[SR-Users] Re: loose_route false when using topos? At which point in the call flow, does topos restore the Route header?

[Karsten Horsmann via sr-users]

Hi,

did you use some kind of db backend for topos like redis or mysql?

Benoit Panizzon via sr-users  schrieb am Mo.,
26. Feb. 2024, 16:28:

> Hi
>
> https://lists.kamailio.org/pipermail/sr-users/2020-July/109801.html
>
> I have the exact same issue.
>
> When the B side is starting a new transaction (UPDATE to refresh the
> session in my case) without topos enabled, that transaction contains
> one or multiple route header and a to_tag.
>
> Therefore loose_route() is true and the call is more or less sent to
> route(RELAY) immediately without further checks.
>
> If topos is enabled, all record-route header are removed and only one
> Via sent to the CPE.
> So when the CPE replies, there is no Route header.
> Therefore loose_route returns false.
>
> This makes me wonder, at which stage is topos restoring the route
> headers? Shouldn't that happen before loose_route is evaluated?
>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel  +41 61 826 93 00
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[SR-Users] Re: Reject TCP SYN

[Karsten Horsmann via sr-users]

Hi David,

I have most of my setups with OPTIONS pings (upstream carrier and myself).
If I want to fade out a Kamailio I set the option response to an not 200
value (like 603 or whatever) and my internal and the external system didn't
bring in new invites but leave the currently running out.

You can use cfg values for that or some kind of htable with value or so.

Maybe an idea for your setup.

David Villasmil via sr-users  schrieb am Fr.,
9. Feb. 2024, 15:04:

>
> Hey, Henning, yeah I thought about that, but thought that maybe there was
> a better way to do it via Kamailio
>
> Thanks!
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Fri, 9 Feb 2024 at 14:08, Henning Westerholt  wrote:
>
>> Hello,
>>
>>
>>
>> what about e.g. just using something like iptables, nftables etc..?
>>
>>
>>
>> iptables -A INPUT -p tcp --syn --destination-port  -j REJECT
>> --reject-with icmp-host-prohibited
>>
>>
>>
>> Cheers,
>>
>>
>>
>> Henning
>>
>>
>>
>> *From:* David Villasmil via sr-users 
>> *Sent:* Donnerstag, 8. Februar 2024 14:28
>> *To:* Kamailio (SER) - Users Mailing List 
>> *Cc:* David Villasmil 
>> *Subject:* [SR-Users] Reject TCP SYN
>>
>>
>>
>> Hello all,
>>
>>
>>
>> Is there any way of actually rejecting (RST) NEW tcp connection attempts,
>> while allowing the ongoing ones to finish naturally?
>>
>>
>>
>> I’m thinking maybe we can add this feature?
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
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[SR-Users] Re: Latest point releases seem to break early dialog UPDATEs from B-side when using topos (again)

[Karsten Horsmann via sr-users]

Hi George,


if you think it's a issue please fill a bug report on Github to let the
Devs investigate. Also the questions in the form ask for all what they
need.

Thank you

George Diamantopoulos via sr-users  schrieb am
Do., 15. Feb. 2024, 16:06:

> Update: I have just downgraded to 5.6.4 using
> https://deb-archive.kamailio.org/repos/kamailio-5.6.4
>
> I have confirmed that topos works correctly with this version, so the
> commit that broke things happened between 5.6.4 and 5.6.5
>
> It might be related to how multi-homed installations interact with topos
> when doing double rr. Tne difference I noticed is that in 5.6.4, the topos
> substitution in Via is:
> * inbound: Via: SIP/2.0/UDP 195.167.21.66:5060
> ;branch=z9hG4bKq27z808ysq87yrzq6vp8m26yq;Role=3;Hpt=8e88_16
> * outbound: Via: SIP/2.0/UDP
> 172.30.154.1;branch=z9hG4bKae0b.fce5449c22b5e79eb01d4f1ab3f7c014.0
>
> That is kamailio replaces the hop with its internal interface. With topos
> on in 5.6.5, the behaviour is different for the UPDATE:
> * inbound: Via: SIP/2.0/UDP 195.167.21.66:5060
> ;branch=z9hG4bK1jo121j8j2h5h3wzh2mzj51q5;Role=3;Hpt=8ea8_16
> * outbound: Via: SIP/2.0/UDP
> 185.73.42.241;branch=z9hG4bKc45b.e6226bcf14264ae3b8674dd25a71191c.0
>
> 185.73.42.241 is an interface on the same instance, but represents its
> "public" interface. Additionally, it happens to reside on a VRF, so the
> 200OK never reaches it because the routing table of the UAS directs it to
> another host with the same IP.
>
> Hopefully this is enough to get an idea of what might have gone wrong?
> Thanks!
>
> On Thu, 15 Feb 2024 at 15:55, George Diamantopoulos 
> wrote:
>
>> Hello all,
>>
>> I've noticed that there seems to be a regression with the topos module,
>> more specifically the redis flavour, but I'm assuming the storage backend
>> shouldn't make a difference.
>> I have confirmed this affects both 5.6.5 and 5.7-nightly, so I'm assuming
>> some backported commit is to blame. Kamailio 5.6.4 used previously, to the
>> best of my memory, was not affected.
>>
>> Early dialog UPDATEs sent from the callee seem to be somehow malformed,
>> since the 200 OK the UAS produces goes directly to the UAC despite having
>> executed record_route() for the original INVITE. I can't imagine how this
>> is possible, I thought responses always honour the Via header and record
>> route shouldn't play a role in this case, right?
>>
>> Simply turning topos off restores the intended behaviour, however, so I
>> can't help but think this is somehow related. Here's two call flows
>> demonstrating the behaviour:
>> * Topos on, 200 OK to UPDATE missing: https://pastebin.com/raw/J0zQeM5g
>> * Topos off, 200 OK to UPDATE routed correctly:
>> https://pastebin.com/raw/49yErezb
>>
>> I was wondering if anyone is aware of any commits that might be
>> responsible for this. Additionally, is there an archive of debian packages
>> with previous point releases so that I can confirm this regression with the
>> latest versions?
>>
>> Thank you!
>>
>> Best regards,
>> George
>>
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[SR-Users] Re: How to know registration status

[Karsten Horsmann via sr-users]

Hi,

I use the rpc stuff for nagios monitoring the uac register status and give
some alerts (I watch you DTAG).

On the other side the in script states are good to reply for example with
503 or whatever sip rc to get this disabled.

Just as Sergiu described build stuff for your needs.



Sergiu Pojoga via sr-users  schrieb am Di., 9.
Jan. 2024, 05:17:

> Have you consulted the UAC module documentation? It's all in there. You
> won't get faster help than that.
>
> BTW, it won't tell you the latest reply status code per se, usually that's
> not a direct indicator of a registration status. If you are looking for
> that specifically - then you'll have to build some kind of [in-memory]
> [htable] storage to keep track of the latest reply status codes for each
> r_uuid, somewhere in the reply route.
>
> RPC
> https://kamailio.org/docs/modules/devel/modules/uac.html#uac.r.uac.reg_info
>
> Within script
>
> https://kamailio.org/docs/modules/devel/modules/uac.html#uac.f.uac_reg_status
>
> On Mon, Jan 8, 2024 at 10:26 PM mm e via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> After using the uac module to initiate registration to the remote end, is
>> there a method to call back to notify the registration result? I want to
>> get the registration result, such as 200 or 403
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[SR-Users] Re: How to filter out xhttp request in event_route[siptrace:msg]

[Karsten Horsmann via sr-users]

Hi Benoit,


how are you measure the traffic you don't want to your Homer? Print stuff
in that Kamailio event route or something else?

Kind regards
Karsten Horsmann



Benoit Panizzon via sr-users  schrieb am Di.,
17. Okt. 2023, 13:54:

> Hi
>
> Further trying to eliminate every possible cause of UDP drops...
>
> We use Homer as HEP server in conjunction with the siptrace module.
>
> So to prevent DQM traffic to be sent to Homer I added:
>
> event_route[siptrace:msg] {
> if(is_method("KDMQ")) {
> drop();
> }
> }
>
> And had a closer look to the traffic sent to homer.
>
> I noticed, also xhttp:requests, as far as I see the replies, not the
> requests, get mirrored to homer.
>
> xhttp is not a sip method I could match I guess.
>
> The event_route[siptrace:msg] docs state only method can be filtered.
> So I can not filter requests on the separate http port I listen on for
> xhttp json-rpc requests.
>
> How can I prevent xhttp traffic to be mirrored to homer by the siptrace
> module?
>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel  +41 61 826 93 00
> CH-4133 PrattelnFax  +41 61 826 93 01
> Schweiz Web  http://www.imp.ch
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[SR-Users] Re: 503(service unavailable)

[Karsten Horsmann via sr-users]

Hi,

I would go with the advise from Daniel to upgrade your Kamailio and if the
error stays then come back or fill a issue on Github.

And your server is really, really good sized for your use. I use VMs with
much less specs and a higher CPS.

Kind regards

Masud via sr-users  schrieb am Fr., 13. Okt.
2023, 09:10:

> Hello.
> Daniel can you help with these questions?)
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[SR-Users] Re: UDP Fragmentation webRTC/UDP Call

[Karsten Horsmann via sr-users]

Hi,

did you pass over the ice stuff from webrtc to the udp side? You could
strip that of with rtpengine options.

Social Boh via sr-users  schrieb am Sa., 14.
Okt. 2023, 01:00:

> Hello,
>
> I'm trying to solve a problem with calls WebRTC/UDP. I think is a
> fragmentation problem because udp/webRTC calls works without problem. The
> INVITE from Kamailio to the UDP device not receive any type of answer.
>
> I'm trying to reduce the SDP annex size using:
> sdp_remove_line_by_prefix
>
> for some lines unsuccessfully. Maybe I don't know where put this function
> in the script.
>
> Or, is there another way to solve this issue?
>
> Thank you in advance
>
> Regards
>
> --
> ---
> I'm SoCIaL, MayBe
>
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[SR-Users] Re: Multiple headers in http client request

[Karsten Horsmann]

Hi,

Maybe an kemi script would be an good choice here? So you can choose from
different script languages that kemi supports (Lua, python, js and many
more that I forgot)? You can also mix classic Kamailio scripting language
with kemi if that http request is the only business logic that need more
complex stuff.



Matteo Brancaleoni  schrieb am Mi., 13. Sept. 2023,
10:31:

> Ok,
>
> seems that the Content-Type is added automatically by libcurl when
> performing a POST request, unless it detects an already present hdr in the
> headers params.
>
> Seems that inverting the order of headers, so building the string with
> "Content-Type: application/json\r\nAuthorization: Bearer sometoken" seems
> to work, probably because libcurl detects the header and does not add it's
> own.
>
> But this is something "magic", not documented and prone to breakage, since
> for libcurl headers should be a linked list of strings.
>
> What do you suggest to properly handle that? Use scripting? Propose a
> patch to split the headers string into a libcurl linked list?
>
> br,
> Mat
>
> On Tue, Sep 12, 2023 at 5:53 PM Matteo Brancaleoni 
> wrote:
>
>> Hello all :)
>>
>> I'm struggling to add two headers to an http client request, but cannot
>> find a proper way to do it.
>>
>> The headers are Authorization and Content-Type.
>>
>> The first try is with http_client module, beside undocumented, is
>> possible to add multiple headers separating them with "\r\n". What I see is
>> the correct Authorization header, and two Content-Type headers, one with my
>> content type the other with x-www-form-urlencoded, which no one sets ?
>>
>> Second try is with http_async_client which supports multiple headers, but
>> since I'm calling it from an xhttp event route, there's no transaction and
>> request cannot be suspended, so I cannot get results before sending out
>> xhttp response...
>>
>> Any hint?
>>
>> br,
>> Mat
>>
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[SR-Users] Re: Kamailio behind TLS-TCP load balancer

[Karsten Horsmann]

Hi Henning,

Yeah TLS offloading is then not an plus but the benefits from DDOS
prevention and clear transport headers should compensate that.


Henning Westerholt  schrieb am Mi., 16. Aug. 2023, 10:09:

> Hi Karsten,
>
>
>
> interesting scenario, thanks. Regarding TLS off-loading its of course less
> useful then.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* Karsten Horsmann 
> *Sent:* Dienstag, 15. August 2023 20:24
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] Re: Kamailio behind TLS-TCP load balancer
>
>
>
> Hi,
>
>
>
> an benefit from using the AWS loadbalancer is the included DDOS
> prevention. Jonas Swiatek gave that as tipp to me.
>
>
>
> He simple setup self-signed certs on the Kamailio behind the NLB
> loadbalancer.
>
> So it's
>
>
>
> Internet -> TLS NLB/AWS loadbalancer -> TLS self-signed Kamailio.
>
>
>
> Should solve your problems.
>
>
>
> Kind regards
>
> Karsten Horsmann
>
> Henning Westerholt  schrieb am Sa., 12. Aug. 2023, 11:09:
>
> Hello David,
>
>
>
> the simplest way is of course to just not use the AWS load-balancer. 
> Do you have performance concerns using Kamailio for that purpose?
>
>
>
> As you probably know, SIP as a protocol is not really suited for this kind
> of cloud balancing infrastructure, which targets more HTTP and other
> protocols. And Kamailio in a load-balancer scenario is usually the first
> TLS/TCP/UDP endpoint to reach from the client point of view.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* David Villasmil 
> *Sent:* Samstag, 12. August 2023 02:55
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] Kamailio behind TLS-TCP load balancer
>
>
>
> Hello all,
>
>
>
> I’m having lots of problems when trying to configure Kamailio behind an
> AWS tls load balancer to offload tls and receive on tcp on Kamailio.
> Everything else inside is UDP.
>
> I found I need to manually add record-route presets every time and invite
> comes in. And when trying to forward an ACK to the client via tls/tcp load
> balancer Kamailio complaint the socket is not TLS so it fails.
>
>
>
> Is there a simpler way of doing this via some parameters I don’t know?
>
>
>
> Thanks for helping me with this!
>
>
>
> David
>
> --
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
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[SR-Users] Re: Kamailio behind TLS-TCP load balancer

[Karsten Horsmann]

Hi,

an benefit from using the AWS loadbalancer is the included DDOS prevention.
Jonas Swiatek gave that as tipp to me.

He simple setup self-signed certs on the Kamailio behind the NLB
loadbalancer.
So it's

Internet -> TLS NLB/AWS loadbalancer -> TLS self-signed Kamailio.

Should solve your problems.

Kind regards
Karsten Horsmann

Henning Westerholt  schrieb am Sa., 12. Aug. 2023, 11:09:

> Hello David,
>
>
>
> the simplest way is of course to just not use the AWS load-balancer. 
> Do you have performance concerns using Kamailio for that purpose?
>
>
>
> As you probably know, SIP as a protocol is not really suited for this kind
> of cloud balancing infrastructure, which targets more HTTP and other
> protocols. And Kamailio in a load-balancer scenario is usually the first
> TLS/TCP/UDP endpoint to reach from the client point of view.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* David Villasmil 
> *Sent:* Samstag, 12. August 2023 02:55
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] Kamailio behind TLS-TCP load balancer
>
>
>
> Hello all,
>
>
>
> I’m having lots of problems when trying to configure Kamailio behind an
> AWS tls load balancer to offload tls and receive on tcp on Kamailio.
> Everything else inside is UDP.
>
> I found I need to manually add record-route presets every time and invite
> comes in. And when trying to forward an ACK to the client via tls/tcp load
> balancer Kamailio complaint the socket is not TLS so it fails.
>
>
>
> Is there a simpler way of doing this via some parameters I don’t know?
>
>
>
> Thanks for helping me with this!
>
>
>
> David
>
> --
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
> __
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[SR-Users] Re: Kamailio failover in aws/azure environment

[Karsten Horsmann]

Hi,

I would setup an additional Kamailio only for the internal stuff that does
what you need.

You can combine it with your external and point all asterisk to the
internal which makes the routing decision asterisk to asterisk or asterisk
to external.

To put everything in one Kamailio is possible but of the cost of
complexity.

Kind regards
Karsten Horsmann

M Arqum CH  schrieb am Do., 15. Juni 2023, 21:48:

> Hi members,
>
> I am setting up kamailio  as loadbalcer for asterisk server farm with
> active-backup conf in Azure cloud using azure loadbalcer, for incoming it
> works fine external incoming traffic with nated external IP,
> can any one guide how to do active-backup configuration for internal
> traffic, as asterisk servers are running on local network.
>
>
> Thanks and regards
>
> Arqum
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[SR-Users] Re: TLS errors

[Karsten Horsmann]

Hi,


First I didn't run the 5.7 right now.

AFAIK 1.0.2k is an additional openssl lib in Centos 7.9. Default is 1.0.1*
and the RPMS (I use also Centos 7.9) from kamailio.org are build agains the
1.0.1* (5.5.x are).

Did you test your certs for example in a httpd or so that they created
right?

Did you try it with an tls.cfg?

Then all other modparms are obsolete except the config param.

loadmodule "tls.so"
# - tls params -
modparam("tls", "config", "/etc/kamailio/tls.cfg")


tls.cfg example:

[server:default]
method = TLSv1.2+
verify_certificate = no
require_certificate = no
private_key = /etc/pki/tls/private/YOURDOMAIN.pem  certificate =
/etc/pki/tls/private/YOURDOMAIN.pem
server_name = yourdomain.example
cipher_list =
HIGH:!3DES:!DES:!aDH:!AECDH:!CAMELLIA128:!CAMELLIA256:!CAMELLIA:!ADH:!SHA1

Kind Regards
Karsten Horsmann

nutxase  schrieb am Do., 15. Juni 2023, 15:48:

> i have definately confirmed the certs are valid and that i am using latest
> openssl for centos 7
>
> perhaps im missing something else?
> i did take the example from
>
> https://github.com/kamailio/kamailio/blob/master/misc/examples/webrtc/websocket.cfg
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> --- Original Message ---
> On Thursday, June 15th, 2023 at 9:36 AM, nutxase 
> wrote:
>
> Hi Karsten and list
>
> I am running
> Centos 7.9
> openssl v1.0.2k
> kamailio 5.7.0
>
>
>
>
> Sent with Proton Mail <https://proton.me/> secure email.
>
> --- Original Message ---
> On Wednesday, June 14th, 2023 at 8:55 PM, nutxase 
> wrote:
>
> Hey All
>
> I have been setting up TLS and websocket for kamailio but i keep getting
> these errors in the log please can someone help me
>
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls
> [tls_util.h:51]: tls_err_ret(): TLS accept:error:1408A0C1:SSL
> routines:ssl3_get_client_hello:no shared cipher (sni: sip.domain.com
> <http://sip.sipalto.com>)
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls
> [tls_server.c:1333]: tls_h_read_f(): src addr: 1.2.3.4.5:57265
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls
> [tls_server.c:1336]: tls_h_read_f(): dst addr: 5.4.3.2.1:8089
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: 
> [core/tcp_read.c:1478]: tcp_read_req(): ERROR: tcp_read_req: error reading
> - c: 0x7f3c8fc16720 r: 0x7f3c8fc16848 (-1)
>
>
> my config is pretty standard so i am not sure what i am missing
> using letsencrypt certs
>
> #!ifdef WITH_TLS
> # - tls params -
> modparam("tls", "tls_method", "TLSv1.2+")
> modparam("tls", "certificate", "/etc/kamailio/tls/cert.pem")
> modparam("tls", "private_key", "/etc/kamailio/tls/ckey.pem")
> modparam("tls", "ca_list", "/etc/kamailio/tls/fullchain.pem")
> #!endif
> Sent with Proton Mail <https://proton.me/> secure email.
>
>
>
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[SR-Users] Re: TLS errors

[Karsten Horsmann]

Hi,

can you please explain your system a bit more?

Kamailio Version, openssl Version and your Linux Distribution would be a
good starting point.

Kind regards
Karsten Horsmann

nutxase  schrieb am Mi., 14. Juni 2023, 22:26:

> Hey All
>
> I have been setting up TLS and websocket for kamailio but i keep getting
> these errors in the log please can someone help me
>
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls
> [tls_util.h:51]: tls_err_ret(): TLS accept:error:1408A0C1:SSL
> routines:ssl3_get_client_hello:no shared cipher (sni: sip.domain.com
> <http://sip.sipalto.com>)
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls
> [tls_server.c:1333]: tls_h_read_f(): src addr: 1.2.3.4.5:57265
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls
> [tls_server.c:1336]: tls_h_read_f(): dst addr: 5.4.3.2.1:8089
> Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: 
> [core/tcp_read.c:1478]: tcp_read_req(): ERROR: tcp_read_req: error reading
> - c: 0x7f3c8fc16720 r: 0x7f3c8fc16848 (-1)
>
>
> my config is pretty standard so i am not sure what i am missing
> using letsencrypt certs
>
> #!ifdef WITH_TLS
> # - tls params -
> modparam("tls", "tls_method", "TLSv1.2+")
> modparam("tls", "certificate", "/etc/kamailio/tls/cert.pem")
> modparam("tls", "private_key", "/etc/kamailio/tls/ckey.pem")
> modparam("tls", "ca_list", "/etc/kamailio/tls/fullchain.pem")
> #!endif
> Sent with Proton Mail <https://proton.me/> secure email.
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[SR-Users] Re: Cannot achieve rtpengine HA

[Karsten Horsmann]

Hi Arsenijs,

the HA-Road to rtpengine is a bit tricky. One Setup could be:

rtpengine-node-1 with redis-server1/redis-db1 (active)
rtpengine-node-2 with redis-server1/redis-db1 (passive, rtpengine disabled).
Some magic (corosync/pacemaker whatever).

On failure of rtpengine-node-1, rtpengine is stopped.
Then *some magic happens* and starts your rtpengine-node-2 rtpengine
process.
On startup rtpengine will use the redis saved calls as its own.
That's the "simplest" way of HA i could found.

There is also a more complex way in rtpengine with keyspace notifications.
https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications.
I didnt finished that one for me.

Am Mo., 12. Juni 2023 um 10:15 Uhr schrieb Arsenijs Kabihno <
a.kabi...@dynatech.lv>:

> Hello,
>
> We are trying to achieve rtpengine HA solely with kamailio and rtpengine
> module.
>
> In the following example call goes to rtpengine node 10.5.0.183:2 and
> then we stop rtpengine service on this node to simulate failure.
> Client is triggering re-INVITE, but, apparently, kamailio sticks to
> initially selected rtpengine, producing the following errors:
>
> Jun 12 06:59:48 ap07stvp.xxx.xxx docker[9373]: 27(33) INFO: {2 2
> INVITE jdl9kq6fv51jcv1agk2e} 

[SR-Users] Re: Dialog DMQ state after restart

[Karsten Horsmann]

Hi Calvin,


Iam not using dmq but there are different aspects.

