[SR-Users] Re: ds_select from xhttp:request

[David Villasmil via sr-users]

This may or not work, but maybe call it from a different route? I.e.: call
another route and execute it there?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com



On Wed, 24 Jul 2024 at 08:08, Alex Balashov via sr-users <
sr-users@lists.kamailio.org> wrote:

> It's not a great fit. I would recommend pulling that info from the RPC
> interface instead, also using HTTP.
>
> > On Jul 24, 2024, at 2:46 AM, Christian Berger via sr-users <
> sr-users@lists.kamailio.org> wrote:
> >
> > Hi,
> >
> > we want to execute ds_select from an xhttp:request event route. The idea
> is to be able to get all active targets of a dispatcher set so our
> monitoring can check that via http which would be convenient for us.
> >
> > So are there any reasons why ds_select wouldn't work in an xhttp:request
> even route?
> >
> > Regards
> >  Christian Berger
> >
> > --
> >
> > Christian Berger - ber...@sipgate.de
> > Telefon: +49 (0)211-63 55 55-0
> > Telefax: +49 (0)211-63 55 55-22
> >
> > sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
> > HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
> > Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
> >
> > www.sipgate.de - www.sipgate.co.uk
> > __
> > Kamailio - Users Mailing List - Non Commercial Discussions
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> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
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[SR-Users] Re: SRV in advertise?

[David Villasmil via sr-users]

hello all,

using only the srv did not work either:

listen=tls:1.2.3.4:443 advertise mydomain.com

logs:

 0(139992) ERROR:  [core/socket_info.c:350]: new_sock_info(): unable
to resolve advertised name mydomain.com
 0(139992) ERROR:  [core/socket_info.c:357]: new_sock_info(): could
not allocate private memory from pkg pool
 0(139992) ERROR:  [core/socket_info.c:784]: new_sock2list():
new_sock_info failed
 0(139992) ERROR:  [core/socket_info.c:876]:
add_listen_advertise_iface(): new_sock2list failed
 0(139992) CRITICAL:  [core/cfg.y:1474]: yyparse(): cfg. parser:
failed to add listen address


Thanks & Regards,

David Villasmil
email: david.villasmil.w...@gmail.com



On Mon, Jul 1, 2024 at 11:38 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Thanks all, I’m using 5.3.9, I’ll also need to upgrade
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
>
>
>
> On Mon, 1 Jul 2024 at 20:47, Daniel-Constantin Mierla via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> Hello,
>>
>> indeed, as mentioned, the presence of a port in a SIP address makes it to
>> skip SRV lookup.
>>
>> Anyhow, I am jumping in to see what is the Kamailio version used in this
>> scenario? I don't recall exactly, but a while ago there was a related
>> commit, at least it looked like when I did some backports today for next
>> planned release in 5.6.x series.
>>
>> Cheers,
>> Daniel
>> On 01.07.24 19:16, Ben Kaufman via sr-users wrote:
>>
>> While I don’t know if it will or won’t cause the failure, you shouldn’t
>> have a port number in something you’re intending to use as a SRV record.
>>
>>
>>
>> - Kaufman
>>
>>
>>
>>
>>
>> *From:* David Villasmil 
>> 
>> *Sent:* Monday, July 1, 2024 9:19 AM
>> *To:* Ben Kaufman  
>> *Cc:* Kamailio (SER) - Users Mailing List 
>> 
>> *Subject:* Re: [SR-Users] SRV in advertise?
>>
>>
>>
>> *CAUTION:* This email originated from outside the organization. *Do not
>> click links or open attachments* unless you recognize the sender and
>> know the content is safe.
>>
>>
>>
>> Hey all,
>>
>>
>>
>> Joel, it’s failing saying the destination is unresolvable and Ben, I’m
>> setting it like
>>
>>
>>
>> Listen=tls:1.2.3.4:5061 advertise mydomain.com:5061
>>
>>
>>
>> My domain.com srv is like
>>
>>
>>
>> _sips._tcp.domain.com resolving to sip.domain.com (among others like
>> udp, etc)
>>
>>
>>
>> I would expect Kamailio to resolve its own srv, or have an option NOT to
>> attempt to resolve and just the set host in the record-route, but maybe
>> that’s just not how it works. Yes, setting sip.domain.com:5061 works but
>> the the record-route would have that (sip.domain.com) exact same host
>> and the client might disconnect and reconnect to that dns instead of the
>> actual SRV.
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>>
>>
>>
>>
>>
>>
>> On Mon, 1 Jul 2024 at 15:46, Ben Kaufman  wrote:
>>
>> Are you using a hostname that should resolve as srv , for example,
>> sip.domain.com, or are you using the underscore, scheme, underscore
>> transport protocol, hostname type like _sip._udp.domain.com ?
>>
>>
>>
>> Regards,
>>
>> Kaufman
>>
>>
>>
>> *From:* David Villasmil via sr-users 
>> *Sent:* Sunday, June 30, 2024 6:02 PM
>> *To:* Kamailio (SER) - Users Mailing List 
>> *Cc:* David Villasmil 
>> *Subject:* [SR-Users] SRV in advertise?
>>
>>
>>
>> *CAUTION:* This email originated from outside the organization. *Do not
>> click links or open attachments* unless you recognize the sender and
>> know the content is safe.
>>
>>
>>
>> Hello guys,
>>
>>
>>
>> I'm trying to use an SRV as an advertised address but kamailio fails to
>> start. Is this not possible?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>>
>>
>>
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
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>> Important: keep the mailing list in the recipients, do not reply only to the 
>> sender!
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>>
>> --
>> Daniel-Constantin Mierla (@ asipto.com)twitter.com/miconda -- 
>> linkedin.com/in/miconda
>> Kamailio Consultancy, Training and Development Services -- asipto.com
>>
>> __
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>
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[SR-Users] Re: SRV in advertise?

[David Villasmil via sr-users]

Thanks all, I’m using 5.3.9, I’ll also need to upgrade

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com



On Mon, 1 Jul 2024 at 20:47, Daniel-Constantin Mierla via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hello,
>
> indeed, as mentioned, the presence of a port in a SIP address makes it to
> skip SRV lookup.
>
> Anyhow, I am jumping in to see what is the Kamailio version used in this
> scenario? I don't recall exactly, but a while ago there was a related
> commit, at least it looked like when I did some backports today for next
> planned release in 5.6.x series.
>
> Cheers,
> Daniel
> On 01.07.24 19:16, Ben Kaufman via sr-users wrote:
>
> While I don’t know if it will or won’t cause the failure, you shouldn’t
> have a port number in something you’re intending to use as a SRV record.
>
>
>
> - Kaufman
>
>
>
>
>
> *From:* David Villasmil 
> 
> *Sent:* Monday, July 1, 2024 9:19 AM
> *To:* Ben Kaufman  
> *Cc:* Kamailio (SER) - Users Mailing List 
> 
> *Subject:* Re: [SR-Users] SRV in advertise?
>
>
>
> *CAUTION:* This email originated from outside the organization. *Do not
> click links or open attachments* unless you recognize the sender and know
> the content is safe.
>
>
>
> Hey all,
>
>
>
> Joel, it’s failing saying the destination is unresolvable and Ben, I’m
> setting it like
>
>
>
> Listen=tls:1.2.3.4:5061 advertise mydomain.com:5061
>
>
>
> My domain.com srv is like
>
>
>
> _sips._tcp.domain.com resolving to sip.domain.com (among others like udp,
> etc)
>
>
>
> I would expect Kamailio to resolve its own srv, or have an option NOT to
> attempt to resolve and just the set host in the record-route, but maybe
> that’s just not how it works. Yes, setting sip.domain.com:5061 works but
> the the record-route would have that (sip.domain.com) exact same host and
> the client might disconnect and reconnect to that dns instead of the actual
> SRV.
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
>
>
>
>
>
>
> On Mon, 1 Jul 2024 at 15:46, Ben Kaufman  wrote:
>
> Are you using a hostname that should resolve as srv , for example,
> sip.domain.com, or are you using the underscore, scheme, underscore
> transport protocol, hostname type like _sip._udp.domain.com ?
>
>
>
> Regards,
>
> Kaufman
>
>
>
> *From:* David Villasmil via sr-users 
> *Sent:* Sunday, June 30, 2024 6:02 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Cc:* David Villasmil 
> *Subject:* [SR-Users] SRV in advertise?
>
>
>
> *CAUTION:* This email originated from outside the organization. *Do not
> click links or open attachments* unless you recognize the sender and know
> the content is safe.
>
>
>
> Hello guys,
>
>
>
> I'm trying to use an SRV as an advertised address but kamailio fails to
> start. Is this not possible?
>
>
>
> Thanks,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
>
>
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
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> Important: keep the mailing list in the recipients, do not reply only to the 
> sender!
> Edit mailing list options or unsubscribe:
>
> --
> Daniel-Constantin Mierla (@ asipto.com)twitter.com/miconda -- 
> linkedin.com/in/miconda
> Kamailio Consultancy, Training and Development Services -- asipto.com
>
> __
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[SR-Users] Re: SRV in advertise?

[David Villasmil via sr-users]

Hey all,

Joel, it’s failing saying the destination is unresolvable and Ben, I’m
setting it like

Listen=tls:1.2.3.4:5061 advertise mydomain.com:5061

My domain.com srv is like

_sips._tcp.domain.com resolving to sip.domain.com (among others like udp,
etc)

I would expect Kamailio to resolve its own srv, or have an option NOT to
attempt to resolve and just the set host in the record-route, but maybe
that’s just not how it works. Yes, setting sip.domain.com:5061 works but
the the record-route would have that (sip.domain.com) exact same host and
the client might disconnect and reconnect to that dns instead of the actual
SRV.

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com



On Mon, 1 Jul 2024 at 15:46, Ben Kaufman  wrote:

> Are you using a hostname that should resolve as srv , for example,
> sip.domain.com, or are you using the underscore, scheme, underscore
> transport protocol, hostname type like _sip._udp.domain.com ?
>
>
>
> Regards,
>
> Kaufman
>
>
>
> *From:* David Villasmil via sr-users 
> *Sent:* Sunday, June 30, 2024 6:02 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Cc:* David Villasmil 
> *Subject:* [SR-Users] SRV in advertise?
>
>
>
> *CAUTION:* This email originated from outside the organization. *Do not
> click links or open attachments* unless you recognize the sender and know
> the content is safe.
>
>
>
> Hello guys,
>
>
>
> I'm trying to use an SRV as an advertised address but kamailio fails to
> start. Is this not possible?
>
>
>
> Thanks,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
>
>
__
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[SR-Users] SRV in advertise?

