Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[Pavithra Mohanraja]

Hi Supreeth,

I have followed your guide to build kamailio IMS

https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/

services are running fine .
when i try to make a call . registration fails with 478 unresolvable
destination

93(25632) INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header
present.
93(25632) NOTICE: 

Re: [SR-Users] Kamailio IMS with ENUM

[BALL SUN]

Hello

I may not describe the problem clearly. When a UE call a number to the
Kamailio IMS core, says, the number is 12345678, the R-URI in the SIP
INVITE is 
"sip:12345678;phone-context=ims.mnc001.mcc001.3gppnetwork@ims.mnc001.mcc001.3gppnetwork.org;user=phone",
Kamailio P-CSCF pass to INVITE to S-CSCF, and then S-CSCF pass to
I-CSCF, I-SCSF trigger LIR to HSS with Public-Identity = tel:12345678,
This IMPU did not provision in HSS, so HSS response  with
"DIAMATER_ERROR_USER_UNKNOWN".

On receiving this LIA, I-CSCF return to S-CSCF with "604 Does not
exist anywhere - HSS User Unknown", and then forward to P-CSCF and the
UE.

My question is why I-CSCF did not trigger a DNS/ENUM query after receiving LIA?

attached is the Kamailio configuration that we are using, please advise.

- RBK

On Wed, Jun 24, 2020 at 11:52 PM Valentin Christoph
 wrote:
>
> I don't know your DNS server nor your addressing plan. Probably to modify / 
> add some zone files
>
> You must be sure to forward the ranges of telephone numbers that you want to 
> forward to the other domain.
>
> Actually, it DID the request to the DNS/ENUM, but the result was "keep the 
> domain as it is" or "replace it by the own domain"
>
> If the DNS/ENUM query did NOT succeed, then you would run into route(PSTN) 
> and the request would be forwarded to "good old CS domain" to find a 
> destination.
>
> -Ursprüngliche Nachricht-
> Von: sr-users  Im Auftrag von BALL SUN
> Gesendet: Mittwoch, 24. Juni 2020 16:49
> An: Kamailio (SER) - Users Mailing List 
> Betreff: Re: [SR-Users] Kamailio IMS with ENUM
>
> yes. when the HSS profile is not found, it did not route to DNS/ENUM to 
> check, which portion of configuration should I check?
>
> On Wed, Jun 24, 2020 at 5:31 PM Valentin Christoph 
>  wrote:
> >
> > This means the HSS had answered the user is not known in this domain 
> > (lir_term_user_unknown), but then the DNS server insists this telephone 
> > number should be handled by our domain ($rd == NETWORKNAME).
> >
> > Just enter the user to the HSS database - if it shall be handled by
> > this domain -
> >
> > Or configure DNS, so that this telephone number is routed to another 
> > domain. - if it shall be handled by a different domain.
> >
> > -Ursprüngliche Nachricht-
> > Von: sr-users  Im Auftrag von
> > BALL SUN
> > Gesendet: Dienstag, 23. Juni 2020 16:40
> > An: Kamailio (SER) - Users Mailing List 
> > Betreff: Re: [SR-Users] Kamailio IMS with ENUM
> >
> > Thanks. I actually keep having "604","Does not exist anywhere - HSS User 
> > Unknown", so by right, I-CSCF should go to check the ENUM, but it didn't so 
> > what configuration I should enable? is it dispatcher list and needed to 
> > enabkle PEERING? below is the routine that related to "604","Does not exist 
> > anywhere - HSS User Unknown" in icscf.
> >
> > - RBK
> >
> > route[lir_term_user_unknown]
> > {
> > if (uri =~ "tel:.*") {
> > # Let's check, if the number can be found in ENUM:
> > if(!enum_query()) {
> > # ENUM failed, send it to the PSTN-Gateway:
> > route(PSTN);
> > break;
> > }
> > # ENUM resolved to another domain
> > if ($rd != NETWORKNAME) {
> > t_on_reply("initial_request_reply");
> > t_on_failure("initial_request_failure");
> > if (!t_relay()) {
> > t_reply("500","Error forwarding to external domain"); exit; }; exit; } else 
> > { t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; }; } 
> > else { # we received a request for our domain (non-tel), but HSS said "User 
> > Unknown"
> > if ($rd != NETWORKNAME) {
> > t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; }
> > else { # try to forward non-tel request to other domain
> > t_on_reply("Initial_Request_reply");
> > t_on_failure("Initial_Request_failure");
> > if (!t_relay()) {
> > t_reply("500","Error forwarding to external domain"); exit; }; exit;
> > }; }; } }
> >
> > On Tue, Jun 23, 2020 at 8:40 PM Valentin Christoph 
> >  wrote:
> > >
> > > The  example I-CSCF and S-CSCF configurations at misc/examples/ims 
> > > contain usage of the enum module, afaik.
> > > This could be easily adapted or taken as an example.
> > >
> > > -Ursprüngliche Nachricht-
> > > Von: sr-users  Im Auftrag von
> > > BALL SUN
> > > Gesendet: Dienstag, 23. Juni 2020 08:53
> > > An: Kamailio (SER) - Users Mailing List
> > > 
> > > Betreff: [SR-Users] Kamailio IMS with ENUM
> > >
> > > Hi
> > >
> > > Is there any information that can help to setup ENUM query in
> > > Kamailio IMS? what we wanted to achieve is to when A in domain A
> > > (the local
> > > domain) makes an INVITE to B, it will check the domain of B from ENUM 
> > > backend.
> > >
> > > if it is not on the local domain, it will route to remote IMS core.
> > >
> > > Thanks
> > >
> > > - RBK
> > >
> > > ___
> > > Kamailio (SER) - Users Mailing List
> > > sr-users@lists.kamailio.org
> > > https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fli
> > > st
> > > 

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)

[Martin W Woscek]

Just to clarify what we are seeing, here is a packet from the INVITE from the 
S-CSCF (6060) to the P-CSCF(4060).  I have eliminated the S-CSCF as this packet 
looks correct.
The P-CSCF is not handling the request URI correctly, and simply sends it back 
to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri 
updated by the S-CSCF with the local IP address of the UE (192.168.20.1).  The 
P-CSCF should send this INVITE to the UE (703222888@192.168.20.1)

No. Arrival Time SourceDestination   Source Port 
Destination Port SIP to tag SIP from tag Method Record-Route Userinfo 
Request-URI Media Port Protocol Length Request-URI Host Port Owner Username 
Media Type Media Format Origin-Host Type   slice_type SIP from address User 
Part SIP from address Host Part SIP to address User Part SIP to address Host 
Part Contact URI User Part Contact URI Host Part Request-URI User Part 
Request-URI Host Part Request-URI Host Port Info
   1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128
192.168.20.12860604060
d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt
sip:703222@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP  2263   
63871 -  audio,video ITU-T G.711 
PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 
  70 asterisk.irisims.org   703222  
 ims522.irisims.org   asterisk  192.168.20.131  
  703222192.168.20.1  63871 Request: 
INVITE sip:703222@192.168.20.1:63871;transport=udp |

Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) 
on interface any, id 0
Linux cooked capture
Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128
User Datagram Protocol, Src Port: 6060, Dst Port: 4060
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:703222@192.168.20.1:63871;transport=udp SIP/2.0
Method: INVITE
Request-URI: sip:703222@192.168.20.1:63871;transport=udp
Request-URI User Part: 703222
Request-URI Host Part: 192.168.20.1
Request-URI Host Port: 63871
[Resent Packet: False]
Message Header
Record-Route: 

Route: 
Via: SIP/2.0/UDP 
192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0
Via: SIP/2.0/UDP 
192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1
Via: SIP/2.0/UDP 
192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335
From: 
;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d
To: 
Contact: 
Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660
[Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660]
CSeq: 19314 INVITE
Route: 
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, 
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 68
User-Agent: Asterisk PBX 16.6.0
Content-Type: application/sdp
Content-Length:  1121
Message Body


From: sr-users  On Behalf Of Martin W 
Woscek
Sent: Wednesday, June 24, 2020 11:21 AM
To: mico...@gmail.com; Kamailio (SER) - Users Mailing List 

Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK 
messages to Asterisk (to tag)

Hi,
The issue is the P should be sending to the UE not back to the I like the 
failed flow:
Asterisk16-[phone]  --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604  HSS user 
unknown]  to Asterisk16[phone]

The successful call should look like:
Asterisk16-[phone]  --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE]

Does anyone have a example P-cscf scripts including the mt.cfg or other script 
file that make this work?

The reverse calls work just fine.

Thanks,
_Martin

From: Daniel-Constantin Mierla mailto:mico...@gmail.com>>
Sent: Wednesday, June 24, 2020 4:53 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>; Martin W 
Woscek mailto:mwos...@mitre.org>>
Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages 
to Asterisk (to tag)


Hello,

Kamailio is not generating the 200ok for INVITE (calls), it just sends out what 
it received from end point, after consuming own Via header (of course, other 
headers can be changed based on config rules, but Kamailio is not in charge of 
setting To-tag).

Cheers,
Daniel
On 19.06.20 21:19, Martin W Woscek wrote:
Hello all,

SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an 
Asterisk registered SIP phone is successful:
[UE]-Kamailio--->INVITE--->Asterisk16-[phone]
[UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone]

But in reverse direction for a 

Re: [SR-Users] Server error on LIR select next S-CSCF

[BALL SUN]

HI

We noticed an issue here, not sure is it related to HSS profile.

the setting is like this
Caller: MSISDN-A, IMSI-A
Called: MSISDN-B, IMSI-B


Both UEs can be registered based on IMSI number. Also the
corresponding tel:MSISDN is added to the corresponding IMPU.

now, when we make a call from UE-A to UE-B, UE-A sends INVITE to
Kamailio IMS,  the R-URI is sip:MSISDN-B, and LIR from I-CSCF sends to
HSS and the check is based on the public identity tel:MSISDN-B, and it
passed, and we noticed that the INVITE send out from P-CSCF to UE-B is
with R-RUI = IMSI-B not MSISDN-B. This make the UE-B did not response
and leading to "Server error on LIR select next S-CSCF'.

anyone can help to point out the problem in where? which part of
configuration in Kamailio or FHoSS?

please help

- RBK


On Mon, Jun 22, 2020 at 12:12 PM BALL SUN  wrote:
>
> Hello
>
> Anyone has an idea?
>
>
> On Fri, Jun 19, 2020 at 5:31 PM BALL SUN  wrote:
> >
> > Hi
> >
> > I have. but still not sure why.
> >
> > On Fri, Jun 19, 2020 at 4:43 PM Pavithra Mohanraja
> >  wrote:
> > >
> > > Hi,
> > >
> > > Have you given your scscf domain in icscf db table ?
> > > Also check ur scscf is running properly
> > > Thanks,
> > > Pavithra
> > >
> > > On Fri, Jun 19, 2020 at 1:22 PM BALL SUN  wrote:
> > >>
> > >> Hi
> > >>
> > >> I have a setup using Kamailio IMS, and when trying to make a SIP call
> > >> between clients, S-CSCF returns " SIP/2.0 500 Server error on LIR
> > >> select next S-CSCF", do you have any idea on how to fix?
> > >>
> > >> - RBK
> > >>
> > >> ___
> > >> Kamailio (SER) - Users Mailing List
> > >> sr-users@lists.kamailio.org
> > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> > >
> > > ___
> > > Kamailio (SER) - Users Mailing List
> > > sr-users@lists.kamailio.org
> > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio Buildroot

[Ovidiu Sas]

I haven't cross compiled in a long time ...
For proper cross-compilation, the CROSS_COMPILE flag must be set to
something, then the LOCALBASE and SYSBASE should point to the staging
directory where the toolchain and all the libraries and include files
for the target platform are present.
Extra compile and link options can be passed via CC_EXTRA_OPTS and
LD_EXTRA_OPTS.