First what you assume it should do.
Second what the module documentation described and last what in reality
happens.

Did you try it in a test setup with your config and if it is "broken" in
your opinion, you can then fill an issue for the devs with an minimal setup
description and how to recreate the issue.

That would I do.

Kind regards
Karsten Horsmann

Calvin E.  schrieb am Fr., 9. Juni 2023, 21:33:

> The question is whether that DMQ synchronization for dialog data is
> sending all dialogs or only delta/incremental data. Some SBCs I've worked
> with only synchronize new dialog data, leaving standby nodes only aware of
> dialogs established since restarting or joining the cluster, not existing
> dialogs. In that circumstance, a failover to the restarted or new standby
> would break any dialogs created prior to the restart.
>
> If Kamailio dialog DMQ synchronization is "complete" then we just need to
> know how to confirm the new standby is synchronized before performing a
> failover.
>
>
> On Thu, Jun 8, 2023 at 12:53 PM Henning Westerholt  wrote:
>
>> Hello,
>>
>>
>>
>> are you aware of the DMQ synchronisation for dialog data? It’s not
>> replicated everything, but certainly a way to seed the dialog information
>> after a restart.
>>
>> Another way is of course the usual shared database, just keep in mind
>> that you need to configure the dialog module (hash table “step” variables”)
>> to properly work.
>>
>>
>>
>> Cheers,
>>
>>
>>
>> Henning
>>
>>
>>
>> --
>>
>> Henning Westerholt – https://skalatan.de/blog/
>>
>> Kamailio services – https://gilawa.com
>>
>>
>>
>> *From:* Calvin E. 
>> *Sent:* Donnerstag, 8. Juni 2023 19:36
>> *To:* Kamailio (SER) - Users Mailing List 
>> *Subject:* [SR-Users] Dialog DMQ state after restart
>>
>>
>>
>> Does the dialog module do anything similar to the htable dmq_init_sync
>> behavior?
>>
>>
>>
>> The scenario is restarting the offline member of an HA pair after
>> modifying the script. We're unsure of the state of the replicated dialogs
>> post-restart. Will the standby become aware of the dialogs prior to the
>> restart? In other words, do we need to wait until the pre-restart calls end
>> before we can safely failover?
>>
>>
>>
>> -Calvin E.
>>
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[SR-Users] Re: using kamailio to send invite to webrtc clients (with freeswitch)

[Karsten Horsmann]

Hi,

as Dilyan mention start small and test and then grow if needed in your
config.

The loop over is for example not needed if you switch on parallel forking.
Then you have new issues with correct branch settings to rtpengine.

And a good idea for webrtc is havfos example

https://github.com/havfo/WEBRTC-to-SIP

Kind Regards
Karsten Horsmann

Дилян Палаузов  schrieb am Di., 9. Mai 2023, 22:41:

>
> Hello,
>
> I suggest to call from WebRTC a non-WebRTC client, talk and then the
> non-WebRTC client hangs up.  Check if the WebRTC client gets the BYE.
>
> Once this simple scenario works, you can try the more complicated scenario
> explained by you.
>
> In my experiments the BYE was not received with topoh.  Without topoh it
> works.
>
>
> Greetings
>   Dilyan
> -Original Message-
> From: shwetank singh 
> Reply-To: Kamailio (SER) - Users Mailing List  >
> To: sr-users@lists.kamailio.org
> Subject: [SR-Users] using kamailio to send invite to webrtc clients (with
> freeswitch)
> Date: 05/09/2023 07:22:40 PM
>
> Hello experts
>
> I have a setup where webrtc client registrations (from behind a NAT / VPN)
> are on kamailio and the INVITE to begin a call is sent by freeswitch to
> kamailio which is then expected to relay it to the subscribed webrtc
> client.
>
> i understand, from documentation and general thinking, that kamailio
> cannot open a new connection to such a client and should reuse an open
> connection
> (a websocket in my case).
>
> i have setup kamailio as generic as it may sound and expect t_relay() to
> do the magic of forwarding the invite, however, from logs i don't see
> anything indicating kamailio is using any existing connection and hence my
> outgoing invite requests are stuck showing pointed towards a vpn gateway
> which was the registered "location" of client (which is the public ip of
> VPN gateway)
>
> please advise what relevant information can i produce here to make the
> query more readable and clear.
>
> headed here after trying just about all the resources available on the
> topic.
>
> e.g. if my location record looks like
>
>   AoR: 1...@mydomain.com
>   Contacts:
>   - Contact:
>   Address: sip:1004@11.22.33.44:49179
>   CFlags: 64
>   CSeq: 3087
>   Call-ID: 189jss21e24ntve1ggsuli
>   Expires: 593
>   Flags: 0
>   Instance: 
>   KA-Roundtrip: 0
>   Keepalive: 1
>   Last-Keepalive: 1683652755
>   Last-Modified: 1683652755
>   Methods: 7071
>   Path:  ;lr;nat=yes>
>   Q: -1
>   Received: sip:11.22.33.44:49179;transport=ws
>   Reg-Id: 1
>   Ruid: uloc-645601c4-131220-3764
>   Server-Id: 0
>   Socket: tls:1.2.3.4:4443
>   State: CS_SYNC
>   Tcpconn-Id: 3916
>   User-Agent: JsSIP 3.9.0
>
> i try to lookup the location and find the address, and then route to that
> address by setting $ru
> in sngrep it does show that kamailio is trying to send INVITE to
> 11.22.33.44:port (which is the VPN gateway) but it isn't exactly clear
> from logs if
> this is being done over an existing connection.
>
> how does kamailio identify if a connection exists for a user (for each
> contact in AOR, if multiple contacts exist for a user -- i.e. different
> browser
> sessions?)
>
>
>  if(reg_fetch_contacts("location", "$ru", "caller")) {
> $var(i) = 0;
> while($var(i) < $(ulc(caller=>count)))
> {
> $ru = $(ulc(caller=>addr)[$var(i)]);
> $var(i) = $var(i) + 1;
> setflag(FLT_DLGINFO);
> dlg_manage();
> t_relay();
> }
>   }
>
> --
> Shwetank
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[SR-Users] Re: Assigning a new RTPEngine for a call for the sake of failover

[Karsten Horsmann]

Hi,

use rtpengine with redis and let it do the work for you with pub/sub the
calls in redis.

https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications


 schrieb am Sa., 15. Apr. 2023, 09:52:

> Hello,
>
> I am trying to implement RTPEngine failover, where if a selected RTPEngine
> node fails in the middle of the call (and not responding NG requests), I
> want to select an alternative node. However, when I call
> rtpengine_manage(), it does not seem to select an alternative node in case
> of a failure. To address this, I have attempted to check if
> $avp(RTP_INSTANCE) is null after calling rtpengine_manage(), and then use
> rtpengine_delete() to forget the selected RTPEngine node for the specific
> call, with the expectation that Kamailio would assign a new RTPEngine node
> for subsequent SIP message retries. However, it appears that
> rtpengine_delete() does not have the desired effect in this regard.
>
> Here is an excerpt of my code for reference:
> ```
> route[RTP] {
> #!ifdef WITH_RTPENGINE
> if (has_body("application/sdp")) {
> if ($avp(x_source) == "INTERNAL") {
> rtpengine_manage("replace-session-connection
> replace-origin direction=internal direction=external");
> }
> else if ($avp(x_source) == "EXTERNAL") {
> rtpengine_manage("replace-session-connection
> replace-origin direction=external direction=internal");
> }
>
> if ($avp(RTP_INSTANCE) == $null) {
> rtpengine_delete();
> drop();
> }
> }
> #!endif
> }
> ```
>
> Can you please advise on how I can achieve the desired failover behavior?
> Is there another function similar to `rtpengine_reset_call` that can make
> Kamailio forget the RTPEngine node for a specific call?
>
> Thank you!
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[SR-Users] Re: TOPOS module: Topology hiding only for a specific host

[Karsten Horsmann]

Hi,

I would suggest just putting another PSTN topos Kamailio in between. This
way the problem is solved and you don't make an all in one complex
configuration.

Kind regards
Karsten Horsmann

Stefan  schrieb am Do., 6. Apr. 2023, 15:14:

> Hi together
>
> Tested now with 5.3.9 and 5.6.4, same result.
> Does anyone use topos only with specific destination hosts?
>
> Cheers
> Stefan
>
> ...
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[SR-Users] Re: Dispatcher behavior

[Karsten Horsmann]

Hi,

You can configure Kamailio with dispatcher to an behavior like you want.

I personal would not set an destination to IP on 500 but that depends on
your needs.

Kind regards

Jawaid Bazyar  schrieb am Di., 14. März 2023, 00:41:

> Hi,
>
> Maybe it would be easier to explain my expectations on how it's supposed
> to work, and then you can tell me where I'm making a wrong assumption __
>
> So, load balancers like haproxy etc perform "health checks" on the service
> being balanced. In this case, I have it setup for Kamailio to send OPTIONS
> and I do in fact see those packets and 200 OK responses. And if haproxy
> sees a host is down, it immediately stops trying to send client requests to
> that host.
>
> In this case, after 10 successful OPTIONS pings, Kamailio marks the
> destination as AP and everything is fine, Kamailio correctly returns the
> destination in ds_select_dst.
>
> I have the settings below, so that if the service misses 5 OPTIONS checks,
> Kamailio should mark that destination as "inactive".
> But, it should keep probing it, so Kamailio can detect when the
> destination comes back up. So as I understand it, that's the status "IP".
>
> And, Kamailio should stop including that destination in the responses from
> ds_select_dst().
>
> And all of that is behaving as expected, EXCEPT that last bit -
> ds_select_dst continues to include the IP (inactive) destination in
> responses.
>
> I added
>
> ds_mark_dst("ip");
>
> in the event of a 500 or branch failure response, but I don't understand
> why this is needed, because the destination is already in "IP" mode.
>
> So seemingly Kamailio knows the destination is inactive, but ds_select_dst
> isn't acting on it unless I manually mark the target also.
>
> Perhaps I am overlooking an option?
>
> Regards,
>
> Jawaid
>
>
> On 3/13/23, 9:51 AM, "Henning Westerholt"  h...@gilawa.com>> wrote:
>
>
> Hello,
>
>
> probing means in this context sending OPTIONS pings, yes.
>
>
> In mode 1 you can disable the probing by setting flag 4 to the destination
> in the dispatcher file, according to the docs.
>
>
> Cheers,
>
>
> Henning
>
>
> -Original Message-
> From: Henning Westerholt mailto:h...@gilawa.com>>
> Sent: Montag, 13. März 2023 15:55
> To: Kamailio (SER) - Users Mailing List  >
> Subject: [SR-Users] Re: Dispatcher behavior
>
>
> Hello,
>
>
> not sure what the "auto-probing" mode is, but you can configure the
> dispatcher OPTIONS ping behaviour with that parameter:
>
>
>
> https://kamailio.org/docs/modules/5.5.x/modules/dispatcher.html#dispatcher.p.ds_probing_mode
> <
> https://kamailio.org/docs/modules/5.5.x/modules/dispatcher.html#dispatcher.p.ds_probing_mode
> >
>
>
> Cheers,
>
>
> Henning
>
>
> -Original Message-
> From: Jawaid Bazyar mailto:baz...@gmail.com>>
> Sent: Montag, 13. März 2023 14:33
> To: Kamailio (SER) - Users Mailing List  >
> Subject: [SR-Users] Re: Dispatcher behavior
>
>
> Hi Alex,
>
>
> I don't see "auto-probing" in the documentation.. here is my dispatcher
> config that I snatched from somewhere on the Internet.
>
>
> modparam("dispatcher", "db_url", "mysql://xxx/kamailio") #Use DBURL
> variable for database parameters
> modparam("dispatcher", "flags", 2) # If flag 2 is set, then the failover
> support is enabled.
> modparam("dispatcher", "table_name", "dispatcher")
> modparam("dispatcher", "ds_ping_interval", 5) #How often to ping
> destinations to check status
> modparam("dispatcher", "ds_ping_method", "OPTIONS") #Send SIP Options ping
> modparam("dispatcher", "ds_probing_threshold", 5) #How many failed pings
> in a row do we need before we consider it down
> modparam("dispatcher", "ds_inactive_threshold", 10) #How many sucessful
> pings in a row do we need before considering it up
> modparam("dispatcher", "ds_ping_latency_stats", 1) #Enables stats on
> latency
> modparam("dispatcher", "ds_probing_mode", 1) #Keeps pinging gateways when
> state is known (to detect change in state)
>
>
>
>
>
>
> On 3/12/23, 8:28 PM, "Alex Balashov"  abalas...@evaristesys.com> >> wrote:
>
>
>
>
> Unless you have auto-probing enabled, yes.
>
>
>
>
> > On Mar 12, 2023, at 9:14 PM, Jawaid Bazyar  baz...@gmail.com> >>
> wrote:
> >
> > Hi, is this expected behavior?
> > I have two target hosts in a dispatcher set, 10.20.20.118 (prio 5) and
> 10.20.20.252 (prio 10).
> > When I have 10.20.20.252 turned off, dispatcher still tries to send
> packets to that host, even when the status flags are IP.
> > root@vs-rs01:/etc/kamailio# kamcmd dispatcher.list {
> > NRSETS: 1
> > RECORDS: {
> > SET: {
> > ID: 1
> > TARGETS: {
> > DEST: {
> > URI: sip:10.20.20.252:5060
> > FLAGS: IP
> > PRIORITY: 10
> > LATENCY: {
> > AVG: 0.00
> > STD: 0.00
> > EST: 0.00
> > MAX: 0
> > TIMEOUT: 62
> > }
> > }
> > DEST: {
> > URI: sip:10.20.20.118:5060
> 

[SR-Users] Re: no autocomplete for kamcmd for latest debian packages

[Karsten Horsmann]

Hi,

as work around you can kamcmd help ¦ grep dispatcher or whatever to get an
hint.
Or if you on Centos ;)

Ovidiu Sas  schrieb am Sa., 4. Feb. 2023, 05:53:

> Hello,
>
> The latest debian packages are built without autocomplete for kamcmd.
> Do we have the libreadline-dev package installed on the build machine?
> I know that we have kamctl rpc  and kamcli, but if we are still
> shipping kamcmd, it should have the autocomplete feature enabled.
>
> Regards,
> Ovidiu Sas
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
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[SR-Users] Re: transport=ws causing kamailio to use wrong listen socket

[Karsten Horsmann]

Hi Nathan,


I use secure websocket and it works with out an issue. Can you provide a
bit more Information? Kamailio version an a bit of your config would help
the list to figure out why it's not working for you.

 schrieb am Fr., 27. Jan. 2023, 16:08:

> Hi all,
>
> We're hitting an issue while integrating secure websockets in our existing
> SIP infrastructure using Kamailio.
>
> When the registration comes in, it causes an entry in our AOR table, with
> ";transport=ws" appended.
> When we want to send a message to this client (using t_relay), kamailio
> seems to take 'ws' as being *unsecure* websockets. In turn, this makes
> Kamailio try to send out the message using a TCP listener - while it should
> have picked the TLS listener.
>
> There are some remarks in the sources about ws vs. wss, so i'm struggling
> to figure out where things go wrong. I've also created github issue #3340
> with more details.
>
> Any help would be appreciated. If this turns out to be a Kamailio bug, i'm
> happy to provide a patch.
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[SR-Users] Re: topos vs topoh, any reason for putting a private IP in the Contact header?

[Karsten Horsmann]

Hi,

I am very happy with Topos and redis for such things. You should give it a
try. And topos had some nice contact header features also.

Benoit Panizzon  schrieb am Do., 26. Jan. 2023,
16:01:

> Hi
>
> I'm trying to work around Route-Header and Via Issues with the two
> topology hiding modules topoh and topos and trying to figure out, which
> one works better for our environment.
>
> My conclusion so far:
>
> topos creates very clean header, but needs a database or redis. I'm
> always reluctant in adding more components which could fail or cause
> load.
>
> topoh also works, but it worries me a bit that according to the manual:
>
> https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip
> a private IP is used to mask the contact header.
>
> I have come across a CPE which, as far as I see, is misbehaving by
> ALWAYS sending a PRACK to the Hostname or IP found in the Contact
> header, and ignoring Route and Via Header.
>
> So why using a private IP as 'mask' and not the IP or Hostname of the
> actual kamailio instance?
>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
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[SR-Users] Re: rtpengine - SRTP <> RTP

[Karsten Horsmann]

Hi,

If you want the signaling that is encrypted see in plaintext you can use
Kamailio module siptrace.
sipdump and sipcapture maybe also good for that, I didn't use these.


Krzysztof Drewicz  schrieb am Mi., 25. Jan.
2023, 16:33:

> pt., 20 sty 2023 o 18:13 Richard Fuchs  napisał(a):
> >
> >
> > You just need one invocation each: one for the offer, one for the
> > answer. (Assuming single offer/answer exchange and no branches.)
> >
> > You can use rtpengine_manage() for either of them if you want to let the
> > module figure out whether the message is an offer or an answer. But you
> > will want to use different flags for the two cases, mostly depending on
> > where the message is going to.
> >
>
> Thanks!
>
> you were right (obviously)
>
> >
> > If the message is going to a plain RTP (non-SRTP, non-ICE, etc) client,
> > use "RTP/AVP ICE=remove" etc. If the message is going to an SRTP,
> > ICE-enabled client, use "RTP/SAVPF ICE=force" etc.
>
> That is 100% true, also pad-crypto.
>
>
> > If you want to manually distinguish between offers and answers instead
> > of using rtpengine_manage(),
>
> Nope, no need for that, just doing simple POC that - it could work.
>
> Still - my encrypted party plays tricks with, me, is there a way - to
> see unencrypted traffic towards TLS endpoint (maybe a simple pcap
> before encryption or dumping in plain text somewhere),
>
> my party needs a R-URI - blahblahblah.byoc.mypurecloud.de, so i do this:
>
>   if(proto != TLS) {
> xlog("L_INFO", "[R-MAIN] from my ciamajda to
> my gienio non tls $ru from $si\n");
>   if (!ds_select_dst("19", "0")) {
> xwarn("I:$var(i) DROP(DOWN!) FWD:$rm
> [$fU->$tU] [SBCVIP] to $du\n");
> sl_send_reply("503", "Destination down");
> exit;
> }
> xlog("L_INFO","Request URI was -> [$ou] [$ru] [$rU]\n");
>
> rewritehostporttrans("blahblahblah.byoc.mypurecloud.de:5061
> ;transport=tls");
> xlog("L_INFO","Request URI changed MAM [$ou] [$ru]
> [$rU]\n");
>
>
>
> in dispach.list:
>
>
> 17  SIP:blahblahblah.byoc.mypurecloud.de:5061;transport=tls 0
> 1 socket=tls:1.2.3.24:5061
> 18  SIP:blahblahblah.byoc.mypurecloud.de:5061;transport=tls
> 19  SIP:blahblahblah.byoc.mypurecloud.de:5061;transport=tls  0 1
> socket=tls:1.2.3.24:5061;ping_from=sip:myfqdnsbc.realhostname.gq
>
> (mod dispatcher - as I believe can and whould give options via TLS -
> right ?)
>
> Any hints? Maybe someone has already tried and succeeded with
> mypurecloud ? (not so popular like o365/ms-teams trunking but still)
>
> thanks,
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[SR-Users] Re: Setting socket_workers does not work if using interface name

[Karsten Horsmann]

Hi Kaufmann,


did you run ipv6 and the extra listener are ipv6?

Kind regards
Karsten Horsmann

Kaufman  schrieb am Fr., 13. Jan. 2023, 04:09:

> At a quick glance it looks like using socket_workers when the listen
> parameter uses the interface name does not work.  For example:
>
>
>
> children = 4
>
> socket_workers=2
>
> listen=udp:10.10.10.10:5060
>
>
>
> Gives two listeners on udp:10.10.10.10:5060
>
>
>
>
>
> children = 4
>
> socket_workers=2
>
> listen=udp:eth0:5060
>
>
>
> Gives four listeners on the socket.
>
>
>
> Note this is just a quick cursory observation.  Is it by design or a
> design limitation?
>
>
>
> *Kaufman*
>
>
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Re: [SR-Users] ds_is_from_list not matching dispatcher listed with hostname

[Karsten Horsmann]

Hi,

actual versions of Kamailio have an cli option to print out the parsed only
config. That's quiet handy for that kind of debugging process.

Maybe an good point for the future.

Benoît Panizzon  schrieb am Fr., 2. Dez. 2022,
16:10:

> > What else could cause this not to match?
>
> Nevermind... being blind...
>
> There was an inactivating #!ifdef around the part matching the
> dispatcher in question.
>
> --
> Mit freundlichen Grüssen
>
> -Benoît Panizzon- @ HomeOffice und normal erreichbar
> --
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> __
>
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Re: [SR-Users] Pipelimit needs restart to reload values

[Karsten Horsmann]

Hi,


I faced the same behavior with pipelimit and solved it for me with cfg_rpc
variable that I raised or lowered to my needs.

https://www.kamailio.org/docs/modules/devel/modules/cfg_rpc.html

Duarte Rocha  schrieb am Fr., 23. Sept. 2022,
14:32:

> Greetings,
>
> I've been using the pipelimit module and the pl_check function to control
> users rates in my proxy.
>
> if (!pl_check("$var(UserID)", "TAILDROP", "$var(UserRate)")) {
>  pl_drop();
>  exit;
> }
>
> In this case  $var(UserID) is the pipe and  $var(UserRate) is the maximum
> rate.
> The value of the rate is retrieved on every call via query to a DB.
>
> If I change the UserRate value to 0 CPS on the DB calls are not rejected
> unless they go over 1 CPS. However, if I restart Kamailio all calls are
> correctly rejected.
>
> Why does this happen? Can I change this behaviour?
>
> Cheers,
>
> Duarte
>
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Re: [SR-Users] update PAI Header User part

[Karsten Horsmann]

Hi,

remove the header(s) and write the new header with your needs.


