date:20220321

Re: [SR-Users] About SMSC multi-byte support

2022-03-21 Thread shigeru ishida

Hi Henning,

I used the following Kamailio 5.5 branch with docker_open5gs.

https://github.com/herlesupreeth/kamailio

As a result, garbled characters occurred in SMS over IMS.

Also, I wrote a little about the VoLTE and SMS settings of docker_open5gs
as follows.

https://github.com/s5uishida/docker_open5gs_volte_sms_config

FYI.

--Shigeru

2022年3月21日(月) 17:11 Henning Westerholt :

> Hello,
>
>
>
> can you share a bit more details how you exactly process the SMS, like
> which modules etc..?
>
>
>
> Also, I would recommend that you use a newer version of kamailio, there
> are also docker images for version 5.5.x available.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* sr-users  *On Behalf Of *shigeru
> ishida
> *Sent:* Sunday, March 20, 2022 3:16 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] About SMSC multi-byte support
>
>
>
> Hi all,
>
>
>
> I have a little question.
>
>
>
> I tried SMS over IMS on my smartphones with docker_open5gs (using Kamailio
> and Open5GS), and the multibyte message (Japanese) was garbled at the
> recipient smartphone.
>
>
>
> docker_open5gs URL: https://github.com/herlesupreeth/docker_open5gs
>
>
>
> Also, docker_open5gs uses Kamailio v5.3.
>
>
>
> I would be very glad if you could tell me about Kamailio's support for
> multi-byte SMS.
>
>
>
> Best regards,
>
>
>
> --Shigeru
>
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Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

2022-03-21 Thread George Diamantopoulos

Hello all,

Any chance this could be affecting early-dialog UPDATE methods as well? I
haven't tried with a different backend (mysql) yet, but loose_route returns
false...

BR,
George

On Fri, 25 May 2018 at 18:00, Daniel Tryba  wrote:

> On Wed, May 23, 2018 at 10:51:37PM +0200, Daniel-Constantin Mierla wrote:
> > I got a bit of time to look at this issue, can you try with the patch
> > from the next commit?
> >
> > ?? -
> >
> https://github.com/kamailio/kamailio/commit/2a3ca5942291d29de05b14338a36cecbec11f129
> >
>
> This off by 1 fix solved the PRACK issue with redis backend.
>
>
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[SR-Users] Kamailio : Parallel Forking

2022-03-21 Thread Paclinio

I've been trying to do parallel forking but i've a problem. It's when the
conversation ends that the user that is in a private network can't send a
bye to kamailio and i don't know why. But the on in public network (4G)
can. I think it's related to record_route() or Via but I'm not sure.

I'm also trying to use lookup on users that are in my domain but the
problem is that I don't know how to go through all my append_branch() and
then t_relay.

My Audio with RTPENGINE works correctly, the only problem is really this
SIP method that doesn't go to kamailio.


Config :


request_route {
route(INVITE);
exit;
}

route[INVITE]{

 route(RECORD_ROUTE);
 route(GROUP_RELAY_SER);

switch($tU)
{
case "112":
xlog("MY_SEVERITY"," -- Calling Group Serial");
route(GROUP_RELAY_SER);
break;
case "113": // This number does not exist (It's not a subscriber).
xlog("MY_SEVERITY"," -- Calling Group Parallel");
route(GROUP_RELAY_PAR);
break;
default:
xlog("MY_SEVERITY"," -- Calling Someone");
route(LOCATION);
route(RELAY);

}

exit;
}



route[RECORD_ROUTE]{
# - remove preloaded route headers
remove_hf("Route");
if(is_method("INVITE|SUBSCRIBE"))
{
# record routing for dialog forming requests (in case they are routed)
record_route();
}

}



route[GROUP_RELAY_PAR]{

t_on_reply("MANAGE_REPLY");

#!ifndef WITH_BRIDGE_ON_FAIL
setbflag(FLB_BRIDGE);
#!endif
route(IPV4V6);
route(NATMANAGE);

append_branch("sip:9919995@192.168.1.24:5066");
append_branch("sip:9919992@192.168.1.26:5066");
append_branch("sip:9919991@192.168.1.10:5067");

// I Would like to do an IP in my domain and then do a lookup but I don't
know how to do it.


t_on_failure("GroupFailureRoute");
#Relay (aka Forward) the request
t_relay();
break;