[David Villasmil via sr-users]

Hello guys,

I'm trying to use an SRV as an advertised address but kamailio fails to
start. Is this not possible?

Thanks,

David Villasmil
email: david.villasmil.w...@gmail.com
__
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[SR-Users] Re: replies using the wrong protocol

[David Villasmil via sr-users]

good that you resolved it.
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com



On Thu, Jun 27, 2024 at 2:32 PM Henning Westerholt via sr-users <
sr-users@lists.kamailio.org> wrote:

> Thanks for reporting back.
>
> Henning
>
> -Original Message-
> From: smartin114--- via sr-users 
> Sent: Tuesday, June 25, 2024 9:34 PM
> To: sr-users@lists.kamailio.org
> Cc: smartin...@me.com
> Subject: [SR-Users] Re: replies using the wrong protocol
>
> In the end it was a very straight forward change.  If the call comes in on
> port 5061 set a TLS flag, then in the relay, if the flag is set call
> t_relay_tls vs t_relay
> __
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[SR-Users] Re: SecSIPID Assistance

[David Villasmil via sr-users]

this is what i do (i have a redirect server receive the INVITEs to be
signed, I add the header and then do 302, the initiating server then add it
to the INVITE and sends the invite out:

 if ($rm=="INVITE") {
$var(rc) = secsipid_add_identity("$(var(from){s.numeric})",
"$(var(to){s.numeric})", "A", "", "
https://pki.domain.com/stir-shaken-cert.pem;,
"/etc/kamailio/ec256-private.pem");

if ( $var(rc) > 0 ) {
msg_apply_changes();
} else {
update_stat("stirshaken_create_identity_failed","+1");
send_reply("503", "Service Unavailable - can not create Identity header");
exit;
}

append_to_reply("Identity: $hdr(Identity)\r\n");
}
sl_send_reply("302", "Redirect");
exit;


hope that helps

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com



On Thu, Jun 20, 2024 at 11:14 PM Blake Ivey via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hmm you are correct. I took it out and it started fine. So what exactly
> would I need for our outbound stirshaken?
>
> Just secsipid_add_identity?
>
> I guess I've been looking at this for too long today. Just lines and lines
> after a while.
>
> On Thu, Jun 20, 2024, 4:47 PM Ben Kaufman  wrote:
>
>> Except for `expire` and `timeout`, those parameters don’t exist for
>> secsip id- at least according to the module documentation:
>> https://kamailio.org/docs/modules/stable/modules/secsipid
>>
>>
>>
>> Regards,
>>
>> Kaufman
>>
>>
>>
>> *From:* Blake Ivey 
>> *Sent:* Thursday, June 20, 2024 3:39 PM
>> *To:* Ben Kaufman 
>> *Cc:* sr-users@lists.kamailio.org
>> *Subject:* Re: [SR-Users] SecSIPID Assistance
>>
>>
>>
>> *CAUTION:* This email originated from outside the organization. *Do not
>> click links or open attachments* unless you recognize the sender and
>> know the content is safe.
>>
>>
>>
>> Sorry for the formatting:
>>
>> ERROR:  [core/modparam.c:185]: set_mod_param_regex(): parameter
>>  of type <1:string> not found in module 
>> kamailio: CRITICAL:  [core/cfg.y:4011]: yyerror_at(): parse error
>> in config file /etc/kamailio/kamailio.cfg, line 71, column 73: Can't set
>> module parameter
>> kamailio: CRITICAL:  [core/cfg.y:4011]: yyerror_at(): parse error
>> in config file /etc/kamailio/kamailio.cfg, line 71, column 70: Can't set
>> module parameter
>> kamailio: ERROR:  [core/modparam.c:185]: set_mod_param_regex():
>> parameter  of type <1:string> not found in module 
>>
>>
>>
>> On Thu, Jun 20, 2024, 4:31 PM Ben Kaufman  wrote:
>>
>> What is the error you’re getting?
>>
>>
>>
>> Regards,
>>
>> Kaufman
>>
>>
>>
>>
>>
>> *From:* Blake Ivey via sr-users 
>> *Sent:* Thursday, June 20, 2024 3:14 PM
>> *To:* Kamailio (SER) - Users Mailing List 
>> *Cc:* Blake Ivey 
>> *Subject:* [SR-Users] SecSIPID Assistance
>>
>>
>>
>> *CAUTION:* This email originated from outside the organization. *Do not
>> click links or open attachments* unless you recognize the sender and
>> know the content is safe.
>>
>>
>>
>> Hi everyone. Wanting to see if someone could point me in the right
>> direction. Still very knew to Kamailio but I am beginning to understand it
>> better. I'm making an outbound proxy and have everything working well
>> besides stir/shaken. I'm looking at the module page and have went back and
>> forth with chatGPT and can't seem to figure this part out. I keep getting
>> errors on the modparam lines.
>>
>>
>>
>> Obviously this is a self signed cert because I'm just testing. I am able
>> to reach and download the cert from the Web server.
>>
>>
>>
>> Thank you for any assistance.
>>
>>
>>
>> # SECSIPID for Stir/Shaken
>>
>> modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key")
>>
>> modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt")
>>
>> modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt")
>>
>> modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2)
>>
>>
>>
>> route[STIRSHAKEN] {
>>
>> if (is_method("INVITE")) {
>>
>> if (!secsipid_add_identity("$fU", "$rU", "A", "", "
>> http://myIPaddress.com/stir_shaken_cert.crt
>> ",
>> "/etc/kamailio/secsipid/private.key")) {
>>
>> xlog("L_ERR", "Failed to sign call with ID: $ci - From:
>> $fU\n");
>>
>> send_reply("500", "Internal Server Error");
>>
>> exit;
>>
>> } else {
>>
>> xlog("L_INFO", "Successfully signed call with ID: $ci - From:
>> $fU\n");
>>
>> }
>>
>> }
>>
>>
>>
>> # Relay the call after signing
>>
>> route(RELAY);
>>
>> }
>>
>>
>>
>> __
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[SR-Users] Re: replies using the wrong protocol

[David Villasmil via sr-users]

is there a UDP socket a well? are you translating TLS<->UDP?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, Jun 20, 2024 at 10:48 PM smartin114--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> Yes, there is a tls socket defined.  I assume since the invite goes out
> TLS and all the cert stuff works, there is a tls socket as well.
> __
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[SR-Users] Re: replies using the wrong protocol

[David Villasmil via sr-users]

check
https://github.com/davidcsi/kamailio-private-public/blob/master/kamailio-tls-to-udp-with-sips-scheme.cfg
that might give you some ideas

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, Jun 20, 2024 at 11:23 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> is there a UDP socket a well? are you translating TLS<->UDP?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Thu, Jun 20, 2024 at 10:48 PM smartin114--- via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> Yes, there is a tls socket defined.  I assume since the invite goes out
>> TLS and all the cert stuff works, there is a tls socket as well.
>> __
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[SR-Users] Re: replies using the wrong protocol

[David Villasmil via sr-users]

you'd need to mark the reply and before forwarding it, force_socket it to
your tls socket
You do have a TLS socket, right?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, Jun 20, 2024 at 7:31 PM smartin114--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> and no, not adding XML to the SDP.
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[SR-Users] Re: replies using the wrong protocol

[David Villasmil via sr-users]

You trying to add the XML to the sdp?
Are you manipulating something somewhere? It is very strange indeed... have
you tried force_socket?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, Jun 20, 2024 at 7:06 PM smartin114--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> it is already in the thread.
>
> 200 OK from carrier
> SIP/2.0 200 OK
> Via:  SIP/2.0/TLS
> KAM_PUB_IP:5061;branch=z9hG4bK18ff.3d856b8b0b007414ab2dec09cbabd574.0;i=a2
> Via:  SIP/2.0/TLS
> FS_PUB_IP:5061;received=FS_PUB_IP;rport=56403;branch=z9hG4bKKaee61yyZ98De
> From:  "+14388006102" ;tag=XjQ5g4Ze5UaZp
> To:  ;tag=gK04d33797
> Call-ID: b99c2b65-a827-123d-1984-4201c0a80193
> CSeq:  84807919 INVITE
> Record-Route:
>
> 
> Accept:  application/sdp, application/isup, application/dtmf,
> application/dtmf-relay,
> multipart/mixed
> Contact:  
> Allow:
>
> INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
> Require:  timer
> Supported:  timer
> Session-Expires:  1800;refresher=uac
> Content-Length:324
> Content-Disposition:  session; handling=required
> Content-Type:  application/sdp
>
> v=0
> o=Sonus_UAC 913845 351585 IN IP4 CARRIER_PUB_IP
> s=SIP Media Capabilities
> c=IN IP4 CARRIER_PUB_IP
> t=0 0
> m=audio 33168 RTP/SAVP 0 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:LfSgFSVqhXNWMSziOtwpEeYmNu0/kGiyuMVS8VXy
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=ptime:20
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[SR-Users] Re: replies using the wrong protocol

[David Villasmil via sr-users]

can you show the 200 OK?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, Jun 20, 2024 at 5:43 PM smartin114--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> I was trying to add an attachment of the call flow, but I don't see that I
> can?
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[SR-Users] Re: Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu

[David Villasmil via sr-users]

fs docker:

https://github.com/davidcsi/docker-freeswitch

that's what i always use. Performance depends upon the underlying hardware.
I've seen 5000+ channels _handling rtp_.

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Fri, Jun 7, 2024 at 5:08 PM Fred Posner via sr-users <
sr-users@lists.kamailio.org> wrote:

>
> > On Jun 7, 2024, at 10:15 AM, Benoît Panizzon via sr-users <
> sr-users@lists.kamailio.org> wrote:
> >
> > = FreeSwitch =
> >
> > Got stuck trying to solve the dependencies to get it to run/build on
> > ubuntu jammy.
> > Has anyone experience in running it in a docker container? Will it
> > perform?
> >
>
> Have you tried from their repo?
>
>
> > = SEMS =
> >
> > Does not seem to be actively developed in the last, couple of years.
> > Looks quite bloated to me for what I need.
>
>
> SEMS is still updated— as recently even as last month.
>
> * https://github.com/sems-server/sems
>
>
>
>
> Regards,
>
> Fred Posner
> p: +1 (352) 664-3733
> https://fred.tel
>
>
>
>
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[SR-Users] Re: Assistance Needed: Converting 11-Digit US Numbers to E.164 in Kamailio

[David Villasmil via sr-users]

Hello guys,

My advice, always use e.164 with "+", your life will be easier if you
normalise to something.


Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Sat, Jun 1, 2024 at 6:10 AM Pavan Kumar via sr-users <
sr-users@lists.kamailio.org> wrote:

> Thank you, everyone, for your responses.
>
> Benoît, thank you for providing the sample code. My concern is that the
> code may become complex as it needs to be both transaction and dialog
> aware. For example, it needs to be transaction-aware when sending a 100 or
> 200 response back and conversion from E.164 to 11 digits should happen. It
> needs to be dialog-aware when FreeSWITCH initiates a BYE request so that
> the conversion to 11 digits occurs. Additionally, we need to handle cases
> where FreeSWITCH (on behalf of the user) initiates a call by sending an
> INVITE. In these instances, I need to identify if the request is going to a
> trunk that doesn't accept the "+" prefix and strip it accordingly,
> potentially using a database to assist in this identification.
>
> Ben, thank you for providing more information on the E.164 format and
> clarifying that the "+" prefix is not mandatory in some cases. Yes, when I
> referred to an 11-digit US number, I meant the NANP without the "+" prefix.
> Replacing "+1" with "1" is sufficient for my needs, and there is no need to
> identify the country.
>
> Thanks & Regards,
> Pavan Kumar
>
> On Fri, May 31, 2024 at 8:05 PM Ben Kaufman  wrote:
>
>> > the phonenum module is pretty decent for helping with NANP numbers and
>>   deciphering if one is let’s say US vs Canada vs Jamaica, etc.
>>
>> Absolutely. And if his use case is to actually get the political country,
>> it's a great choice. With that said, I'm guessing that in his case the
>> vendor wants the + removed for all NANP numbers, so my point was to only
>> use libphonenumber for this if was critical to differentiate the USA
>> from the rest of the NANP.
>>
>>
>>
>> -Original Message-
>> From: Fred Posner 
>> Sent: Friday, May 31, 2024 8:35 AM
>> To: Kamailio (SER) - Users Mailing List 
>> Cc: Pavan Kumar ; Ben Kaufman > >
>> Subject: Re: [SR-Users] Assistance Needed: Converting 11-Digit US Numbers
>> to E.164 in Kamailio
>>
>> CAUTION: This email originated from outside the organization. Do not
>> click links or open attachments unless you recognize the sender and know
>> the content is safe.
>>
>>
>> > On May 31, 2024, at 9:20 AM, Ben Kaufman via sr-users <
>> sr-users@lists.kamailio.org> wrote:
>> >
>> > Answered in order of easiest answer to most complex:
>> >
>> >> Does it start with a 0?
>> >  No.  I've never encountered any US number requiring a prefix of "0"
>> > nor a US number being  written that way.
>> >
>> >
>> >> US numbers are fixed lenght 11 digits?
>> >
>> >  The USA is in the North American Numbering Plan (NANP).  This
>> > includes USA, Canada, much of the  Caribbean (Cuba, Jamaica, Barbados
>> > for example), some US territories in the Pacific (Guam,  American
>> > Samoa).  The NANP's "Country Code" is 1.  In the USA and Canada (I
>> > think the entirety  of the NANP) this is followed with a 3 digit area
>> code (National Destination Number) followed  by a 7 digit subscriber number.
>> >
>> >
>> >> What is the usual representation of a 'local' US number?
>> >  There is no consistent answer.  Consider that the oldest phone
>> > network is in the USA, and the  unplanned initial growth and many
>> > historic regional governing bodies it varies massively.  It  has also
>> > changed greatly over time. GENERALLY speaking people will use the 11
>> > digit number or  the 10 digit number (without the "1" country code),
>> but there's not a lot of rhyme or reason  as to which is preferred in one
>> case vs another.
>>
>>
>>
>> It was mentioned previously, and requires libphonenumber, but the
>> phonenum module is pretty decent for helping with NANP numbers and
>> deciphering if one is let’s say US vs Canada vs Jamaica, etc.
>>
>>
>> Regards,
>>
>> Fred Posner
>> p: +1 (352) 664-3733
>> https://fred.tel/
>>
>>
>>
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[SR-Users] Re: Via header force change protocol to TLS?

[David Villasmil via sr-users]

trying to use

msg_apply_changes();
subst_hf("Via","/TCP/TLS/","f");

But as i don't seem to have access the top-most via, i can'it doesn't
work... i cam change the incoming Via, but not the outgoing...
I need this urgently.. any help anyone?



> Sergey,
>
> Thanks for pointing me to the PR but that's not exactly what i need, let
> me explain:
>
> We have an AWS an NLB on which our clients connect to and terminate TLS
> connections and from then on into our proxies it is TCP. Now when clients
> send invites (with TLS in their VIA as protocol) and we reply to them
> (because the socket is TCP) I need to set the protocol on the Via to be
> TLS. Otherwise, the client would not understand. This message or will
> believe that it's not. about the same connection
>
> I haven't figured out a way of doing this since it can't get the actual
> VIA that will be sent out. I've tried on the onsend_route (enabling
> onsend_route_reply)
>
> Help is greatly appreciated!
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Thu, May 30, 2024 at 11:44 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> HEllo Sergey,
>>
>> i can send one. yes.
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Thu, May 23, 2024 at 5:14 PM Sergey Safarov 
>> wrote:
>>
>>> Hi David
>>> Could you send PCAP for an inbound call via TCP connection?
>>>
>>> Sergey
>>>
>>> On Thu, May 23, 2024 at 5:53 PM David Villasmil <
>>> david.villasmil.w...@gmail.com> wrote:
>>>
>>>> it's still in progress though.
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.w...@gmail.com
>>>> phone: +34669448337
>>>>
>>>>
>>>> On Thu, May 23, 2024 at 4:51 PM David Villasmil <
>>>> david.villasmil.w...@gmail.com> wrote:
>>>>
>>>>> Thanks, I'll check it out!
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.w...@gmail.com
>>>>> phone: +34669448337
>>>>>
>>>>>
>>>>> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov 
>>>>> wrote:
>>>>>
>>>>>> We have tested this PR using the Linphone app.
>>>>>> So your case will be resolved using this PR.
>>>>>> Need to enable HAproxy protocol headers.
>>>>>>
>>>>>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov 
>>>>>> wrote:
>>>>>>
>>>>>>> Please try Kamailio PR
>>>>>>> https://github.com/kamailio/kamailio/pull/3731
>>>>>>>
>>>>>>> We have developed this PR for use case you have described.
>>>>>>> We have tested Route and Record-Route headers not Via.
>>>>>>> So will provide some review for this PR then will be fine.
>>>>>>>
>>>>>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil <
>>>>>>> david.villasmil.w...@gmail.com> wrote:
>>>>>>>
>>>>>>>> Hello Sergey,
>>>>>>>>
>>>>>>>> Thanks for the suggestion. Not sure if his is what i'm looking for,
>>>>>>>> allow me to explain further:
>>>>>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This
>>>>>>>> Load balancer is a TLS listener on the outside and a TCP connection to 
>>>>>>>> the
>>>>>>>> proxy inside.
>>>>>>>> So when sending an INVITE to the connected client, the via has a
>>>>>>>> TCP protocol like
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP
>>>>>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0
>>>>>>>>
>>>>>>>> and the client (linphone) disconnects and tries to contact the
>>>>>>>> proxy on that address on a TCP socket, which doesn't exist. I tried 
>>>>>>>> many
>>>>>>>> solutions none of which actually work... last one setting $du =$du +
>>>>>>>> ";transport=tls" and forcing the socket to the T

[SR-Users] Re: Via header force change protocol to TLS?

[David Villasmil via sr-users]

HEllo Sergey,

i can send one. yes.

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, May 23, 2024 at 5:14 PM Sergey Safarov  wrote:

> Hi David
> Could you send PCAP for an inbound call via TCP connection?
>
> Sergey
>
> On Thu, May 23, 2024 at 5:53 PM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> it's still in progress though.
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Thu, May 23, 2024 at 4:51 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>> Thanks, I'll check it out!
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
>>>
>>>
>>> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov 
>>> wrote:
>>>
>>>> We have tested this PR using the Linphone app.
>>>> So your case will be resolved using this PR.
>>>> Need to enable HAproxy protocol headers.
>>>>
>>>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov 
>>>> wrote:
>>>>
>>>>> Please try Kamailio PR
>>>>> https://github.com/kamailio/kamailio/pull/3731
>>>>>
>>>>> We have developed this PR for use case you have described.
>>>>> We have tested Route and Record-Route headers not Via.
>>>>> So will provide some review for this PR then will be fine.
>>>>>
>>>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil <
>>>>> david.villasmil.w...@gmail.com> wrote:
>>>>>
>>>>>> Hello Sergey,
>>>>>>
>>>>>> Thanks for the suggestion. Not sure if his is what i'm looking for,
>>>>>> allow me to explain further:
>>>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This
>>>>>> Load balancer is a TLS listener on the outside and a TCP connection to 
>>>>>> the
>>>>>> proxy inside.
>>>>>> So when sending an INVITE to the connected client, the via has a TCP
>>>>>> protocol like
>>>>>>
>>>>>> Via: SIP/2.0/TCP
>>>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0
>>>>>>
>>>>>> and the client (linphone) disconnects and tries to contact the proxy
>>>>>> on that address on a TCP socket, which doesn't exist. I tried many
>>>>>> solutions none of which actually work... last one setting $du =$du +
>>>>>> ";transport=tls" and forcing the socket to the TCP socket to the load
>>>>>> balancer, but of course i'm getting warnings about this.
>>>>>>
>>>>>> is this something that PR (not merged) would be addressing, i didn't
>>>>>> see that.
>>>>>> If not, is there a way of doing this without any trickery?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> David Villasmil
>>>>>> email: david.villasmil.w...@gmail.com
>>>>>> phone: +34669448337
>>>>>>
>>>>>>
>>>>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov 
>>>>>> wrote:
>>>>>>
>>>>>>> Probable you need this PR
>>>>>>> https://github.com/kamailio/kamailio/pull/3810
>>>>>>>
>>>>>>> Or you can try
>>>>>>> https://github.com/kamailio/kamailio/pull/3731
>>>>>>> In this PR we faced the same issue and solved this.
>>>>>>>
>>>>>>>
>>>>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users <
>>>>>>> sr-users@lists.kamailio.org> wrote:
>>>>>>>
>>>>>>>> Hello Anthony, did you solve this problem? I'm facing the same
>>>>>>>> problem
>>>>>>>>
>>>>>>>> Thanks!
>>>>>>>> Regards,
>>>>>>>>
>>>>>>>> David Villasmil
>>>>>>>> email: david.villasmil.w...@gmail.com
>>>>>>>> phone: +34669448337
>>>>>>>

[SR-Users] Re: Inbound Identity Verification Stir/Shaken

[David Villasmil via sr-users]

Hello,

For outgoing calls _from your DIDs_ to your providers, you need to sign the
identify header.
For incoming calls from your providers, IF THEY SIGN them, then you verify
them, Otherwise they just have no attestation. If they do provide it, it
possible you need to let them know you want it, but i'd assume they'd send
it automatically -if they receive it from upstream-.



Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, May 30, 2024 at 10:45 PM tfayomi--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> Yes, I have done an sngrep and the Identity header isn't shown at all on
> any incoming requests. That's partially why I was wondering if I need to
> request the identity header from the terminating carrier or if they need to
> provide it.
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[SR-Users] core dump on AWS with ~4000 TCP sockets

[David Villasmil via sr-users]

Hello all,
I'm seeing core dumps (can't find them!) on AWS with lots of TCP sockets
open (not related to load), kamailio dies randomly with:

*May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743739]: CRITICAL: 
[core/pass_fd.c:277]: receive_fd(): EOF on 142 May 30 14:31:03 sip-proxy
/usr/sbin/kamailio[743542]: ALERT:  [main.c:788]: handle_sigs():
child process 743672 exited by a signal 11 May 30 14:31:03 sip-proxy
/usr/sbin/kamailio[743542]: ALERT:  [main.c:791]: handle_sigs(): core
was generated May 30 14:32:03 sip-proxy /usr/sbin/kamailio[743542]:
CRITICAL:  [main.c:698]: sig_alarm_abort(): shutdown timeout
triggered, dying...*

before this happens, there's a ton of

May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.56.51.197:34846
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 98.196.165.187:38593
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:49372
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 97.101.162.40:45248
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.59.213.36:57074
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 207.195.63.14:2521
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:22155
failed (timeout)
May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: 
[core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:10339
failed (timeout)

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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[SR-Users] Re: Inbound Identity Verification Stir/Shaken

[David Villasmil via sr-users]

Hello,

not everyone has implemented this. You need to check whether it's there and
validate if present.

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, May 30, 2024 at 9:11 PM tfayomi--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hi All,
>
> I have a quick question regarding verification on incoming calls regarding
> the stir/shaken policy. I'm using the secsipid library to check the
> identity header, but the problem I'm running into is that there is no
> identity header in the invite from the incoming call (It's showing up as
> null).  Have most carriers enabled passing along an identity header, or is
> this something that is still in the works for some of them.
>
> Would I need to request and get the certificate from the terminating
> carrier? I've tried using the auth_identity module to see if i need to use
> the verification functions to get/request the certificate, but I have had
> no luck so far.
>
> Would anyone be able to provide direction on how to request/get/pass along
> the identity header from an incoming call with stir/shaken.
>
> (I am using kamailio v5.6 and debian 12)
>
> Thanks in advance,
> Temi
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[SR-Users] Re: libstirshaken installation issues

[David Villasmil via sr-users]

Secsipid works flawlessly, been in production for 3 years already

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Wed, 29 May 2024 at 20:44, tfayomi--- via sr-users <
sr-users@lists.kamailio.org> wrote:

> Thanks for the direction! I knew it wasn't related to Kamailio, but I
> wasn't too sure where to ask for guidance as it seems the libstirshaken
> library hasn't been updated in a bit. I'll go ahead and try the secsipid
> module since I'm unable to find and replace all the deprecated functions
> for libstirshaken/openssl 3.0.
>
> Thanks again!
> Temi
> __
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[SR-Users] Re: Via header force change protocol to TLS?

[David Villasmil via sr-users]

it's still in progress though.
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, May 23, 2024 at 4:51 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Thanks, I'll check it out!
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov 
> wrote:
>
>> We have tested this PR using the Linphone app.
>> So your case will be resolved using this PR.
>> Need to enable HAproxy protocol headers.
>>
>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov 
>> wrote:
>>
>>> Please try Kamailio PR
>>> https://github.com/kamailio/kamailio/pull/3731
>>>
>>> We have developed this PR for use case you have described.
>>> We have tested Route and Record-Route headers not Via.
>>> So will provide some review for this PR then will be fine.
>>>
>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil <
>>> david.villasmil.w...@gmail.com> wrote:
>>>
>>>> Hello Sergey,
>>>>
>>>> Thanks for the suggestion. Not sure if his is what i'm looking for,
>>>> allow me to explain further:
>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This Load
>>>> balancer is a TLS listener on the outside and a TCP connection to the proxy
>>>> inside.
>>>> So when sending an INVITE to the connected client, the via has a TCP
>>>> protocol like
>>>>
>>>> Via: SIP/2.0/TCP
>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0
>>>>
>>>> and the client (linphone) disconnects and tries to contact the proxy on
>>>> that address on a TCP socket, which doesn't exist. I tried many
>>>> solutions none of which actually work... last one setting $du =$du +
>>>> ";transport=tls" and forcing the socket to the TCP socket to the load
>>>> balancer, but of course i'm getting warnings about this.
>>>>
>>>> is this something that PR (not merged) would be addressing, i didn't
>>>> see that.
>>>> If not, is there a way of doing this without any trickery?
>>>>
>>>> Thanks!
>>>>
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.w...@gmail.com
>>>> phone: +34669448337
>>>>
>>>>
>>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov 
>>>> wrote:
>>>>
>>>>> Probable you need this PR
>>>>> https://github.com/kamailio/kamailio/pull/3810
>>>>>
>>>>> Or you can try
>>>>> https://github.com/kamailio/kamailio/pull/3731
>>>>> In this PR we faced the same issue and solved this.
>>>>>
>>>>>
>>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users <
>>>>> sr-users@lists.kamailio.org> wrote:
>>>>>
>>>>>> Hello Anthony, did you solve this problem? I'm facing the same problem
>>>>>>
>>>>>> Thanks!
>>>>>> Regards,
>>>>>>
>>>>>> David Villasmil
>>>>>> email: david.villasmil.w...@gmail.com
>>>>>> phone: +34669448337
>>>>>>
>>>>>>
>>>>>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba 
>>>>>> wrote:
>>>>>>
>>>>>>> I have kamailio behind a TLS termination proxy so the sockets are
>>>>>>> correctly deduced to be TCP. However the clients only talk TLS to the 
>>>>>>> proxy
>>>>>>> and are confused when the top Via header added by Kamailio is TCP. Is 
>>>>>>> there
>>>>>>> a way for Kamailio to forcibly pretend its protocol is TLS? Like
>>>>>>> advertised_address but "advertised_protocol"  instead.
>>>>>>>
>>>>>>> (With pjsip testing: it has a flag use_tls which ignores TCP from
>>>>>>> Kamailio and continues to use the persistent TLS transport to proxy.
>>>>>>> Linphone fails because it tries to honor TCP in Via and is unable to
>>>>>>> establish TCP transport).
>>>>>>>
>>>>>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to the
>>>>>>> proxy as TCP even though the contact addresses specify transport=TLS.
>>>>>>> ___
>>>>>>> Kamailio (SER) - Users Mailing List
>>>>>>> sr-users@lists.kamailio.org
>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>> __
>>>>>> Kamailio - Users Mailing List - Non Commercial Discussions
>>>>>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
>>>>>> Important: keep the mailing list in the recipients, do not reply only
>>>>>> to the sender!
>>>>>> Edit mailing list options or unsubscribe:
>>>>>>
>>>>>
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[SR-Users] Re: Via header force change protocol to TLS?

[David Villasmil via sr-users]

Thanks, I'll check it out!
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, May 23, 2024 at 4:16 PM Sergey Safarov  wrote:

> We have tested this PR using the Linphone app.
> So your case will be resolved using this PR.
> Need to enable HAproxy protocol headers.
>
> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov 
> wrote:
>
>> Please try Kamailio PR
>> https://github.com/kamailio/kamailio/pull/3731
>>
>> We have developed this PR for use case you have described.
>> We have tested Route and Record-Route headers not Via.
>> So will provide some review for this PR then will be fine.
>>
>> On Wed, May 22, 2024 at 2:22 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>>> Hello Sergey,
>>>
>>> Thanks for the suggestion. Not sure if his is what i'm looking for,
>>> allow me to explain further:
>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This Load
>>> balancer is a TLS listener on the outside and a TCP connection to the proxy
>>> inside.
>>> So when sending an INVITE to the connected client, the via has a TCP
>>> protocol like
>>>
>>> Via: SIP/2.0/TCP
>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0
>>>
>>> and the client (linphone) disconnects and tries to contact the proxy on
>>> that address on a TCP socket, which doesn't exist. I tried many
>>> solutions none of which actually work... last one setting $du =$du +
>>> ";transport=tls" and forcing the socket to the TCP socket to the load
>>> balancer, but of course i'm getting warnings about this.
>>>
>>> is this something that PR (not merged) would be addressing, i didn't see
>>> that.
>>> If not, is there a way of doing this without any trickery?
>>>
>>> Thanks!
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.w...@gmail.com
>>> phone: +34669448337
>>>
>>>
>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov 
>>> wrote:
>>>
>>>> Probable you need this PR
>>>> https://github.com/kamailio/kamailio/pull/3810
>>>>
>>>> Or you can try
>>>> https://github.com/kamailio/kamailio/pull/3731
>>>> In this PR we faced the same issue and solved this.
>>>>
>>>>
>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users <
>>>> sr-users@lists.kamailio.org> wrote:
>>>>
>>>>> Hello Anthony, did you solve this problem? I'm facing the same problem
>>>>>
>>>>> Thanks!
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.w...@gmail.com
>>>>> phone: +34669448337
>>>>>
>>>>>
>>>>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba 
>>>>> wrote:
>>>>>
>>>>>> I have kamailio behind a TLS termination proxy so the sockets are
>>>>>> correctly deduced to be TCP. However the clients only talk TLS to the 
>>>>>> proxy
>>>>>> and are confused when the top Via header added by Kamailio is TCP. Is 
>>>>>> there
>>>>>> a way for Kamailio to forcibly pretend its protocol is TLS? Like
>>>>>> advertised_address but "advertised_protocol"  instead.
>>>>>>
>>>>>> (With pjsip testing: it has a flag use_tls which ignores TCP from
>>>>>> Kamailio and continues to use the persistent TLS transport to proxy.
>>>>>> Linphone fails because it tries to honor TCP in Via and is unable to
>>>>>> establish TCP transport).
>>>>>>
>>>>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to the
>>>>>> proxy as TCP even though the contact addresses specify transport=TLS.
>>>>>> ___
>>>>>> Kamailio (SER) - Users Mailing List
>>>>>> sr-users@lists.kamailio.org
>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>> __
>>>>> Kamailio - Users Mailing List - Non Commercial Discussions
>>>>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
>>>>> Important: keep the mailing list in the recipients, do not reply only
>>>>> to the sender!
>>>>> Edit mailing list options or unsubscribe:
>>>>>
>>>>
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[SR-Users] Re: Via header force change protocol to TLS?