Patrick Karton  schrieb am Di., 21. Juni 2022,
10:27:

> hello is there a cleaner/fastest way to update PAI header user part other
> than using subst
>
> i come up with
>
> subst('/^P-Asserted-Identity:(.*)(sip|sips|tel|tels):([^@]*)@([a-zA-Z0-9.]+)(.*)$/P-Asserted-Identity:\1\2:$var(newPAIUser)@\4\5/ig');
>
> Thanks.
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Re: [SR-Users] Recommended openSSL version

[Karsten Horsmann]

Hi Igor,

I jumped from 5.3 to 5.5.x so I read carefull the changelog and migrate
steps.

https://www.kamailio.org/wiki/features/new-in-5.5.x

Show a bit about tls.

Igor Olhovskiy  schrieb am Mi., 22. Juni 2022,
21:08:

> Karsten,
>
> Thanks for your answer!
>
> Out of your head, were there any significant changes in TCP/TLS on 5.4 ->
> 5.5 change?
>
> Regards,
> Igor
>
> Le 22.06.2022 à 18:11, Karsten Horsmann a écrit :
>
> Hi Igor,
>
> I also use CentOS 7 with the same openssl version and between 1000 up to
> 2000 tls/wss connections.
>
> Works for me. Main difference I use Kamailio 5.5.x
>
> Kind regards
> Karsten Horsmann
>
> Igor Olhovskiy  schrieb am Mi., 22. Juni 2022,
> 10:36:
>
>> Hello!
>>
>> Due to I still experience irregular Kamailio 5.4 crashes (like 1/month)
>> related to SSL (using websockets and SIPS) I'm wondering, could openSSL
>> upgrade change the situation?
>> As of now in CentOS 7 I have 1.0.2k version.
>>
>> Does anyone have experience to fix crash-related to TLS problems with
>> openSSL upgrade?
>>
>> Or maye some tuneup of TCP parameters can help here?My current setup is
>> quite simple:
>>
>> children=4
>> enable_tls=yes
>> tcp_accept_no_cl=yes
>> tcp_connection_lifetime=600
>> tcp_max_connections=998976 # 100 - 1024, so we're leaving 1k for
>> system reserve
>> tls_max_connections=998976
>>
>> Number of clients ~ 200 constantly connected to websocket.
>> --
>> Best regards,
>> Igor
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Re: [SR-Users] Recommended openSSL version

[Karsten Horsmann]

Hi Igor,

I also use CentOS 7 with the same openssl version and between 1000 up to
2000 tls/wss connections.

Works for me. Main difference I use Kamailio 5.5.x

Kind regards
Karsten Horsmann

Igor Olhovskiy  schrieb am Mi., 22. Juni 2022,
10:36:

> Hello!
>
> Due to I still experience irregular Kamailio 5.4 crashes (like 1/month)
> related to SSL (using websockets and SIPS) I'm wondering, could openSSL
> upgrade change the situation?
> As of now in CentOS 7 I have 1.0.2k version.
>
> Does anyone have experience to fix crash-related to TLS problems with
> openSSL upgrade?
>
> Or maye some tuneup of TCP parameters can help here?My current setup is
> quite simple:
>
> children=4
> enable_tls=yes
> tcp_accept_no_cl=yes
> tcp_connection_lifetime=600
> tcp_max_connections=998976 # 100 - 1024, so we're leaving 1k for
> system reserve
> tls_max_connections=998976
>
> Number of clients ~ 200 constantly connected to websocket.
> --
> Best regards,
> Igor
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Re: [SR-Users] Kamailio vs FreeSWITCH as SBC

[Karsten Horsmann]

Hi Unai,

you can take a look at sbcOS https://github.com/voiceboys/sbcOS
and use only the kamailio/rtpengine config parts as inspiration or use the
(a littlebit outdated) image.

Also you can combine kamailio + freeswitch (kamailio in front of a group of
freeswitches) with dispatcher.

Kind regards
Karsten Horsmann

Am Di., 31. Mai 2022 um 11:31 Uhr schrieb Henning Westerholt :

> Hello,
>
>
>
> this depends on what you are expecting from the term SBC. Have a look e.g.
> on this page for an overview:
>
> http://www.evaristesys.com/blog/kamailio-as-an-sbc-five-years-on/
>
>
>
> Kamailio can fulfil many SBC duties, but its not a 100% replacement for a
> FreeSwitch. Usually Kamailio is used together with a PBX like FreeSwitch or
> Asterisk in infrastructure.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> *From:* sr-users  *On Behalf Of *Unai
> Rodriguez
> *Sent:* Tuesday, May 31, 2022 11:24 AM
> *To:* sr-users@lists.kamailio.org
> *Subject:* [SR-Users] Kamailio vs FreeSWITCH as SBC
>
>
>
> Dear List,
>
> I’m looking at using Kamailio as SBC.
>
> This would replace our current FreeSWITCH-based SBCs.
>
> What are your thoughts on this move? Would there be any known reasons why
> Kamailio would be superior to FreeSWITCH as an SBC?
>
> Thank you so much
> Unai Rodriguez
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-- 
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*Karsten Horsmann*
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Re: [SR-Users] Relay wss to Asterisk

[Karsten Horsmann]

Hi Michal,

then i would configure (WSS) kamailio/rtpengine SIP ---> asterisk (and talk
only plain SIP "internal").
Works like a charme with kamailio/rtpengine in WSS audio only.

Am Mi., 18. Mai 2022 um 13:14 Uhr schrieb Michal Hajek <
michal.ha...@daktela.com>:

> Hi, thanks for your reply.
> I would like modify SDP to use my RTPEngine.
> I have SBC Kamailio on public IP to routing on asterisk with internal LAN.
>
> Thanks
>
> st 18. 5. 2022 v 12:44 odesílatel Karsten Horsmann 
> napsal:
>
>> Hi,
>>
>>
>> if its only forwarding/upgrading http to wss - for that i would use
>> haproxy or nginx.
>>
>> Am Di., 17. Mai 2022 um 22:32 Uhr schrieb Michal Hajek <
>> michal.ha...@daktela.com>:
>>
>>> Hello,
>>> can i configure kamailio for relay wss to asterisk/pjsip?
>>> WebRTC client ( chrome )  --> WSS -> Kamailio --> WSS -> Asterisk with
>>> pjsip
>>> Asterisk accept wss://ip_address:wss_port/ws
>>>
>>> Thanks
>>> Michal Hajek
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>>
>>
>> --
>> Mit freundlichen Grüßen
>> *Karsten Horsmann*
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Re: [SR-Users] Relay wss to Asterisk

[Karsten Horsmann]

Hi,


if its only forwarding/upgrading http to wss - for that i would use haproxy
or nginx.

Am Di., 17. Mai 2022 um 22:32 Uhr schrieb Michal Hajek <
michal.ha...@daktela.com>:

> Hello,
> can i configure kamailio for relay wss to asterisk/pjsip?
> WebRTC client ( chrome )  --> WSS -> Kamailio --> WSS -> Asterisk with
> pjsip
> Asterisk accept wss://ip_address:wss_port/ws
>
> Thanks
> Michal Hajek
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-- 
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Re: [SR-Users] Manual usage and example kamailio kamctl

[Karsten Horsmann]

Hi,

as Fred said. kamcmd help show you command that are currently possible
based on your kamailio.cfg.

Fred Posner  schrieb am Fr., 1. Apr. 2022, 14:50:

> On 4/1/22 06:48, bayu p wrote:
> > [snip]
> >
> > but, i cannot found further tutorial how use kamctl.
> > Could you please share, the detail/further  tutorial and example of
> > kamctl usage..
> > [...]
> It looks like there are some aspects of the commands you're hitting
> incorrectly as well as perhaps not having the modules loaded or configured.
>
> You can also try with kamcmd... such as kamcmd core.uptime or kamcmd
> core.info
>
> Fred Posner | palner.com
> Matrix: https://matrix.to/#/@fred:matrix.lod.com
> o: +1 (212) 937-7844
>
>
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Re: [SR-Users] RTPENGINE : Kamailio - Get status if max sessions reached

[Karsten Horsmann]

Hi Youssef,

a good starting point is to write your Kamailio version and the parts of
the kamailio.cfg that is interesting.

Kind regards
Karsten Horsmann
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Re: [SR-Users] (no subject)

[Karsten Horsmann]

Hi,

Fred's mid register example could be a starting point
https://github.com/fredposner/scripts/blob/master/kamailio/mid-registrar.cfg

So you forward your registrations to your pbx. If I understand you right.

Needs tls but for starting and understanding plain communication helps you
with tools like sngrep / tcpdump etc.

Then you can add tls

Hope that helps

jaflong jaflong  schrieb am Fr., 25. März 2022, 18:06:

> Hi Kamailio community,
>
> Any advise on how to archive this
>
>   sip.mydomain.comsip.mypbx.com
> sip_phone sip/tls> KAMAILIO ===sip/udp> PBX
>
>
> sip_phone should register to Kamilio with tls
> Kamailio should proxy register message to PBX with udp
> sip_phone invites should be proxied to PBX
>
> in sip_client setting he will point to sip.mydomain.com as his server
>
> Regards
> __
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Re: [SR-Users] SIPS Errors on Kamailio

[Karsten Horsmann]

Hi,


are you sure the Kamailio tls module is on your system? Check the module
path for tls.so like this or if you have build it from source?

rpm -ql kamailio-tls

/usr/lib64/kamailio/modules/auth_identity.so
/usr/lib64/kamailio/modules/tls.so
/usr/lib64/kamailio/openssl_mutex_shared
/usr/lib64/kamailio/openssl_mutex_shared/openssl_mutex_shared.so
/usr/share/doc/kamailio/modules/README.auth_identity
/usr/share/doc/kamailio/modules/README.tls

Christopher Vincent  schrieb am Fr., 18. März 2022,
12:37:

> Hi,
>
>
>
> Kamailio / RTPEngine was set up on CentOS 8 running SIP to SIPS and RTP to
> SDES SRTP conversion. This worked as expected
>
>
>
> Attempted to duplicate the setup on RHEL but errors were seen. These
> errors were present on both RHEL 7 / RHEL 8.
>
>
>
> The errors seen were as below
>
>
>
> kamailio -c
>
> loading modules under config path: /usr/lib64/kamailio/modules/
>
> 0(9165) ERROR: tls [tls_init.c:611]: tls_pre_init(): Unable to set the
> memory allocation functions
>
> 0(9165) ERROR: tls [tls_init.c:613]: tls_pre_init(): libssl current mem
> functions - m: 0x7f7a77c367a0 r: 0x7f7a77c367f0 f: 0x7f7a77c36770
>
> 0(9165) ERROR: tls [tls_init.c:615]: tls_pre_init(): module mem functions
> - m: 0x7f7a72db7653 r: 0x7f7a72db769f f: 0x7f7a72db76fc
>
> 0(9165) ERROR: tls [tls_init.c:617]: tls_pre_init(): Be sure tls module is
> loaded before any other module using libssl (can be loaded first to be safe)
>
> 0(9165) ERROR:  [core/sr_module.c:590]: load_module():
> /usr/lib64/kamailio/modules/tls.so: mod_register failed
>
> 0(9165) CRITICAL:  [core/cfg.y:3683]: yyerror_at(): parse error in
> config file /etc/kamailio/kamailio.cfg, line 137, column 12-19: failed to
> load module
>
> 0(9165) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16
>
> 0(9165) ERROR:  [core/modparam.c:181]: set_mod_param_regex(): No
> module matching  found
>
> 0(9165) CRITICAL:  [core/cfg.y:3686]: yyerror_at(): parse error in
> config file /etc/kamailio/kamailio.cfg, line 249, column 72: Can't set
> module parameter
>
> 0(9165) ERROR:  [core/modparam.c:181]: set_mod_param_regex(): No
> module matching  found
>
> 0(9165) CRITICAL:  [core/cfg.y:3686]: yyerror_at(): parse error in
> config file /etc/kamailio/kamailio.cfg, line 250, column 72: Can't set
> module parameter
>
> 0(9165) ERROR:  [core/modparam.c:181]: set_mod_param_regex(): No
> module matching  found
>
> 0(9165) CRITICAL:  [core/cfg.y:3686]: yyerror_at(): parse error in
> config file /etc/kamailio/kamailio.cfg, line 251, column 68: Can't set
> module parameter
>
> 0(9165) ERROR:  [core/modparam.c:181]: set_mod_param_regex(): No
> module matching  found
>
> 0(9165) CRITICAL:  [core/cfg.y:3686]: yyerror_at(): parse error in
> config file /etc/kamailio/kamailio.cfg, line 256, column 39: Can't set
> module parameter
>
> ERROR: bad config file (5 errors) (parsing code: 0)
>
> 0(9165) INFO:  [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API
> not initialized
>
>
>
>
>
>
>
>
>
>
>
> The kamailio config was exactly the same as on the CentOS systems and
> started as below
>
>
>
> /* Server ports: */
>
> #!substdef "!SIP_PORT!5060!g"
>
> #!substdef "!SIPS_PORT!5061!g"
>
>
>
> /* Listen addresses */
>
> #!substdef "!UDP_LOCAL_ADDR!udp:SERVER_IP_ADDR:SIP_PORT!g"
>
> #!substdef "!TCP_LOCAL_ADDR!tcp:SERVER_IP_ADDR:SIPS_PORT!g"
>
>
>
> /* Server connections: */
>
> #!ifndef MAX_CONNECTIONS
>
> #!define MAX_CONNECTIONS 8192
>
> #!endif
>
>
>
>
>
> ##!define WITH_DEBUG
>
>
>
> /* Transaction and branch flags:
>
>   FLT_ - per transaction (message) flags
>
>   FLB_ - per branch flags
>
> */
>
> #!define FLT_ACC 1
>
> #!define FLT_ACCMISSED 2
>
> #!define FLT_ACCFAILED 3
>
> #!define FLT_NATS 5
>
> #!define FLT_OUT 8
>
> #!define FLB_NATB 6
>
> #!define FLB_NATSIPPING 7
>
>
>
> #!define KAMAILIODBURL1 "mysql://kamailio:kamailiorw@localhost/kamailio"
>
>
>
> #!define WITH_TLS
>
> enable_tls=1
>
> listen=tls::5062
>
>
>
> ### Global Parameters #
>
> ### LOG Levels: ALERT=-5, BUG=-4, CRIT=-3, ERR=-1, WARN=0, NOTICE=1,
> INFO=2, DBG=3
>
> #!ifdef WITH_DEBUG
>
> debug=4
>
> log_stderror=no
>
> #!else
>
> debug=2
>
> log_stderror=no
>
> #!endif
>
>
>
> memdbg=5
>
> memlog=5
>
>
>
> log_facility=LOG_LOCAL0
>
>
>
> /* display memory usage on exit */
>
> mem_summary=15
>
>
>
> /* join free memory fragments */
>
> mem_join=1
>
>
>
> /* proxy will fork and run in daemon mode */
>
> /* one process will be created for each network interface the proxy
> listens to and for each protocol (TCP/UDP), multiplied with the value of
> 'children' parameter */
>
> fork=yes
>
> children=8
>
>
>
> listen=TCP_LOCAL_ADDR
>
> listen=UDP_LOCAL_ADDR
>
>
>
> /* life time of TCP connection when there is no traffic
>
>- a bit higher than registration expires to cope with UA behind NAT */
>
> tcp_connection_lifetime=3605
>
>
>
> /* sip over websockets may not specify a content length header */
>
> tcp_accept_no_cl=yes
>
>
>
> /* buffer size used for tcp reads, limits the 

Re: [SR-Users] Kamailio + KEMI + topos + change_reply_status

[Karsten Horsmann]

Hi Marrold,

not tested but could it be that you check an string with an int?

Kind regards
Karsten Horsmann

Marrold  schrieb am Di., 8. März 2022, 23:46:

> Hi,
>
> I'm using Kamailio 5.5.3 with a KEMI / Python based configuration on
> Debian 11.
>
> In my reply route I am trying to convert a 183 to a 180 with: 
> KSR.textopsx.change_reply_status(180,
> "Ringing")
>
> I've noticed with TOPOS enabled, it doesn't work and I still see a 183 go
> out. But with it disabled, it works as expected and it's converted to a
> 180. I get the same behaviour with / without msg_apply_changes()
>
> Config looks like this:
>
> def ksr_onreply_manage(self, msg):
>
> KSR.info(f"ksr_onreply_manage\r\n")
>
> scode = KSR.pv.get("$rs")
>
> if KSR.permissions.allow_source_address_group()<0:
> if scode == 183:
>KSR.info(f"ksr_onreply_manage: Converting 183 to 180\r\n")
>KSR.textopsx.change_reply_status(180, "Ringing")
>KSR.textopsx.msg_apply_changes()
>
> if scode>100 and scode<299 :
> self.ksr_route_natmanage(msg)
>
> return 1
>
> The debug log doesn't reveal anything too obvious, I can share if its of
> any use.
>
> Does anyone have any suggestions?
>
> Thanks
> Matthew
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Re: [SR-Users] Homer7 and Kamailio 5.3

[Karsten Horsmann]

Hi James an Angelo,

most of the time my homer or hepic servers are in different subnets.
So that arp think sounds good to me.

@Angelo Take a look at the HEP Fidelity Proxy, it's capable to be in the
middle between your kamailio and the hep-server and can store the HEP
packets if your hep-server is unavailable.
After the hep-server (homer 7, hepic, whatever) is available again it sends
the stored data automatically to the hep-server.

https://github.com/ivlovric/HFP/



James Browne  schrieb am Do., 3. Feb. 2022, 12:26:

> I've encountered a very similar problem in the past.The siptrace module
> uses UDP and does not expect a response, so it can be confusing that
> kamailio's behavior changes depending on whether the homer server is online.
>
> The problem I've seen in the past is that when kamailio and a statsd
> server are in the same IP subnet, and the statds server goes offline, an
> ARP request is made whenever traffic is to be sent, and this ARP request is
> blocking until it times out (a few seconds). It blocks all workers because
> they all want to send a UDP packet and all have to wait.
>
> If this is what's going on, then moving the siptrace server to a different
> subnet should resolve the problem.
>
> James
>
>
> On Wed, 2 Feb 2022 at 21:43, Angelo Sipper  wrote:
>
>> Hi,
>>
>> I have the exact same configuration with two workers and same syntax.
>> It works fine until I shut down homer. After that Kamailio stop
>> precessing any traffic and all clients try to register as they are
>> disconnected.
>>
>> Kind Regards,
>> Angelo
>>
>> On Wed, Feb 2, 2022, 8:45 PM Karsten Horsmann 
>> wrote:
>>
>>> Hi,
>>>
>>> try to define an extra sendsocket.
>>>
>>> And add an listen directive for that sendsocket.  Sorry for the
>>> formating wrote this on my smartphone.
>>>
>>> Works for me with an similar setup and 5.3
>>>
>>> # capture node or webhomer target
>>> #!ifdef WITH_HOMER
>>> #!substdef "!MY_HOMER_CAPTURE!sip:MY_HOMER_IP_ADDR:MY_HOMER_PORT!g"
>>> #!substdef "!MY_LOCAL_SENDSOCK!sip:MY_PRV_IP:MY_LOCAL_SEND_PORT!g"
>>> #!substdef "!MY_LOCAL_SENDLISTEN!udp:MY_PRV_IP:MY_LOCAL_SEND_PORT!g"
>>> #!endif
>>>
>>>
>>> #!ifdef WITH_HOMER
>>> #Siptrace
>>> modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK")
>>> # local interface from where to send the duplicated traffic
>>> modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
>>> modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer
>>> encapsulate mode
>>> modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture
>>> agent ID will be included in the HEP header
>>> modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture
>>> agent id Limitation: 32-bit for HEPv3.
>>> modparam("siptrace", "trace_to_database", 0) # trace to database
>>> modparam("siptrace", "trace_on", 1) # enable/disable trace
>>> modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont
>>> need call siptrace flag or sip_trace()
>>> #!endif
>>>
>>> # sendsocket for homer
>>> #!ifdef WITH_HOMER
>>> socket_workers=2
>>> listen=MY_LOCAL_SENDLISTEN
>>> #!endif
>>>
>>> Henning Westerholt  schrieb am Mi., 2. Feb. 2022, 18:44:
>>>
>>>> Hello,
>>>>
>>>>
>>>>
>>>> any feedback regarding my previous question about log files, errors?
>>>>
>>>>
>>>>
>>>> Cheers,
>>>>
>>>>
>>>>
>>>> Henning
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Henning Westerholt – https://skalatan.de/blog/
>>>>
>>>> Kamailio services – https://gilawa.com
>>>>
>>>>
>>>>
>>>> *From:* sr-users  *On Behalf Of 
>>>> *Angelo
>>>> Sipper
>>>> *Sent:* Wednesday, February 2, 2022 5:45 PM
>>>> *To:* Kamailio (SER) - Users Mailing List 
>>>> *Subject:* Re: [SR-Users] Homer7 and Kamailio 5.3
>>>>
>>>>
>>>>
>>>> Hi,
>>>>
>>>>
>>>>
>>>> Memory on the system is 8GB and on Kamailio we have SHM 4096 and PKG
>>>> 32. Each o

Re: [SR-Users] REGISTER with http_async_client ?

[Karsten Horsmann]

Hi Cyril,

This Kamailio module could imho do the same

https://www.kamailio.org/docs/modules/devel/modules/auth_ephemeral.html


Cyril Ramière  schrieb am Do., 27. Jan. 2022,
08:04:

> Hi there,
>
> I have a weird issue with kamailio (latest docker
> image kamailio-ci:5.5.2-alpine) and http_async_client.
>
> Before posting a lot of logs, let me describe what I want to achieve.
>
> I have a Kamailio and a SIP Phone.
>
> The SIP phone sends a REGISTER to kamailio, then in my routing block, I
> check if I have an Authorization header.
>
> Since I don't have an Authorization (first message), I
> use "www_challenge()".
> This replies to the SIP phone, and then the SIP phone sends a new REGISTER
> with the correct Authorization header.
>
> So far so good.
>
> Now, when I get the REGISTER with Authorization header, I want to ask an
> HTTP endpoint if this user is allowed to connect and check the password
> using http_async_query().
>
> The problem is that when the transaction resumes, the tmx module is
> unhappy and throws this error :
>
> 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked
> branch (-1) for a final response in MODE_ONFAILURE
>
> And a 500 error is sent back to the sip phone.
> The AUTH_REPLY route is still called and I can use the $http* values.
>
> Do you see something that I am doing wrong or missing in my logic?
> Is pausing/resuming to use the async http client is allowed if I'm
> handling a REGISTER transaction?
>
> Here's a simplified version of my routing block (not far from reality):
>
> # SNIP
>
> request_route{
>
> route(AUTH);
>
>
> route[AUTH]{
>
> if (is_method("REGISTER"){
>
> if(no_auth_header){
>
> www_challenge("$td","1");
>
> exit;
>
> }
>
> else{
>
> t_newtran();
>
> http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar;, 
> "AUTH_REPLY");
>
> }
>
> }
>
> }
>
>
> route[AUTH_REPLY]{
>
> xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n");
>
> }
>
>
> }
>
> # END SNIP
>
>
> Best regards!
> __
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Re: [SR-Users] Homer7 and Kamailio 5.3

[Karsten Horsmann]

Hi,

try to define an extra sendsocket.