An old example for optware can be found here:
https://raw.githubusercontent.com/nslu2-linux/Optware/master/make/kamailio.mk

Some modules have a proper Makefile that will use the pkg-config from
staging directory and detect proper compile and link flags while other
modules require manual tinkering with the Makefile.
It's good to enable the noisy build (Q=0) and check that proper
include files and libs are used (from the staging directory and not
from the build machine).

Regards,
Ovidiu Sas

On Wed, Jun 24, 2020 at 4:52 AM Daniel-Constantin Mierla
 wrote:
>
> Hello,
>
> I haven't cross compiled myself, but there is a makefile flag CROSS_COMPILE 
> that you can set in such cases -- the Makefile of the modules sets the path 
> to includes and libs based on it. You may need to set other Makefile 
> variables to get it done, look in the src/Makefile.defs to see most of them.
>
> Ovidiu Sas seemed to be active in this space some time ago, maybe he can give 
> further guidelines.
>
> Cheers,
> Daniel
>
> On 19.06.20 14:11, Benjamin Marty wrote:
>
> Hello
>
> I would like to ask if someone already tried to build Kamailio for the 
> Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio 
> is not that easy to cross compile.
>
> Thanks
>
> Benjamin
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com

___
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[SR-Users] Sql_result_free() not cleaning properly

[Duarte Rocha]

Greetings,

I'm trying to clean a SQL result using "sql_result_free()" but i'm not
having success.

Can you please verify if i'm doing anything wrong here?

This is my code :

xerr("Pre clean [0] = Iterator 0 =
$xavp(Cli_PICKER_Default_Search_Result[0]=>Code)");
sql_result_free("Cli_PICKER_Default_Search_Result");
xerr("Pos clean [0] = Iterator 0 =
$xavp(Cli_PICKER_Default_Search_Result[0]=>Code)");

The result is the following :

ERROR: 

Re: [SR-Users] Kamailio IMS with ENUM

[Valentin Christoph]

I don't know your DNS server nor your addressing plan. Probably to modify / add 
some zone files

You must be sure to forward the ranges of telephone numbers that you want to 
forward to the other domain.

Actually, it DID the request to the DNS/ENUM, but the result was "keep the 
domain as it is" or "replace it by the own domain"

If the DNS/ENUM query did NOT succeed, then you would run into route(PSTN) and 
the request would be forwarded to "good old CS domain" to find a destination.

-Ursprüngliche Nachricht-
Von: sr-users  Im Auftrag von BALL SUN
Gesendet: Mittwoch, 24. Juni 2020 16:49
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] Kamailio IMS with ENUM

yes. when the HSS profile is not found, it did not route to DNS/ENUM to check, 
which portion of configuration should I check?

On Wed, Jun 24, 2020 at 5:31 PM Valentin Christoph 
 wrote:
>
> This means the HSS had answered the user is not known in this domain 
> (lir_term_user_unknown), but then the DNS server insists this telephone 
> number should be handled by our domain ($rd == NETWORKNAME).
>
> Just enter the user to the HSS database - if it shall be handled by 
> this domain -
>
> Or configure DNS, so that this telephone number is routed to another domain. 
> - if it shall be handled by a different domain.
>
> -Ursprüngliche Nachricht-
> Von: sr-users  Im Auftrag von 
> BALL SUN
> Gesendet: Dienstag, 23. Juni 2020 16:40
> An: Kamailio (SER) - Users Mailing List 
> Betreff: Re: [SR-Users] Kamailio IMS with ENUM
>
> Thanks. I actually keep having "604","Does not exist anywhere - HSS User 
> Unknown", so by right, I-CSCF should go to check the ENUM, but it didn't so 
> what configuration I should enable? is it dispatcher list and needed to 
> enabkle PEERING? below is the routine that related to "604","Does not exist 
> anywhere - HSS User Unknown" in icscf.
>
> - RBK
>
> route[lir_term_user_unknown]
> {
> if (uri =~ "tel:.*") {
> # Let's check, if the number can be found in ENUM:
> if(!enum_query()) {
> # ENUM failed, send it to the PSTN-Gateway:
> route(PSTN);
> break;
> }
> # ENUM resolved to another domain
> if ($rd != NETWORKNAME) {
> t_on_reply("initial_request_reply");
> t_on_failure("initial_request_failure");
> if (!t_relay()) {
> t_reply("500","Error forwarding to external domain"); exit; }; exit; } else { 
> t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; }; } else 
> { # we received a request for our domain (non-tel), but HSS said "User 
> Unknown"
> if ($rd != NETWORKNAME) {
> t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; } 
> else { # try to forward non-tel request to other domain 
> t_on_reply("Initial_Request_reply");
> t_on_failure("Initial_Request_failure");
> if (!t_relay()) {
> t_reply("500","Error forwarding to external domain"); exit; }; exit; 
> }; }; } }
>
> On Tue, Jun 23, 2020 at 8:40 PM Valentin Christoph 
>  wrote:
> >
> > The  example I-CSCF and S-CSCF configurations at misc/examples/ims contain 
> > usage of the enum module, afaik.
> > This could be easily adapted or taken as an example.
> >
> > -Ursprüngliche Nachricht-
> > Von: sr-users  Im Auftrag von 
> > BALL SUN
> > Gesendet: Dienstag, 23. Juni 2020 08:53
> > An: Kamailio (SER) - Users Mailing List 
> > 
> > Betreff: [SR-Users] Kamailio IMS with ENUM
> >
> > Hi
> >
> > Is there any information that can help to setup ENUM query in 
> > Kamailio IMS? what we wanted to achieve is to when A in domain A 
> > (the local
> > domain) makes an INVITE to B, it will check the domain of B from ENUM 
> > backend.
> >
> > if it is not on the local domain, it will route to remote IMS core.
> >
> > Thanks
> >
> > - RBK
> >
> > ___
> > Kamailio (SER) - Users Mailing List
> > sr-users@lists.kamailio.org
> > https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fli
> > st
> > s.kamailio.org%2Fcgi-bin%2Fmailman%2Flistinfo%2Fsr-usersdata=02
> > %7
> > C01%7Cchristoph.valentin%40kontron.com%7Cc6c88e565e844b41467208d8178
> > 36 
> > ada%7C8c9d3c973fd941c8a2b1646f3942daf1%7C0%7C0%7C637285200482063836&
> > am
> > p;sdata=YphpOloEosgWB9Gyt5xXH1COcafnRqMBDrLzhnRsM1o%3Dreserved=
> > 0 ___
> > Kamailio (SER) - Users Mailing List
> > sr-users@lists.kamailio.org
> > https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Fli
> > st
> > s.kamailio.org%2Fcgi-bin%2Fmailman%2Flistinfo%2Fsr-usersdata=02
> > %7
> > C01%7Cchristoph.valentin%40kontron.com%7Cc6c88e565e844b41467208d8178
> > 36 
> > ada%7C8c9d3c973fd941c8a2b1646f3942daf1%7C0%7C0%7C637285200482063836&
> > am
> > p;sdata=YphpOloEosgWB9Gyt5xXH1COcafnRqMBDrLzhnRsM1o%3Dreserved=
> > 0
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://eur04.safelinks.protection.outlook.com/?url=https%3A%2F%2Flist
> s.kamailio.org%2Fcgi-bin%2Fmailman%2Flistinfo%2Fsr-usersdata=02%7
> 

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)

[Martin W Woscek]

Hi,
The issue is the P should be sending to the UE not back to the I like the 
failed flow:
Asterisk16-[phone]  --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604  HSS user 
unknown]  to Asterisk16[phone]

The successful call should look like:
Asterisk16-[phone]  --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE]

Does anyone have a example P-cscf scripts including the mt.cfg or other script 
file that make this work?

The reverse calls work just fine.

Thanks,
_Martin

From: Daniel-Constantin Mierla 
Sent: Wednesday, June 24, 2020 4:53 AM
To: Kamailio (SER) - Users Mailing List ; Martin W 
Woscek 
Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages 
to Asterisk (to tag)


Hello,

Kamailio is not generating the 200ok for INVITE (calls), it just sends out what 
it received from end point, after consuming own Via header (of course, other 
headers can be changed based on config rules, but Kamailio is not in charge of 
setting To-tag).

Cheers,
Daniel
On 19.06.20 21:19, Martin W Woscek wrote:
Hello all,

SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an 
Asterisk registered SIP phone is successful:
[UE]-Kamailio--->INVITE--->Asterisk16-[phone]
[UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone]

But in reverse direction for a call, Kamailio does not return the SIP OK so no 
to_tag is sent so call fails to ring and complete:
[phone]-Asterisk16 --->INVITE--->Kamailio

Where the UE device never receives the INVITE, Asterisk never gets and 200 OK 
message with the to_tag from the I-CSCF, the call flow itself gets lost in 
Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 
604 HSS user unknown message is sent back to Asterisk from the I-CSCF.

Basically Im using the default "sample" configs for both the P and the I-CSCF.  
Our sauce is in the S-CSCF for out going calls that originate by a registered 
UE.

Any insight or sample Kamailio configuration that Im lacking?
Has anyone done this and could share the asterisk and Kamailio script snippets 
that make it possible.