[David Villasmil via sr-users]

Hello Sergey,

Thanks for the suggestion. Not sure if his is what i'm looking for, allow
me to explain further:
We set up an NetworkLoadBalancer on AWS to offload tls on it. This Load
balancer is a TLS listener on the outside and a TCP connection to the proxy
inside.
So when sending an INVITE to the connected client, the via has a TCP
protocol like

Via: SIP/2.0/TCP
mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0

and the client (linphone) disconnects and tries to contact the proxy on
that address on a TCP socket, which doesn't exist. I tried many
solutions none of which actually work... last one setting $du =$du +
";transport=tls" and forcing the socket to the TCP socket to the load
balancer, but of course i'm getting warnings about this.

is this something that PR (not merged) would be addressing, i didn't see
that.
If not, is there a way of doing this without any trickery?

Thanks!

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Wed, May 22, 2024 at 12:16 PM Sergey Safarov  wrote:

> Probable you need this PR
> https://github.com/kamailio/kamailio/pull/3810
>
> Or you can try
> https://github.com/kamailio/kamailio/pull/3731
> In this PR we faced the same issue and solved this.
>
>
> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> Hello Anthony, did you solve this problem? I'm facing the same problem
>>
>> Thanks!
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
>>
>>
>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba 
>> wrote:
>>
>>> I have kamailio behind a TLS termination proxy so the sockets are
>>> correctly deduced to be TCP. However the clients only talk TLS to the proxy
>>> and are confused when the top Via header added by Kamailio is TCP. Is there
>>> a way for Kamailio to forcibly pretend its protocol is TLS? Like
>>> advertised_address but "advertised_protocol"  instead.
>>>
>>> (With pjsip testing: it has a flag use_tls which ignores TCP from
>>> Kamailio and continues to use the persistent TLS transport to proxy.
>>> Linphone fails because it tries to honor TCP in Via and is unable to
>>> establish TCP transport).
>>>
>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to the
>>> proxy as TCP even though the contact addresses specify transport=TLS.
>>> ___
>>> Kamailio (SER) - Users Mailing List
>>> sr-users@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
>> Edit mailing list options or unsubscribe:
>>
>
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[SR-Users] Re: Via header force change protocol to TLS?

[David Villasmil via sr-users]

Hello Anthony, did you solve this problem? I'm facing the same problem

Thanks!
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba  wrote:

> I have kamailio behind a TLS termination proxy so the sockets are
> correctly deduced to be TCP. However the clients only talk TLS to the proxy
> and are confused when the top Via header added by Kamailio is TCP. Is there
> a way for Kamailio to forcibly pretend its protocol is TLS? Like
> advertised_address but "advertised_protocol"  instead.
>
> (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio
> and continues to use the persistent TLS transport to proxy. Linphone fails
> because it tries to honor TCP in Via and is unable to establish TCP
> transport).
>
> BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy
> as TCP even though the contact addresses specify transport=TLS.
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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[SR-Users] Re: Add XML to INVITE

[David Villasmil via sr-users]

Thanks for replying!

thanks for this, it worked perfectly. I'm still wondering why
set_body_multipart
and append_body_part don't work together...

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Tue, 5 Mar 2024 at 20:25, Valentin Christoph <
christoph.valen...@kontron.com> wrote:

> Hi David,
>
>
>
> I am always very cautious when using msg_apply_changes() – e.g. there are
> some implications with Record-Route, afaik
>
>
>
> I do it this way:
>
>
>
> set_body("--outer\r\nContent-Type:application/sdp\r\n\r\n$rb" +
>
> 
> "\r\n--outer\r\nContent-Type:application/vnd.3gpp.mcptt-info+xml\r\n\r\n"
> +
>
> "\r\n"
> +
>
> 
>
> "\t\r\n" +
>
> "\r\n--outer--\r\n",
>
> "multipart/mixed;boundary=\"outer\"");
>
>
>
> Hope this helps, and sorry for slow reaction. I am scanning the mailing
> list in rather low frequency.
>
>
>
> Kr
>
>
>
> *From:* David Villasmil via sr-users 
> *Sent:* Tuesday, March 5, 2024 7:08 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Cc:* David Villasmil 
> *Subject:* [SR-Users] Re: Add XML to INVITE
>
>
>
> Hello guys,
>
>
>
> Anyone on this?
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
>
>
>
> On Sun, Mar 3, 2024 at 4:01 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
> Hello guys,
>
>
>
> I'm trying to add an xml with a boundary to an outgoing INVITE. But if i
> do:
>
>
>
> set_body_multipart("$rb", "application/sdp", "delimiter");
> msg_apply_changes();
> append_body_part("$var(something)", "application/pidf+xml");
> msg_apply_changes();
>
>
>
> but kamailio adds the SDP, but NOT the second $var(something). If i switch
> them and add firs the var, and then the SDP. kamailio adds the contents of
> $var(something) but NOT the SDP point is it doesn't add the second
> append
>
>
>
>
>
> Thanks!!!
>
>
>
>
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.w...@gmail.com
>
> phone: +34669448337
>
>
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[SR-Users] Re: Add XML to INVITE

[David Villasmil via sr-users]

Hello guys,

Anyone on this?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Sun, Mar 3, 2024 at 4:01 AM David Villasmil <
david.villasmil.w...@gmail.com> wrote:

> Hello guys,
>
> I'm trying to add an xml with a boundary to an outgoing INVITE. But if i
> do:
>
> set_body_multipart("$rb", "application/sdp", "delimiter");
> msg_apply_changes();
> append_body_part("$var(something)", "application/pidf+xml");
> msg_apply_changes();
>
> but kamailio adds the SDP, but NOT the second $var(something). If i switch
> them and add firs the var, and then the SDP. kamailio adds the contents of
> $var(something) but NOT the SDP point is it doesn't add the second
> append
>
>
> Thanks!!!
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
__
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[SR-Users] Add XML to INVITE

[David Villasmil via sr-users]

Hello guys,

I'm trying to add an xml with a boundary to an outgoing INVITE. But if i do:

set_body_multipart("$rb", "application/sdp", "delimiter");
msg_apply_changes();
append_body_part("$var(something)", "application/pidf+xml");
msg_apply_changes();

but kamailio adds the SDP, but NOT the second $var(something). If i switch
them and add firs the var, and then the SDP. kamailio adds the contents of
$var(something) but NOT the SDP point is it doesn't add the second
append


Thanks!!!


Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
__
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[SR-Users] Re: Reject TCP SYN

[David Villasmil via sr-users]

Thanks Karsten,

That won’t work for me with TLS clients, which is what I’m trying to do.
I’m making a PR with a new tcp param.

Thanks!

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Thu, 15 Feb 2024 at 17:42, Karsten Horsmann  wrote:

> 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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>> the sender!
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>>
>
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[SR-Users] Re: High CPU with DMQ on ARM64 (graviton) - Tuning values for DMQ worker_usleep

[David Villasmil via sr-users]

What is it synchronizing with? Bear in mind you can’t mix Kamailio versions
when doing DMQ

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Fri, 9 Feb 2024 at 17:17, Ben Kaufman via sr-users <
sr-users@lists.kamailio.org> wrote:

> Just checking to see if anyone had any ideas about tuning on this feature.
>
>
>
>
>
> *From:* Ben Kaufman via sr-users 
> *Sent:* Thursday, February 1, 2024 8:38 AM
> *To:* Kamailio (SER) - Users Mailing List 
> *Cc:* Ben Kaufman 
> *Subject:* [SR-Users] High CPU with DMQ on ARM64 (graviton) - Tuning
> values for DMQ worker_usleep
>
>
>
> *CAUTION:* This email originated from outside the organization. *Do not
> click links or open attachments* unless you recognize the sender and know
> the content is safe.
>
>
>
> I encountered a behavior with DMQ using excessive CPU when running
> kamailio 5.8 on AWS Graviton processors (running in Docker on Amazon Linux
> 2023), even on an otherwise idle system.  Finding the issue below pointed
> me towards using polling. I set the value for worker_usleep to 1000 as a
> test, and that seems to have resolved the CPU issue, but I have no idea of
> what a “good” value is for this setting, and 1000 was simply the example
> present in the documentation.  My specific use case for DMQ is
> synchronizing USRLOC data for ~7,500 users, registering with expiration
> times of 60 – 300 seconds with 7 – 9 nodes on the DMQ bus.  Any general
> recommendations on tuning values for this setting?
>
>
>
> https://github.com/kamailio/kamailio/issues/822
>
>
>
> Regards,
>
> Kaufman
> __
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[SR-Users] Re: Reject TCP SYN

[David Villasmil via sr-users]

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] Reject TCP SYN

[David Villasmil via sr-users]

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: Get the amount of open TCP connection from routing logic

[David Villasmil via sr-users]

Yeah, problem comes when calls draining last longer than the drain time
(which in aws is max 3600). Then I need to figure out something different.
Like iptables or something like that.