And add an listen directive for that sendsocket.  Sorry for the formating
wrote this on my smartphone.

Works for me with an similar setup and 5.3

# capture node or webhomer target
#!ifdef WITH_HOMER
#!substdef "!MY_HOMER_CAPTURE!sip:MY_HOMER_IP_ADDR:MY_HOMER_PORT!g"
#!substdef "!MY_LOCAL_SENDSOCK!sip:MY_PRV_IP:MY_LOCAL_SEND_PORT!g"
#!substdef "!MY_LOCAL_SENDLISTEN!udp:MY_PRV_IP:MY_LOCAL_SEND_PORT!g"
#!endif


#!ifdef WITH_HOMER
#Siptrace
modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK")
# local interface from where to send the duplicated traffic
modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate
mode
modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture
agent ID will be included in the HEP header
modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent
id Limitation: 32-bit for HEPv3.
modparam("siptrace", "trace_to_database", 0) # trace to database
modparam("siptrace", "trace_on", 1) # enable/disable trace
modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need
call siptrace flag or sip_trace()
#!endif

# sendsocket for homer
#!ifdef WITH_HOMER
socket_workers=2
listen=MY_LOCAL_SENDLISTEN
#!endif

Henning Westerholt  schrieb am Mi., 2. Feb. 2022, 18:44:

> Hello,
>
>
>
> any feedback regarding my previous question about log files, errors?
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sr-users  *On Behalf Of *Angelo
> Sipper
> *Sent:* Wednesday, February 2, 2022 5:45 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] Homer7 and Kamailio 5.3
>
>
>
> Hi,
>
>
>
> Memory on the system is 8GB and on Kamailio we have SHM 4096 and PKG 32.
> Each of these VM handles only a few hundred of concurrent calls and each VM
> has a few hundred of TLS user registrations.
>
> Kamailio uses IP to send the traffic to Homer and not domain name.
>
>
>
> Any ideas how to solve this issue?
>
>
>
> Kind Regards,
>
> Angelo
>
>
>
>
>
>
>
> On Tue, Feb 1, 2022, 8:43 PM Nick F  wrote:
>
> Hello,
>
>
>
> Do use domain name as destination for homer?
>
>
>
>
>
> On Mon, Jan 31, 2022, 23:12 Henning Westerholt  wrote:
>
> Hello,
>
>
>
> Interesting observation. Do you see anything related to this error period
> in the log files? Maybe out of memory errors etc..?
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sr-users  *On Behalf Of *Angelo
> Sipper
> *Sent:* Monday, January 31, 2022 12:32 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] Homer7 and Kamailio 5.3
>
>
>
> Hi,
>
>
>
> Any ideas on this topic?
>
> Anyone with same behavior?
>
>
>
> Regards,
>
> Angelo
>
>
>
> On Thu, Jan 27, 2022, 9:18 PM Angelo Sipper  wrote:
>
> Hi to all,
>
>
>
> We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace
> module with parameters force_send_sock and hep_mode_on,1 and hep_version,3
> and trace_mode,1. Everything works fine getting all traffic to Homer7 but,
> in case Homer7 is not reachable, Kamailio then no longer accepts traffic or
> registrations and it is like it crashes. When we revive Homer7 and is it
> again reachable accepting traffic, Kamailio then works again fine without
> any restart or actions.
>
> Any ideas why this may be happening? Is there anything we need to add to
> the config to avoid this behavior as having Homer7 now is a single point of
> failure for Kamailio?
>
>
>
> Thank you!
>
>
>
> Kind Regards,
>
> Angelo
>
> __
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Re: [SR-Users] Kamailio as a SIP registrar

[Karsten Horsmann]

Hi Aditya,

you can try to add Kamailio db Schemata to your database VOIP and then try
to create an mysql view from your VoipSubscribers to subscriber.

Try that in an test database with an test Kamailio to run.

Kind regards
Karsten Horsmann

Aditya Uppuluri  schrieb am Mi., 26. Jan. 2022,
11:53:

> Hello Team,
> Thank you for the amazing product. I am a newbie to kamailio and currently
> trying to use kamailio as a SIP Registrar (for a multidomain). Here is the
> requirement
>
> 1. We already have a mysql database (name: VOIP) with a table named
> VoipSubscribers which contains the userid, password and domain information
> of all the users.
> 2. We want the ability in kamailio to use the same VoipSubscribers table
> for user registrations and also store the location information in the same
> database (VOIP)
>
> May I know how to achieve this without altering the table schema?
> Thank you in advance.
>
> Regards,
> Aditya Uppuluri
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[SR-Users] Question about siptrace destination/source port capture in HEPv3 mode

[Karsten Horsmann]

Hello List,

I observe an interesting behavior. We use siptrace as an active capture
agent for QXIP HOMER/HEPIC installations.
The destination port and the source port in the HEPv3 Header are not
correct if you use transport tcp for example.

My Question: is that intended or a bug and i should fill up an issue?

On udp, it's filled with 5060 or whatever your listening directive says.

I tested that with kamailio 5.5 and kamailio master.

I attached an HEPv3 capture from two kamailios (master) speaking with each
other.
You need the https://github.com/sipcapture/hep-wireshark lua dissector to
read the capture.

10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode)
192.168.50.4 (kamailio as registrar)

Frame 1 shows an tcp transport register Request from 10.0.2.15 to
192.168.50.4 in HEPv3.
The Source Port in the HEP3 Protocol is not correct (its an highport /
45419).

Frame 4 shows the 401 answer and the Destination port is 0.
Frame 7 also had this behavior.


- params (nearly at the end):
modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
modparam("siptrace", "hep_mode_on", 1)
modparam("siptrace", "hep_version", 3)
modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent
id Limitation: 32-bit for HEPv3.
modparam("siptrace", "trace_to_database", 0)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need
call siptrace flag or sip_trace()



-- 
Mit freundlichen Grüßen / Kind Regards
*Karsten Horsmann*


dumphep3.pcap
Description: Binary data
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Re: [SR-Users] uac module - use htable instead of database

[Karsten Horsmann]

Hi Markus,

with an db its easier and re-register is working fine automatically.

But https://kamailio.org/docs/modules/5.5.x/modules/uac.html#uac.f.uac_auth
for authentication INVITES
and uac_req_send (to form REGISTER messages) and a route to challenge
response maybe possible.

Its then "building" the stuff uac normally did automatically manual.
Didnt test that but - its an kamailio script - so the sky is the limit.

Am Fr., 17. Dez. 2021 um 12:37 Uhr schrieb Markus Bönke :

> Hello all,
>
> we want to use the "remote registrations feature" of the uac module to let
> kamailio register with other sip providers. As far as I see the module
> works only with a database backend. As our kamailio script should be
> independent from a database, a solution could be to load the configuration
> on startup (and refresh it periodically) from a database view into a htable
> and then trigger the remote registrations based on the expiration of the
> htable columns.
>
> Could something like this be implemented by using the uac module or by an
> other workaround ? Or does it require some enhancements in the uac module ?
>
> Thanks and regards
>
> Markus
>
>
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*Karsten Horsmann*
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[SR-Users] wireshark decode kamailio / rtpengine ng messages

[Karsten Horsmann]

Hi List,

maybe someone wrote a wireshark helper (didnt know the right term) to view
kamailio to rtpengine communication in human form?
Some hints would be great.

-- 
Mit freundlichen Grüßen
*Karsten Horsmann*
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Re: [SR-Users] Manipulation of request-line

[Karsten Horsmann]

Hi Alex,

my hope was that this would write the whole line then. Didn't test that.

Am Di., 7. Dez. 2021 um 23:19 Uhr schrieb Alex Balashov <
abalas...@evaristesys.com>:

> That will manipulate the request URI, but not the request line.
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
> > On Dec 7, 2021, at 4:16 PM, Karsten Horsmann 
> wrote:
> >
>
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-- 
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Re: [SR-Users] Manipulation of request-line

[Karsten Horsmann]

Hi,


$ru = "sip:" + $rU + "@" + $td;

Or simple $ru = $tu;

could be the trick.

Kind regards
Karsten Horsmann

Mihovil Kolaric  schrieb am Di., 7. Dez. 2021,
20:18:

> Hi,
>
> I have an issue with a SIP-client, that occasionally puts a blank between
> the version SIP/2.0 and the CRLF at the and of the request-line. (i.e.
> sends "ACK  SIP/2.0 \r\n" instead of "ACK  SIP/2.0\r\n").
> Kamailio (as well as most SIP UAs) does not care about that error and
> successfully processes/forwards the request (also when using
> sanity_check()). But I also deal with a client that fails to parse that
> request. (Furthermore, wireshark's dissector fails to recognize such
> malformed messages as SIP, which makes debugging even harder).
>
> I tried to fix this issue using textops, but all functions I tried operate
> only on headers or message body, but do not seem to affect the request
> line.
> Is there any possibility to modify the request-line, or to replace it with
> a new one?
>
> Best Regards,
> Mihovil
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Re: [SR-Users] topos xavu_field_contact_host kamailio 5.5

[Karsten Horsmann]

Hi Jonathan,


I placed a new route and set the user part

$xavu(_tps_=>a_contact) = "sbc_a";$xavu(_tps_=>b_contact) =
"sbc_b";

And for special cases I set the to user part to the To user for some silly
pbx boxes.
Can also be that in this mode without setting a/b_contact the normal
contact user part passes through, didn't test that.


Kind regards
Karsten Horsmann

Jonathan Hunter  schrieb am Fr., 3. Dez. 2021, 20:03:

> Hi Karsten,
>
>
>
> Thank you for your reply.
>
>
>
> Thats interesting I will test it, so I see in the documentation it states;
>
>
>
>
>
> $xavu(_tps_=>a_contact) = "...";
>
>
>
> In your case what did you enter so that it just sets the user part?
>
>
>
> I assume I could just set it to $sel(contact.uri.user) for example and it
> would then add that and the appropriate listen/ socket value ?
>
>
>
> Many thanks!
>
>
>
> Jon
>
> Sent from Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for
> Windows
>
>
>
> *From: *Karsten Horsmann 
> *Sent: *03 December 2021 18:49
> *To: *Kamailio (SER) - Users Mailing List 
> *Subject: *Re: [SR-Users] topos xavu_field_contact_host kamailio 5.5
>
>
>
> Hi Jonathan,
>
>
>
>
>
>
>
> I use topos in mode 2 (rewrite user part)
>
> and then it sets the domain part to the listen / socket value AFAIK if you
> only set user part and did not specify the contact host.
>
>
>
>
>
>
>
> #!ifdef WITH_TOPOS
>
> modparam("topos", "storage", "redis")
>
> modparam("topos_redis", "serverid", "CFG_REDIS_NAME")
>
> modparam("topos", "branch_expire", 14400) modparam("topos",
> "dialog_expire", 14400)
>
> modparam("topos", "clean_interval", 60)
>
> #!ifdef WITH_TOPOS_CONTACTmodparam("topos",
> "contact_mode", 2)
>
> modparam("topos", "xavu_cfg", "_tps_")modparam("topos",
> "xavu_field_contact_host", "contact_host")
> modparam("topos", "xavu_field_a_contact", "a_contact")
>
> modparam("topos", "xavu_field_b_contact", "b_contact")
>
> #!endif
>
> #!endif
>
>
>
>
>
> Jonathan Hunter  schrieb am Fr., 3. Dez. 2021,
> 10:50:
>
> Hello Henning,
>
>
>
> Ok appreciate the confirmation, could this be a possible option moving
> forwards with the module?
>
>
>
> In the meantime I assume other approach is hash tables as I have done
> before?
>
>
>
> Thanks again appreciate your response.
>
>
>
> Jon
>
>
>
> Sent from Mail
> <https://emea01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgo.microsoft.com%2Ffwlink%2F%3FLinkId%3D550986=04%7C01%7C%7C0c191eafe0044b48f33d08d9b68d9ba4%7C84df9e7fe9f640afb435%7C1%7C0%7C637741541592586515%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000=YoP2dawL1mtfkJEF7%2FodNV%2F4ZqR59S4hP8vrxoDLAYo%3D=0>
> for Windows
>
>
>
> *From: *Henning Westerholt 
> *Sent: *03 December 2021 09:36
> *To: *Kamailio (SER) - Users Mailing List 
> *Cc: *Jonathan Hunter 
> *Subject: *RE: topos xavu_field_contact_host kamailio 5.5
>
>
>
> Hello Jonathan,
>
>
>
> I see, so you want basically configure different contact addresses for
> A-side and B-side of the call.
>
> This is probably not supported from topos right now, its only possible for
> the user name.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> *From:* Jonathan Hunter 
> *Sent:* Friday, December 3, 2021 10:07 AM
> *To:* Henning Westerholt ; Kamailio (SER) - Users Mailing
> List 
> *Subject:* RE: topos xavu_field_contact_host kamailio 5.5
>
>
>
> Hello Henning,
>
>
>
> Yes your comments make perfect sense.
>
>
>
> Essentially I want to change the contact in-dialog as essentially the
> initial INVITE comes in on a private interface from the carrier, and then
> goes out on a public interface to the subscriber, so the subscriber only
> see’s public IP address in the SIP contact host and responds to it, and the
> carrier only see’s the private IP address in the contact host so they
> respond to the appropriate IP address, so its topology hiding IP addresses
> across interfaces.
>
>
>
> I have done a similar thing with hash tables before and using
> record_route_preset but was hoping I could do a similar thing with  topos
> and leave it to manage the dialog.
>
>
>
&g

Re: [SR-Users] topos xavu_field_contact_host kamailio 5.5

[Karsten Horsmann]

Hi Jonathan,



I use topos in mode 2 (rewrite user part)
and then it sets the domain part to the listen / socket value AFAIK if you
only set user part and did not specify the contact host.



#!ifdef WITH_TOPOS
modparam("topos", "storage", "redis")
modparam("topos_redis", "serverid", "CFG_REDIS_NAME")
modparam("topos", "branch_expire", 14400) modparam("topos",
"dialog_expire", 14400)
modparam("topos", "clean_interval", 60)
#!ifdef WITH_TOPOS_CONTACTmodparam("topos",
"contact_mode", 2)
modparam("topos", "xavu_cfg", "_tps_")modparam("topos",
"xavu_field_contact_host", "contact_host")
modparam("topos", "xavu_field_a_contact", "a_contact")
modparam("topos", "xavu_field_b_contact", "b_contact")
#!endif
#!endif


Jonathan Hunter  schrieb am Fr., 3. Dez. 2021, 10:50:

> Hello Henning,
>
>
>
> Ok appreciate the confirmation, could this be a possible option moving
> forwards with the module?
>
>
>
> In the meantime I assume other approach is hash tables as I have done
> before?
>
>
>
> Thanks again appreciate your response.
>
>
>
> Jon
>
>
>
> Sent from Mail  for
> Windows
>
>
>
> *From: *Henning Westerholt 
> *Sent: *03 December 2021 09:36
> *To: *Kamailio (SER) - Users Mailing List 
> *Cc: *Jonathan Hunter 
> *Subject: *RE: topos xavu_field_contact_host kamailio 5.5
>
>
>
> Hello Jonathan,
>
>
>
> I see, so you want basically configure different contact addresses for
> A-side and B-side of the call.
>
> This is probably not supported from topos right now, its only possible for
> the user name.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> *From:* Jonathan Hunter 
> *Sent:* Friday, December 3, 2021 10:07 AM
> *To:* Henning Westerholt ; Kamailio (SER) - Users Mailing
> List 
> *Subject:* RE: topos xavu_field_contact_host kamailio 5.5
>
>
>
> Hello Henning,
>
>
>
> Yes your comments make perfect sense.
>
>
>
> Essentially I want to change the contact in-dialog as essentially the
> initial INVITE comes in on a private interface from the carrier, and then
> goes out on a public interface to the subscriber, so the subscriber only
> see’s public IP address in the SIP contact host and responds to it, and the
> carrier only see’s the private IP address in the contact host so they
> respond to the appropriate IP address, so its topology hiding IP addresses
> across interfaces.
>
>
>
> I have done a similar thing with hash tables before and using
> record_route_preset but was hoping I could do a similar thing with  topos
> and leave it to manage the dialog.
>
>
>
> Does that make sense?
>
>
>
> Jon
>
>
>
> Sent from Mail
> 
> for Windows
>
>
>
> *From: *Henning Westerholt 
> *Sent: *03 December 2021 08:55
> *To: *Kamailio (SER) - Users Mailing List 
> *Cc: *Jonathan Hunter 
> *Subject: *RE: topos xavu_field_contact_host kamailio 5.5
>
>
>
> Hello Jonathan,
>
>
>
> Usually, you want to set the appropriate contact host in the initial
> INVITE and then the dialog stays “with” this contact. This can be done by
> just setting the xavp and is the way the topology module usually works.
>
>
>
> Maybe you can elaborate why you want to change the contact in-dialog, or
> even in one transaction (onreply_route example below).
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
> 
>
> Kamailio services – https://gilawa.com
> 
>
>
>
> *From:* Jonathan Hunter 
> *Sent:* Friday, December 3, 2021 9:28 AM
> *To:* Henning Westerholt ; Kamailio (SER) - Users Mailing
> List 
> *Subject:* RE: topos xavu_field_contact_host kamailio 5.5
>
>
>
> HI Henning,
>
>
>
> Yes sorry late night  and thank you for reply.
>
>
>
> Yes I conditionally want to set it, I have tried this but not sure it has
> worked to this point hence why I am wondering if I am missing something.
>
>
>
> Say for example I have request coming in from carrier on 10.10.10.10
> interface and want to route to 

Re: [SR-Users] NIST advisory

[Karsten Horsmann]

Hi David,

and for "unpatched" systems you can use remove_hf_re instead of remove_hf
to avoid that.

Am Do., 2. Dez. 2021 um 10:17 Uhr schrieb Gerry | Rigatta <
gjacob...@rigatta.com>:

> Hi David,
>
> there has been a very interesting discussion here about this
> https://lists.kamailio.org/pipermail/sr-users/2020-September/110325.html
>
> Unfortunately there is no dedicated security page on kamailio.org or the
> wiki therefore Google indexes Kamailio security issues badly.
>
> Best
>
> Gerry
>
>
>
> On 1 Dec 2021, at 23:13, David Villasmil 
> wrote:
>
> Anyone knows about this?
>
> https://nvd.nist.gov/vuln/detail/CVE-2020-28361
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
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-- 
Mit freundlichen Grüßen
*Karsten Horsmann*
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Re: [SR-Users] probles with listen advertise address trying to be resolved

[Karsten Horsmann]

Hi Richard,


i play around with your examples and this combination(s) works for me:
version: kamailio 5.5.3

#!substdef "!LISTEN_EXTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g"
#!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4!g"
#!substdef "!TLS_PORT!5061!g"
#!substdef "!LISTEN_NAME!ext_tls!g"

*listen = LISTEN_EXTERNAL_SOCKET_TLS advertise
ADVERTISE_NAT_ADDRESS_TLS:TLS_PORT name "ext_tls"*
or
*listen = LISTEN_EXTERNAL_SOCKET_TLS advertise
ADVERTISE_NAT_ADDRESS_TLS:TLS_PORT name "LISTEN_NAME"*



"/etc/kamailio/kamailio.cfg" 1626L, 48756C geschrieben
# kamailio -c -f /etc/kamailio/kamailio.cfg
loading modules under config path: /usr/lib64/kamailio/modules/
 0(17553) INFO: tls [tls_init.c:503]: init_tls_compression(): disabling
compression...
Listening on
 t*ls: 10.225.73.16 [10.225.73.16]:5061 name ext_tls advertise
1.2.3.4:5061 *
Aliases:

config file ok, exiting...

Hope this helps

Kind regards
Karsten
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Re: [SR-Users] probles with listen advertise address trying to be resolved

[Karsten Horsmann]

Hi Richard,

seems that you forgot the equal signs in the listen directive. Did kamailio
-c -f /etc/kamailio/kamailio.cfg return something?
http://www.kamailio.org/wiki/cookbooks/devel/core#listen

I did the same with puppet. And I only manipulate my
/etc/kamailio/kamailio-local.cfg with the configuration management tool.
Small example from my templating.

grep MY_PUB_IP /etc/kamailio/kamailio-local.cfg
#!substdef "!MY_PUB_IP!1.2.3.4!g"

egrep 'MY_HTTPS_ADDR|MY_PUB_IP|_PORT' /etc/kamailio/kamailio.cfg
import_file "kamailio-local.cfg"

#!substdef "!MY_HTTPS_ADDR!tls:MY_PUB_IP:MY_WSS_HTTPS!g"

listen=udp:MY_PUB_IP:5060
listen=tcp:MY_PUB_IP:5060
listen=tls:MY_PUB_IP:5061
listen=MY_HTTPS_ADDR

Kind regards
Karsten

Am Di., 30. Nov. 2021 um 16:57 Uhr schrieb Richard Robson <
rich...@rikrobson.co.uk>:

> I have a problem when trying to add an advertised address in Kamailio 5.4.2
>
>
> we have a checf config that is populating various variable. some of these
> are for the Listeners
>
>
>
> we have this working and Im getting all protocols for the listeners based
> on the non _TLS Vars but the TLS config is trying to resolve
> ADVERTISE_NAT_ADDRESS_TLS. If I try and quote the variable
> "ADVERTISE_NAT_ADDRESS_TLS" id get an error saying 1.2.3.4:5061 is not
> resolvable
>
>
> therefroe I have udp,tcp and tls listeners all on the default ports where
> i need the tls on port 5061 ans not 5060
>
>
> listen LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS name = "ext"
>
> listen LISTEN_INTERNAL_SOCKET  name = "int"
>
> listen LISTEN_EXTERNAL_SOCKET_TLS advertise  name = "ext-tls"
>
> listen LISTEN_INTERNAL_SOCKET_TLS  name = "int-tls"
>
> the Vars are defined as
>
> #!define TOPO_NAT
>
> #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g"
>
> #!substdef "!LISTEN_INTERNAL_SOCKET!10.10.10.10:5080!g"
>
> #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g"
>
>
>
> #!define ENABLE_TLS
>
> #!define TLS_CONFIG_FILE "chef-managed/tls.cfg"
>
> #!substdef "!LISTEN_EXTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g"
>
> #!substdef "!LISTEN_INTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g"
>
> #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g"
>
> REgards,
>
>
> Richard
>
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Re: [SR-Users] Kamailio and RTPEngine acting as SBC : wrong IP address returned

[Karsten Horsmann]

Hi Cyril,

There is only one listen directive


listen=udp:172.17.203.102:5060 advertise 3.99.99.99:5060

And that causes that every communication is send out with the advertise ip.