}



# User location service
route[GROUP_LOCATION] {
$avp(oexten) = $rU;

if (lookup_branches("location")){
return (1);
}
else
{
return (-1);
}
}



failure_route[GroupFailureRoute]{
t_next_contacts();
while(!route(GROUP_LOCATION)){ // Check if the next user (in serial) is
logged in)
if (!t_next_contacts()) {
send_reply("408","Nobody available");
exit;
}
}
t_on_failure("GroupFailureRoute");
t_relay();
}




# Manage incoming replies in transaction context
onreply_route[MANAGE_REPLY] {
if(status=~"[12][0-9][0-9]") {
if(has_body("application/sdp")){
setbflag(FLB_BRIDGE);
}
route(NATMANAGE);
}
}



-

Sip Trace :

2022/03/21 18:36:46.469257 149.154.195.121:9665 -> 192.168.1.41:5060
INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---4108e717927214fd;rport
Max-Forwards: 70
Contact: 
To: 
From: "codec3-public";tag=ed51991c
Call-ID: GlECSiXh3tJ37DqYLLvm_g..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.17.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 226

v=0
o=Zoiper 53798437 1 IN IP4 100.79.133.159
s=Z
c=IN IP4 100.79.133.159
t=0 0
m=audio 8000 RTP/AVP 0 101 8 3 97
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=sendrecv


2022/03/21 18:36:46.470066 192.168.1.41:5060 -> 149.154.195.121:9665
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---4108e717927214fd;rport=9665;received=149.154.195.121
To: ;tag=cbfe6543b9079987bce3b5b9a0d1195a.6dfd9117
From: "codec3-public";tag=ed51991c
Call-ID: GlECSiXh3tJ37DqYLLvm_g..
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm="192.168.1.41",
nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/"
Server: kamailio (5.5.4 (x86_64/linux))
Content-Length: 0



2022/03/21 18:36:46.500471 149.154.195.121:9665 -> 192.168.1.41:5060
ACK sip:113@192.168.1.41:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---4108e717927214fd;rport
Max-Forwards: 70
To: ;tag=cbfe6543b9079987bce3b5b9a0d1195a.6dfd9117
From: "codec3-public";tag=ed51991c
Call-ID: GlECSiXh3tJ37DqYLLvm_g..
CSeq: 1 ACK
Content-Length: 0



2022/03/21 18:36:46.549993 149.154.195.121:9665 -> 192.168.1.41:5060
INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---75020835629f7238;rport
Max-Forwards: 70
Contact: 
To: 
From: "codec3-public";tag=ed51991c
Call-ID: GlECSiXh3tJ37DqYLLvm_g..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest
username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri="sip:113@Public_IP
;transport=TCP",response="c8654099b37ca083ace2b245f22ebc71",algorithm=MD5
User-Agent: Zoiper v2.10.17.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 226

v=0
o=Zoiper 53798437 1 IN

[SR-Users] Kamailio : Serial forking

2022-03-21 Thread Paclinio

I've been trying to do serial forking, I've configured something that is
working but when I'm looking on github and other sources it seems like my
code is a little too complicated and I would like to know if there is a
simpler config that I can use.
To have more precision, I'm using RTPENGINE and I need to be able to go
through NAT.


My Config :

request_route {
route(INVITE);
exit;
}




route[INVITE]{

 route(RECORD_ROUTE);
 route(GROUP_RELAY_SER);

switch($tU)
{
case "112": // This number does not exist (It's not a subscriber).
xlog("MY_SEVERITY"," -- Calling Group Serial");
route(GROUP_RELAY_SER);
break;
case "113":
xlog("MY_SEVERITY"," -- Calling Group Parralelle");
route(GROUP_RELAY_PAR);
break;
default:
xlog("MY_SEVERITY"," -- Calling Someone");
route(LOCATION);
route(RELAY);

}

exit;
}



route[RECORD_ROUTE]{
# - remove preloaded route headers
remove_hf("Route");
if(is_method("INVITE|SUBSCRIBE"))
{
# record routing for dialog forming requests (in case they are routed)
record_route();
}