Thanks,
_Martin



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Re: [SR-Users] Kamailio IMS using returned SRV records for transport slelection

[BALL SUN]

Hi

can I simply change all of the t_relay to t_relay_to_tcp in P-CSCF,
I-CSCF and S-CSCF?

how I can make sure it will not affect "SIP client <-> P-CSCF (either
UDP or TCP, based on the transport tag) "?



On Sun, Jun 7, 2020 at 6:21 PM Henning Westerholt  wrote:
>
> Hello,
>
> there is not a central switch to configure TCP transport etc.. But check your 
> configuration, I guess that it uses t_relay() for forwarding. You can specify 
> the wanted protocol in the t_relay call, there are special function 
> (t_relay_to_tcp() etc..) for this. 
> http://kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_relay
>
> About the SRV query - check it e.g. with tcpdump if Kamailio does the SRV 
> query. Is there is a port specified in the forwarding target, it will not do 
> it.
>
> Cheers,
>
> Henning
>
> --
> Henning Westerholt – https://skalatan.de/blog/
> Kamailio services – https://gilawa.com
>
> -Original Message-
> From: sr-users  On Behalf Of BALL SUN
> Sent: Friday, June 5, 2020 10:36 AM
> To: sr-users@lists.kamailio.org
> Subject: [SR-Users] Kamailio IMS using returned SRV records for transport 
> slelection
>
> Hi All
>
> I am new to Kamailio IMS, and I just setup a testing environment where having 
> P-CSCF, I-CSCF and S-CSCF.
>
> I would like to use make below works
> 1. SIP client <-> P-CSCF (either UDP or TCP, based on the transport tag) 2. 
> P-CSCF <-> I-CSCF (TCP by default) 3. I-CSCF <-> S-CSCF (TCP by default)
>
> For point 2, I observed that P-CSCF actually will send DNS SRV query to DNS 
> server for resolving _sip._udp. and _sip._tcp. I believed we can just disable 
> the "_sip._udp." so that P-CSCF will always use TCP to communication wih 
> I-CSCF.
>
> However, for point 3, I didnt notice I-CSCF send DNS SRV query,
>
> My question is how we can force I-CSCF to do that or is there any 
> configuration that we should enable such that I-CSCF always use TCP to S-CSCF?
>
> - RBK
>
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Re: [SR-Users] SECFILTER reload from Siremis

[Social Boh]

thank you. I'll check soon.

Regards

---
I'm SoCIaL, MayBe

On 24/06/20 3:57, Daniel-Constantin Mierla wrote:

Hello,

for the records, jsonrpc command panel has to be installed/configured:

   - http://kb.asipto.com/siremis:install53x:jsonrpc-commands

Cheers,
Daniel

On 12.06.20 21:32, Henning Westerholt wrote:

Hello,

look for the option in the GUI to execute arbitrary RPC commands. You might 
need to configure it before, if it gives only an error.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

-Original Message-
From: sr-users  On Behalf Of Social Boh
Sent: Friday, June 12, 2020 9:14 PM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] SECFILTER reload from Siremis

Hello list,

is posible to reload SECFILTER data directly from Siremis?

or the only way is:

kamcmd secfilter.reload

Regards

--
---
I'm SoCIaL, MayBe   


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Re: [SR-Users] SIP OPTIONS between UE and P-CSCF

[BALL SUN]

Hi

I can see that it is not in the IPSEC tunnel. below is the SIP OPTIONS
from two UEs with P-CSCF.

10:43:04.072360 IP 192.168.11.66.5060 > 192.168.10.42.9701: SIP:
OPTIONS sip:45400237@192.168.10.42:9700 SIP/2.0
10:43:04.072592 IP 192.168.11.66.5060 > 192.168.10.44.6602: SIP:
OPTIONS sip:45400236@192.168.10.44:6600 SIP/2.0
10:43:04.411901 IP 192.168.10.44.6602 > 192.168.11.66.5060: SIP: SIP/2.0 200 OK
10:43:04.443886 IP 192.168.10.42.9701 > 192.168.11.66.5060: SIP: SIP/2.0 200 OK

On Wed, Jun 24, 2020 at 4:48 PM Daniel-Constantin Mierla
 wrote:
>
> Hello,
>
> is Kamailio listening on other sockets towards UE other than the IPSec
> tunnel?
>
> Cheers,
> Daniel
>
> On 17.06.20 04:49, BALL SUN wrote:
> > Hi
> >
> > We have setup UE with Kamailio IMS, and after the registration is
> > successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through
> > IPSEC tunnel.
> >
> > But we found that  the SIP OPTIONS (I believed it is a health check)
> > from P-CSCF to UE is not through the IPSEC tunnel.
> >
> > our question is
> > 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside?
> > 2. if it can be configured to run inside the IPSEC tunnel, what
> > configuration can we change to achieve it?
> >
> > - RBK
> >
> > ___
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> > sr-users@lists.kamailio.org
> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>

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Re: [SR-Users] Kamailio IMS with ENUM

[BALL SUN]

yes. when the HSS profile is not found, it did not route to DNS/ENUM
to check, which portion of configuration should I check?

On Wed, Jun 24, 2020 at 5:31 PM Valentin Christoph
 wrote:
>
> This means the HSS had answered the user is not known in this domain 
> (lir_term_user_unknown), but then the DNS server insists this telephone 
> number should be handled by our domain ($rd == NETWORKNAME).
>
> Just enter the user to the HSS database - if it shall be handled by this 
> domain -
>
> Or configure DNS, so that this telephone number is routed to another domain. 
> - if it shall be handled by a different domain.
>
> -Ursprüngliche Nachricht-
> Von: sr-users  Im Auftrag von BALL SUN
> Gesendet: Dienstag, 23. Juni 2020 16:40
> An: Kamailio (SER) - Users Mailing List 
> Betreff: Re: [SR-Users] Kamailio IMS with ENUM
>
> Thanks. I actually keep having "604","Does not exist anywhere - HSS User 
> Unknown", so by right, I-CSCF should go to check the ENUM, but it didn't so 
> what configuration I should enable? is it dispatcher list and needed to 
> enabkle PEERING? below is the routine that related to "604","Does not exist 
> anywhere - HSS User Unknown" in icscf.
>
> - RBK
>
> route[lir_term_user_unknown]
> {
> if (uri =~ "tel:.*") {
> # Let's check, if the number can be found in ENUM:
> if(!enum_query()) {
> # ENUM failed, send it to the PSTN-Gateway:
> route(PSTN);
> break;
> }
> # ENUM resolved to another domain
> if ($rd != NETWORKNAME) {
> t_on_reply("initial_request_reply");
> t_on_failure("initial_request_failure");
> if (!t_relay()) {
> t_reply("500","Error forwarding to external domain"); exit; }; exit; } else { 
> t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; }; } else 
> { # we received a request for our domain (non-tel), but HSS said "User 
> Unknown"
> if ($rd != NETWORKNAME) {
> t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; } else { # 
> try to forward non-tel request to other domain 
> t_on_reply("Initial_Request_reply");
> t_on_failure("Initial_Request_failure");
> if (!t_relay()) {
> t_reply("500","Error forwarding to external domain"); exit; }; exit; }; }; } }
>
> On Tue, Jun 23, 2020 at 8:40 PM Valentin Christoph 
>  wrote:
> >
> > The  example I-CSCF and S-CSCF configurations at misc/examples/ims contain 
> > usage of the enum module, afaik.
> > This could be easily adapted or taken as an example.
> >
> > -Ursprüngliche Nachricht-
> > Von: sr-users  Im Auftrag von
> > BALL SUN
> > Gesendet: Dienstag, 23. Juni 2020 08:53
> > An: Kamailio (SER) - Users Mailing List 
> > Betreff: [SR-Users] Kamailio IMS with ENUM
> >
> > Hi
> >
> > Is there any information that can help to setup ENUM query in Kamailio
> > IMS? what we wanted to achieve is to when A in domain A (the local
> > domain) makes an INVITE to B, it will check the domain of B from ENUM 
> > backend.
> >
> > if it is not on the local domain, it will route to remote IMS core.
> >
> > Thanks
> >
> > - RBK
> >
> > ___
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Re: [SR-Users] dialog's track_cseq_updates seems to break uac_auth() in this configuration

[George Diamantopoulos]

Hello Daniel,

Thank you for the reply. Sure, I'll send one to your private email address.
Thanks!

BR,
George

On Wed, 24 Jun 2020 at 12:06, Daniel-Constantin Mierla 
wrote:

> Hello,
>
> can you send a pcap with a call reproducing such case? I can check it with
> the c code and see if something is mismatched.
>
> Cheers,
> Daniel
> On 09.06.20 13:00, George Diamantopoulos wrote:
>
> Hello again,
>
> This might turn out to be a bug, but I'm posting here first because my
> config is fairly complicated and it might not be the sanest it could be.
> The problem manifests as follows: with the sole change being the enabling
> of track_cseq_updates for the dialog module, kamailio fails to process the
> 100 Trying it receives following authentication initiated by the uac_auth()
> function. Without the track_cseq_updates option enabled, the call proceeds
> successfully, albeit with the CSeq for the auth-carrying INVITE having the
> same value as the original.
>
> Here's a simplified version of the configuration routes involved in this:
>
> request_route {
> ...
> route(DISPATCH_PROVIDER);
> ...
> }
>
> route[DISPATCH_PROVIDER] {
> if ( !ds_select_dst("10", "8") ) {
> t_send_reply("503", "Downstream carrier unavailable");
> exit;
> }
>
> t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER");
> t_on_branch("PROVIDER_BRANCH");
> t_on_failure("DISPATCH_PROVIDER_FAILURE");
>
> route(RELAY);
> exit;
> }
>
> failure_route[DISPATCH_PROVIDER_FAILURE] {
> if (t_is_canceled()) {
> exit;
> }
> if ( t_check_status("401|407") ) {
> $avp(arealm) = "authrealm.com";
> $avp(auser) = "authusername";
> $avp(apass) = "verys3kr1t";
> uac_auth();
>
> t_on_branch("PROVIDER_BRANCH");
> t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER");
> t_on_failure("DISPATCH_PROVIDER_FAILURE");
>
> $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain
> why this is here later in this e-mail
> t_relay();
> exit;
> }
> }
>
> branch_route[PROVIDER_BRANCH] {
> uac_replace_from("$dlg_var(my_new_from)");
> uac_replace_to("$dlg_var(my_new_to)");
> $rU = "my new RURI user";
> }
>
> event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] {
> if (t_is_canceled()) {
> exit;
> }
> if ( t_check_status("401|407") ) {
> return;
> # next DST - only for 5xx or local timeout
> } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() &&
> !t_branch_replied()) ) {
> if ( ds_next_dst() ) {
> t_on_branch("PROVIDER_BRANCH");
> t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER");
> t_on_failure("DISPATCH_PROVIDER_FAILURE");
> route(RELAY);
> exit;
> } else {
> xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER:
> Failed to route request to PROVIDER! Giving Up. Negative response will be
> sent upstream.\n");
> }
> }
> }
>
> One will rightfully wonder why use both branch-failure routes and
> failure_route to handle things. Well, it's not obvious from this because
> the relevant parts are removed for brevity, but I'm generally using branch
> failure event routes to try other destinations in the same destination set,
> and failure routes to additionally try other downstream "providers" in case
> all entries is the current destination set have failed. I don't think these
> actions/routes are relevant because the problem manifests without any of
> these failover mechanisms (of switching over to the next provider) engaging.
>
> Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in
> the failure_route, which might seem a little odd, this was added because
> without it, uac_auth() will send the auth-carrying INVITE always to the
> first destination in the destination set even if ds_next_dst has been
> called from the branch-failure event route. Unfortunately I haven't been
> able to determine if this actually helped, because I have only received a
> 5xx error once from the downstream peer before the "fix" and haven't been
> able to reproduce this since then. But that's another issue unrelated to
> the one I'm asking about here.
>
> So what happens with track_cseq_updates enabled is, after uac_auth()
> successfully sends the authenticated INVITE and the peer starts sending
> provisional responses, kamailio doesn't seem to acknowledge them. Instead,
> it will retransmit the auth INVITE as if there was a firewall preventing
> the 1xx responses from being admitted. Then, the timeout will engage and
> the configuration script will behave as if t_branch_timeout returns true
> (it will do ds_next_dst).
>
> So I was wondering if this is to be expected with this configuration, or
> if this should be reported as a bug. Thanks!
>
> Best regards,
> George
>
>
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Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[Pavithra Mohanraja]