I personally would add an second listen directive without advertising and
an different port. Then use force sendsocket to make sure that you use the
right socket for sending. You can use of the new name directive to label
the sockets with useful names in that place.

Kind regards

Karsten Horsmann


Cyril Ramière  schrieb am Mi., 24. Nov. 2021,
08:22:

> Hi there,
>
> I'm already using Kamailio as a simple proxy and it worked flawlessly.
>
> This time, I need to have a new setup like this:
>
> [UAC (sip phone)] <--- register & has to deal only with > | PUBLIC |
> [Kamailio + RtpEngine] | PRIVATE | <> [Asterisk]
> Client = 37.100.100.100 (public)
> Asterisk = 172.17.200.20 (private)
> Kamailio/RtpEngine = 172.17.203.102 (private, advertising 3.99.99.99
> public IP)
>
> Kamailio and RtpEngine will behave as a SBC/B2BUA, the client will deal
> only with kamailio using public access (internet), and asterisk will handle
> the calls but stays 100% private behind kamailio.
>
> Kamailio will handle registrations to reduce the load put on the asterisk
> boxes.
>
> The setup is running on AWS, my Kamailio has 1 Private IP address tied to
> 1 Public IP address (an Elastic public IP) so, no multi-homing, one network
> interface.
>
> I made a "simple" work in progress configuration, the registration is
> working as a PoC, and my calls are *almost* working but with some nonsense.
>
> My Main issue is that when my UAC (SIP phone) make a call, to let's
> say +1000, the call enters kamailio, then kamailio handle the invite,
> trigger rtpEngine, but the IP addresses in Record-Route and Via headers are
> wrong.
>
> From Kamailio to UAC Sip Phone, it seems OK.
> But from Kamailio to Asterisk, the record-route/via headers are set with
> Kamailio Public IP when it should be the Kamailio private IP.
>
> So, when Asterisk sends a BYE, it tries to send it to the public IP of
> Kamailio, which is wrong.
> There are probably other quirks in the configuration but this one, I can't
> figure out what's wrong.
>
> I'm attaching a capture file, a log file and my configuration file (some
> informations are redacted).
> I am missing something and sadly I can't figure out what, a little help
> would be appreciated :)
>
> Regards
>
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Re: [SR-Users] Call limiting with Kamailio

[Karsten Horsmann]

Hi,

using dialog or dlg makes your life easier.

-> Counting calls with htable and replicating with DMQ to other node
(can the data survive from a restart?)

htable can write back to the database. But that solution needs a bit more
(getting the data - manually or via dialog/dlg and replicate that htable,
how to work with htable)?

As an untested idea - you can define two htables. One htable_nodeA the
other htable_nodeB.
Node A writes only into htable_nodeA and vice versa. So there should no
race condition.
And then you may check htable_nodeA customerX calls and htable_nodeB
customerX calls (sum that) on new calls.

Many ways.

Kind regards
Karsten


Am Fr., 26. Nov. 2021 um 13:26 Uhr schrieb Henning Westerholt :

> Hello,
>
> you could also use dialog profile functionality (single server or also
> with DMQ sync), also probably also the new dlgs module (single server)
>
> Cheers,
>
> Henning
>
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Re: [SR-Users] Problem implementing Outbound Proxy

[Karsten Horsmann]

Hi Christoph,

you can also use the "topos" module to "hide" all the stuff:

https://kamailio.org/docs/modules/5.5.x/modules/topos.html

Kind regards
Karsten

Am Fr., 26. Nov. 2021 um 13:04 Uhr schrieb Christoph Russow :

> yeah the only thing different are the private ips and the record-route
> headers.
>
> the carrier needs a special request thats why the +e164 to and e164
> everywhere else and the e164 in invite header.
>
> if i start the call directly from freeswitch to the carrier (without
> kamailio proxy) it works.
>
> so i think maybe (as this is our main carrier) i need some other way to
> solve that.
>
> Best regards
> Christoph
>
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Re: [SR-Users] Problem implementing Outbound Proxy

[Karsten Horsmann]

Hi Christoph,

sorry i overlooked that. Seems okay to me. So IMHO its "okay" in SIP.

Your Carrier didnt like it.
First - i would try to ask the Carrier -
The Error Message "503 Service Unavailable-No Bandwidth Available" seems to
me like a blocking rule on the carrier side.
Could be mixed +e164 (in To) vs e164 in From/PAI/whatever - the
Contact-Header or something else.


The other way is to capture a working outbound call and then try to check
what's different.

Best
Karsten


Am Fr., 26. Nov. 2021 um 11:18 Uhr schrieb Christoph Russow :

> hi,
>
> thats what i did on our application servers (freeswitch gateway config).
> kamailio routes the calls as wanted but the carrier rejects it (see pcap
> file in first email).
>
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Re: [SR-Users] Problem implementing Outbound Proxy

[Karsten Horsmann]

Hi Christoph,

its easier for the mailinglist members to help you,
if they see that you tried something and show for example your config
snippet where you have issues.

Right now its very hard to tell what to do.

So "i use configuration like in the shipped one with kamailio X.Y.Z and i
didnt get the point how to identify my freeswitch as trusted for outbound
and i used the default config
https://github.com/kamailio/kamailio/blob/5.5.3/etc/kamailio.cfg#L566 and
didnt get pstn route working."

As an example.

Kind regards
Karsten

Am Fr., 26. Nov. 2021 um 10:16 Uhr schrieb Christoph Russow :

> hi karsten,
>
> yeah thats what i thought but i'm a bit without a clue here how to
> achieve this. any hints?
>
> Best regards
> Christoph
>
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Re: [SR-Users] Problem implementing Outbound Proxy

[Karsten Horsmann]

Hi Christoph,


so you have an trusted connection "SIPTRUNK" to kamailio (and then
dispatcher to application/freeswitch B2BUA).
I would create an now an vice-versa flow in your kamailio like this:
FreeSWITCH (bridge/internal/kamailio-ip) to kamailio to SIPTRUNK.


Kind regards
Karsten

Am Fr., 26. Nov. 2021 um 09:01 Uhr schrieb Christoph Russow :

> Hi all,
>
> can no one help here? do you need more informations?
>
> Best regards
> Christoph
>
> Am 23.11.21 um 15:26 schrieb Christoph Russow:
> > Hi all,
> >
> > im currently trying to add a Outbound Proxying to our system which right
> > now does inbound loadbalancing for our application servers.
> >
> > Current:
> > Inbound SIP Trunk -> Kamailio -> dispatcher -> ApplicationServer 1..n
> >
> > i now want to add the ability for our internal Application Servers to
> > terminate their outbound calls through the kamailio system.
> >
> > currently i've setup a gateway in the (freeswitch) application server
> > that has the kamailio system set as outbound-proxy. this leads to sip
> > calls started with
> >
> > originate sofia/gateway/kamailio/@
> >
> > to be sent to the kamailio system which then forwards it to the given
> > carrier IP. Sadly our main carrier answers with "503 service unavailable
> > - no bandwith available". (maybe due to the internal ip-addresses within
> > some of the headers?)
> >
> > i've attached our current kamailio configuration and also a pcap of a
> > call i tried to start with our main carrier.
> >
> > i tried this way as it seemed to be the easiest way so our application
> > servers still can select the outbound carrier within their flow.
> >
> > maybe i'm on the wrong way here and need to solve this problem in a
> > completely different way i don't know.
> >
> > TLDR: sip proxying does not work as expected. need a way to achieve
> > outbound calls routed through kamailio from internal application servers
> >
> > any hint/help into the right direction is appreciated
> >
> > Best Regards
> > Christoph Russow
> >
> >
> > __
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>
> --
> 
>
> EMTEX GmbH
> Christoph Russow
> Software Engineer
>
> Bischof-Otto-Weg 9
> D-91086 Aurachtal
>Geschäftsführer:   Markus Enzinger
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Re: [SR-Users] How to not write logs to journald with kamailio

[Karsten Horsmann]

lution was to increase some journald/rsyslog settings 
> (RateLimitBurst=100 / $imjournalRatelimitInterval 1 
> $imjournalRatelimitBurst 50).
>
>
>
> But it is somehow a workaround.
>
>
>
> I’m looking to know if it should be possible for kamailio to write directly 
> the logs to the syslog only.
>
>
>
> Thanks in advance to those can help me.
>
>  Regards
>
> Patrick Ginhoux
>
>
>
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> --
>
> Daniel-Constantin Mierla -- www.asipto.com
>
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
> Kamailio Advanced Training - Online
>
>   Feb 21-24, 2022 (America Timezone)
>
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
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Re: [SR-Users] Suppressing part of the xlog line

[Karsten Horsmann]

Hi,

In kamailio.cfg place

log_name="kamailio"

to have only kamailio as log stuff

And just for the records - how to get rid off the rsyslog prefix time/date
stuff:
Depends on your syslog facility in config

# /etc/rsyslog.d/kamailio.conf
# drop :msg from rtpengine with "timer run time" - debugging only:
:msg, contains, "timer run time" ~
# Template only msg:
$template myFormat,"%msg%\n"
# log only to local* and NOT to /var/log/messages
local6.*;local6.!=info -/var/log/platform/voip/kamailio/rtpengine.log
local6.* stop
local5.* -/var/log/platform/voip/kamailio/rtpengine-cdr.log; myFormat
local5.* stop
local4.* -/var/log/platform/voip/kamailio/kamailio.log
local4.* stop
local2.* -/var/log/platform/voip/kamailio/kamailio-cdr.log; myFormat
local2.* stop
local1.* -/var/log/platform/voip/kamailio/kamailio-cdr-json.log; myFormat

Hope that helps a bit.

Steve Wilkins  schrieb am Do., 4. Nov. 2021, 18:07:

> Hello,
>
>
>
> Is there a way to suppress the printing  “date & time, host, and Kamailio
> execution path”  at the begging of each xlog line?
>
>
>
> Example current xlog line:
>
> *Nov  4 16:54:52 myserver  /usr/local/sbin/kamailio[7021]: INFO:
> 

[SR-Users] how to prevent the log_prefix in syslog acc cdrs?

[Karsten Horsmann]

Hi List,

i thought there could be a solution to disable the log_prefix for acc
module - but i didn't found it. Is there a solution to supress log_prefix
for acc like acc_json did that?

I use kamailio 5.5.2 with that settings:

log_name="kamailio"
log_facility=LOG_LOCAL4
log_prefix="{$hdr(CSeq) $ci} "

the acc module then logs the "log_prefix" - plus some module/function stuff:

INFO: {1 INVITE SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050} acc
[acc.c:285]: acc_log_request(): ACC: transaction answered:
timestamp=1635843776;utctime=2021-11-02
09:02:56;method=INVITE;from_tag=SDkk50d01-e8bc0cda;to_tag=XBr6apQ9HN4HD;call_id=SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050;code=200;reason=OK;src_user=bob;src_domain=sip100;src_ip=172.20.120.56;dst_ouser=alice;dst_user=echo;dst_domain=172.20.120.59;user_agent=;x_t_call_id=;x_routing=;x_billing=
INFO: {1 ACK SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050} acc
[acc.c:285]: acc_log_request(): ACC: request acknowledged:
timestamp=1635843776;utctime=2021-11-02
09:02:56;method=ACK;from_tag=SDkk50d01-e8bc0cda;to_tag=XBr6apQ9HN4HD;call_id=SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050;code=200;reason=OK;src_user=bob;src_domain=sip100;src_ip=172.20.120.56;dst_ouser=alice;dst_user=alice;dst_domain=172.20.120.51;user_agent=;x_t_call_id=;x_routing=;x_billing=
INFO: {2 BYE SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050} acc
[acc.c:285]: acc_log_request(): ACC: transaction answered:
timestamp=1635843833;utctime=2021-11-02
09:03:53;method=BYE;from_tag=SDkk50d01-e8bc0cda;to_tag=XBr6apQ9HN4HD;call_id=SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050;code=200;reason=OK;src_user=bob;src_domain=sip100;src_ip=172.20.120.56;dst_ouser=alice;dst_user=alice;dst_domain=172.20.120.51;user_agent=;x_t_call_id=;x_routing=;x_billing=

The acc_json module is able to log acc without log_prefix:

{"time": "2021/11/02 10:02:56", "method": "INVITE", "from_tag":
"SDkk50d01-e8bc0cda", "to_tag": "XBr6apQ9HN4HD", "callid":
"SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050", "sip_code": "200",
"sip_reason": "OK", "src_user": "bob", "src_domain": "sip100", "src_ip":
"172.20.120.56", "dst_ouser": "alice", "dst_user": "echo", "dst_domain":
"172.20.120.59", "user_agent": "", "x_t_call_id": "", "x_routing": "",
"x_billing": ""}
{"time": "2021/11/02 10:03:53", "method": "BYE", "from_tag":
"SDkk50d01-e8bc0cda", "to_tag": "XBr6apQ9HN4HD", "callid":
"SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050", "sip_code": "200",
"sip_reason": "OK", "src_user": "bob", "src_domain": "sip100", "src_ip":
"172.20.120.56", "dst_ouser": "alice", "dst_user": "alice", "dst_domain":
"172.20.120.51", "user_agent": "", "x_t_call_id": "", "x_routing": "",
"x_billing": ""}

And just for the records - how to get rid off the rsyslog prefix time/date
stuff:

# /etc/rsyslog.d/kamailio.conf
# drop :msg from rtpengine with "timer run time" - debugging only:
:msg, contains, "timer run time" ~
# Template only msg:
$template myFormat,"%msg%\n"
# log only to local* and NOT to /var/log/messages
local6.*;local6.!=info
 -/var/log/platform/voip/kamailio/rtpengine.log
local6.*stop
local5.*
 -/var/log/platform/voip/kamailio/rtpengine-cdr.log; myFormat
local5.*stop
local4.*
 -/var/log/platform/voip/kamailio/kamailio.log
local4.*stop
local2.*
 -/var/log/platform/voip/kamailio/kamailio-cdr.log; myFormat
local2.*stop
local1.*
 -/var/log/platform/voip/kamailio/kamailio-cdr-json.log; myFormat
local1.*stop

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Re: [SR-Users] kamailio didnt run with acc_json (without dialog module) - kamailio 5.5.2

[Karsten Horsmann]

Hi Daniel,

now its clear for me - if you want to use cdr, dialog module is needed.
Then you become one-line cdrs as plus.
Thanks.

Am Mo., 1. Nov. 2021 um 14:15 Uhr schrieb Daniel-Constantin Mierla <
mico...@gmail.com>:

> Hello,
>
> I updated a bit the text in the section, but feel free to make a pull
> request to clarify further.
>
> Cheers,
> Daniel
>
>
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Re: [SR-Users] kamailio didnt run with acc_json (without dialog module) - kamailio 5.5.2

[Karsten Horsmann]

Hi List,

okay - i found it myself after reading the code a bit. cdr_enable must be 0
if i didnt want to use dialog.so.
A bit confusing what that parameter described in the documentation.

3.11. cdr_enable (str)

Enable Call Data Record generation.

Default value is 0 (disabled).

Am Mo., 1. Nov. 2021 um 13:22 Uhr schrieb Karsten Horsmann <
khorsm...@gmail.com>:

> Hi List,
>
> i want to try out acc_json (to syslog). But if i set up  "cdr_enable", 1 -
> it breaks the startup of kamailio with that error (dialog module is not
> listed as dependency).
>
> Any hints?
>
> Nov 01 13:13:59 siprouter2 kamailio[22397]: INFO: acc_json
> [acc_json_mod.c:135]: mod_init(): janson version : 2.10
> Nov 01 13:13:59 siprouter2 kamailio[22397]: ERROR: acc_json
> [acc_json_mod.c:204]: mod_init(): can't load dialog API
> Nov 01 13:13:59 siprouter2 kamailio[22397]: ERROR: 
> [core/sr_module.c:943]: init_mod(): Error while initializing module
> acc_json (/usr/lib64/kamailio/modules/acc_json.so)
>
>
> #!ifdef WITH_DBTEXT
> loadmodule "db_text.so"
> #!endif
> #!ifndef WITH_DBTEXT
> loadmodule "db_mysql.so"
> #!endif
>
> loadmodule "jsonrpcs.so"
> loadmodule "kex.so"
> loadmodule "corex.so"
> loadmodule "tm.so"
> loadmodule "tmx.so"
> loadmodule "sl.so"
> loadmodule "rr.so"
> loadmodule "pv.so"
> loadmodule "maxfwd.so"
> loadmodule "textops.so"
> loadmodule "siputils.so"
> loadmodule "xlog.so"
> loadmodule "sanity.so"
> loadmodule "ctl.so"
> loadmodule "cfg_rpc.so"
> loadmodule "acc.so"
> loadmodule "dispatcher.so"
> loadmodule "permissions.so"
> loadmodule "avpops.so"
> #!ifdef WITH_ACCJSON
> loadmodule "acc_json.so"
> #!endif
> #!ifdef WITH_DIALPLAN
> loadmodule "dialplan.so"
> modparam("dialplan", "db_url", DBURL)
> modparam("dialplan", "fetch_rows", 1000)
> modparam("dialplan", "attrs_pvar", "$avp(s:dest)")
> #!endif
>
> #!ifdef WITH_JSONSTATS
> loadmodule "xhttp.so"
> loadmodule "jansson.so"
> loadmodule "janssonrpcc.so"
> #!endif
>
> #!ifdef WITH_HOMER
> loadmodule "siptrace.so"
> #!endif
>
> loadmodule "pipelimit.so"
> loadmodule "htable.so"
>
> 
>
> # - acc params -
> modparam("acc", "log_flag", FLT_ACC)
> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
> # - acc params -
> #!ifdef WITH_ACCSYSLOG
> modparam("acc", "log_facility", "LOG_LOCAL2")
> modparam("acc", "log_level", 2)
> #!endif
> /* what special events should be accounted ? */
> modparam("acc", "time_mode", 4)
> modparam("acc", "time_attr", "utctime")
> modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S")
> modparam("acc", "acc_time_column", "localtime")
> ## --- to syslog
> modparam("acc", "early_media", 1)
> modparam("acc", "report_ack", 1)
> modparam("acc", "report_cancels", 1)
> /* by default ww do not adjust the direct of the sequential requests.
>  * if you enable this parameter, be sure the enable "append_fromtag"
>  * in "rr" module */
> modparam("acc", "detect_direction", 0)
> /* account triggers (flags) */
> modparam("acc", "log_flag", FLT_ACC)
> modparam("acc", "log_missed_flag", FLT_ACCMISSED)
> modparam("acc", "log_extra",
> "src_user=$fU;src_domain=$fd;src_ip=$si;"
>
> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;user_agent=$ua;x_t_call_id=$avp(x-t-call-id);x_routing=$avp(x_routing);x_billing=$avp(x_billing)")
> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
> /* enhanced DB accounting */
> #!ifdef WITH_ACCDB
> modparam("acc", "db_flag", FLT_ACC)
> modparam("acc", "db_missed_flag", FLT_ACCMISSED)
> modparam("acc", "db_url", DBURL)
> modparam("acc", "db_extra",
> "src_user=$fU;src_domain=$fd;src_ip=$si;"
> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
> #!endif
>
> #!ifdef WITH_ACCJSON
> modparam("acc_json", "acc_missed_flag", 3)
> modparam("acc_json", "acc_time_mode&

[SR-Users] kamailio didnt run with acc_json (without dialog module) - kamailio 5.5.2

[Karsten Horsmann]

Hi List,

i want to try out acc_json (to syslog). But if i set up  "cdr_enable", 1 -
it breaks the startup of kamailio with that error (dialog module is not
listed as dependency).

Any hints?

Nov 01 13:13:59 siprouter2 kamailio[22397]: INFO: acc_json
[acc_json_mod.c:135]: mod_init(): janson version : 2.10
Nov 01 13:13:59 siprouter2 kamailio[22397]: ERROR: acc_json
[acc_json_mod.c:204]: mod_init(): can't load dialog API
Nov 01 13:13:59 siprouter2 kamailio[22397]: ERROR: 
[core/sr_module.c:943]: init_mod(): Error while initializing module
acc_json (/usr/lib64/kamailio/modules/acc_json.so)


#!ifdef WITH_DBTEXT
loadmodule "db_text.so"
#!endif
#!ifndef WITH_DBTEXT
loadmodule "db_mysql.so"
#!endif

loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"
loadmodule "permissions.so"
loadmodule "avpops.so"
#!ifdef WITH_ACCJSON
loadmodule "acc_json.so"
#!endif
#!ifdef WITH_DIALPLAN
loadmodule "dialplan.so"
modparam("dialplan", "db_url", DBURL)
modparam("dialplan", "fetch_rows", 1000)
modparam("dialplan", "attrs_pvar", "$avp(s:dest)")
#!endif

#!ifdef WITH_JSONSTATS
loadmodule "xhttp.so"
loadmodule "jansson.so"
loadmodule "janssonrpcc.so"
#!endif

#!ifdef WITH_HOMER
loadmodule "siptrace.so"
#!endif

loadmodule "pipelimit.so"
loadmodule "htable.so"



# - acc params -
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
# - acc params -
#!ifdef WITH_ACCSYSLOG
modparam("acc", "log_facility", "LOG_LOCAL2")
modparam("acc", "log_level", 2)
#!endif
/* what special events should be accounted ? */
modparam("acc", "time_mode", 4)
modparam("acc", "time_attr", "utctime")
modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S")
modparam("acc", "acc_time_column", "localtime")
## --- to syslog
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
 * if you enable this parameter, be sure the enable "append_fromtag"
 * in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"

"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;user_agent=$ua;x_t_call_id=$avp(x-t-call-id);x_routing=$avp(x_routing);x_billing=$avp(x_billing)")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif

#!ifdef WITH_ACCJSON
modparam("acc_json", "acc_missed_flag", 3)
modparam("acc_json", "acc_time_mode", 1)
modparam("acc_json", "cdr_output_syslog", 1)
modparam("acc_json", "acc_time_format", "%Y/%m/%d %H:%M:%S")
modparam("acc_json", "acc_log_facility", "LOG_LOCAL1")
modparam("acc_json", "acc_log_level", 2) # Set acc_log_level to 2 (L_INFO)
modparam("acc_json", "cdr_enable", 1)
#!endif


version: kamailio 5.5.2 (x86_64/linux) 55e232
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE,
USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,
TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024,
BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 55e232
compiled on 14:20:31 Aug 25 2021 with gcc 4.8.5

-- 
Mit freundlichen Grüßen
*Karsten Horsmann*
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Re: [SR-Users] tls connections

[Karsten Horsmann]

Hi Richard,


last time I saw something with tcp window problems was an problem with
incompatible systems that didn't handle this well (for example SMTP
servers, Helo works, sending not).