}



route[GROUP_RELAY_SER]{

t_on_reply("MANAGE_REPLY");

 route(IPV4V6);

route(NATMANAGE); (RTP Routing is working correctly)


append_branch("sip:9919995@192.168.1.41","1");
append_branch("sip:9919992@192.168.1.41", "0.2");
append_branch("sip:75575757@192.168.1.41", "0.4");
append_branch("sip:9919991@192.168.1.41", "0.5");

t_load_contacts();

t_next_contacts();

while(!route(GROUP_LOCATION))
{
if (!t_next_contacts()) {
send_reply("408","Timeout or nobody available");
exit;
}
}

t_on_failure("GroupFailureRoute");

#Relay (aka Forward) the request
t_relay();
break;
}



# User location service
route[GROUP_LOCATION] {
$avp(oexten) = $rU;

if (lookup_branches("location")){
return (1);
}
else
{
return (-1);
}
}



failure_route[GroupFailureRoute]{
t_next_contacts();
while(!route(GROUP_LOCATION)){
if (!t_next_contacts()) {

send_reply("408","Nobody available");
exit;
}
}

t_on_failure("GroupFailureRoute");

t_relay();
}




# Manage incoming replies in transaction context
onreply_route[MANAGE_REPLY] {
if(status=~"[12][0-9][0-9]") {
if(has_body("application/sdp")){
setbflag(FLB_BRIDGE);
}
route(NATMANAGE);
}
}



-

Here's The SIP Trace :



2022/03/21 18:28:45.265009 149.154.195.121:9665 -> 192.168.1.41:5060
INVITE sip:112@192.168.1.41:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport
Max-Forwards: 70
Contact: 
To: 
From: "codec3-public";tag=35ce2334
Call-ID: RtH8fBIrIgUYPAkxlxjI4Q..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.17.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 226

v=0
o=Zoiper 53721112 1 IN IP4 100.79.133.159
s=Z
c=IN IP4 100.79.133.159
t=0 0
m=audio 8000 RTP/AVP 0 101 8 3 97
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=sendrecv


2022/03/21 18:28:45.274652 192.168.1.41:5060 -> 149.154.195.121:9665
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport=9665;received=149.154.195.121
To: ;tag=cbfe6543b9079987bce3b5b9a0d1195a.064b0d7d
From: "codec3-public";tag=35ce2334
Call-ID: RtH8fBIrIgUYPAkxlxjI4Q..
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm="192.168.1.41",
nonce="Yji2+WI4tc0uQkJ6GbQ4I9VeSUQmR6Gk"
Server: kamailio (5.5.4 (x86_64/linux))
Content-Length: 0



2022/03/21 18:28:45.335876 149.154.195.121:9665 -> 192.168.1.41:5060
ACK sip:112@192.168.1.41:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport
Max-Forwards: 70
To: ;tag=cbfe6543b9079987bce3b5b9a0d1195a.064b0d7d
From: "codec3-public";tag=35ce2334
Call-ID: RtH8fBIrIgUYPAkxlxjI4Q..
CSeq: 1 ACK
Content-Length: 0



2022/03/21 18:28:45.385686 149.154.195.121:9665 -> 192.168.1.41:5060
INVITE sip:112@192.168.1.41:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 100.79.133.159:5060
;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport
Max-Forwards: 70
Contact: 
To: 
From: "codec3-public";tag=35ce2334
Call-ID: RtH8fBIrIgUYPAkxlxjI4Q..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest
username="9919993",realm="192.168.1.41",nonce="Yji2+WI4tc0uQkJ6GbQ4I9VeSUQmR6Gk",uri="sip:112@Public_IP
;transport=TCP",response="54837810a66298e4542486bb6ab8ca04",algorithm=MD5
User-Agent: Zoiper v2.10.17.3-mod
Allow-Events: presence, kpml, talk
Content-Length: 226

v=0
o=Zoiper 53721112 1 IN IP4 100.79.133.159
s=Z
c=IN IP4 100.79.133.159
t=0 0

[SR-Users] dispatcher socket attrs binding

2022-03-21 Thread David Villasmil

Hello guys,

Im starting kamailio with listen=0.0.0.0:5060 and in my dispatcher list i
have in the attrs something like: socket=udp:172.17.0.2:5060
That IP address belongs to kamailio's host and it is binging to it.

Is it right i can't set a specific IP in the socket attribute if listen is
0.0.0.0?
maybe i should be able to?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
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Re: [SR-Users] Can Kamailio send an ACK to a final response?