Hi,

I am not sure if this issue comes in 5.3.4 version .

The below error i am facing in pcscf .
0(4152) ERROR: 

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[supreeth herle]

Please try to keep the mailing list in loop so that it helps others as well
facing this issue.

I believe this was resolved in 5.3.3 branch not sure why you are facing
this issue. Anyways, make sure that modules loaded in kamailio.cfg of PCSCF
is in following order

loadmodule "ims_dialog"
loadmodule "ims_usrloc_pcscf"
#!ifdef WITH_IPSEC
loadmodule "ims_ipsec_pcscf"
#!endif
loadmodule "ims_registrar_pcscf"

Best Regards,
Supreeth

On Wed, 24 Jun 2020 at 13:36, Pavithra Mohanraja 
wrote:

> Hi Supreeth,
>
> I am getting the below error
>
> ms_usrloc_pcscf
> [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_usrloc_pcscf.so]
>  0(455) DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]:
> mod_init(): Successfully bound to PCSCF Usrloc module
>  0(455) DEBUG:  [core/sr_module.c:632]: find_mod_export_record():
> found export of  in module ims_ipsec_pcscf
> [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_ipsec_pcscf.so]
>  0(455) ERROR: ims_ipsec_pcscf [cmd.c:85]: bind_ipsec_pcscf():
> configuration error - trying to bind to ipsec pscscf module before being
> initialized
>  0(455) ERROR:  [core/sr_module.c:849]: init_mod(): Error while
> initializing module ims_registrar_pcscf
> (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_registrar_pcscf.so)
> ERROR: error while initializing modules
>  0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:305]: mod_destroy():
> Error destroying spi generator
>  0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy():
> Error destroying port generator
>  0(455) DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start
>  0(455) DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table
>  0(455) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores
>  0(455) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists
>
>
> Kindly help
>
> On Wed, Jun 24, 2020 at 5:03 PM supreeth herle 
> wrote:
>
>> There is no other IPSec configuration needed apart from running Kamailio
>> P-CSCF and UE sending a SIP Register with Security-Client information for
>> IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using
>> Netlink library.
>>
>> Best Regards,
>> Supreeth
>>
>> On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja 
>> wrote:
>>
>>> Hi ,
>>>
>>> Could you please tell how to configure ipsec (general configuration not
>>> related to kamailio).
>>>
>>> Thanks,
>>> Pavithra
>>>
>>> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle 
>>> wrote:
>>>
 reason=404 (Not Found - destination user not found on this S-CSCF)]
>

 Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or
 any other HSS you are using to enter IMS user details). SCSCF is not able
 to find a registered user with that IMPU (i am assuming that you are
 getting this response while calling)

 Increase the debug level on S-CSCF and check out which IMPU is being
 referred by SCSCF

 Best Regards,
 Supreeth

 On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja <
 pavimohan3...@gmail.com> wrote:

> Hi,
>
> I am facing below error in kamailio-ims 5.3.4 version.
>
> reason=404 (Not Found - destination user not found on this S-CSCF)]
>
> could you please help me how should i proceed with and where the
> registration details will be stored in kamailio ims.
>
> It fails at this line in kamailio.cfg file in scscf .
>
> route[FINAL_TERM] {
> if (lookup("location")) {
> if (uri==myself) {
> if (!t_newtran()) {
> sl_reply_error();
> exit;
> }
>  xlog("Destination Not Found \n");
> t_reply("404","Not Found - destination user
> not found on this S-CSCF");
> exit;
> }
>
> here lookup(location) means what it means. Could you please help me
>
>
>
> Thanks,
> Pavithra
>
> On Thu, May 14, 2020 at 9:59 PM supreeth herle <
> herlesupre...@gmail.com> wrote:
>
>> Hello Pavithra,
>>
>> I have written a guide for how to integrate an IMS using Kamailio
>> with open5gs EPC, here is the link
>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can
>> find steps to setup rtpengine in that guide.
>>
>> Regarding requirement of rtpengine, its mandatory if you want to use
>> the scripts in that example folder without modification. Also, read the
>> READE of sipwise/rtpengine github repo, its very informative (in a gist
>> rtpengine offer much more feature than rtpproxy)
>>
>> why can't we have sip client to sip client rtps flowing like how it
>>> is happening without NAT.
>>>
>>
>> For this, i would definitely suggest to read about SIP working. This
>> is a perfect link 

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[Antony Stone]

On Wednesday 24 June 2020 at 13:29:37, Pavithra Mohanraja wrote:

> Hi ,
> 
> Could you please tell how to configure ipsec (general configuration not
> related to kamailio).

Surely this question is better asked on a Freeswan / Strongswan (or whatever 
name your operating system uses for its IPsec implementation) list - you'll 
get far more expertise there.


Regards,


Antony.

-- 
I conclude that there are two ways of constructing a software design: One way 
is to make it so simple that there are _obviously_ no deficiencies, and the 
other way is to make it so complicated that there are no _obvious_ 
deficiencies.

 - C A R Hoare

   Please reply to the list;
 please *don't* CC me.

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Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[supreeth herle]

There is no other IPSec configuration needed apart from running Kamailio
P-CSCF and UE sending a SIP Register with Security-Client information for
IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using
Netlink library.

Best Regards,
Supreeth

On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja 
wrote:

> Hi ,
>
> Could you please tell how to configure ipsec (general configuration not
> related to kamailio).
>
> Thanks,
> Pavithra
>
> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle 
> wrote:
>
>> reason=404 (Not Found - destination user not found on this S-CSCF)]
>>>
>>
>> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or any
>> other HSS you are using to enter IMS user details). SCSCF is not able to
>> find a registered user with that IMPU (i am assuming that you are getting
>> this response while calling)
>>
>> Increase the debug level on S-CSCF and check out which IMPU is being
>> referred by SCSCF
>>
>> Best Regards,
>> Supreeth
>>
>> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja 
>> wrote:
>>
>>> Hi,
>>>
>>> I am facing below error in kamailio-ims 5.3.4 version.
>>>
>>> reason=404 (Not Found - destination user not found on this S-CSCF)]
>>>
>>> could you please help me how should i proceed with and where the
>>> registration details will be stored in kamailio ims.
>>>
>>> It fails at this line in kamailio.cfg file in scscf .
>>>
>>> route[FINAL_TERM] {
>>> if (lookup("location")) {
>>> if (uri==myself) {
>>> if (!t_newtran()) {
>>> sl_reply_error();
>>> exit;
>>> }
>>>  xlog("Destination Not Found \n");
>>> t_reply("404","Not Found - destination user not
>>> found on this S-CSCF");
>>> exit;
>>> }
>>>
>>> here lookup(location) means what it means. Could you please help me
>>>
>>>
>>>
>>> Thanks,
>>> Pavithra
>>>
>>> On Thu, May 14, 2020 at 9:59 PM supreeth herle 
>>> wrote:
>>>
 Hello Pavithra,

 I have written a guide for how to integrate an IMS using Kamailio with
 open5gs EPC, here is the link
 https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can
 find steps to setup rtpengine in that guide.

 Regarding requirement of rtpengine, its mandatory if you want to use
 the scripts in that example folder without modification. Also, read the
 READE of sipwise/rtpengine github repo, its very informative (in a gist
 rtpengine offer much more feature than rtpproxy)

 why can't we have sip client to sip client rtps flowing like how it is
> happening without NAT.
>

 For this, i would definitely suggest to read about SIP working. This is
 a perfect link https://github.com/onmyway133/blog/issues/284 which
 offers extensive material (short and very interesting to read with
 examples) related to SIP.

 Best Regards,
 Supreeth


 On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja <
 pavimohan3...@gmail.com> wrote:

> Hi ,
>
> I didn't configure rtpengine ..instead of that only I used
> rrpproxy..will it not work?
>
> If not please tell me how to configure rtpengine and also tell me if
> it is really madatory .. why can't we have sip client to sip client rtps
> flowing like how it is happening without NAT.
>
> That's my doubt ..please clarify ..
>
> Thanks.
>
> On Thu, May 14, 2020, 8:57 PM supreeth herle 
> wrote:
>
>> HI,
>>
>> Hold on a moment,
>>
>>
>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in pcscf.cfg
>>> (!#define WITH_NAT) and configured rtpproxy)
>>> 10.40.10.12 - icscf in port 4070
>>> 10.40.10.5 - scscf in port 4080
>>> 10.40.10.30 - hss
>>>
>>
>> So i assume you are using the examples provided in the ims folder of
>> kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have 
>> you
>> modified the P-CSCF configuration to work with RTPProxy?
>>
>> Best regards,
>> Supreeth
>>
>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja <
>> pavimohan3...@gmail.com> wrote:
>>
>>> Hi ,
>>> I have attached working rtp pcap file. But it is getting reverted
>>> back to me .
>>>
>>> As you said , i have tried restarting rtpproxy several times. but it
>>> didnt work .
>>>
>>>
>>>
>>> On Thu, May 14, 2020 at 7:11 PM supreeth herle <
>>> herlesupre...@gmail.com> wrote:
>>>
 Thanks for the pcap. Sorry, to ask this but can you also send pcap
 of scenario where you see RTP packets?