Unless Daniel has a better way from within Kamailio?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Tue, 6 Feb 2024 at 14:35, Sebastian Damm via sr-users <
sr-users@lists.kamailio.org> wrote:

> Yes, that's a successful connection first. But it helps to get rid of the
> existing connections as well, since there is no new SYN for
> re-registrations. And yes, I want the clients to reconnect. If have
> disabled the node in the loadbalancer before enabling maintenance mode,
> There won't be any new requests, and the existing connections will drain
> slowly.
> --
> *From:* David Villasmil 
> *Sent:* Tuesday, February 6, 2024 13:49
> *To:* Sebastian Damm 
> *Cc:* Kamailio (SER) - Users Mailing List 
>
> *Subject:* Re: [SR-Users] Get the amount of open TCP connection from
> routing logic
>
> Thanks for the tips. But wouldn’t that be considered by the load balancer
> a successful connection since there was an ACK to the SYN in order to send
> the 503 data? Or does that work for you because you’re relying on the
> client to retry and the load balancer sending to a different server?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, 6 Feb 2024 at 13:45, Sebastian Damm  wrote:
>
> We usually already have a high enough number of max tcp connections.
> However, there are certain scenarios when restarting some of our servers
> the loadbalancer sends all the clients previously connected to these
> servers to the remaining ones, leading to really full and really empty
> nodes. And sometimes we see Kamailio rejecting connections, and sometimes
> maybe the client can still connect, but the outgoing connection to the next
> hop is not possible anymore.
>
> But maybe you're right. Manually dropping the connection would probably
> not be any better than what Kamailio does after really reaching the limit.
>
> We have already implemented the maintenance mode draining, this would work
> as follows:
>
> Add a shared var named "MAINTENANCE_MODE", set it to 0 in config. Then,
> when handling the REGISTER, add something like this:
> if msg.is_method("REGISTER"):
> if KSR.pv.get("$shv(MAINTENANCE_MODE)") == 1:
> msg.log("In maintenance mode, rejecting registration")
> KSR.set_reply_close()
> KSR.hdr.append_to_reply("Retry-After: 3\r\n")
> KSR.sl.sl_send_reply(503, "Draining")
> return False
>
> You can set the value of the shared var to 1 via kamcmd. After that, on
> the next re-registration, the client will be disconnected, ending up
> hopefully on a different node after reconnect.
>
> Regards,
> Sebastian
>
>
> --
> *From:* David Villasmil 
> *Sent:* Tuesday, February 6, 2024 13:11
> *To:* Kamailio (SER) - Users Mailing List 
> *Cc:* Sebastian Damm 
> *Subject:* Re: [SR-Users] Get the amount of open TCP connection from
> routing logic
>
> Why not just increase the max tcp connections? Also, once you hit it,
> Kamailio will reset the syn and the load balancer should try on a different
> server.
>
> I am look for something like what you’re looking for but reset the sub
> on-demand so I can put the server in maintenance mode.
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, 6 Feb 2024 at 12:47, Sebastian Damm via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
> Hello,
>
> My plan is to reject new inbound requests if there is already a
> certain number of open connections on one Kamailio server. I could answer
> with a 503 and directly close the connection forcing the client to
> reconnect and the loadbalancer to send traffic to a different node on the
> next connect. It's just a last-resort measure to prevent Kamailio from
> running into the configured TCP connection limit (workarounding a
> poorly-implemented external load balancer).
>
> Since I didn't find anything in the docs: Is there a pseudovariable
> containing the current number of open tcp/tls connections on this Kamailio?
> I can (and already do) query them via kamcmd or rpc, but can I access the
> variable from inside the routing script as well?
>
> Regards,
> Sebastian
> __
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[SR-Users] Re: Get the amount of open TCP connection from routing logic

[David Villasmil via sr-users]

Thanks for the tips. But wouldn’t that be considered by the load balancer a
successful connection since there was an ACK to the SYN in order to send
the 503 data? Or does that work for you because you’re relying on the
client to retry and the load balancer sending to a different server?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Tue, 6 Feb 2024 at 13:45, Sebastian Damm  wrote:

> We usually already have a high enough number of max tcp connections.
> However, there are certain scenarios when restarting some of our servers
> the loadbalancer sends all the clients previously connected to these
> servers to the remaining ones, leading to really full and really empty
> nodes. And sometimes we see Kamailio rejecting connections, and sometimes
> maybe the client can still connect, but the outgoing connection to the next
> hop is not possible anymore.
>
> But maybe you're right. Manually dropping the connection would probably
> not be any better than what Kamailio does after really reaching the limit.
>
> We have already implemented the maintenance mode draining, this would work
> as follows:
>
> Add a shared var named "MAINTENANCE_MODE", set it to 0 in config. Then,
> when handling the REGISTER, add something like this:
> if msg.is_method("REGISTER"):
> if KSR.pv.get("$shv(MAINTENANCE_MODE)") == 1:
> msg.log("In maintenance mode, rejecting registration")
> KSR.set_reply_close()
> KSR.hdr.append_to_reply("Retry-After: 3\r\n")
> KSR.sl.sl_send_reply(503, "Draining")
> return False
>
> You can set the value of the shared var to 1 via kamcmd. After that, on
> the next re-registration, the client will be disconnected, ending up
> hopefully on a different node after reconnect.
>
> Regards,
> Sebastian
>
>
> --
> *From:* David Villasmil 
> *Sent:* Tuesday, February 6, 2024 13:11
> *To:* Kamailio (SER) - Users Mailing List 
> *Cc:* Sebastian Damm 
> *Subject:* Re: [SR-Users] Get the amount of open TCP connection from
> routing logic
>
> Why not just increase the max tcp connections? Also, once you hit it,
> Kamailio will reset the syn and the load balancer should try on a different
> server.
>
> I am look for something like what you’re looking for but reset the sub
> on-demand so I can put the server in maintenance mode.
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, 6 Feb 2024 at 12:47, Sebastian Damm via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
> Hello,
>
> My plan is to reject new inbound requests if there is already a
> certain number of open connections on one Kamailio server. I could answer
> with a 503 and directly close the connection forcing the client to
> reconnect and the loadbalancer to send traffic to a different node on the
> next connect. It's just a last-resort measure to prevent Kamailio from
> running into the configured TCP connection limit (workarounding a
> poorly-implemented external load balancer).
>
> Since I didn't find anything in the docs: Is there a pseudovariable
> containing the current number of open tcp/tls connections on this Kamailio?
> I can (and already do) query them via kamcmd or rpc, but can I access the
> variable from inside the routing script as well?
>
> Regards,
> Sebastian
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
> To unsubscribe send an email to sr-users-le...@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
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> Edit mailing list options or unsubscribe:
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>
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[SR-Users] Re: Get the amount of open TCP connection from routing logic

[David Villasmil via sr-users]

Why not just increase the max tcp connections? Also, once you hit it,
Kamailio will reset the syn and the load balancer should try on a different
server.

I am look for something like what you’re looking for but reset the sub
on-demand so I can put the server in maintenance mode.

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Tue, 6 Feb 2024 at 12:47, Sebastian Damm via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hello,
>
> My plan is to reject new inbound requests if there is already a
> certain number of open connections on one Kamailio server. I could answer
> with a 503 and directly close the connection forcing the client to
> reconnect and the loadbalancer to send traffic to a different node on the
> next connect. It's just a last-resort measure to prevent Kamailio from
> running into the configured TCP connection limit (workarounding a
> poorly-implemented external load balancer).
>
> Since I didn't find anything in the docs: Is there a pseudovariable
> containing the current number of open tcp/tls connections on this Kamailio?
> I can (and already do) query them via kamcmd or rpc, but can I access the
> variable from inside the routing script as well?
>
> Regards,
> Sebastian
> __
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[SR-Users] install location when compiling

[David Villasmil via sr-users]

Hello guys,

i compiled kamailio and i want it installed on debian's deb location, i.e.:
/etc/kamailio
/usr/lib/x86_64-linux-gnu/kamailio/modules
etc..

I know if i do

make prefix=/ all
make prefix=/ install

the config will end up on /etc/kamailio, but then the modules will be on
/lib64...

i don't see a way of doing this, is there?

Thanks!

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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[SR-Users] Re: STIR/SHAKEN

[David Villasmil via sr-users]

Ditto, i do first approach.

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users <
sr-users@lists.kamailio.org> wrote:

> These are all valid approaches, depending on preference. Catching 302s and
> extracting exactly the desired info might be easier with Kamailio, so I
> suppose, ceteris paribus, I'd recommend the first one.
>
> Dispatcher is a great approach!
>
> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users <
> sr-users@lists.kamailio.org> wrote:
> >
> > Hello,
> >
> > We're attempting to add identities to our invites. I believe we want to
> send our invites to the NSS, and it'll reply with a 302'd invite with an
> identity header. We currently use FreeSWITCH for dialplans and media and
> Kamailio as our SBC.
> >
> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS
> (302) -> Kamailio -> PSTN
> >
> > or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH
> -> Kamailio -> PSTN
> >
> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right
> approach? We currently use it for our two Kamailios and our FreeSWITCHs.
> We're going to have two NSS servers, one for failover.
> >
> > Any thoughts would be greatly appreciated, thanks!
> >
> > --
> > Anthony Wittig
> > Mango Voice Developer
> > __
> > Kamailio - Users Mailing List - Non Commercial Discussions
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> > Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> > Edit mailing list options or unsubscribe:
>
> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
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[SR-Users] Re: refer-to to new INVITE

[David Villasmil via sr-users]

https://sr-users.sip-router.narkive.com/1lg6Ov5U/problem-initiating-a-call-with-dlg-bridge


On Sat, 25 Nov 2023 at 06:39, Alex Balashov via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hi,
>
> You may wish to explore this function, and its RPC correlate:
>
> dialog Module
> 
> kamailio.org
> 
> [image: favicon.ico]
> 
> 
>
>
> -- Alex
>
> —
> Sent from mobile, apologies for brevity and errors.
>
> On Nov 22, 2023, at 7:17 PM, Social Boh via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
> Hello,
>
>
>
> I'd like to know if there is a way, from Kamailio side, take the refer-to
> header Kamailio receive in a REFER request and create a new INVITE whit
> this data; then bridge the call between the originating user and the new
> destination user.
>
> Thank you
>
> Regards
>
> --
> ---
> I'm SoCIaL, MayBe
>
> __
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favicon.ico
Description: Binary data
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[SR-Users] Re: sips to sip

[David Villasmil via sr-users]

Interesting, thanks for the info!