2022-03-21 Thread Yuriy Gorlichenko

 Here are some methods You can use like:
- t_uac_send from TM
- uac_req_send from UAC
- send_data from COREX
Depending on your needs

On Sat, 19 Mar 2022, 12:36 Marrold,  wrote:

> Hello,
>
> I'm exploring using Kamailio as a dumb edge proxy, and as such I want to
> keep things as lean as possible.
>
> In some scenarios I would like the "core" proxies to signal to the edge
> that a request should be dropped without a response. This could be with a
> custom header in the reply or perhaps a 444 response ( nginx uses this
> internally to drop a request without responding )
>
> I have discovered that if I drop the response in the reply_route, the edge
> proxy doesn't send an ACK and keeps retransmitting the request until it
> times out, at which point it sends a 408 back to the UA.
>
> Is there any way to ACK the response from the core and destroy the
> transaction?
>
> Thanks
> Matthew
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Re: [SR-Users] pseudo var for From and To sip uri port

2022-03-21 Thread Patrick Karton

Thanks for uac tip 

De : Henning Westerholt 
Envoyé : lundi 21 mars 2022 09:13
À : Kamailio (SER) - Users Mailing List 
Cc : Patrick Karton 
Objet : RE: pseudo var for From and To sip uri port


Hello,



you can get the header port with “transformations”.



If you want to set the port for From/To, you should not use the PVs but the uac 
module functions for this. You should also think about why you want to do this, 
as the From/To are coming from the user agent and are not set from the proxy 
usually. 



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: sr-users  On Behalf Of Patrick Karton
Sent: Sunday, March 20, 2022 2:57 PM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] pseudo var for From and To sip uri port



Hello,



is there a pseudo variable for setting only sip uri port for From and To 
Headers like  $rp for R-URI ?






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Re: [SR-Users] $ru vs. $fu as r/w pseudo variables

2022-03-21 Thread Daniel-Constantin Mierla

Operations to change the headers and body are not applied immediately:

  *
https://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers_or

Daniel

On 20.03.22 07:32, Juha Heinanen wrote:
> If I set $ru in config file and then print $ru to log, the new value is
> printed.  If I do the same with $fu, the old value is printed.
>
> I haven't found in Pseudo-Variables Wiki any documentation about this.
> Does it exist somewhere else?
>
> -- Juha
>
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Re: [SR-Users] pseudo var for From and To sip uri port

2022-03-21 Thread Henning Westerholt

Hello,

you can get the header port with “transformations”.

If you want to set the port for From/To, you should not use the PVs but the uac 
module functions for this. You should also think about why you want to do this, 
as the From/To are coming from the user agent and are not set from the proxy 
usually. 

Cheers,

Henning

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

From: sr-users  On Behalf Of Patrick Karton
Sent: Sunday, March 20, 2022 2:57 PM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] pseudo var for From and To sip uri port

Hello,

is there a pseudo variable for setting only sip uri port for From and To 
Headers like  $rp for R-URI ?



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Re: [SR-Users] About SMSC multi-byte support

2022-03-21 Thread Henning Westerholt

Hello,

can you share a bit more details how you exactly process the SMS, like which 
modules etc..?

Also, I would recommend that you use a newer version of kamailio, there are 
also docker images for version 5.5.x available.

Cheers,

Henning

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

From: sr-users  On Behalf Of shigeru ishida
Sent: Sunday, March 20, 2022 3:16 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] About SMSC multi-byte support

Hi all,

I have a little question.

I tried SMS over IMS on my smartphones with docker_open5gs (using Kamailio and 
Open5GS), and the multibyte message (Japanese) was garbled at the recipient 
smartphone.

docker_open5gs URL: https://github.com/herlesupreeth/docker_open5gs

Also, docker_open5gs uses Kamailio v5.3.

I would be very glad if you could tell me about Kamailio's support for 
multi-byte SMS.

Best regards,

--Shigeru
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Re: [SR-Users] About SMSC multi-byte support

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

[SR-Users] Kamailio : Parallel Forking

[SR-Users] Kamailio : Serial forking

[SR-Users] dispatcher socket attrs binding

Re: [SR-Users] Can Kamailio send an ACK to a final response?

Re: [SR-Users] pseudo var for From and To sip uri port

Re: [SR-Users] $ru vs. $fu as r/w pseudo variables

Re: [SR-Users] pseudo var for From and To sip uri port

Re: [SR-Users] About SMSC multi-byte support

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