 Also, did you restart the rtpproxy after altering the
 /etc/default/rtpproxy ?

 i.e.

 $ systemctl restart rtpproxy

 Best Regards,
 

Re: [SR-Users] multiple domains with microsoft team

[Alexey Vasilyev]

Hi
Reusing of TLS ok only in the case if you have the wildcard certificate and
all your domains are subdomains.
Otherwise it will not work. MS checks if RR FQDN matches your TLS
certificate.

ср, 24 июн. 2020 г. в 11:14, Daniel-Constantin Mierla :

> Hello,
>
> MS Teams uses the Record-Route header to match the PBX/group domain. In my
> experience testing with MS Teams so far, reusing the TLS connection is ok
> when having multiple domains.
>
> Cheers,
> Daniel
> On 12.06.20 22:28, Leanderson Correia wrote:
>
> I'm testing here to use multiple domains with microsoft teams and I'm
> having trouble making and receiving calls. It seems that when connecting a
> call, kamailio keeps the tls connection active and reuses it, making the
> domain parameters used the wrong ones for new call. Can someone help me, or
> give a tip?
>
> --
> Att,
> Lean
>
>
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>
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> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
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>
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-- 
Best regards
Alexey Vasilyev
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Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[Pavithra Mohanraja]

Hi ,

Could you please tell how to configure ipsec (general configuration not
related to kamailio).

Thanks,
Pavithra

On Wed, Jun 24, 2020 at 2:12 PM supreeth herle 
wrote:

> reason=404 (Not Found - destination user not found on this S-CSCF)]
>>
>
> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or any
> other HSS you are using to enter IMS user details). SCSCF is not able to
> find a registered user with that IMPU (i am assuming that you are getting
> this response while calling)
>
> Increase the debug level on S-CSCF and check out which IMPU is being
> referred by SCSCF
>
> Best Regards,
> Supreeth
>
> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja 
> wrote:
>
>> Hi,
>>
>> I am facing below error in kamailio-ims 5.3.4 version.
>>
>> reason=404 (Not Found - destination user not found on this S-CSCF)]
>>
>> could you please help me how should i proceed with and where the
>> registration details will be stored in kamailio ims.
>>
>> It fails at this line in kamailio.cfg file in scscf .
>>
>> route[FINAL_TERM] {
>> if (lookup("location")) {
>> if (uri==myself) {
>> if (!t_newtran()) {
>> sl_reply_error();
>> exit;
>> }
>>  xlog("Destination Not Found \n");
>> t_reply("404","Not Found - destination user not
>> found on this S-CSCF");
>> exit;
>> }
>>
>> here lookup(location) means what it means. Could you please help me
>>
>>
>>
>> Thanks,
>> Pavithra
>>
>> On Thu, May 14, 2020 at 9:59 PM supreeth herle 
>> wrote:
>>
>>> Hello Pavithra,
>>>
>>> I have written a guide for how to integrate an IMS using Kamailio with
>>> open5gs EPC, here is the link
>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can find
>>> steps to setup rtpengine in that guide.
>>>
>>> Regarding requirement of rtpengine, its mandatory if you want to use the
>>> scripts in that example folder without modification. Also, read the READE
>>> of sipwise/rtpengine github repo, its very informative (in a gist rtpengine
>>> offer much more feature than rtpproxy)
>>>
>>> why can't we have sip client to sip client rtps flowing like how it is
 happening without NAT.

>>>
>>> For this, i would definitely suggest to read about SIP working. This is
>>> a perfect link https://github.com/onmyway133/blog/issues/284 which
>>> offers extensive material (short and very interesting to read with
>>> examples) related to SIP.
>>>
>>> Best Regards,
>>> Supreeth
>>>
>>>
>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja <
>>> pavimohan3...@gmail.com> wrote:
>>>
 Hi ,

 I didn't configure rtpengine ..instead of that only I used
 rrpproxy..will it not work?

 If not please tell me how to configure rtpengine and also tell me if it
 is really madatory .. why can't we have sip client to sip client rtps
 flowing like how it is happening without NAT.

 That's my doubt ..please clarify ..

 Thanks.

 On Thu, May 14, 2020, 8:57 PM supreeth herle 
 wrote:

> HI,
>
> Hold on a moment,
>
>
>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in pcscf.cfg
>> (!#define WITH_NAT) and configured rtpproxy)
>> 10.40.10.12 - icscf in port 4070
>> 10.40.10.5 - scscf in port 4080
>> 10.40.10.30 - hss
>>
>
> So i assume you are using the examples provided in the ims folder of
> kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have 
> you
> modified the P-CSCF configuration to work with RTPProxy?
>
> Best regards,
> Supreeth
>
> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja <
> pavimohan3...@gmail.com> wrote:
>
>> Hi ,
>> I have attached working rtp pcap file. But it is getting reverted
>> back to me .
>>
>> As you said , i have tried restarting rtpproxy several times. but it
>> didnt work .
>>
>>
>>
>> On Thu, May 14, 2020 at 7:11 PM supreeth herle <
>> herlesupre...@gmail.com> wrote:
>>
>>> Thanks for the pcap. Sorry, to ask this but can you also send pcap
>>> of scenario where you see RTP packets?
>>>
>>> Also, did you restart the rtpproxy after altering the
>>> /etc/default/rtpproxy ?
>>>
>>> i.e.
>>>
>>> $ systemctl restart rtpproxy
>>>
>>> Best Regards,
>>> Supreeth
>>>
>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja <
>>> pavimohan3...@gmail.com> wrote:
>>>
 Hi Supreeth,
 I have attached pcap files . kindly check and let me know.
 filter of rtp packets and check.

 *filter : sip || rtp*


 On Thu, May 14, 2020 at 5:15 PM supreeth herle <
 herlesupre...@gmail.com> wrote:

> Hi Pavithra,
>
> Thanks for the 

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[Pavithra Mohanraja]

Hi Supreeth,

I have registered the user bob .. and it is already stored in hss database
also . Registration is successful with that user .

When i make a call from alice to bob .. in call establishment , it is
throwing this error.

In scscf logs , I am able to see registered user not found. But in hss , it
is there .

In kamailio.cfg file in scscf, as i already mentioned, it is checking for
the condition ,

if(lookup(location))

I am not sure what it tries to do..

Kindly help .


Thanks,
Pavithra

On Wed, Jun 24, 2020 at 2:12 PM supreeth herle 
wrote:

> reason=404 (Not Found - destination user not found on this S-CSCF)]
>>
>
> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or any
> other HSS you are using to enter IMS user details). SCSCF is not able to
> find a registered user with that IMPU (i am assuming that you are getting
> this response while calling)
>
> Increase the debug level on S-CSCF and check out which IMPU is being
> referred by SCSCF
>
> Best Regards,
> Supreeth
>
> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja 
> wrote:
>
>> Hi,
>>
>> I am facing below error in kamailio-ims 5.3.4 version.
>>
>> reason=404 (Not Found - destination user not found on this S-CSCF)]
>>
>> could you please help me how should i proceed with and where the
>> registration details will be stored in kamailio ims.
>>
>> It fails at this line in kamailio.cfg file in scscf .
>>
>> route[FINAL_TERM] {
>> if (lookup("location")) {
>> if (uri==myself) {
>> if (!t_newtran()) {
>> sl_reply_error();
>> exit;
>> }
>>  xlog("Destination Not Found \n");
>> t_reply("404","Not Found - destination user not
>> found on this S-CSCF");
>> exit;
>> }
>>
>> here lookup(location) means what it means. Could you please help me
>>
>>
>>
>> Thanks,
>> Pavithra
>>
>> On Thu, May 14, 2020 at 9:59 PM supreeth herle 
>> wrote:
>>
>>> Hello Pavithra,
>>>
>>> I have written a guide for how to integrate an IMS using Kamailio with
>>> open5gs EPC, here is the link
>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can find
>>> steps to setup rtpengine in that guide.
>>>
>>> Regarding requirement of rtpengine, its mandatory if you want to use the
>>> scripts in that example folder without modification. Also, read the READE
>>> of sipwise/rtpengine github repo, its very informative (in a gist rtpengine
>>> offer much more feature than rtpproxy)
>>>
>>> why can't we have sip client to sip client rtps flowing like how it is
 happening without NAT.

>>>
>>> For this, i would definitely suggest to read about SIP working. This is
>>> a perfect link https://github.com/onmyway133/blog/issues/284 which
>>> offers extensive material (short and very interesting to read with
>>> examples) related to SIP.
>>>
>>> Best Regards,
>>> Supreeth
>>>
>>>
>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja <
>>> pavimohan3...@gmail.com> wrote:
>>>
 Hi ,

 I didn't configure rtpengine ..instead of that only I used
 rrpproxy..will it not work?

 If not please tell me how to configure rtpengine and also tell me if it
 is really madatory .. why can't we have sip client to sip client rtps
 flowing like how it is happening without NAT.

 That's my doubt ..please clarify ..

 Thanks.

 On Thu, May 14, 2020, 8:57 PM supreeth herle 
 wrote:

> HI,
>
> Hold on a moment,
>
>
>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in pcscf.cfg
>> (!#define WITH_NAT) and configured rtpproxy)
>> 10.40.10.12 - icscf in port 4070
>> 10.40.10.5 - scscf in port 4080
>> 10.40.10.30 - hss
>>
>
> So i assume you are using the examples provided in the ims folder of
> kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have 
> you
> modified the P-CSCF configuration to work with RTPProxy?
>
> Best regards,
> Supreeth
>
> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja <
> pavimohan3...@gmail.com> wrote:
>
>> Hi ,
>> I have attached working rtp pcap file. But it is getting reverted
>> back to me .
>>
>> As you said , i have tried restarting rtpproxy several times. but it
>> didnt work .
>>
>>
>>
>> On Thu, May 14, 2020 at 7:11 PM supreeth herle <
>> herlesupre...@gmail.com> wrote:
>>
>>> Thanks for the pcap. Sorry, to ask this but can you also send pcap
>>> of scenario where you see RTP packets?
>>>
>>> Also, did you restart the rtpproxy after altering the
>>> /etc/default/rtpproxy ?
>>>
>>> i.e.
>>>
>>> $ systemctl restart rtpproxy
>>>
>>> Best Regards,
>>> Supreeth
>>>
>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja <
>>> 

Re: [SR-Users] Kamailio IMS with ENUM

[Valentin Christoph]

This means the HSS had answered the user is not known in this domain 
(lir_term_user_unknown), but then the DNS server insists this telephone number 
should be handled by our domain ($rd == NETWORKNAME).

