P-SIPREG-1 opensips[2521]:
>> INFO:core:fm_malloc: unable to alloc a big enough fragment!
>> 2019-12-03T11:48:15.652218+00:00 TH-P-SIPREG-1 opensips[2521]:
>> ERROR:rtpengine:rtpe_function_call: failed to decode bencoded reply from
>> proxy: d7:createdi1575373650e10:created_usi505483e11:last
>> signali1575373670e4:SSRCd10:3831331386d11:average
>> MOSd3:MOSi44e15:round-trip timei6754e6:jitteri0e11:packet
>> lossi0e7:samplesi9ee10:lowest MOSd3:MOSi44e15:round-trip
>> timei6175e6:jitteri0e11:packet lossi0e11:reported ati1575373656ee11:highest
>> MOSd3:MOSi44e15:round-trip timei6175e6:jitteri0e11:packet
>> lossi0e11:reported ati1575373656ee15:MOS
>> progressiond8:intervali3e7:entriesld3:MOSi44e15:round-trip
>> timei6175e6:jitteri0e11:packet lossi0e11:reported
>> ati1575373656eed3:MOSi44e15:round-trip timei6307e6:jitteri0e11:packet
>> lossi0e11:reported ati1575373660eed3:MOSi44e15:round-trip
>> timei6884e6:jitteri0e11:packet lossi0e11:reported
>> ati1575373664eed3:MOSi44e15:round-trip timei7016e6:jitteri0e11:packet
>> lossi0e11:reported ati1575373668eed3:MOSi44e15:round-trip
>> timei6368e6:jitteri0e11:packet lossi0e11:reported
>> ati1575373674eed3:MOSi44e15:round-trip timei6929e6:jitteri0e11:packet
>> lossi0e11:reported ati1575373679eed3:MOSi44e15:round-trip
>> timei6884e6:jitteri0e11:packet lossi0e11:reported
>> ati1575373684eed3:MOSi44e15:round-trip timei7241e6:jitteri0e11:packet
>> lossi0e11:reported ati1575373689eed3:MOSi44e15:round-trip
>> timei6988e6:jitteri0e11:packet lossi0e11:reported
>> ati1575373694e10:1028099660d11:average MOSd3:MOSi43e15:round-trip
>> timei29936e6:jitteri0e11:packet lossi0e7:samplesi7ee10:lowest
>> MOSd3:MOSi43e15:round-trip timei29515e6:jitteri0e11:packet
>> lossi0e11:reported ati15

Re: [OpenSIPS-Users] openSIPS- Dialplan Manipulation Help

2019-12-04 Thread Callum Guy
Why not just glue together the strings?

$var(a) = "972" + $rU;

Seems pretty effective to me :)

On Tue, 3 Dec 2019 at 22:02, VOIP Security via Users <
users@lists.opensips.org> wrote:

> Hi,
>
> I am struggling with openSIPS regex rules to append some prefix before
> regex. I have this regex rule setup for Israel numbers -
>
> Matching Regular Expression -  ^0[0-9]{8}$
>
> So this rule is catching all the 9 digits DID from my Israeli carrier but
> now I want to add 972 in front of any DID, for example DID - 021458547
> After dp_translate I want to manipulate that DID and add 972 so new DID
> gonna be - 972021458547
>
> Please help me setting up this dial plan rule. I will appreciate that.
>
> Thanks and regards
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] Memory Leak - RTP Engine?