You can AFAIK only adjust that on your Linux system via sysctl.



Richard Robson  schrieb am Mi., 20. Okt. 2021,
18:01:

> I'm trying in vain to get a tls connection to  a carrier.
>
>
> ~Without too much detail we are exchanging certificates and they are
> saying we are triggering a reset of the connection because we are
> requesting to chage the tcp window size.
>
>
> there are requiring us to have a window size of 65535. there we are
> sending a smaller window size. as far as I understand this should be OK.
> I have tried to force out linux install to set the window size to 65535
> but we are still sing the smallerwindow in the tcp packet.
>
>
> Is there any settings in Kamailio that could force the window size? we
> are using Kamailio 5.4.2
>
>
> Thanks in advance Richard
>
>
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Re: [SR-Users] Use constants in attrs column of the dispatcher module

[Karsten Horsmann]

Hi Markus,

not testet but looked at latest doc 'sockname' instead of socket could be a
solution.
So you label your sockets with "name" in the listen directive (in config
file) and use 'sockname' in dispatcher table.
https://kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatcher.ex.attributes

listen=udp:1.2.3.4:5060 name "sock1"


Am Di., 19. Okt. 2021 um 15:13 Uhr schrieb Markus Bönke :

> Hello all,
>
> I’ve a question regarding the dispatcher module. So far we use on every
> server a separate database table, where the attribute column is set to the
> local internal or external socket like "socket=udp:192.168.1.70:5060“ or
> socket=udp:80.239.xxx.xxx:5060. Is there a way to use constants here like
> socket=udp:IPADDR_INTERN:5060 or socket=udp:IPADDR_EXTERN:5060 which are
> replaced when the data is loaded from the DB?
>
>
> Thanks and regards
>
> Markus
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-- 
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*Karsten Horsmann*
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Re: [SR-Users] help

[Karsten Horsmann]

Hi Said,

if no one is answering your questions then you should ask more specific
what is the issue and what you try to solve that.

That could result in more answers.

Kind regards
Karsten Horsmann

Said Hassani  schrieb am Fr., 30. Juli 2021, 08:11:

> Dear Team,
>
> I would like to know how to link Kamailio with openIMSCore?
> I would like to use Kamailio as an Application server.
> Please guide me or if there is someone who has worked on this subject, I
> would like to benefit from his help.
>
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Re: [SR-Users] which is better for B2BUA

[Karsten Horsmann]

Hello,

just for the records:

You can also bypass media with FreeSWITCH if you want. The overall handling
of many parallel calls is still not so high like in SEMS.

https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
This is also possible on a per Call base.

And you can use different profiles as well in FreeSWITCH.

Asterisk Experts will have more knowledge what is possible there, and maybe
someone told us whats going on with Yate.

Kind regards
Karsten

Am Di., 13. Juli 2021 um 10:11 Uhr schrieb Mojtaba :

>
> Hello there,
> According to ,
> https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html,
> which talked about  B2BUA (just signalling) in Kamailio.
> As i have experienced working with SEMS, freeswitch and Kamailio while
> using B2BUA feature, Each of them have pros and cons:
> 1- The sems is a light sip engine server with several applications (like
> as sbc) for using b2bua. All incoming and outgoing calls could go to sems
> server for doing b2bua like this:
>
> Incoming<===>Kamailio<>Sems<>Kamailio<===>outgoing
>
> 2- In sems, you could disable rtp realying. It forces sems to work just as
> b2bua without anchoring RTP
> 3- Easy to use different active profiles in routing.
>
>
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Re: [SR-Users] Create accounta with multiple DIDs

[Karsten Horsmann]

Hi,

You can also map your number blocks with an avp from the dialplan module
and then feed location for example.

Kind regards
Karsten Horsmann

John Tuxies  schrieb am So., 18. Apr. 2021, 18:30:

> Dear Daniel, thanks a lot for the reply.
>
> Here is a diagram of what i have and what i try to achieve. In a isolated
> environment there is a single network (192.168.0.0/24) and one extension
> number is registered to one Asterisk server as a trunk. I would like to
> "push" DIDs in/out to these trunks(1000-1003) and eg
>
>- Asterisk's_1 server user which has the DID +493087710024 to be able
>and dial the number +49308771360 and reach another user on Asterisk Server
>4. All the communication will go through the Server (Kamailio).
>
> I am new to the field and i tried to search on alis db module, but i got
> more confused.
>
>
>
>
> [image: ITSP_Scenario.jpg]
>
>
>
> On Fri, Apr 16, 2021 at 4:47 PM Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> On 16.04.21 08:18, John Tuxies wrote:
>> > I am novice in the area and i have a Kamailio basic setup where i
>> > create users and each user can call each other.
>> > No connection to an ITSP. i am trying to simulate things and need to
>> > send multiple numbers to each user.
>> > For example i have users 1000-1003. I need each user to have a range
>> > of 100 numbers (fake since it ia not connected anywhere) and routed to
>> > all the other users as well. The idea is to connect Asterisk servers
>> > and each one have a registered account 1000-1003. Then each Asterisk's
>> > extension have a particular DID  and call another Asterisk server to a
>> > different DID. Off course this will go through Kamailio.
>> > Now i call from one Asterisk to the other by using the 1000-1003 and
>> > it works. But i need the DID routing
>>
>>
>> Somehow I do not really get what you try to achieve, how you want the
>> mapping between between DIDs and user IDs. But, by a long shot, look at
>> alias_db module, might be what you look for.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla -- www.asipto.com
>> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>> Kamailio Advanced Training - Online
>> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>>
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Re: [SR-Users] monitor uac registration status

[Karsten Horsmann]

Hi,


if you want to go the golang way maybe the binrpc implementation here helps
you.

But imho binrpc is nowadays.
https://github.com/florentchauveau/go-kamailio-binrpc

Kind regards
Karsten Horsmann

Arsen Semenov  schrieb am Fr., 5. März 2021, 12:06:

> Hi,
>
> just as an idea I would suggest to have a service which will periodically
> query kamailio with json-rpc uac.reg_info and process the response, it
> could be written in golang for example
>
> On Fri, 5 Mar 2021 at 11:37 AM, Davide Zanichelli <
> dav.zaniche...@gmail.com> wrote:
>
>> Hello
>>
>> is there a way to monitor (write to db) uacreg status ?
>>
>> Thanks You
>>
>> Davide
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Re: [SR-Users] kamailio+rtpengine - number of calls

[Karsten Horsmann]

Hi Marek,


Could be that your rtpengine sessions are not deleted correctly in some
cases.

You can list the sessions with "rtpengine-ctl list sessions all" and check
the callids there with your cdrs and check the "mark for deletion" row.

rtpengine AFAIK keeps the deleted sessions in the list and remove it then
from time to time.

Which is normally not a long period (not sure every 30 seconds or longer).

Anyhow if the number of rtpengine sessions grows up and your dialog did not
match with that, than it could be an issue with your config how you control
rtpengine.

Or your dialog counter is not corrected.

Kind regards


marek  schrieb am Mi., 24. Feb. 2021, 14:16:

> there is some problem problem with rtpengine-ctl list numsessions and
> caching.
>
> i have
>
> Current sessions own: 719
> Current sessions foreign: 0
> Current sessions total: 719
>
> but current calls is maybe around 50
>
> i wil check the prometheus module
>
> thanks
>
>
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Re: [SR-Users] kamailio+rtpengine - number of calls

[Karsten Horsmann]

Hi Marek,

depends on your needs.

There is an Prometheus module if you on that side or I personally use
dialog module with profiles in out total and fetch that with shell scripts
by telegraf which feeds them into influxdb. Also that parsed rtpengine-ctl.

But rtpengine had also an grafana export that I do not use at this time.

And there is an lighter dlg module maybe that can count also. And the
statistics of Kamailio itself brings many numbers.

There is no turnkey read solution more ideas how to solve the question.

Kind regards
Karsten Horsmann

marek  schrieb am Fr., 19. Feb. 2021, 13:58:

> hi,
>
> i'm using kamailio+rtpengine as SBC. I need info about
>
> - current number of total calls
>
> - current number of incoming calls
>
> - current number of outgoing calls
>
> what's you preferred way? (kamailio snmp, statistics
> module,rtpengine-ctl list numsessions,...)
>
> Marek
>
>
>
>
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Re: [SR-Users] FW: re I am trying to get a webrtc setup going. Here is what I have

[Karsten Horsmann]

Hi Daniel,

The Github howto from havfo based on Debian buster and the shipped Kamailio
version AFAIK is 5.2. You tried 5.4.x.

So you check the Kamailio.org change wiki sites 5.2.x to 5.3.x and 5.3.x to
5.4.x to adapt the config until the errors are gone.

Or try the older version to see if it works generally and then try to adapt
the configuration.

Kind regards
Karsten Horsmann


Daniel Siemens  schrieb am Di., 16. Feb. 2021, 22:57:

>
>
> I have tried to follow the instructions from the site,
>
>
>
> Kamaillo however won’t startup a number of errors below. Is the
> version on the web rtc site specific to a version?
>
> Site is https://github.com/havfo/WEBRTC-to-SIP
>
>
>
>
>
>
>
>
>
> [root@ec2-sipproxy01 ~]# service kamailio status
>
> Redirecting to /bin/systemctl status kamailio.service
>
> â— kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server
>
>Loaded: loaded (/usr/lib/systemd/system/kamailio.service; enabled;
> vendor preset: disabled)
>
>Active: failed (Result: start-limit) since Fri 2021-02-12 14:52:06 CST;
> 4 days ago
>
> Main PID: 5329 (code=exited, status=255)
>
>
>
> Feb 12 14:52:05 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service: main process exited, code=exited, status=255/n/a
>
> Feb 12 14:52:05 ec2-sipproxy01.multiservice.com systemd[1]: Unit
> kamailio.service entered failed state.
>
> Feb 12 14:52:05 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service failed.
>
> Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service holdoff time over, scheduling restart.
>
> Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: Stopped
> Kamailio (OpenSER) - the Open Source SIP Server.
>
> Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: start request
> repeated too quickly for kamailio.service
>
> Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: Failed to
> start Kamailio (OpenSER) - the Open Source SIP Server.
>
> Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]: Unit
> kamailio.service entered failed state.
>
> Feb 12 14:52:06 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service failed.
>
> [root@ec2-sipproxy01 ~]# service kamailio start
>
> Redirecting to /bin/systemctl start kamailio.service
>
> [root@ec2-sipproxy01 ~]# journalctl -xe
>
> --
>
> -- The start-up result is done.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12905]: ERROR:
> bad config file (4 errors)
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12905]: loading
> modules under config path: /usr/lib64/kamailio/modules/
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service: main process exited, code=exited, status=255/n/a
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Unit
> kamailio.service entered failed state.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service failed.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service holdoff time over, scheduling restart.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Stopped
> Kamailio (OpenSER) - the Open Source SIP Server.
>
> -- Subject: Unit kamailio.service has finished shutting down
>
> -- Defined-By: systemd
>
> -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
>
> --
>
> -- Unit kamailio.service has finished shutting down.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Started
> Kamailio (OpenSER) - the Open Source SIP Server.
>
> -- Subject: Unit kamailio.service has finished start-up
>
> -- Defined-By: systemd
>
> -- Support: http://lists.freedesktop.org/mailman/listinfo/systemd-devel
>
> --
>
> -- Unit kamailio.service has finished starting up.
>
> --
>
> -- The start-up result is done.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12908]: ERROR:
> bad config file (4 errors)
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com kamailio[12908]: loading
> modules under config path: /usr/lib64/kamailio/modules/
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service: main process exited, code=exited, status=255/n/a
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Unit
> kamailio.service entered failed state.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service failed.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]:
> kamailio.service holdoff time over, scheduling restart.
>
> Feb 16 15:35:13 ec2-sipproxy01.multiservice.com systemd[1]: Stopped
> Kamail

[SR-Users] filter_body("application/sdp") on multipart/mixed; boundary with double Content-Length

[Karsten Horsmann]

Hi List,

I have some weird  multipart/mixed;boundary issues.
I use kamailio 5.3.6 with rtpengine 7.x.x (same with 8.x.x).

If i try to remove a multipart/mixed and leave only the sdp with
filter_body() that works except one thing.

The sender used IMHO an non RFC compliant version of rfc5621 and wrote the
Content-Length Header twice (one in the header, one in the sdp).
After manipulating the SDP filter_body() lets Content-Length in the sdp (i
guess).
And rtpengine is not happy with the sdp after that.
rtpengine [rtpengine.c:2588]: rtpp_function_call(): proxy replied with
error: Failed to parse SDP

Any ideas how to solve this?

Kind regards
Karsten Horsmann

https://tools.ietf.org/html/rfc5621#section-3.1

route[REMOVE_MULTIPART] {
if (has_body("multipart/mixed")) {
if ($sel(cfg_get.remove.multipart) == 1) {
if (filter_body("application/sdp")) {
remove_hf("Content-Type");
append_hf("Content-Type:
application/sdp\r\n");
msg_apply_changes();
xlog("L_WARN", "[$cfg(route)] Body part
multipart/mixed with application/sdp found\n");
} else {
xlog("L_ERROR", "[$cfg(route)] Body part
multipart/mixed without application/sdp found\n");
}
} else {
xlog("L_INFO", "[$cfg(route)] Body part
multipart/mixed found - manipulation disabled\n");
}
}
}

---
INVITE sip:+49@172.20.120.57:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.120.51:5061;branch=z9hG4bK-6589-1-0
From: sipp ;tag=6589SIPpTag001
To: sut 
Call-ID: 1-6589@172.20.120.51
CSeq: 1 INVITE
Contact: sip:sipp@172.20.120.51:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: multipart/mixed;boundary=sonus-content-delim
Content-Length:  1185

--sonus-content-delim
Content-Type: application/sdp
Content-Length: 305

v=0
o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8
s=SIP Media Capabilities
c=IN IP4 8.8.8.8
t=0 0
m=audio 26088 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
a=rtcp:26089
a=ptime:20

--sonus-content-delim
Content-Type: application/pidf+xml

http://btd.orange-business.com; entity="
pres:geotar...@btip.orange-business.com">DESome
StateSomewhereSomewhereSomestreet1XXX
--sonus-content-delim--

-
INVITE sip:+49@foo SIP/2.0
Via: SIP/2.0/TCP
212.XX.XX.XX;branch=z9hG4bKc228.6b8fdf20d79998d1d355feb2dcf5ba63.0
From: sipp ;tag=6589SIPpTag001
To: sut 
Call-ID: 1-6589@172.20.120.51
CSeq: 1 INVITE
Max-Forwards: 69
Content-Length:  347
Content-Type: application/sdp
P-Asserted-Identity: 
Contact: 

Content-Length: 305

v=0
o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8
s=SIP Media Capabilities
c=IN IP4 8.8.8.8
t=0 0
m=audio 26088 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
a=rtcp:26089
a=ptime:20


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Re: [SR-Users] (no subject)

[Karsten Horsmann]

Hi Daniel,


Take a look at havfo nice config example with all you need to play around
with RTC and websocket. There are many moving parts to cover.

https://github.com/havfo/WEBRTC-to-SIP

Kind regards
Karsten Horsmann

Daniel Hermann N'don  schrieb am Fr., 8.
Jan. 2021, 11:08:

> Hello everyone,
> I'm a beginner in kamailio technology, and i want to use websockets module
> in kamailio 5.4 to exchange from the web to kamailio, but i don't know how
> to use it?
> Can someone help me please?
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Re: [SR-Users] Establish connection between MsTeams and Kamailio.

[Karsten Horsmann]

Hi,

If I take a look at the howto from Henning and your dispatcher state I miss
the rest of the ms teams geo redundante dispatcher targets.


https://skalatan.de/de/blog/kamailio-sbc-teams


The OPTION ping works for ms teams AFAIK remember that you must also ping
sip, sip2, sip3.pstnhub

Kind regards
Karsten Horsmann



Willy Valles Rios  schrieb am Di., 26. Jan. 2021,
20:31:

> Hello community,
>
> Please your support, I tell you that I am integrating Kamailio with
> MsTeams, after some time reviewing this, I finally achieved the connection
> from my Kamailio server to MsTeams and I can validate that the connection
> to MsTeams is in AP.
>
> [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI |
> FLAGS"
> URI: sip:
> sip.pstnhub.microsoft.com; transport = tls
> FLAGS: AP
>
> However, from the admin panel of MsTeams (Direct Routing) I see that the
> connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity
> status in "Active" but the SIP options status is in "Warning".
>
> I have made calls from MsTeams thinking that the SIP options status would
> change to "active" but it is still in "Warning" state. On the other hand, I
> have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from
> kamailio are being sent in the following format to MsTeams.
>
> OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0
> Via: SIP / 2.0 / TLS 161.35.44.66:5061
> ;branch=z9hG4bKea07.52224687.0
> To: 
> From: ; tag =
> d3569c818b500aeb8c373426e76c2884-81763c71
> CSeq: 10 OPTIONS
> Call-ID: 13ea237a751e0c48-9148@161.35.44.66
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: kamailio (5.4.0 (x86_64 / linux))
>
> As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not
> contain the "Contact" field, which in theory said "Contact" field should
> have been added by Kamailio according to the configuration added in
> kamailio.cfg
>
> event_route [tm: local-request] {
> sip_trace ();
> if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") {
>append_hf ("Contact:  transport = tls> \ r \ n");
> }
> xlog ("L_INFO", "Sent out tm request: $ mb \ n");
> }
>
> As additional information, I inform you that I also managed to observe the
> SIP OPTIONS that MsTeams sends to Kamailio.
>
> OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0
> FROM: ; tag =
> f1bdeb5f-662f-4544-a436-e9aa9ad78da4
> TO: 
> CSEQ: 1 OPTIONS
> CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a
> MAX-FORWARDS: 70
> VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be
> CONTACT: 
> CONTENT-LENGTH: 0
> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10
> ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY
>
> However I don't see the 200 OK SIP responses from Kamailio to MsTeams.
>
> I think this may be the reason why I see the SIP OPTIONS status in
> "Warning" from the MsTeams panel. Maybe the contact field is not being
> added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for
> that reason I don't see 200OK responses from MsTeams.
>
> Could you help me solve this please.
>
> Cheers
>
> Saludos Cordiales
> --
> *Willy Valles Rios*
> *Unified Communications Specialist*
>
> phone: +51955747343
> em@il: willyvalle...@gmail.com
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Re: [SR-Users] UACREG to many Asterisk

[Karsten Horsmann]

Hi Javier,

Not knowing your config I could imagine that you can maybe use
ds_next_dst()
In a while loop to iterate over all setid members of dispatcher. That's not
tested and most people here would strongly suggest to use Kamailio as
Registrar.

Or you try some configuration that spreads the registration around to your
backends
Take a look at this thread

http://sip-router.1086192.n5.nabble.com/MID-Registrar-Example-td192277.html#a192339

Not sure if this works for you (due the path missing in asterisk 11 you
mentioned) but maybe gives some inspiration.


Kind regards
Karsten Horsmann

Javier Valencia  schrieb am Mi., 27. Jan. 2021, 11:34:

> Hi there!
>
> Go ahead I apologize for my English.
>
> I have a Kamailio 5.4 with the DISPATCHER module and UAC enabled (for
> remote REGISTER), the DISPATCHER has several SETIDs, and they have one or
> more Asterisk balanced behind it.
>
> In Kamalio script, when an endpoint does REGISTER, I enable the REGISTER
> of the UAC module. And when it expires or expires=0 I disable the REGISTER
> in the UAC module.
>
> In the UACREG table in the database, L_UUID (unique key), L_USERNAME,
> R_USERNAME, and AUTH_USERNAME have the same value. For example CST100EXT100.
>
> Knowing that the DISPATCHER and UAC modules have no relationship, but that
> when a SETID has several balanced Asterisks, how can I make the UAC module
> send the REGISTER to all the Asterisks of a SETID?
>
> Clarifications;
>
> 1.- I cannot update the Asterisk, which are currently in version 11. They
> do not support PATH.
>
> 2.- I can modify the content of the UACREG table so that l_uuid has a
> suffix '_X'.
>
> 3.- Call jsonrpc_exec several times to enable the REGISTER in the UAC
> module.
>
> (diff emulated)
> < jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable",
> "params": ["l_uuid", "s: $ au"]}') ;
> > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable",
> "params": ["l_uuid", "s: $ au_1"]}') ;
> > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable",
> "params": ["l_uuid", "s: $ au_2"]}') ;
> > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable",
> "params": ["l_uuid", "s: $ au_3"]}') ;
>
> But I want to know if there is any more elegant way to do this.
>
> Cheers, JV
> --
> [image: Logo] <http://www.voiper.es/>
> Javier Valencia | CTO
> Centro de Negocios Martín Buendía
> Camino de las Cañadas, nº 1C, Portal 1, 2ºG
> 29649 Mijas (Málaga)
> [image: #] 951562080 (T) <951562080> | 687486759 (M) <687486759>
> [image: #] www.voiper.es
> --
>
> AVISO LEGAL
> ---
>
> Este mensaje contiene información confidencial destinada para ser leída
> exclusivamente por el destinatario. Queda prohibida la reproducción,
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> inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo
> por Internet.
>
> Las opiniones, conclusiones y demás información incluida en este mensaje que
> no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se
> entenderá que nunca se ha dado, ni está respaldado por el mismo.
>
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> 
> SMART RECARGAS , SL
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> 29651 MIJAS COSTA (MALAGA)
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Re: [SR-Users] ndb_redis with keydb master/master

[Karsten Horsmann]

Hi David,

ah okay. I used the full way of an redis sentinel cluster with haproxy for
dead end detection.
And this redis setup is for my Kamailio Topos and my rtpengine.