Just enter the user to the HSS database - if it shall be handled by this domain 
-

Or configure DNS, so that this telephone number is routed to another domain. - 
if it shall be handled by a different domain.

-Ursprüngliche Nachricht-
Von: sr-users  Im Auftrag von BALL SUN
Gesendet: Dienstag, 23. Juni 2020 16:40
An: Kamailio (SER) - Users Mailing List 
Betreff: Re: [SR-Users] Kamailio IMS with ENUM

Thanks. I actually keep having "604","Does not exist anywhere - HSS User 
Unknown", so by right, I-CSCF should go to check the ENUM, but it didn't so 
what configuration I should enable? is it dispatcher list and needed to enabkle 
PEERING? below is the routine that related to "604","Does not exist anywhere - 
HSS User Unknown" in icscf.

- RBK

route[lir_term_user_unknown]
{
if (uri =~ "tel:.*") {
# Let's check, if the number can be found in ENUM:
if(!enum_query()) {
# ENUM failed, send it to the PSTN-Gateway:
route(PSTN);
break;
}
# ENUM resolved to another domain
if ($rd != NETWORKNAME) {
t_on_reply("initial_request_reply");
t_on_failure("initial_request_failure");
if (!t_relay()) {
t_reply("500","Error forwarding to external domain"); exit; }; exit; } else { 
t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; }; } else { 
# we received a request for our domain (non-tel), but HSS said "User Unknown"
if ($rd != NETWORKNAME) {
t_reply("604","Does not exist anywhere - HSS User Unknown"); exit; } else { # 
try to forward non-tel request to other domain 
t_on_reply("Initial_Request_reply");
t_on_failure("Initial_Request_failure");
if (!t_relay()) {
t_reply("500","Error forwarding to external domain"); exit; }; exit; }; }; } }

On Tue, Jun 23, 2020 at 8:40 PM Valentin Christoph 
 wrote:
>
> The  example I-CSCF and S-CSCF configurations at misc/examples/ims contain 
> usage of the enum module, afaik.
> This could be easily adapted or taken as an example.
>
> -Ursprüngliche Nachricht-
> Von: sr-users  Im Auftrag von 
> BALL SUN
> Gesendet: Dienstag, 23. Juni 2020 08:53
> An: Kamailio (SER) - Users Mailing List 
> Betreff: [SR-Users] Kamailio IMS with ENUM
>
> Hi
>
> Is there any information that can help to setup ENUM query in Kamailio 
> IMS? what we wanted to achieve is to when A in domain A (the local
> domain) makes an INVITE to B, it will check the domain of B from ENUM backend.
>
> if it is not on the local domain, it will route to remote IMS core.
>
> Thanks
>
> - RBK
>
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Re: [SR-Users] dialog's track_cseq_updates seems to break uac_auth() in this configuration

[Daniel-Constantin Mierla]

Hello,

can you send a pcap with a call reproducing such case? I can check it
with the c code and see if something is mismatched.

Cheers,
Daniel

On 09.06.20 13:00, George Diamantopoulos wrote:
> Hello again,
>
> This might turn out to be a bug, but I'm posting here first because my
> config is fairly complicated and it might not be the sanest it could
> be. The problem manifests as follows: with the sole change being the
> enabling of track_cseq_updates for the dialog module, kamailio fails
> to process the 100 Trying it receives following authentication
> initiated by the uac_auth() function. Without the track_cseq_updates
> option enabled, the call proceeds successfully, albeit with the CSeq
> for the auth-carrying INVITE having the same value as the original.
>
> Here's a simplified version of the configuration routes involved in this:
>
> request_route {
>     ...
>     route(DISPATCH_PROVIDER);
>     ...
> }
>
> route[DISPATCH_PROVIDER] {
>     if ( !ds_select_dst("10", "8") ) {
>         t_send_reply("503", "Downstream carrier unavailable");
>         exit;
>     }
>
>     t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER");
>     t_on_branch("PROVIDER_BRANCH");
>     t_on_failure("DISPATCH_PROVIDER_FAILURE");
>
>     route(RELAY);
>     exit;
> }
>
> failure_route[DISPATCH_PROVIDER_FAILURE] {
>     if (t_is_canceled()) {
>         exit;
>     }
>     if ( t_check_status("401|407") ) {
>         $avp(arealm) = "authrealm.com ";
>         $avp(auser) = "authusername";
>         $avp(apass) = "verys3kr1t";
>         uac_auth();
>
>         t_on_branch("PROVIDER_BRANCH");
>         t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER");
>         t_on_failure("DISPATCH_PROVIDER_FAILURE");
>
>         $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll
> explain why this is here later in this e-mail
>         t_relay();
>         exit;
>     }
> }
>
> branch_route[PROVIDER_BRANCH] {
>     uac_replace_from("$dlg_var(my_new_from)");
>     uac_replace_to("$dlg_var(my_new_to)");
>     $rU = "my new RURI user";
> }
>
> event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] {
>     if (t_is_canceled()) {
>         exit;
>     }
>     if ( t_check_status("401|407") ) {
>         return;
>     # next DST - only for 5xx or local timeout
>     } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout()
> && !t_branch_replied()) ) {
>         if ( ds_next_dst() ) {
>             t_on_branch("PROVIDER_BRANCH");
>             t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER");
>             t_on_failure("DISPATCH_PROVIDER_FAILURE");
>             route(RELAY);
>             exit;
>         } else {
>             xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER:
> Failed to route request to PROVIDER! Giving Up. Negative response will
> be sent upstream.\n");
>         }
>     }
> }
>
> One will rightfully wonder why use both branch-failure routes and
> failure_route to handle things. Well, it's not obvious from this
> because the relevant parts are removed for brevity, but I'm generally
> using branch failure event routes to try other destinations in the
> same destination set, and failure routes to additionally try other
> downstream "providers" in case all entries is the current destination
> set have failed. I don't think these actions/routes are relevant
> because the problem manifests without any of these failover mechanisms
> (of switching over to the next provider) engaging.
>
> Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);"
> in the failure_route, which might seem a little odd, this was added
> because without it, uac_auth() will send the auth-carrying INVITE
> always to the first destination in the destination set even if
> ds_next_dst has been called from the branch-failure event route.
> Unfortunately I haven't been able to determine if this actually
> helped, because I have only received a 5xx error once from the
> downstream peer before the "fix" and haven't been able to reproduce
> this since then. But that's another issue unrelated to the one I'm
> asking about here.
>
> So what happens with track_cseq_updates enabled is, after uac_auth()
> successfully sends the authenticated INVITE and the peer starts
> sending provisional responses, kamailio doesn't seem to acknowledge
> them. Instead, it will retransmit the auth INVITE as if there was a
> firewall preventing the 1xx responses from being admitted. Then, the
> timeout will engage and the configuration script will behave as if
> t_branch_timeout returns true (it will do ds_next_dst).
>
> So I was wondering if this is to be expected with this configuration,
> or if this should be reported as a bug. Thanks!
>
> Best regards,
> George
>
>
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Re: [SR-Users] Presence and call queueing - Kamailio and Asterisk

[Daniel-Constantin Mierla]

Hello,

I suggest to run kamailio with debug=3 in kamailio.cfg to see if the
ringing state is published by dialog/pua_dialoginfo during the call setup.

On the other hand, the dialoginfo is a different event than user/device
presence state, so the endpoints have to subscribe for both dialog info
and presence states to get notifications on both cases. Eventually you
can enhance the config (or code) to generate presence states on
dialog-info events -- iirc, with pua (or pua_json), you can push any
PUBLISH (or presence states).

Cheers,
Daniel

On 11.06.20 18:43, Pover A.V. wrote:
>
> Summary: Using Kamailio to load balance between Asterisk servers.
> Trouble getting call queuing working because can’t get device state
> hints to work reliably.
>
> —
>
> Hello
>
>  
>
> I’d be grateful for any advice that can be offered for this problem to
> do with presence and call queueing.
>
>  
>
> I have 3 Kamailio (v5.3.4) servers that load-balance incoming traffic
> from various sources: a legacy PBX, soft phone clients, and a SIP
> trunking service.
>
>  
>
> Sitting behind the 3 Kamailio servers are 2 Asterisk (v16 LTS) servers.
>
>  
>
> Kamailio acts as a registrar and user location service. Asterisk is
> routing calls and (should be) providing device state information for
> each extension.
>
>  
>
> An external script builds a list of PJSIP endpoints based on the
> content of the Kamailio subscriber table, and deploys it to the
> Asterisk servers whenever a change is detected. The AOR for each
> record is configured to point to Kamailio. No authentication details
> are attached to these endpoints, as Kamailio is doing authentication
> before dispatching a request to Asterisk.
>
>  
>
> Whenever Kamailio receives an INVITE, it passes this to an Asterisk
> server using the dispatcher module (round-robin algorithm ‘4’);
> asterisk then processes the request.
>
>  
>
> At the moment, everything is working except for presence/device states
> and call queueing.
>
>  
>
> I’ve tried to set up ‘presence’, ‘pua’, ‘dialog’ and
> ‘(presence/pua)_dialoginfo’ modules on Kamailio with limited success.
> The presence state when a user toggles ‘do not disturb’ on their soft
> phone is passed to other users who are subscribed (watching), however
> when a user makes a call, no presence updates are seen by subscribed
> users (the extensions show as ‘Idle’ or ‘Available’ instead of ‘In
> Use’ or ‘Busy’).
>
>  
>
> Asterisk is able to report that an extension is ‘Ringing’ or ‘InUse’
> when a call is made to that extension, because an INVITE to an
> internal extension matches a pattern in the dialplan and Asterisk
> routes it to a PJSIP endpoint with the same name. However, when
> somebody makes a call, Asterisk is unable to track the state of the
> extension making the call because the call is going via Kamailio so
> Asterisk is not seeing the caller as an internal extension. We need a
> way to make Asterisk think the call is coming from an internal device
> if the caller ID matches a PJSIP endpoint.
>
>  
>
> Both Asterisk servers are configured to share device state updates
> with each other, and this is working.
>
>  
>
> Without being able to see device state for an internal calling party,
> we will be unable to reliably set up call queues in Asterisk.
>
>  
>
> Using the ‘mohqueue’ module on Kamailio is not something we want to do
> because of its reliance on a database, which is creating a single
> point of failure – if the database server is down for any reason,
> Kamailio refuses to start.
>
>  
>
> Ideally, we would achieve some form of distributed call queueing,
> however the simplest option seems to be setting up a queue on one
> Asterisk server and having an active-passive high availability
> configuration, should a server go down.
>
>  
>
> I’m looking for advice on how best to keep track of what is happening
> in a call, and for Asterisk to know the states of each extension and
> make these available in a dialplan hint. Or any other suggestions that
> might help to achieve call queueing in this setup.
>
>  
>
> Thank you
>
>  
>
>
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Re: [SR-Users] Dialog Profiles

[Ilie Soltanici]

Very good idea with rpc command or json module, never mind about it.
For the moment I will try with that, but any way will be good to have a
function in the dialog module for that.