On Fri, 24 Nov 2023 at 03:50, Ihor Olkhovskyi via sr-users <
sr-users@lists.kamailio.org> wrote:

> Just to add for a history.
>
> I've found that Asterisk normally accepts sips schema over UDP and keeps
> sending responses regardless on schema specified also in UDP.
>
> So, in my case only thing that was needed is to change the $du, smth like
>
> request_route{
>
> ...
> if (is_method("CANCEL")) {
> if (t_check_trans()) {
> route(SIPS_PROCESS);
> if (!t_relay()) {
> send_reply("500", "Relay failed");
> }
> }
> exit;
> }
>
> }
>
> route[RELAY] {
>
> route(SIPS_PROCESS);
>
> t_relay();
>
> }
>
> route[SIPS_PROCESS] {
> # If device is using "sips:" dialing scheme, force it with $du
> pointing to "sip:" scheme
> # so Kamailio will not try to send it over TLS to internal services.
> # Applies to requests coming FROM external devices, like Poly OBI ATA's
> if (route(FROMASTERISK)) {
> return;
> }
> if ($du == $null && $rz != 'sips') {
> return;
> }
> if ($du != $null) {
> if ($(du{uri.scheme}) == 'sips') {
> xlog("$var(debug_level)", "[SIPS_PROCESS]: Modifying <$du> to
> sip scheme");
>
> $du = 'sip:' + $dd + ':' + $dp;
> }
> return;
> }
> if ($rz != 'sips') {
> return;
> }
> $du = 'sip:' + $rd + ':' + $rp;
>
> xlog("$var(debug_level)", "[SIPS_PROCESS]: Setting <$du> for packet to
> <$ru> to avoid sips routing");
> }
>
> Le 22/11/2023 à 01:53, David Villasmil via sr-users a écrit :
>
> thanks all, i figured it out with help from Alfonso who found
> https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srtp-bridge.cfg
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, Nov 21, 2023 at 2:34 PM Henning Westerholt  wrote:
>
>> Hi David,
>>
>>
>>
>> at least from the logs the dialog is matched:
>>
>>
>>
>> DEBUG: dialog [dlg_hash.c:885]: internal_get_dlg(): dialog
>> callid='aa157c08df1109c4655ca64d0ed67' found on entry 1431, dir=2
>> to-tag='ZDDN09pNH37XB'
>> DEBUG: dialog [dlg_profile.c:541]: set_current_dialog(): setting current
>> dialog [1431:511]
>> DEBUG: dialog [dlg_handlers.c:343]: dlg_iuid_sfree(): freeing dlg iuid
>> [3314:1253] (0x7f6d9b2afe20)
>> DEBUG: dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7f6d9b2f8240
>> changed from state 5 to state 5, due event 7 (ref 2)
>>
>>
>>
>> The dialog is already deleted at this point, not sure why.
>>
>>
>>
>> Please note that your rewrite method can cause problems in dialogs, as
>> for example the From/To header will be not matching to the expectations of
>> the UAs. Consider using the methods I mentioned earlier to prevent this.
>>
>>
>>
>> Cheers,
>>
>>
>>
>> Henning
>>
>>
>>
>> *From:* David Villasmil 
>> *Sent:* Dienstag, 21. November 2023 13:58
>> *To:* Henning Westerholt 
>> *Cc:* Kamailio (SER) - Users Mailing List 
>> *Subject:* Re: [SR-Users] sips to sip
>>
>>
>>
>> thanks Henning,
>>
>>
>>
>> Already tried that with:
>>
>>
>>
>>
>> if (is_method("INVITE")) {
>> $var(fline) = $msg(fline);
>> $var(hdrs) = $msg(hdrs);
>> $var(body) = $msg(body);
>>
>> $var(fline) = $(var(fline){s.replace,sips,sip});
>> #{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
>> $var(hdrs) = $(var(hdrs){s.replace,sips,sip});
>> #{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
>> $var(body) = $(var(body){s.replace,sips,sip});
>> #{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
>>
>> msg_set_buffer("$var(fline)\r\n$var(hdrs)\r\n\r\n$var(body)");
>> msg_apply_changes();
>> }
>>
>>
>>
>> but then the BYE doesn't work, kamailio can't match it to a dialog
>>
>>
>>
>> 2023/11/21 12:48:41.962062 FREESWITCH-IP:5080 -> 10.0.3.141:5060
>> BYE sip:+FROM-NUMBER@CLIENT-IP;transport=TLS SIP/2.0
>> Via: SIP/2.0/UDP FREESWITCH-IP:5080;rport;branch=z9hG4bK34FNtg4e31B8K
>> Route: 
>> Route: 
>> Max-Forwards: 70
>> From: ;tag=133B764ZS9QvD
>> To: ;tag=df6b235d
>> Call-ID: 75fc6fcaaa32bcd7655ca72452eac
>> CSeq: 75731668 BYE
>> User-Agent: C3S-v1.1
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, path, replaces
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>> Content-Length: 0
>>
>>
>>
>> DEBUG:  [core/receive.c:392]: receive_msg(): --- received sip
>> message - request - call-id: [aa157c08df1109c4655ca64d0ed67] - cseq:
>> [75731561 BYE]
>> DEBUG:  [core/receive.c:263]: ksr_evrt_pre_routing(): event route
>> core:pre-routing not defined
>> DEBUG:  [core/receive.c:474]: receive_msg(): preparing to run
>> routing scripts...
>> ERROR: 

[SR-Users] Re: refer-to to new INVITE

[David Villasmil via sr-users]

AFAIK that’s a job for a B2BUA, not a proxy. You want something like
Freeswitch or Asterisk

On Thu, 23 Nov 2023 at 01:26, Social Boh via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hello,
>
> I'd like to know if there is a way, from Kamailio side, take the
> refer-to header Kamailio receive in a REFER request and create a new
> INVITE whit this data; then bridge the call between the originating user
> and the new destination user.
>
> Thank you
>
> Regards
>
> --
> ---
> I'm SoCIaL, MayBe
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
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> Important: keep the mailing list in the recipients, do not reply only to
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>
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[SR-Users] Re: sips to sip

[David Villasmil via sr-users]

thanks all, i figured it out with help from Alfonso who found
https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srtp-bridge.cfg
Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337


On Tue, Nov 21, 2023 at 2:34 PM Henning Westerholt  wrote:

> Hi David,
>
>
>
> at least from the logs the dialog is matched:
>
>
>
> DEBUG: dialog [dlg_hash.c:885]: internal_get_dlg(): dialog
> callid='aa157c08df1109c4655ca64d0ed67' found on entry 1431, dir=2
> to-tag='ZDDN09pNH37XB'
> DEBUG: dialog [dlg_profile.c:541]: set_current_dialog(): setting current
> dialog [1431:511]
> DEBUG: dialog [dlg_handlers.c:343]: dlg_iuid_sfree(): freeing dlg iuid
> [3314:1253] (0x7f6d9b2afe20)
> DEBUG: dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7f6d9b2f8240
> changed from state 5 to state 5, due event 7 (ref 2)
>
>
>
> The dialog is already deleted at this point, not sure why.
>
>
>
> Please note that your rewrite method can cause problems in dialogs, as for
> example the From/To header will be not matching to the expectations of the
> UAs. Consider using the methods I mentioned earlier to prevent this.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> *From:* David Villasmil 
> *Sent:* Dienstag, 21. November 2023 13:58
> *To:* Henning Westerholt 
> *Cc:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] sips to sip
>
>
>
> thanks Henning,
>
>
>
> Already tried that with:
>
>
>
>
> if (is_method("INVITE")) {
> $var(fline) = $msg(fline);
> $var(hdrs) = $msg(hdrs);
> $var(body) = $msg(body);
>
> $var(fline) = $(var(fline){s.replace,sips,sip});
> #{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
> $var(hdrs) = $(var(hdrs){s.replace,sips,sip});
> #{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
> $var(body) = $(var(body){s.replace,sips,sip});
> #{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
>
> msg_set_buffer("$var(fline)\r\n$var(hdrs)\r\n\r\n$var(body)");
> msg_apply_changes();
> }
>
>
>
> but then the BYE doesn't work, kamailio can't match it to a dialog
>
>
>
> 2023/11/21 12:48:41.962062 FREESWITCH-IP:5080 -> 10.0.3.141:5060
> BYE sip:+FROM-NUMBER@CLIENT-IP;transport=TLS SIP/2.0
> Via: SIP/2.0/UDP FREESWITCH-IP:5080;rport;branch=z9hG4bK34FNtg4e31B8K
> Route: 
> Route: 
> Max-Forwards: 70
> From: ;tag=133B764ZS9QvD
> To: ;tag=df6b235d
> Call-ID: 75fc6fcaaa32bcd7655ca72452eac
> CSeq: 75731668 BYE
> User-Agent: C3S-v1.1
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Reason: Q.850;cause=16;text="NORMAL_CLEARING"
> Content-Length: 0
>
>
>
> DEBUG:  [core/receive.c:392]: receive_msg(): --- received sip
> message - request - call-id: [aa157c08df1109c4655ca64d0ed67] - cseq:
> [75731561 BYE]
> DEBUG:  [core/receive.c:263]: ksr_evrt_pre_routing(): event route
> core:pre-routing not defined
> DEBUG:  [core/receive.c:474]: receive_msg(): preparing to run
> routing scripts...
> ERROR: 

[SR-Users] Re: sips to sip

[David Villasmil via sr-users]

thanks Henning,

Already tried that with:


if (is_method("INVITE")) {
$var(fline) = $msg(fline);
$var(hdrs) = $msg(hdrs);
$var(body) = $msg(body);

$var(fline) = $(var(fline){s.replace,sips,sip});
#{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
$var(hdrs) = $(var(hdrs){s.replace,sips,sip});
#{s.replace,;transport=TLS,}{s.replace,;transport=tls,});
$var(body) = $(var(body){s.replace,sips,sip});
#{s.replace,;transport=TLS,}{s.replace,;transport=tls,});

msg_set_buffer("$var(fline)\r\n$var(hdrs)\r\n\r\n$var(body)");
msg_apply_changes();
}

but then the BYE doesn't work, kamailio can't match it to a dialog

2023/11/21 12:48:41.962062 FREESWITCH-IP:5080 -> 10.0.3.141:5060
BYE sip:+FROM-NUMBER@CLIENT-IP;transport=TLS SIP/2.0
Via: SIP/2.0/UDP FREESWITCH-IP:5080;rport;branch=z9hG4bK34FNtg4e31B8K
Route: 
Route: 
Max-Forwards: 70
From: ;tag=133B764ZS9QvD
To: ;tag=df6b235d
Call-ID: 75fc6fcaaa32bcd7655ca72452eac
CSeq: 75731668 BYE
User-Agent: C3S-v1.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0



DEBUG:  [core/receive.c:392]: receive_msg(): --- received sip message
- request - call-id: [aa157c08df1109c4655ca64d0ed67] - cseq: [75731561 BYE]
DEBUG:  [core/receive.c:263]: ksr_evrt_pre_routing(): event route
core:pre-routing not defined
DEBUG:  [core/receive.c:474]: receive_msg(): preparing to run routing
scripts...
ERROR: 

[SR-Users] sips to sip

[David Villasmil via sr-users]

Hello guys,

I have this setup where one side is TLS and the other UDP. Normally this
works fine, but we have this provider sending sips as the schema everywhere
(from, to, rr, contacts), kamailio sends the same sips to the upstream usp
freeswitch.