But you can subscribe to one redis dB 1 and replicate it to the other side.
And vice versa. With Kamailio cfg pointing to the local or cross over redis
and the right db it should possible.

I did tried that it's more an idea from the rtpengine folks.

Also Topos will work with other backends IMHO like mysql.

Cheers
Karsten Horsmann

David VILLAUME  schrieb am Mo., 21. Dez. 2020,
14:02:

> Hello,
>
>
>
> Thanks for your response, cross subscription mechanism in video is native
> rtpengine, not on Kamailio side.
>
>
>
> My goal is to have redundant topos in Kamailio without having to build a
> full redis cluster with sentinel, but I’m not sure if  there is a way to
> declare two hosts in ndb_redis with the same name (and any parameter to
> set them either as active/passive either active/active).
>
>
>
> Regards,
>
> David
>
>
>
> *From:* sr-users  * On Behalf Of *Karsten
> Horsmann
> *Sent:* Sunday, December 20, 2020 10:14 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] ndb_redis with keydb master/master
>
>
>
> Hi,
>
>
>
> Maybe you got some inspiration from Yufei Tao presentation
>
>
>
> https://youtu.be/p56F2fw483I
>
>
>
> about an HA Setup with Kamailio. There is redis in a cross subscription
> manner in use.
>
>
>
>
>
> Kind regards
>
> Karsten Horsmann
>
>
>
> David VILLAUME  schrieb am Mi., 16. Dez. 2020,
> 09:18:
>
> Hello,
>
>
>
> I consider using keydb-server in an active/action setup without sentinel.
>
> Do you know if there is a way to declare both hosts in ndb_redis with the
> same server name (target use : topos) ?
>
>
>
> Regards,
>
> David
>
>
>
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Re: [SR-Users] ndb_redis with keydb master/master

[Karsten Horsmann]

Hi,

Maybe you got some inspiration from Yufei Tao presentation

https://youtu.be/p56F2fw483I

about an HA Setup with Kamailio. There is redis in a cross subscription
manner in use.


Kind regards
Karsten Horsmann

David VILLAUME  schrieb am Mi., 16. Dez. 2020,
09:18:

> Hello,
>
>
>
> I consider using keydb-server in an active/action setup without sentinel.
>
> Do you know if there is a way to declare both hosts in ndb_redis with the
> same server name (target use : topos) ?
>
>
>
> Regards,
>
> David
>
>
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Re: [SR-Users] sngrep TLS connection

[Karsten Horsmann]

Hi,

you can also setup an Homer docker instance and use Kamailio as active hep
client (homer encapsulation protocoll).

https://github.com/sipcapture/homer/wiki
So Kamailio sends copies of the signaling traffic via hep to Homer and you
can investigate it.

Cheers
Karsten Horsmann

Agiftel  schrieb am Sa., 12. Dez. 2020, 17:30:

> Hi again, I sang victory too soon :-(
>
> It's working perfectly if SIP calls are UDP or TCP based. If are TLS, very
> strange thing, i can see all trace inside file .data created by siptrace
> but
> if I open that file using sngrep, TLS calls are never displayed.
> Any idea?
> If not, anyway, I would like to ask you what is the best way to
> troubleshoot
> TLS calls?
> What do you use guys?
>
> Best regards
>
>
>
> --
> Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
>
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Re: [SR-Users] how to reload kemi scripts? kamailio 5.3.x or 5.4.x

[Karsten Horsmann]

Hello Daniel and Henning,

I am fine with the rpc reload command.
I try to understand how the mechanism works.

That python 2/3 rpc reloading also works (since which version?) is very
nice.

I assume to find a hint for "reload your scripts with the rpc reload
command for your kemi-scripting language, description are in the module
docu"
or something like that in the KEMI documentation.

Thanks for the quick response.

Cheers
Karsten


Am Di., 8. Sept. 2020 um 09:27 Uhr schrieb Daniel-Constantin Mierla <
mico...@gmail.com>:

> Hello,
>
> Henning pointed in the other response that an RPC command has to be
> executed for reload, giving the link to the one for app_lua module.
>
> I am jumping in here to say that Python2/3 module have now reload
> capabilities as well, what I presented in 2018 is no longer valid for those
> modules. If you see the app_puthon.reload in the list of exported rpc
> commands, then that version has reload support.
>
> Regarding the possibility of "drop in a new file and that's it", I
> explicitly didn't want this mode of operations for a few reasons:
>
>   * requires to check the last update timestamp for each sip packet routed
> with the script
>   * I typically do intermediary saves when updating the files, and this
> can lead to reloading a version before everything is finished. Of course,
> one can edit in other place and the push it to config folder, but somehow I
> didn't like the approach.
>
> If someone wants something like this, it has to do it with a modparam
> option.
>
> Cheers,
> Daniel
> On 08.09.20 09:14, Karsten Horsmann wrote:
>
> Hello List,
>
> i read Daniels Presentation
> https://www.kamailio.org/events/2018-KamailioWorld/Day0/W07-Daniel-Constantin.Mierla-KEMI-Scripting.pdf
> and there is an hint about python(2/3) - that kemi script reloading is not
> supported with python kemi scripts.
>
> Is that correct for the actual versions (5.3.x and 5.4.x)?
> And if i use for example lua as kemi script language, how i restart that
> script only?
> Drop in a new file and thats it (like FreeSWITCH did that)?
>
> Thanks in advance
>
> Cheers
> Karsten
>
> --
> *Karsten Horsmann*
>
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> www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
>

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[SR-Users] how to reload kemi scripts? kamailio 5.3.x or 5.4.x

[Karsten Horsmann]

Hello List,

i read Daniels Presentation
https://www.kamailio.org/events/2018-KamailioWorld/Day0/W07-Daniel-Constantin.Mierla-KEMI-Scripting.pdf
and there is an hint about python(2/3) - that kemi script reloading is not
supported with python kemi scripts.

Is that correct for the actual versions (5.3.x and 5.4.x)?
And if i use for example lua as kemi script language, how i restart that
script only?
Drop in a new file and thats it (like FreeSWITCH did that)?

Thanks in advance

Cheers
Karsten

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Re: [SR-Users] Kamailio v5.4.0 RPM Availability

[Karsten Horsmann]

Hi,

sorry my fault. I checkt the obs service a few days ago and then I read
only the this thread without investigation of the other major distro
versions.


Thanks for the correction.

Cheers
Karsten

Daniel-Constantin Mierla  schrieb am Di., 11. Aug. 2020,
17:43:

> Hello,
>
> I see 7 listed as directory at:
>
>   - http://rpm.kamailio.org/centos/
>
> Cheers,
> Daniel
> On 11.08.20 17:39, Karsten Horsmann wrote:
>
> Hi there,
>
> Thanks for your work.
>
> Are there plans for Centos 7, since Centos 8 is not widely adopted in
> production and 7 is still in support would be nice to see Kamailio 5.4 also
> there.
>
> Sergey Safarov  schrieb am Di., 11. Aug. 2020, 16:15:
>
>> Kamailio RPM you can see here
>> http://rpm.kamailio.org/centos/8/5.4/5.4.0/x86_64/Packages/k/
>>
>> To install
>>
>> dnf -y install dnf-plugins-core
>> dnf config-manager --add-repo http://rpm.kamailio.org/centos/kamailio.repo
>> dnf install kamailio
>>
>>
>> To team.
>> How about officially start to use rpm.kamailio.org?
>>
>> Sergey
>>
>> On Tue, Aug 11, 2020 at 3:54 PM Asgaroth <00asgarot...@gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> I would just like to check in and find out if/when the Kamailio v5.4.0
>>> rpm's will be available on the OpenSUSE build service.
>>>
>>> Thanks to all involved for all the hard work in this new release!
>>>
>>> Thanks
>>>
>>> There are two major products that came out of Berkeley: LSD and UNIX. We
>>> don't believe this to be a coincidence. -- Jeremy S. Anderson
>>> 00asgarot...@gmail.com
>>> <https://link.getmailspring.com/link/146de51a-e8c2-4126-b8d5-ea45e62cb...@getmailspring.com/0?redirect=mailto%3A00asgaroth00%40gmail.com=c3ItdXNlcnNAbGlzdHMua2FtYWlsaW8ub3Jn>
>>> [image: Sent from Mailspring]
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> www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
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Re: [SR-Users] Kamailio v5.4.0 RPM Availability

[Karsten Horsmann]

Hi there,

Thanks for your work.

Are there plans for Centos 7, since Centos 8 is not widely adopted in
production and 7 is still in support would be nice to see Kamailio 5.4 also
there.

Sergey Safarov  schrieb am Di., 11. Aug. 2020, 16:15:

> Kamailio RPM you can see here
> http://rpm.kamailio.org/centos/8/5.4/5.4.0/x86_64/Packages/k/
>
> To install
>
> dnf -y install dnf-plugins-core
> dnf config-manager --add-repo http://rpm.kamailio.org/centos/kamailio.repo
> dnf install kamailio
>
>
> To team.
> How about officially start to use rpm.kamailio.org?
>
> Sergey
>
> On Tue, Aug 11, 2020 at 3:54 PM Asgaroth <00asgarot...@gmail.com> wrote:
>
>> Hi all,
>>
>> I would just like to check in and find out if/when the Kamailio v5.4.0
>> rpm's will be available on the OpenSUSE build service.
>>
>> Thanks to all involved for all the hard work in this new release!
>>
>> Thanks
>>
>> There are two major products that came out of Berkeley: LSD and UNIX. We
>> don't believe this to be a coincidence. -- Jeremy S. Anderson
>> 00asgarot...@gmail.com
>> 
>> [image: Sent from Mailspring]
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Re: [SR-Users] Use-case does Kamailio support PBX - webrtc (web sockets)

[Karsten Horsmann]

Hi Johnny,

an example Integration could be something like havfo did. Helps a bit to
understand the moving parts of Kamailio (sip and websockets, control of
rtpengine) and webrtc stuff around it (stun, turn server for clients that
didn't allowed udp for example).

It will not cover 100% what you need, but you get an idea.


https://github.com/havfo/WEBRTC-to-SIP

Cheers
Karsten

Johnny Ritzer  schrieb am Do., 6. Aug. 2020, 08:08:

> We have a PBX that doesn’t support websockets .
>
>
>
> But we want a PWA webaapp ctxSIP that uses sip.js to work .
>
>
>
> Due to its not SIP generic do we setup like this
>
>
>
> PBX ßà Kamailio ßàwebapp
>
>
>
> Im looking  for developers for android build via react native or ionic
> framework but not sure what solution/archtectiure
>
>
>
> Question. Does Kamailio register the SIP user/pass on the Kamailio server
> meaning it keeps it registered. And thus the ctxSIP client app just handles
> the pass thru for that connection.
>
>
>
> What about RTP do I need a separate handler for that.
>
>
>
> Tha nks
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[SR-Users] kamailio and rtpengine_manage() / rtpengine_delete()

[Karsten Horsmann]

 Hi list,

I use rtpengine 7.2.1 branch with kamailio 5.2.x and 5.3.x in
different setups.
Due to some issues with my 5.2.x config I saw the "Total timed-out sessions
via TIMEOUT" grow but  "Total regular terminated sessions never"  goes up
from 0.
That was an indicator for an non-existing rtpengine_manage() or
rtpengine_delete() at the end of dialogs. Okay.

Now I want to observe that a bit more.
I use parallel forking to webrtc clients, that i guess will be the "more
Offers" than Answers.
Any hints on what numbers are important?

rtpengine-ctl -ip 127.0.0.1: list totals

Total statistics (does not include current running sessions):

 Uptime of rtpengine :27726 seconds
 Total managed sessions  :3435
 Total rejected sessions :0
 Total timed-out sessions via TIMEOUT:0
 Total timed-out sessions via SILENT_TIMEOUT :0
 Total timed-out sessions via FINAL_TIMEOUT  :0
 Total timed-out sessions via OFFER_TIMEOUT  :0
 Total regular terminated sessions   :3435
 Total forced terminated sessions:0
 Total relayed packets   :69127802
 Total relayed packet errors :5063
 Total number of streams with no relayed packets :128
 Total number of 1-way streams   :8
 Average call duration   :233.321876


Graphite interval statistics (last reported values to graphite):
 Total calls duration:0.00

 Min managed sessions:0
 Max managed sessions:0
 Min/Max/Avg offer processing delay
 :0.00/0.00/0.00 sec
 Min/Max/Avg answer processing delay
:0.00/0.00/0.00 sec
 Min/Max/Avg delete processing delay
:0.00/0.00/0.00 sec
 Min/Max/Avg offer requests per second   :0/0/0 per sec
 Min/Max/Avg answer requests per second  :0/0/0 per sec
 Min/Max/Avg delete requests per second  :0/0/0 per sec


Control statistics:

Proxy |  Offer | Answer | Delete |   Ping |
  List |  Query |   StartRec |StopRec | Errors |BlkDTMF
|  UnblkDTMF
127.0.0.1 |   3896 |   3757 |   3499 |  0 |
 0 |  0 |  0 |  0 |  0 |  0
|  0



Kind Regards
-- 
Mit freundlichen Grüßen
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Re: [SR-Users] Setting up MSFT Teams SBC <> SIP trunk but forwarding ACK and CANCEL messages

[Karsten Horsmann]

Hi,


use siptrace to see what Kamailio sees on the tls side or use the message
buffer pv $mb

Like this (please check Syntax at your own, copy pasted that with my
smartphone so no guarantees)

It shows the raw message (unencrypted) before tls module it handles or
after its decrypted AFAIK.

event_route[network:msg] {
if (is_incoming()) {
 xlog("L_INFO", "Received message '$mb' \n");
} else {

xlog("L_INFO", "Sending message '$mb' \n");
}
}


Kind regards
Karsten Horsmann


tangd122  schrieb am Sa., 1. Aug. 2020, 19:40:

> Thank you for your replies! Outbound calling is now completely working.
>
> For incoming calls from SIP trunk to Teams there's still no ACK and CANCEL
> messages. Which causes disconnect in 20 seconds and no audio.
>
> I think there's something wrong with the routing or via headers in the 200
> OK but don't know what.
>
> The complete sip dump: https://paste.projectdev.org/imiluwecaf
>
> Unfortunately i can't see the sip messages from the SIP provider side.
>
>
>
> --
> Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
>
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Re: [SR-Users] Manipulating SDP IP for Inbound Calls

[Karsten Horsmann]

Hi Edward,


if your source freeswitch didn't use your settings then you should check
the sofia profile settings. Imho there are 4 ip settings for each profile
(just what I remember whiteout checking it right now) and then your
freeswitch will write the correct ips in the messages.

On the other hand if you want to go the Kamailio way you can use different
types of logic to identify the traffic.

Like "it's the source ip freeswitch then do that with sdp" and take a look
at routes the sdp goes. They are serval modules for manipulate the sdp
maybe some from the list point them.

Cheers
Karsten

Edward Romanenco  schrieb am Sa., 1. Aug. 2020,
10:25:

> Hey guys,
>
>
>
> I am working on a project involving Kamailio dockerezation, which is meant to 
> run alongside Freeswitch and RTPEngine containers, on the basis of a 
> Docker-Compose file which is launched on top of a CentOS 7.7 host system.
>
>
>
> Anyway, I would love to know if there is any way to manipulate / mask the IP 
> addresses that are being appended to a status 183 response for an incoming 
> invite.
>
>
>
> For some reason which I am trying to figure out in parallel, Freeswitch uses 
> the local network bridge subnet instead of the defined external RTP IPs, and 
> I was wondering - Can I manipulate them using Kamailio? I know that Mangler 
> module can do it for outbound calls, but can I do the same for inbound?
>
>
>
> v=0.
>
> o=FreeSWITCH 1595974788 1595974789 IN IP4 172.18.0.40.
>
> s=FreeSWITCH.
>
> c=IN IP4 172.18.0.40.
>
> t=0 0.
>
> m=audio 45878 RTP/AVP 8 101.
>
> a=rtpmap:8 PCMA/8000.
>
> a=rtpmap:101 telephone-event/8000.
>
> a=fmtp:101 0-16
>
>
>
> Edward
>
>
>
>
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Re: [SR-Users] [Help] Kamailio not receive bye requests

[Karsten Horsmann]

Hi,

I would start with tcpdump or ngrep for basic is ip a to ip b
communicating.
Because if the sip request is broken or not correct responded or whatever
sngrep didn't shows all on the wire in such cases.

Sngrep is great for "its working and I want to see dialogs" imho, not for
fundamental network problems.

Cheers
Karsten

Mark Boyce  schrieb am Fr., 31. Juli 2020, 09:45:

> Hi
>
> I’d start with sngrep so establish if it’s not receiving the BYE
> (client/network/nat/firewall/etc issue) or not kamailio is not seeing the
> BYE.
>
> Mark
>
> On 31 Jul 2020, at 03:33, Duy Phan  wrote:
>
> Hi guy!
>  I use ec2 ubuntu on aws. But kamailio not receive bye requests.
> But when I install Kmailio in vultr is OK.
> can you help me?
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> --
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Re: [SR-Users] how to achive destination number based routing before falling back to dispatcher-module

[Karsten Horsmann]

Hi Christoph,

the dialplan module could also taken for that purpose and it loads the data
on startup or command.

So place an route to check the number is matching in dialplan, do stuff
then or if not matching just return from your dialplan route and go further
to the dispatcher logic.

Cheers
Karsten

Christoph Russow  schrieb am Do., 23. Juli 2020, 15:02:

> Hi all,
>
> im fairly new to Kamailio and currently setting up our new sip-server
> systems so, hello everyone!
>
> right now i have a Kamailio SBC which routes all calls using the
> dispatcher module to our internally setup application servers (freeswitch).
>
> i just got the idea that we might want specific routing based on
> destination phone number in addition to simple loadbalancing.
>
> can anyone point me in the right direction where i should look for
> something i can place before the dispatcher routing to route calls based
> on some entries in a database like destination number XYZ goes to server
> A, number ABC goes to server Z and so on.
>
> Or is there a way to select the dispatcher group id from the database
> based on destination number?
>
> best regards
> Christoph
>
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Re: [SR-Users] Kamailio & Freeswitch as media and SBC

[Karsten Horsmann]

Hi,

I mean that module (correct the 5.2.x to your version, 5.3.x if 5.3)

https://kamailio.org/docs/modules/5.2.x/modules/xlog.html

https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables

With the cookbook you can log everything you want to watch in the logic of
kamailio.cfg

Like echo in shell scripts or more logger in scripts as an example to image
that more.

Cheers
Karsten

John Tuxies  schrieb am Fr., 17. Juli 2020, 09:33:

> Hi Karsten.
> Regarding Freeswitch, i have done the changes on the Freeswitch.
> Regarding the xlog you mention, i am not sure what to do.
>
> On Fri, Jul 17, 2020 at 9:49 AM Karsten Horsmann 
> wrote:
>
>> Hi John,
>>
>> On Kamailio side you can use xlog to print out log messages from config
>> to debug. That helps a lot to understand what is going on. You should
>> configure also syslog for Kamailio correctly.
>>
>> Also the howto described changes on freeswitch side whare you can use
>> fs_cli to trace the call if you enable the correct debug level.
>>
>> It seems for me that your call is not be answered by your freeswitch, but
>> that's guessing.
>>
>> Cheers
>> Karsten
>>
>> John Tuxies  schrieb am Fr., 17. Juli 2020, 07:41:
>>
>>> Thanks for the replies.
>>> By following the command kamailio -c -f kamailio i managed to correct
>>> all the mistakes and now i have Kamailio running with Siremis.
>>> Unfortunately i am not able to place any call anywhere. No error is
>>> reported, but the call hangs there. Even if i press end call, softphones
>>> get hanged for around 1 minute!
>>> i do not receive 200OK. Here is the print screen of the call.
>>> 192.168.1.7 is one of the softphones and 192.168.1.213 is the Kamailio with
>>> Freeswitch system.
>>> [image: image.png]
>>>
>>> I am attaching the kamailio.cfg
>>> Not able to make calls between extensions.
>>>
>>>
>>> On Thu, Jul 16, 2020 at 10:51 PM Karsten Horsmann 
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> Between the howto version and today actual version of Kamailio are many
>>>> changes that can generate errors (due for example rename or new module
>>>> params in kamailio). So copy paste an howto works not out of the box (if
>>>> it's not exactly what the author used).
>>>>
>>>> Start with the shipped default config, then try to adopt the logic step
>>>> by step with
>>>> kamailio -c -f /etc/kamailio/kamailio.cfg is a great place to check
>>>> Syntax fails.
>>>>
>>>> And which Kamailio version you use is also a good way to to describe
>>>> your issue.
>>>>
>>>> Cheers
>>>> Karsten
>>>>
>>>>
>>>> David Villasmil  schrieb am Do., 16.
>>>> Juli 2020, 15:33:
>>>>
>>>>> What we need are the errors :)
>>>>>
>>>>> On Thu, 16 Jul 2020 at 13:46, John Tuxies  wrote:
>>>>>
>>>>>> Trying to setup Kamailio with Freeswitch in a Debian machine as in
>>>>>> the
>>>>>> http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc,
>>>>>> but Kamailio fails to start and gives a ton of errors.
>>>>>> Before of the alterations Kamailio and Siremis start correctly and i
>>>>>> can easily create users through Siremis. After the changes to include
>>>>>> Freeswitch's blocks, it fails to start. All run in the same machine with 
>>>>>> IP
>>>>>> 192.168.1.213.
>>>>>> I am attaching kamailio.cfg (named as test1.txt) if someone could
>>>>>> spot the error, please. Also i am attaching the 
>>>>>> kamailio.cfg(Original1.txt)
>>>>>> when it was working fine with Siremis.
>>>>>>
>>>>>>
>>>>>>
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>>>>>>
>>>>> --
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.w...@gmail.com
>>>>> phone: +34669448337
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Re: [SR-Users] Kamailio & Freeswitch as media and SBC

[Karsten Horsmann]

Hi John,

On Kamailio side you can use xlog to print out log messages from config to
debug. That helps a lot to understand what is going on. You should
configure also syslog for Kamailio correctly.