Thanks.

În mie., 24 iun. 2020 la 09:45, Daniel-Constantin Mierla 
a scris:

> Hello,
>
> if there is a rpc command for it, then maybe you can leverage the
> jsonrpc_exec() from jsonrpcs module and walk through json using jansson.
>
> If not, then such feature has to be added within the c code and exported
> as function to config.
>
> Cheers,
> Daniel
> On 18.06.20 13:44, Ilie Soltanici wrote:
>
> Hello,
>
> Is there any way in Kamailio to get the profile values the current dialog
> is part of?
>
> For example, I have one dialog with the following profiles:
>
> profiles: {
> {
> test: abcd
> }
> test: 1234
> }
>
> I need in the reply route to get the values for the test dialog profile.
> (abcd, 1234). I know I can use get_profile_size, but that will give the
> number of profiles of the current dialog, but I need the values to loop
> through.
>
> Thanks.
>
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>
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> www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
>
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Re: [SR-Users] SECFILTER reload from Siremis

[Daniel-Constantin Mierla]

Hello,

for the records, jsonrpc command panel has to be installed/configured:

  - http://kb.asipto.com/siremis:install53x:jsonrpc-commands

Cheers,
Daniel

On 12.06.20 21:32, Henning Westerholt wrote:
> Hello,
>
> look for the option in the GUI to execute arbitrary RPC commands. You might 
> need to configure it before, if it gives only an error.
>
> Cheers,
>
> Henning
>
> -- 
> Henning Westerholt – https://skalatan.de/blog/
> Kamailio services – https://gilawa.com 
>
> -Original Message-
> From: sr-users  On Behalf Of Social Boh
> Sent: Friday, June 12, 2020 9:14 PM
> To: sr-users@lists.kamailio.org
> Subject: [SR-Users] SECFILTER reload from Siremis
>
> Hello list,
>
> is posible to reload SECFILTER data directly from Siremis?
>
> or the only way is:
>
> kamcmd secfilter.reload
>
> Regards
>
> -- 
> ---
> I'm SoCIaL, MayBe 
>
>
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Re: [SR-Users] multiple domains with microsoft team

[Daniel-Constantin Mierla]

Hello,

MS Teams uses the Record-Route header to match the PBX/group domain. In
my experience testing with MS Teams so far, reusing the TLS connection
is ok when having multiple domains.

Cheers,
Daniel

On 12.06.20 22:28, Leanderson Correia wrote:
> I'm testing here to use multiple domains with microsoft teams and I'm
> having trouble making and receiving calls. It seems that when
> connecting a call, kamailio keeps the tls connection active and reuses
> it, making the domain parameters used the wrong ones for new call. Can
> someone help me, or give a tip?
>
> -- 
> Att,
> Lean
>
>
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Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)

[Daniel-Constantin Mierla]

Hello,

Kamailio is not generating the 200ok for INVITE (calls), it just sends
out what it received from end point, after consuming own Via header (of
course, other headers can be changed based on config rules, but Kamailio
is not in charge of setting To-tag).

Cheers,
Daniel

On 19.06.20 21:19, Martin W Woscek wrote:
>
> Hello all,
>
>  
>
> SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to
> an Asterisk registered SIP phone is successful:
>
> [UE]-Kamailio--->INVITE--->Asterisk16-[phone]
>
> [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone]
>
>  
>
> But in reverse direction for a call, Kamailio does not return the SIP
> OK so no to_tag is sent so call fails to ring and complete:
>
> [phone]-Asterisk16 --->INVITE--->Kamailio
>
>  
>
> Where the UE device never receives the INVITE, Asterisk never gets and
> 200 OK message with the to_tag from the I-CSCF, the call flow itself
> gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF
> and and ultimately a 604 HSS user unknown message is sent back to
> Asterisk from the I-CSCF.
>
>  
>
> Basically Im using the default “sample” configs for both the P and the
> I-CSCF.  Our sauce is in the S-CSCF for out going calls that originate
> by a registered UE.
>
>  
>
> Any insight or sample Kamailio configuration that Im lacking? 
>
> Has anyone done this and could share the asterisk and Kamailio script
> snippets that make it possible.
>
>  
>
> Thanks,
>
> _Martin
>
>
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Re: [SR-Users] Kamailio ims video calls

[Daniel-Constantin Mierla]

Hello,

Kamailio is agnostic of the media content type. If both endpoints
support video, then the call just works. The rtpproxy or rtpengine is
needed if you have to deal with nat traversal.

For voice conferencing you need sip media servers such as asterisk,
freeswitch or sems (to list a few from open source space).

Cheers,
Daniel

On 14.06.20 18:03, Pavithra Mohanraja wrote:
> Hi,
> Could anyone please give me some idea on how video call works in IMS. 
> Does it need rtp proxy to work?
> Or is it the same way how voice call works  ?
> It would be good , if anybody share some links on video calls as well
> voice conferencing.
>
> Thanks,
> pavithra
>
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Re: [SR-Users] SIP OPTIONS between UE and P-CSCF

[Daniel-Constantin Mierla]

Hello,

is Kamailio listening on other sockets towards UE other than the IPSec
tunnel?

Cheers,
Daniel

On 17.06.20 04:49, BALL SUN wrote:
> Hi
>
> We have setup UE with Kamailio IMS, and after the registration is
> successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through
> IPSEC tunnel.
>
> But we found that  the SIP OPTIONS (I believed it is a health check)
> from P-CSCF to UE is not through the IPSEC tunnel.
>
> our question is
> 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside?
> 2. if it can be configured to run inside the IPSEC tunnel, what
> configuration can we change to achieve it?
>
> - RBK
>
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Re: [SR-Users] kamailio 5.2.2 enabling hashing_type

[Daniel-Constantin Mierla]

Hello,

I do not work with IMS modules, but if you find some inconsitency
between the code and the documentation, just make a PR to fix the
documentation.

Cheers,
Daniel

On 17.06.20 14:46, Martin W Woscek wrote:
>
> Hello,
>
>  
>
> I’m revisiting some initial issues I had with P-CSCF.  I could not set
> the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it
> out and moved on. 
>
>  
>
> But now, I may need it as the P-CSCF is not handling INVITES coming
> from I-CSCF that originated  from external SIP server, where the
> endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE
> back to the S rather that the registered UE and fails.
>
>  
>
> kamailio.cfg:
>
> modparam("ims_usrloc_pcscf", "hashing_type", 1)
>
>  
>
> So with that, when I set the hashing_type to 1,1 or 2 I get the
> following starting the pcscf:
>
> Starting the P-CSCF:
>
> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]:  * Not starting Kamailio
> IMS P-CSCF: invalid configuration file!
> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]:  *
> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]:  *  0(5766) ERROR: 
> [core/modparam.c:140]: set_mod_param_regex(): parameter 
> of type <2> not found in module 
> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]:  0(5766) CRITICAL: 
> [core/cfg.y:3510]: yyerror_at(): parse error in config file
> /etc/kamailio/pcscf/kamailio.cfg, line 364, column 47: Can't set
> module parameter
>
>  
>
> Am I missing some module that I did not include at compile time? 
> According to the docs, it has no additional dependencies other than
> the ims_usrloc_pcscf.so
>
> https://kamailio.org/docs/modules/5.0.x/modules/ims_usrloc_pcscf.html#idp35204148
>   
>
>  
>
> Any help is appreciated.
>
>  
>
> Thanks,
>
> _Martin
>
>
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Re: [SR-Users] Dialog Profiles

[Daniel-Constantin Mierla]

Hello,

if there is a rpc command for it, then maybe you can leverage the
jsonrpc_exec() from jsonrpcs module and walk through json using jansson.

If not, then such feature has to be added within the c code and exported
as function to config.

Cheers,
Daniel

On 18.06.20 13:44, Ilie Soltanici wrote:
> Hello,
>
> Is there any way in Kamailio to get the profile values the current
> dialog is part of?
>
> For example, I have one dialog with the following profiles:
>
> profiles: {
> {
>     test: abcd
> }
>     test: 1234
> }
>
> I need in the reply route to get the values for the test dialog
> profile. (abcd, 1234). I know I can use get_profile_size, but that
> will give the number of profiles of the current dialog, but I need the
> values to loop through.
>
> Thanks.
>
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Re: [SR-Users] Kamailio Buildroot

[Daniel-Constantin Mierla]

Hello,

I haven't cross compiled myself, but there is a makefile flag
CROSS_COMPILE that you can set in such cases -- the Makefile of the
modules sets the path to includes and libs based on it. You may need to
set other Makefile variables to get it done, look in the
src/Makefile.defs to see most of them.

Ovidiu Sas seemed to be active in this space some time ago, maybe he can
give further guidelines.