2019-12-03 Thread Callum Guy
irmed10:kernelizede4:SSRCi3831331386e5:statsd7:packetsi2018e5:bytesi347096e6:errorsi0eeed10:local
porti34227e8:endpointd6:family4:IPv47:address15:192.168.163.2224:porti11485ee19:advertised
endpointd6:family4:IPv47:address15:192.168.163.2224:porti11485ee11:last
packeti1575373694e5:flagsl4:RTCP6:filled9:confirmed10:kernelized17:no
kernel
supporte4:SSRCi3831331386e5:statsd7:packetsi9e5:bytesi1008e6:errorsi05:flagsl11:initialized4:send4:recv15:ICE
controllinge6:totalsd3:RTPd7:packetsi4063e5:bytesi698836e6:errorsi0ee4:RTCPd7:packetsi17e5:bytesi2248e6:errorsi0eee6:result2:oke


On Sat, 30 Nov 2019 at 22:51, Callum Guy  wrote:

> Hi Ben,
>
> Thank you for your reply and insight here, very helpful to know you're
> running a drastically different setting for the package memory.
>
> I presumed if it preallocated that I would have seen some issues during
> testing, hence I've ended up with figures that were intended to provide 75%
> of the system memory to the application.
>
> Memory usage had been creeping up all day at the time of writing however
> migrations to this platform had been on hold since the initial capture of
> memory usage although call traffic would have been relatively even during
> daytime hours where the increase continued. On that basis I'm still
> concerned that there is an issue with my config causing this growth however
> I've now increased available memory and restarted so I should have ample
> time to investigate this week, I'll report back any findings for the
> community benefit. I will give some serious thought to lowering the package
> allocation value once I've got to grips with the situation.
>
> Usefully this implementation shares a lot of common components to another
> variant which acts as a pure proxy and does not deal with registrations
> where I'm not seeing this issue so that will narrow down the search area
> somewhat.
>
> Thanks again for your time,
>
> Callum
>
> On Sat, 30 Nov 2019, 15:46 Ben Newlin,  wrote:
>
>> Callum,
>>
>>
>>
>> It’s my understanding that OpenSIPS does not release memory back to the
>> OS, but it also pre-allocates all memory at startup into its private pool
>> and then allocates from that internally. Normally shared memory should be
>> significantly higher than package memory. For reference, on our system we
>> run with “-m 1024 -M 64” and that is sufficient for us to process very high
>> traffic volume. We don’t do registration though, so that may affect the
>> sizes you need.
>>
>>
>>
>> You are setting your package memory size to 4G, so that will allocate 4G
>> memory for every package (process) that loads and then 2G for shared
>> memory. That will use up all the memory on your machine extremely quickly
>> for sure. The statistics you provided seem like the memory increase is
>> consistent with higher traffic levels on the second reading. You can see in
>> your case that all of your “pkmem” processes have an extremely high amount
>> of free memory (~3GB!). But that memory is still allocated from the OS, so
>> you are instructing OpenSIPS to allocate much more than your system memory
>> right at startup.
>>
>>
>>
>> Your shared memory also has just under 2GB free, so you have a lot of
>> headroom there too. Since OpenSIPS pre-allocates, the amount of memory
>> being used by the system overall should be fairly steady; if it is
>> continuously increasing that implies a leak somewhere. IIRC there are a few
>> processes/modules/commands in OpenSIPS or libraries it uses that do
>> allocate memory directly from the system and not from OpenSIPS’ pool. You
>> may need to investigate some of those to find out where your memory is
>> going, or look at other processes/daemons you have running that could be
>> using that memory.
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users  on behalf of Callum Guy
>> 
>> *Reply-To: *OpenSIPS users mailling list 
>> *Date: *Friday, November 29, 2019 at 10:57 AM
>> *To: *OpenSIPS users mailling list 
>> *Subject: *[OpenSIPS-Users] Memory Leak - runtime flags?
>>
>>
>>
>> Hi All,
>>
>>
>>
>> I have recently deployed a new registrar and have been seeing a gradual
>> increase in the memory footprint - enough that I'm having to expand the RAM
>> (its virtualised) to ensure it doesn't run out.
>>
>>
>>
>> You can see a diff of the statistics collected last night at 11pm and
>> today at 3pm here:
>> https://gist.github.com/spacetourist/2103503674e134bd598c7f1e3a82674c/revisions
>>
>>
>>
>> Processes 5-9

  1   2   >