Also the howto described changes on freeswitch side whare you can use
fs_cli to trace the call if you enable the correct debug level.

It seems for me that your call is not be answered by your freeswitch, but
that's guessing.

Cheers
Karsten

John Tuxies  schrieb am Fr., 17. Juli 2020, 07:41:

> Thanks for the replies.
> By following the command kamailio -c -f kamailio i managed to correct all
> the mistakes and now i have Kamailio running with Siremis. Unfortunately i
> am not able to place any call anywhere. No error is reported, but the call
> hangs there. Even if i press end call, softphones get hanged for around 1
> minute!
> i do not receive 200OK. Here is the print screen of the call. 192.168.1.7
> is one of the softphones and 192.168.1.213 is the Kamailio with Freeswitch
> system.
> [image: image.png]
>
> I am attaching the kamailio.cfg
> Not able to make calls between extensions.
>
>
> On Thu, Jul 16, 2020 at 10:51 PM Karsten Horsmann 
> wrote:
>
>> Hi,
>>
>> Between the howto version and today actual version of Kamailio are many
>> changes that can generate errors (due for example rename or new module
>> params in kamailio). So copy paste an howto works not out of the box (if
>> it's not exactly what the author used).
>>
>> Start with the shipped default config, then try to adopt the logic step
>> by step with
>> kamailio -c -f /etc/kamailio/kamailio.cfg is a great place to check
>> Syntax fails.
>>
>> And which Kamailio version you use is also a good way to to describe your
>> issue.
>>
>> Cheers
>> Karsten
>>
>>
>> David Villasmil  schrieb am Do., 16.
>> Juli 2020, 15:33:
>>
>>> What we need are the errors :)
>>>
>>> On Thu, 16 Jul 2020 at 13:46, John Tuxies  wrote:
>>>
>>>> Trying to setup Kamailio with Freeswitch in a Debian machine as in the
>>>> http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc,
>>>> but Kamailio fails to start and gives a ton of errors.
>>>> Before of the alterations Kamailio and Siremis start correctly and i
>>>> can easily create users through Siremis. After the changes to include
>>>> Freeswitch's blocks, it fails to start. All run in the same machine with IP
>>>> 192.168.1.213.
>>>> I am attaching kamailio.cfg (named as test1.txt) if someone could spot
>>>> the error, please. Also i am attaching the kamailio.cfg(Original1.txt) when
>>>> it was working fine with Siremis.
>>>>
>>>>
>>>>
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>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
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Re: [SR-Users] Kamailio & Freeswitch as media and SBC

[Karsten Horsmann]

Hi,

Between the howto version and today actual version of Kamailio are many
changes that can generate errors (due for example rename or new module
params in kamailio). So copy paste an howto works not out of the box (if
it's not exactly what the author used).

Start with the shipped default config, then try to adopt the logic step by
step with
kamailio -c -f /etc/kamailio/kamailio.cfg is a great place to check Syntax
fails.

And which Kamailio version you use is also a good way to to describe your
issue.

Cheers
Karsten


David Villasmil  schrieb am Do., 16. Juli
2020, 15:33:

> What we need are the errors :)
>
> On Thu, 16 Jul 2020 at 13:46, John Tuxies  wrote:
>
>> Trying to setup Kamailio with Freeswitch in a Debian machine as in the
>> http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc, but
>> Kamailio fails to start and gives a ton of errors.
>> Before of the alterations Kamailio and Siremis start correctly and i can
>> easily create users through Siremis. After the changes to include
>> Freeswitch's blocks, it fails to start. All run in the same machine with IP
>> 192.168.1.213.
>> I am attaching kamailio.cfg (named as test1.txt) if someone could spot
>> the error, please. Also i am attaching the kamailio.cfg(Original1.txt) when
>> it was working fine with Siremis.
>>
>>
>>
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>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
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Re: [SR-Users] xHTTP_PROM with older Kamailio

[Karsten Horsmann]

Hi Marat,

You could also reach your goal with an golang program that provides
prometheus for example


https://github.com/florentchauveau/kamailio_exporter

AFAIK there are also an implementation from pacom.

Cheers
Karsten

Marat Gareev  schrieb am Mo., 13. Juli 2020, 14:31:

> I found that the export module interface was changed (
> https://www.kamailio.org/w/2018/09/new-module-exports-interface/).
>
> I downgraded to old-style version this structure
>
>> struct module_exports exports = {
>> "xhttp_prom",
>> DEFAULT_DLFLAGS, /* dlopen flags */
>> cmds,
>> params,
>> 0,  /* exported RPC methods */
>> 0,  /* exported pseudo-variables */
>> 0,  /* response function */
>> mod_init,   /* module initialization function */
>> 0,  /* per child init function */
>> mod_destroy /* destroy function */
>> };
>
> to
>
>> struct module_exports exports = {
>> "xhttp_prom",
>> DEFAULT_DLFLAGS, /* dlopen flags */
>> cmds,
>> params,
>> 0,
>> 0,  /* exported MI functions */
>> 0,  /* exported pseudo-variables */
>> 0,  /* extra processes */
>> mod_init,   /* module initialization function */
>> 0,  /* per child init function */
>> mod_destroy, /* destroy function */
>> NULL  /* per child init function */
>> };
>
> The problem is that *mod_init* was zero, so it was not called.
> And now mod_init is called and I can use this module.
>
> Many thanks 
>
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Re: [SR-Users] Setting up Kamailio as Teams SBC

[Karsten Horsmann]

Hi,

Yeah they told you that. But I got it working with letsencrypt. It's an
easy and harmless try before you bumping your head on the desk in case of
tls debugging.


BTW I remember that you can sniff ms teams ssl/tls handshake with ssldump.

And if teams is happy with there option pings to you the direct routing
shows up as okay (AFAIK).

Роман С.  schrieb am Mi., 8. Juli 2020, 09:07:

> Hm, letsencrypt is out of supported CA list :/ I will give it a try and
> roll over to sipdump if it fails. Thank you guys.
>
> вт, 7 июл. 2020 г. в 21:19, Karsten Horsmann :
>
>> Hi there,
>>
>> my teams tls problems with wildcard certs are gone after I did the an
>> letsencrypt cert fqdn based cn.
>>
>> Did you tried that?
>>
>> Cheers
>> Karsten
>>
>> Роман С.  schrieb am Di., 7. Juli 2020, 11:46:
>>
>>> Hello.
>>> I'm trying to set up Kamailio as SBC for Teams using
>>> https://skalatan.de/en/blog/kamailio-sbc-teams.
>>> Setup is completely default (except things mentioned at article), but I
>>> use wildcard certificate for TLS. Well, I can't even pass dispatcher:
>>>
>>> kamcmd dispatcher.list | egrep "RI|FLAG"
>>> URI: sip:
>>> sip.pstnhub.microsoft.com;transport=tls
>>> FLAGS: IP
>>> PRIORITY: 3
>>> URI: sip:
>>> sip2.pstnhub.microsoft.com;transport=tls
>>> FLAGS: IP
>>> PRIORITY: 2
>>> URI: sip:
>>> sip3.pstnhub.microsoft.com;transport=tls
>>> FLAGS: IP
>>> PRIORITY: 1
>>>
>>> Where do I start to dig?
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>>>
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Re: [SR-Users] Setting up Kamailio as Teams SBC

[Karsten Horsmann]

Hi there,

my teams tls problems with wildcard certs are gone after I did the an
letsencrypt cert fqdn based cn.

Did you tried that?

Cheers
Karsten

Роман С.  schrieb am Di., 7. Juli 2020, 11:46:

> Hello.
> I'm trying to set up Kamailio as SBC for Teams using
> https://skalatan.de/en/blog/kamailio-sbc-teams.
> Setup is completely default (except things mentioned at article), but I
> use wildcard certificate for TLS. Well, I can't even pass dispatcher:
>
> kamcmd dispatcher.list | egrep "RI|FLAG"
> URI: sip:sip.pstnhub.microsoft.com
> ;transport=tls
> FLAGS: IP
> PRIORITY: 3
> URI: sip:
> sip2.pstnhub.microsoft.com;transport=tls
> FLAGS: IP
> PRIORITY: 2
> URI: sip:
> sip3.pstnhub.microsoft.com;transport=tls
> FLAGS: IP
> PRIORITY: 1
>
> Where do I start to dig?
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Re: [SR-Users] RTP voice failover

[Karsten Horsmann]

Hi,

The Github repo contains an spec file for building RPMS. Take a look in the
el folder.

https://github.com/sipwise/rtpengine/tree/master/el

Cheers
Karsten Horsmann

Joel Serrano  schrieb am Sa., 27. Juni 2020, 21:26:

> Not sure about RPM, but for sure you can find DEB packages in the SIPwise
> APT repo..
>
> On Sat, Jun 27, 2020 at 5:25 AM Evgeniy  wrote:
>
>> Thanks a lot !
>> I will try to implement it with Cloud Load Balance (Azure, AWS) and
>> rtpproxy + Redis cache.
>> BTW - can i install rtpengine with RPM or DEB packages ? Can someone give
>> me please a link to the repository ?
>> Many thanks
>>
>> On 26.06.20 18:48, Alex Balashov wrote:
>>
>> I would agree about the marginal utility.
>>
>> —
>> Sent from mobile, with due apologies for brevity and errors.
>>
>> On Jun 26, 2020, at 11:00 AM, Sergiu Pojoga 
>>  wrote:
>>
>> 
>> The most thorough presentation about the topic that I know of was 1&1's
>> where BGP is used.
>> The task of detecting such a failure within a couple of seconds so that
>> user's won't hang up intuitively, avoid the failse-positive failure
>> detection, makes the entire task pretty daunting and marginally useful,
>> IMO.
>>
>>
>> https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High-Quality-Telephony-Using-A-Fail-Safe-Media-Relay-Setup.pdf
>>
>>
>> On Fri, Jun 26, 2020 at 9:39 AM Evgeniy  wrote:
>>
>>> Is that possible to implement RTP voice fail-over?
>>>
>>> In case RTP node goes down - the call should not  be interrupted.
>>> This is kinda strange - but client want to have super redundant system.
>>> For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2
>>> nodes (yes - network traffic will increase)
>>> Or maybe is a way to recover UDP connections after a crash and replace
>>> node state.
>>> Maybe use a Kubernetes tools for that purpose ?
>>> Here what i found in mailing-list -
>>> https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html
>>>
>>> Many thanks for any hints to this topic.
>>>
>>>
>>>
>>>
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Re: [SR-Users] RTP voice failover

[Karsten Horsmann]

Hi Evgeniy,

you can for example save rtpengine sessions informations in redis and do an
active / passive setup for media nodes.

Rtpengine is capable to fetch up the session on startup from redis if you
point it to the same redis dB

Just an idea.

Cheers
Karsten

Evgeniy  schrieb am Fr., 26. Juni 2020, 15:39:

> Is that possible to implement RTP voice fail-over?
>
> In case RTP node goes down - the call should not  be interrupted.
> This is kinda strange - but client want to have super redundant system.
> For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2
> nodes (yes - network traffic will increase)
> Or maybe is a way to recover UDP connections after a crash and replace
> node state.
> Maybe use a Kubernetes tools for that purpose ?
> Here what i found in mailing-list -
> https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html
>
> Many thanks for any hints to this topic.
>
>
>
>
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Re: [SR-Users] Which mode to capture "written to network" traffic with homer and topos

[Karsten Horsmann]

Hi Daniel,

in that case it works (also with topos and then sipcapture). With your
suggestion.

In my other case (as webrtc and tls registrar) I needed the function to see
Kamailio self created traffic and so, imho.

Thanks for the hints.

Cheers
Karsten

Daniel-Constantin Mierla  schrieb am Do., 11. Juni 2020,
13:00:

> Hello,
>
> to get what is received from/sent to network, you have to use:
>
> modparam("siptrace", "trace_mode", 1)
>
> And no longer use siptrace flag or functions in configuration file. In the
> configuration file the sip message is the one after topos handled the
> incoming message and before topos handles the outgoing message.
>
> Cheers,
> Daniel
> On 11.06.20 11:35, Karsten Horsmann wrote:
>
> Hello Daniel,
>
> thanks for the fast reply.
>
> i changed the loadmodule order (siptrace.so before topos.so) and its the
> same behavior. With  sip_trace_mode("t");
>
> For better understanding I attached the loading-part of my kamailio.cfg
> with siptrace and topos.
> The howl request route is not included, but didnt matter imho.
>
> Cheers
> Karsten
>
> [...]
> # vim: set tabstop=4 softtabstop=4 shiftwidth=4 noexpandtab:
>
> Am Do., 11. Juni 2020 um 10:25 Uhr schrieb Daniel-Constantin Mierla <
> mico...@gmail.com>:
>
>> Hello,
>>
>> try to load siptrace module before the topos module, iirc they use the
>> same event hooks. If doesn't work, then I will check the code.
>>
>> Cheers,
>> Daniel
>> On 11.06.20 10:05, Karsten Horsmann wrote:
>>
>> Hi Mailinglist,
>>
>> some of my kamailio-version is 5.3.4 with topos.
>>
>> During the last discussion about sip_trace_mode("t") i found new
>> behaivor. With this tracingmode i see the "internal" before topos writing.
>>
>> Here my questions:
>>
>> Can i change it to "whats on the wire" (topos rewritten mode)? I
>> didnt checked all modes.
>> And if so, can i switch that on-the-fly with $var for example? For topos
>> debugging its fine with sip_trace_mode("t") - not for "hey carrier, you
>> made some mistakes".
>>
>>
>> #!ifdef WITH_HOMER
>> #Siptrace
>> modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK")
>> # local interface from where to send the duplicated traffic
>> modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
>> modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate
>> mode
>> modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture
>> agent ID will be included in the HEP header
>> modparam("siptrace", "trace_to_database", 0) # trace to database
>> modparam("siptrace", "trace_on", 1) # enable/disable trace
>> modparam("siptrace", "trace_mode", 0) # default 0, if 1 then you dont
>> need call siptrace flag or sip_trace()
>> modparam("siptrace", "hep_capture_id", 8) # capture agent id Limitation:
>> 32-bit for HEPv3.
>> #!endif
>>
>>
>> #!ifdef WITH_HOMER
>> # Set the tracing mode: message, transaction or dialog. Need 5.3.x
>> sip_trace_mode("t");
>> #!endif
>>
>> *** captured on the topos kamailio:
>>
>> 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060
>> INVITE sip:+49@172.24.52.3:5060 SIP/2.0
>> Record-Route:
>> 
>> Record-Route:
>> 
>> Via: SIP/2.0/UDP
>> 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74
>> Via: SIP/2.0/TCP
>> AA.AAA.AAA.AAA:5060;rport=5060;branch=z9hG4bKeca7a7f789c9f647560353e3cf1cd592.ba7930dc
>> Max-Forwards: 52
>> To: 
>> From: ;tag=1ea69f36
>> Call-ID: 3af87db089119...@aa.aaa.aaa.aaa
>> Contact: 
>> Supported: histinfo,replaces
>> CSeq: 647169 INVITE
>> Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE
>> P-Asserted-Identity: 
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 298
>> X-Group-SBC: 2006
>> X-Routing-SBC: un2tr-cgn-trunk
>>
>> ---
>> *** captured on the kamailio behind 172.24.52.3
>>
>> 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060
>> INVITE sip:+49@172.24.52.3:5060 SIP/2.0
>> Via: SIP/2.0/UDP
>> 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74
>> Max-Forwards: 52
>> To: 
>&

Re: [SR-Users] Which mode to capture "written to network" traffic with homer and topos

[Karsten Horsmann]

, "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_from", "")
modparam("dispatcher", "ds_ping_interval", 30)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_probing_threshold", 2)
modparam("dispatcher", "setid_pvname", "$var(setid)")
modparam("dispatcher", "attrs_pvname", "$var(attrs)")
modparam("dispatcher", "ds_ping_reply_codes", "class=2")
modparam("dispatcher", "ds_ping_latency_stats", 1)
#!ifdef WITH_500_ERROR
modparam("dispatcher", "flags", 2)
#!endif

#TOPOS
#modparam("ndb_redis", "server",
"name=srv8;addr=172.20.160.155;port=6379;db=8")
modparam("ndb_redis", "server",
"name=CFG_REDIS_NAME;addr=CFG_REDIS_ADDR;port=6379;db=CFG_REDIS_DB")
modparam("ndb_redis", "connect_timeout", 1500) # default 1000ms/1sec
modparam("ndb_redis", "cmd_timeout", 1500) # default 1000ms/1sec


 $mb message debugging (see event-routes)
#!ifdef WITH_MB_DEBUG
modparam("corex", "network_io_intercept", 1)
modparam("corex", "min_msg_len", 32)
#!endif

#!ifdef WITH_TOPOS
modparam("topos", "storage", "redis")
modparam("topos_redis", "serverid", "CFG_REDIS_NAME")
modparam("topos", "branch_expire", 14400)
modparam("topos", "dialog_expire", 14400)
modparam("topos", "clean_interval", 60)
#!endif

#dont' restore
##modparam("uac","restore_mode","none") # SBC-OS
### modparam("uac","restore_dlg", 1) # needs dialog.so
#modparam("uac","restore_mode","manual")
#modparam("uac","restore_from_avp","$avp(s:original_uri_from)") # needed if
you dont activate dialog.so and set restore_mode auto!
#modparam("uac","restore_to_avp","$avp(s:original_uri_to)")


## UAC REGISTER
#!ifdef WITH_UAC_REGISTER
modparam("uac", "reg_contact_addr", "CFG_PROD_IP")
modparam("uac", "reg_timer_interval", 10)
modparam("uac", "reg_retry_interval", 10)
modparam("uac", "reg_db_url", "DBURL")
modparam("uac", "restore_mode", "auto") # auto is default
modparam("uac", "auth_username_avp", "$avp(AVP_AUTH_USERNAME)")
modparam("uac", "auth_password_avp", "$avp(AVP_AUTH_PASSWORD)")
modparam("uac", "auth_realm_avp", "$avp(AVP_AUTH_REALM)")
modparam("uac", "reg_keep_callid", 1) # set to 1 - RFC3261 conform
modparam("uac", "default_socket", "CFG_PROD_IP:5060") # tcp_reuse_port=yes
MUSS gesetzt werden.
#!endif


#!ifdef WITH_HOMER
#Siptrace
modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK")
# local interface from where to send the duplicated traffic
modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate
mode
modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture
agent ID will be included in the HEP header
modparam("siptrace", "trace_to_database", 0) # trace to database
modparam("siptrace", "trace_on", 1) # enable/disable trace
modparam("siptrace", "trace_mode", 0) # default 0, if 1 then you dont need
call siptrace flag or sip_trace()
modparam("siptrace", "hep_capture_id", 8) # capture agent id Limitation:
32-bit for HEPv3.
#!endif

#!ifdef WITH_MYSQL
modparam("sqlops","sqlcon","db=>DBURL")
#!endif

### Routing Logic 

include_file "kamailio-hmr.cfg"

remove.prack = 1 desc "1 true - 0 false"
mylog.logint = 0 desc "on the fly logging"

# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {

#!ifdef WITH_HOMER
# Set the tracing mode: message, transaction or dialog. Need 5.3.x
sip_trace_mode("t");
#!endif
  ...
}
# vim: set tabstop=4 softtabstop=4 shiftwidth=4 noexpandtab:

Am Do., 11. Juni 2020 um 10:25 Uhr schrieb Daniel-Constantin Mierla <
mico...@gmail.com>:

> Hello,
>
> try to load siptrace module before the topos module, iirc they use the
> same event hooks. If doesn't work, then I will check the code.
>
> Cheers,
> Daniel
> On 11.06.20 10:05, Karsten

[SR-Users] Which mode to capture "written to network" traffic with homer and topos

[Karsten Horsmann]

Hi Mailinglist,

some of my kamailio-version is 5.3.4 with topos.

During the last discussion about sip_trace_mode("t") i found new behaivor.
With this tracingmode i see the "internal" before topos writing.

Here my questions:

Can i change it to "whats on the wire" (topos rewritten mode)? I
didnt checked all modes.
And if so, can i switch that on-the-fly with $var for example? For topos
debugging its fine with sip_trace_mode("t") - not for "hey carrier, you
made some mistakes".


#!ifdef WITH_HOMER
#Siptrace
modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK")
# local interface from where to send the duplicated traffic
modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE")
modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate
mode
modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture
agent ID will be included in the HEP header
modparam("siptrace", "trace_to_database", 0) # trace to database
modparam("siptrace", "trace_on", 1) # enable/disable trace
modparam("siptrace", "trace_mode", 0) # default 0, if 1 then you dont need
call siptrace flag or sip_trace()
modparam("siptrace", "hep_capture_id", 8) # capture agent id Limitation:
32-bit for HEPv3.
#!endif


#!ifdef WITH_HOMER
# Set the tracing mode: message, transaction or dialog. Need 5.3.x
sip_trace_mode("t");
#!endif

*** captured on the topos kamailio:

2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060
INVITE sip:+49@172.24.52.3:5060 SIP/2.0
Record-Route:

Record-Route:

Via: SIP/2.0/UDP
172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74
Via: SIP/2.0/TCP
AA.AAA.AAA.AAA:5060;rport=5060;branch=z9hG4bKeca7a7f789c9f647560353e3cf1cd592.ba7930dc
Max-Forwards: 52
To: 
From: ;tag=1ea69f36
Call-ID: 3af87db089119...@aa.aaa.aaa.aaa
Contact: 
Supported: histinfo,replaces
CSeq: 647169 INVITE
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE
P-Asserted-Identity: 
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 298
X-Group-SBC: 2006
X-Routing-SBC: un2tr-cgn-trunk

---
*** captured on the kamailio behind 172.24.52.3

2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060
INVITE sip:+49@172.24.52.3:5060 SIP/2.0
Via: SIP/2.0/UDP
172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74
Max-Forwards: 52
To: 
From: ;tag=1ea69f36
Call-ID: 3af87db089119...@aa.aaa.aaa.aaa
Supported: histinfo,replaces
CSeq: 647169 INVITE
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE
P-Asserted-Identity: 
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 298
X-Group-SBC: 2006
X-Routing-SBC: un2tr-cgn-trunk
Contact: 



-- 
Cheers
*Karsten Horsmann*
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