Cheers,
Daniel

On 19.06.20 14:11, Benjamin Marty wrote:
> Hello
>
> I would like to ask if someone already tried to build Kamailio for the
> Embedded Distribution Buildroot. I tried it myself but it looks like
> Kamailio is not that easy to cross compile.
>
> Thanks
>
> Benjamin
>
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Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[supreeth herle]

>
> reason=404 (Not Found - destination user not found on this S-CSCF)]
>

Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or any
other HSS you are using to enter IMS user details). SCSCF is not able to
find a registered user with that IMPU (i am assuming that you are getting
this response while calling)

Increase the debug level on S-CSCF and check out which IMPU is being
referred by SCSCF

Best Regards,
Supreeth

On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja 
wrote:

> Hi,
>
> I am facing below error in kamailio-ims 5.3.4 version.
>
> reason=404 (Not Found - destination user not found on this S-CSCF)]
>
> could you please help me how should i proceed with and where the
> registration details will be stored in kamailio ims.
>
> It fails at this line in kamailio.cfg file in scscf .
>
> route[FINAL_TERM] {
> if (lookup("location")) {
> if (uri==myself) {
> if (!t_newtran()) {
> sl_reply_error();
> exit;
> }
>  xlog("Destination Not Found \n");
> t_reply("404","Not Found - destination user not
> found on this S-CSCF");
> exit;
> }
>
> here lookup(location) means what it means. Could you please help me
>
>
>
> Thanks,
> Pavithra
>
> On Thu, May 14, 2020 at 9:59 PM supreeth herle 
> wrote:
>
>> Hello Pavithra,
>>
>> I have written a guide for how to integrate an IMS using Kamailio with
>> open5gs EPC, here is the link
>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can find
>> steps to setup rtpengine in that guide.
>>
>> Regarding requirement of rtpengine, its mandatory if you want to use the
>> scripts in that example folder without modification. Also, read the READE
>> of sipwise/rtpengine github repo, its very informative (in a gist rtpengine
>> offer much more feature than rtpproxy)
>>
>> why can't we have sip client to sip client rtps flowing like how it is
>>> happening without NAT.
>>>
>>
>> For this, i would definitely suggest to read about SIP working. This is a
>> perfect link https://github.com/onmyway133/blog/issues/284 which offers
>> extensive material (short and very interesting to read with examples)
>> related to SIP.
>>
>> Best Regards,
>> Supreeth
>>
>>
>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja 
>> wrote:
>>
>>> Hi ,
>>>
>>> I didn't configure rtpengine ..instead of that only I used
>>> rrpproxy..will it not work?
>>>
>>> If not please tell me how to configure rtpengine and also tell me if it
>>> is really madatory .. why can't we have sip client to sip client rtps
>>> flowing like how it is happening without NAT.
>>>
>>> That's my doubt ..please clarify ..
>>>
>>> Thanks.
>>>
>>> On Thu, May 14, 2020, 8:57 PM supreeth herle 
>>> wrote:
>>>
 HI,

 Hold on a moment,


> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in pcscf.cfg
> (!#define WITH_NAT) and configured rtpproxy)
> 10.40.10.12 - icscf in port 4070
> 10.40.10.5 - scscf in port 4080
> 10.40.10.30 - hss
>

 So i assume you are using the examples provided in the ims folder of
 kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have you
 modified the P-CSCF configuration to work with RTPProxy?

 Best regards,
 Supreeth

 On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja <
 pavimohan3...@gmail.com> wrote:

> Hi ,
> I have attached working rtp pcap file. But it is getting reverted back
> to me .
>
> As you said , i have tried restarting rtpproxy several times. but it
> didnt work .
>
>
>
> On Thu, May 14, 2020 at 7:11 PM supreeth herle <
> herlesupre...@gmail.com> wrote:
>
>> Thanks for the pcap. Sorry, to ask this but can you also send pcap of
>> scenario where you see RTP packets?
>>
>> Also, did you restart the rtpproxy after altering the
>> /etc/default/rtpproxy ?
>>
>> i.e.
>>
>> $ systemctl restart rtpproxy
>>
>> Best Regards,
>> Supreeth
>>
>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja <
>> pavimohan3...@gmail.com> wrote:
>>
>>> Hi Supreeth,
>>> I have attached pcap files . kindly check and let me know.
>>> filter of rtp packets and check.
>>>
>>> *filter : sip || rtp*
>>>
>>>
>>> On Thu, May 14, 2020 at 5:15 PM supreeth herle <
>>> herlesupre...@gmail.com> wrote:
>>>
 Hi Pavithra,

 Thanks for the clarification. Can you please attach a pcap of the
 call where you are not seeing the RTP packets? And, btw is the call 
 getting
 dropped or something in that scenario?

 Regards,
 Supreeth


 On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja <
 pavimohan3...@gmail.com> wrote:

> 

Re: [SR-Users] kamailio5.2.2 calling t_relay(host, port)

[Daniel-Constantin Mierla]

Hello,

the code seems to have it. Can you provide the full log messages printed
in such case?

Cheers,
Daniel

On 19.06.20 15:46, Martin W Woscek wrote:
>
> Hello all,
>
>  
>
> Im trying to call t_relay() with host and port as documented in:
>
> https://www.kamailio.org/docs/modules/5.2.x/modules/tm.html
>
>  
>
> However, I get a subst_str(): no match on P-CSCF start and therefore
> fails.
>
> Is t_relay(host, port) supported as documented?
>
> Am I missing something?
>
>  
>
> Thanks,
>
> _Martin
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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Re: [SR-Users] Expired transaction

[Daniel-Constantin Mierla]

Hello,

I never faced it myself, the functionality was added based on a report
from the community more for troubleshooting purposes to see if the
transactions can still be freed after timer was elapsed and they were no
longer cleaned.

IIRC, the report said that there was no activity reported in the logs by
the server for 15min or so, eventually overlapping with taking a
snapshot of the virtual machine (thus, you may need to freeze the
virtual machine). You can check the commit history and see the date when
the code was added, then search the mailing lists archives for
discussions on that topic.

Cheers,
Daniel

On 21.06.20 18:32, Marat Gareev wrote:
> Hello!
> I'm trying to manually reproduce expired transactions which couldn't
> be cleared.
>
> I had one idea, but it was unsuccessful. I tried to send SIGSTOP to a
> group of Kamailio processes while the transaction is live. The idea is
> to freeze processes for a while, then continue to work, so that by
> this time the transaction deletion time was missed. But after a few
> seconds, the transaction is still cleared.
>
> Do you have any ideas for reproduction?
>
> P.S. I'm sorry If you received my incoherent mails. These were
> attempts to reply to the thread, seems to be my mail client or
> sr-users mailing list doesn't support In-Reply-To header.
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working

[Pavithra Mohanraja]

Hi,

I am facing below error in kamailio-ims 5.3.4 version.

reason=404 (Not Found - destination user not found on this S-CSCF)]

could you please help me how should i proceed with and where the
registration details will be stored in kamailio ims.

It fails at this line in kamailio.cfg file in scscf .

route[FINAL_TERM] {
if (lookup("location")) {
if (uri==myself) {
if (!t_newtran()) {
sl_reply_error();
exit;
}
 xlog("Destination Not Found \n");
t_reply("404","Not Found - destination user not
found on this S-CSCF");
exit;
}

here lookup(location) means what it means. Could you please help me



Thanks,
Pavithra

On Thu, May 14, 2020 at 9:59 PM supreeth herle 
wrote:

> Hello Pavithra,
>
> I have written a guide for how to integrate an IMS using Kamailio with
> open5gs EPC, here is the link
> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can find
> steps to setup rtpengine in that guide.
>
> Regarding requirement of rtpengine, its mandatory if you want to use the
> scripts in that example folder without modification. Also, read the READE
> of sipwise/rtpengine github repo, its very informative (in a gist rtpengine
> offer much more feature than rtpproxy)
>
> why can't we have sip client to sip client rtps flowing like how it is
>> happening without NAT.
>>
>
> For this, i would definitely suggest to read about SIP working. This is a
> perfect link https://github.com/onmyway133/blog/issues/284 which offers
> extensive material (short and very interesting to read with examples)
> related to SIP.
>
> Best Regards,
> Supreeth
>
>
> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja 
> wrote:
>
>> Hi ,
>>
>> I didn't configure rtpengine ..instead of that only I used rrpproxy..will
>> it not work?
>>
>> If not please tell me how to configure rtpengine and also tell me if it
>> is really madatory .. why can't we have sip client to sip client rtps
>> flowing like how it is happening without NAT.
>>
>> That's my doubt ..please clarify ..
>>
>> Thanks.
>>
>> On Thu, May 14, 2020, 8:57 PM supreeth herle 
>> wrote:
>>
>>> HI,
>>>
>>> Hold on a moment,
>>>
>>>
 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in pcscf.cfg
 (!#define WITH_NAT) and configured rtpproxy)
 10.40.10.12 - icscf in port 4070
 10.40.10.5 - scscf in port 4080
 10.40.10.30 - hss

>>>
>>> So i assume you are using the examples provided in the ims folder of
>>> kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have you
>>> modified the P-CSCF configuration to work with RTPProxy?
>>>
>>> Best regards,
>>> Supreeth
>>>
>>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja <
>>> pavimohan3...@gmail.com> wrote:
>>>
 Hi ,
 I have attached working rtp pcap file. But it is getting reverted back
 to me .

 As you said , i have tried restarting rtpproxy several times. but it
 didnt work .



 On Thu, May 14, 2020 at 7:11 PM supreeth herle 
 wrote:

> Thanks for the pcap. Sorry, to ask this but can you also send pcap of
> scenario where you see RTP packets?
>
> Also, did you restart the rtpproxy after altering the
> /etc/default/rtpproxy ?
>
> i.e.
>
> $ systemctl restart rtpproxy
>
> Best Regards,
> Supreeth
>
> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja <
> pavimohan3...@gmail.com> wrote:
>
>> Hi Supreeth,
>> I have attached pcap files . kindly check and let me know.
>> filter of rtp packets and check.
>>
>> *filter : sip || rtp*
>>
>>
>> On Thu, May 14, 2020 at 5:15 PM supreeth herle <
>> herlesupre...@gmail.com> wrote:
>>
>>> Hi Pavithra,
>>>
>>> Thanks for the clarification. Can you please attach a pcap of the
>>> call where you are not seeing the RTP packets? And, btw is the call 
>>> getting
>>> dropped or something in that scenario?
>>>
>>> Regards,
>>> Supreeth
>>>
>>>
>>> On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja <
>>> pavimohan3...@gmail.com> wrote:
>>>
 Hi Supreeth,

 I have kamailio ims installed in 4 vms inside openstack.

 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in
 pcscf.cfg (!#define WITH_NAT) and configured rtpproxy)
 10.40.10.12 - icscf in port 4070
 10.40.10.5 - scscf in port 4080
 10.40.10.30 - hss

 so all these vms are running in same subnet.
 I have a floating ip associated to pcscf.
 Now I have two zoiper clients outside openstack
 zoiper1 - 10.0.2.15 (NAT Enabled Network adaptor)
 zoiper2 - 10.0.2.15

 when i am making a call from zoiper1 to zoiper2, call is going
 properly