My problem is when FS sends back a 200OK and kamailio forwards it back to
the provider, the provider sends an ACK and kamailio can't match it with
the dialog and doesn't know where to forward it.

i think this is happening because FS when is sees SIPS is setting the
contact port as 5081 instead of the usual 5080...


Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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[SR-Users] Re: How to check for presence of a file from a route? (Or any other way to tell kamailio not to accept new calls)

[David Villasmil via sr-users]

Or better yet use an htable

On Mon, 13 Nov 2023 at 13:31, Henning Westerholt via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hello,
>
> just to add another common method, you could also just use a cfg file
> variable that you can modify later with kamcmd, kamcli etc.. There is a
> prepared variable (pstn..) in the default cfg.
>
> Cheers,
>
> Henning
>
>
> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://gilawa.com
>
>
>
> > -Original Message-
> > From: Daniel-Constantin Mierla via sr-users  >
> > Sent: Montag, 13. November 2023 11:46
> > To: Kamailio (SER) - Users Mailing List 
> > Cc: Daniel-Constantin Mierla 
> > Subject: [SR-Users] Re: How to check for presence of a file from a
> route? (Or
> > any other way to tell kamailio not to accept new calls)
> >
> > On 13.11.23 11:17, Benoit Panizzon via sr-users wrote:
> > > Hi
> > >
> > > I'm still looking for a better way to tell Kamailio that we want to
> > > enter 'maintenance' and STAY in maintenance after a restart.
> > >
> > > Maintenance is: Reject all messages without totag with 503 to prevent
> > > creating new dialogues.
> > >
> > > I could use a shared pv and the use kamcmd pv.shvSet to toggle it.
> > >
> > > But when kamailio is restarted, that pv is also reset to it's initial
> > > state. Not good, if I want to make sure the node stays in maintenance
> > > mode after a restart as for example after pushing a config change via
> > > ansible.
> > >
> > > So what comes to my mind is to check for the presence of a file.
> > >
> > > something like:
> > >
> > > route[CHECK_MAINTENANCE]
> > > {
> > > if (!has_totag() && is_method("INVITE")) {
> > > if (file_exist("/etc/kamailio/maintenance.flag")) {
> > > xlog("L_ERR", "Maintenance flag present! Rejecting
> > INVITE\n");
> > > t_send_reply("503", "Maintenance mode - no new
> > calls accepted");
> > > exit;
> > > }
> > > }
> > > }
> >
> > I would really avoid checking/reading a file for every new sip call,
> especially on
> > production system with decent volume of calls. But if suits your needs
> and
> > system, look at corex module, there is a function to read the content of
> a file.
> >
> > I would rather find a solution where the state is kept in memory, like
> still using
> > $shv(...) and change kamailio.cfg to init in maintenance mode via pv
> > parameter which can set $shv(...) at startup, or using htable with
> database
> > backend that loads at startup.
> >
> > Cheers,
> > Daniel
> >
> > --
> > Daniel-Constantin Mierla (@ asipto.com)
> > twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and
> > Development Services
> >
> > __
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[SR-Users] Re: STIR/SHAKEN with Kamailio

[David Villasmil via sr-users]

Agreed

On Sun, 22 Oct 2023 at 23:22, Oleg Belousov via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hi, Ben.
> Inline is how we implemented mentioned points in our project, If that will
> be helpful.
> Not sure if SS helps much to prevent spoofed calls, but there could be
> other bonuses like rcd/branded calling which sounds like a promising cnam
> extension.
> --
> obelousov.tel
>
>
> On Fri, 20 Oct 2023 at 16:25, Ben Kaufman via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> My point was simply that there's more challenge in the bureaucracy than
>> technical implementation.
>>
>> From a technical standpoint, the corner cases to consider are:
>>
>> 1. Number validity. Sure things that fit into an e.164 and/or
>> recognizable number patterns are simple.  What happens when someone sends a
>> From: URI of `sip:anonym...@domain.com` - IIRC, the orig_tn field within
>> the identity header is supposed to be numeric.  Do you reject the call?
>> Attest it as "C" and provide this in the `orig_tn` field or in a separate
>> field?
>>
> [Oleg] We are using P-Asserted-Identity instead of From, it keeps the
> number even if CLIR is enabled. As per ATIS-174
> The ‘orig’ claim ‘tn’ value shall be derived using the following rules:
> - The P-Asserted-Identity header field value shall be used as the
> telephone identity, if present, otherwise the From header field value shall
> be used.
> - If there are two P-Asserted-Identity header field values, the
> authentication service shall have logic to choose the most appropriate one
> based on local service provider policy.
> - The action taken when neither the P-Asserted-Identity header field value
> nor the From header contain tel URI identities is outside the scope of the
> SHAKEN framework.
>
>>
>> 2. Handling of forwarded calls - If you're sending a Diversion: header,
>> do you also add an Identity header with a `div` passport?  Rewrite the From
>> header?  How do you determine the attestation in that case?
>>
>  [Oleg] For the forwarded calls (with enabled diversion) we are checking
> if Identity is present. If yes - send call as it is, otherwise assign
> attestation C,
>
>>
>> 3. Known customers sending numbers for which you're not the provider?
>> Strictly speaking this should attest as "B", but supposing that you're a
>> secondary vendor for the customer, and they're sending their primary number
>> which is with a different provider?  Do you then allow them to submit an
>> LOA (or whatever your jurisdictional equivalent is) and attest as A?
>>
>  [Oleg] If A-number does nor belong to operator - we assign attestation B
>
>>
>> The questions above are strictly for STI Authentication.  Verification
>> has some other idiosyncrasies.   Consider that there's three attestation
>> levels for authentication, and normally as a carrier it is not desirable to
>> pass the Identity header to the customer (consider if Privacy: is on).  The
>> general practice is to assign this to a verstat parameter to the user
>> portion of a PAI header's **USER** field, which is syntactically awkward in
>> Kamailio.  Also strictly speaking AFAIK, the verstat only has two values -
>> passed or failed - so there's three possible attestation levels but they
>> only map to two verification levels.  Therea are suggestions on how to deal
>> with this, but I'm not sure on their official status.
>>
>> This brings up the final complexity:  It's a rapidly evolving system
>> without a high degree of consistency vendor to vendor, so there's as much
>> of a challenge of staying on top of things as anything else.
>>
>> -Original Message-
>> From: Olle E. Johansson via sr-users 
>> Sent: Friday, October 20, 2023 2:08 AM
>> To: Kamailio (SER) - Users Mailing List 
>> Cc: Olle E. Johansson 
>> Subject: [SR-Users] Re: STIR/SHAKEN with Kamailio
>>
>> CAUTION: This email originated from outside the organization. Do not
>> click links or open attachments unless you recognize the sender and know
>> the content is safe.
>>
>>
>> > On 19 Oct 2023, at 18:46, Alex Balashov via sr-users <
>> sr-users@lists.kamailio.org> wrote:
>> >
>> > Would join Kaufman here to say that free-range STIR/SHAKEN
>> > implementations in the US are limited by the small number of certified
>> > authentication providers, but presumably the EU version will to some
>> > extent avoid US-style Guilded Age corporate welfare...
>> Sadly that's my view of the US implementation. I can't say if it solved
>> the problem, but I can see that a lot of new and old actors got an
>> oppurtunity to earn more money.
>>
>> There's no EU-wide implementation or regulation at this point. I am aware
>> of France. There are certainly discussions.
>> /O
>> >
>> > -- Alex
>> >
>> >> On 19 Oct 2023, at 09:33, Ben Kaufman via sr-users <
>> sr-users@lists.kamailio.org> wrote:
>> >>
>> >> Like some of the other posters here, we've implemented it as a
>> 302-redirect server. This was the primary reason for using the secsipid
>> rather than stirshaken 

[SR-Users] Re: STIR/SHAKEN with Kamailio

[David Villasmil via sr-users]

I implemented it with Kamailio. I forward all invites to a Kamailio
instance that does the stir/shaken stuff and returns it in a 302 which is
then added to the INVITE to continue normally. OpenSIPS has a nice how-to
and examples on how to do it.

Hope that helps

David

On Wed, 18 Oct 2023 at 10:03, Olle E. Johansson via sr-users <
sr-users@lists.kamailio.org> wrote:

> Hello Kamailians!
>
> As STIR/SHAKEN seems to cross the ocean and arrive on the European shores,
> I’m curious on how you’ve implemented it with Kamailio. I asked on our
> Matrix chat and got two responses that seems to not use the Kamailio
> STIR/SHAKEN support but rather a 3rd party service that they’ve integrated
> using restful APIs.
>
> Please help me and share how you’re implementing STIR/SHAKEN with Kamailio.
>
> I’m not shaken, nor stirred. yet.
>
> Regards,
> /O
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[SR-Users] Re: Multi-dimensional data in htable

[David Villasmil via sr-users]

Why not just use a hash and separate fields by ; ? You can then just join
or split as you set or get?

On Fri, 22 Sep 2023 at 08:20, Ben Kaufman via sr-users <
sr-users@lists.kamailio.org> wrote:

> Thank you.  I’ll look into that!
>
>
>
> Kaufman
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