Re: [SR-Users] Branching and rejects.

[Kjeld Flarup]

What could change this behavior? I see that the first 404 I get return 
is passed on.


Could it be the t_newtran(). I Honestly don't remember why we put it there.
if ( !lookup("doorlocation") ) {
                $var(rc) = $rc;
                t_newtran();
                switch ($var(rc)) {


The doorlocation is also a "fake" location table, not used for 
registrations, but to make a branch to either a fixed terminal, or a 
mobile phone, which uses a second location table.
The fixed terminal can right away return 404, but the mobile phone, 
first has to receive a push, which can take some seconds.

Can this time difference in reply cause the issue?

  Kjeld Flarup

2018-04-16 9:23 GMT+02:00 Daniel-Constantin Mierla >:


   Hello,

   you do not need to discard the branch replies at all. Kamailio sends
   only one reply back, even if you do many outgoing branches, kamailio is
   going to wait until all of the get a final reply and then selects the
   one with highest priority to send back.

   Cheers,
   Daniel


   On 14.04.18 23:05, Kjeld Flarup wrote:
> Hello
>
> What is the correct way to handle, if a branched call reaches a
   404 on
> one of the branches.
>
> Currently I discard all 404 in onreply_route, but if all branches has
> 404, then this is never send to the caller.
>

   -- 
   Daniel-Constantin Mierla

   www.twitter.com/miconda  --
   www.linkedin.com/in/miconda 
   Kamailio Advanced Training - April 16-18, 2018, Berlin -
   www.asipto.com 
   Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
   


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Re: [SR-Users] Failure Route Issue

[Alex Balashov]

Oh, you're referring to the end-to-end ACK following the 200 OK.

What is the Contact URI in the 200 OK received by the calling UA? Does
the domain portion reflect the "SIP service", or the "Media server"? Is
there a Record-Route set?

On Tue, Apr 17, 2018 at 11:41:14PM -0400, m...@dopensource.com wrote:

> Thanks for the info Alex!
> 
> But, I don’t seem to get that behavior.  My sngrep graphical flow located 
> here: https://snag.gy/qMWfvk.jpg  shows that 
> Kamailio absorbs that negative ACK and then does an INVITE to the media 
> server, which is what I want it to do (because I have a failure route 
> defined)  But, 200 OK from the media server is being passed back to the UA 
> and then the ACK from the UA is being sent to the SIP service versus the 
> media server, which doesn’t make any sense to me.  I must be looking at this 
> wrong.

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Re: [SR-Users] Failure Route Issue

[mack]

Thanks for the info Alex!

But, I don’t seem to get that behavior.  My sngrep graphical flow located here: 
https://snag.gy/qMWfvk.jpg  shows that Kamailio 
absorbs that negative ACK and then does an INVITE to the media server, which is 
what I want it to do (because I have a failure route defined)  But, 200 OK from 
the media server is being passed back to the UA and then the ACK from the UA is 
being sent to the SIP service versus the media server, which doesn’t make any 
sense to me.  I must be looking at this wrong.___
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Re: [SR-Users] TCP packet relay delayed (Kamailio 4.1.5)

[Andrew Chen]

Thanks Alex for your response.  I honestly forgot about these parameters
and will play with them a bit.

Is there anyway to log how much is written or stored in the buffer?

On Tue, Apr 17, 2018 at 7:35 PM, Alex Balashov 
wrote:

> Hello,
>
> Could it be that your send queue or other parameters regulating outgoing
> flows are too low?
>
> Stuff like this:
>
> https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_conn_wq_max
> https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_wq_max
>
> -- Alex
>
> On Tue, Apr 17, 2018 at 05:00:28PM -0700, Andrew Chen wrote:
>
> > Hi Kamailio community,
> >
> > I need your urgent help regarding a situation where the Kamailio TCP
> packet
> > delayed in relaying SIP packets.  They delay between the time Kamailio
> logs
> > the relay to the time it left the system was ~20 seconds difference,
> which
> > is huge in SIP world.
> >
> > Any ideas, thoughts or suggestions on how I can narrow down to the
> > problem?  We have about 30 cases of this today
> >
> > Thanks in advance.
> >
> > --
> > Andy Chen
> > Sr. Telephony Lead Engineer
> > achen@ fuze.com
> >
> > --
> > *Confidentiality Notice: The information contained in this e-mail and any
> >
> > attachments may be confidential. If you are not an intended recipient,
> you
> >
> > are hereby notified that any dissemination, distribution or copying of
> this
> >
> > e-mail is strictly prohibited. If you have received this e-mail in error,
> >
> > please notify the sender and permanently delete the e-mail and any
> >
> > attachments immediately. You should not retain, copy or use this e-mail
> or
> >
> > any attachment for any purpose, nor disclose all or any part of the
> >
> > contents to any other person. Thank you.*
>
> > ___
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>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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>



-- 
Andy Chen
Sr. Telephony Lead Engineer
415 516 5535 (M)
achen@ fuze.com

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Re: [SR-Users] TCP packet relay delayed (Kamailio 4.1.5)

[Alex Balashov]

Hello,

Could it be that your send queue or other parameters regulating outgoing
flows are too low?

Stuff like this:

https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_conn_wq_max
https://www.kamailio.org/wiki/cookbooks/5.1.x/core#tcp_wq_max

-- Alex

On Tue, Apr 17, 2018 at 05:00:28PM -0700, Andrew Chen wrote:

> Hi Kamailio community,
> 
> I need your urgent help regarding a situation where the Kamailio TCP packet
> delayed in relaying SIP packets.  They delay between the time Kamailio logs
> the relay to the time it left the system was ~20 seconds difference, which
> is huge in SIP world.
> 
> Any ideas, thoughts or suggestions on how I can narrow down to the
> problem?  We have about 30 cases of this today
> 
> Thanks in advance.
> 
> -- 
> Andy Chen
> Sr. Telephony Lead Engineer
> achen@ fuze.com
> 
> -- 
> *Confidentiality Notice: The information contained in this e-mail and any
> 
> attachments may be confidential. If you are not an intended recipient, you
> 
> are hereby notified that any dissemination, distribution or copying of this
> 
> e-mail is strictly prohibited. If you have received this e-mail in error,
> 
> please notify the sender and permanently delete the e-mail and any
> 
> attachments immediately. You should not retain, copy or use this e-mail or
> 
> any attachment for any purpose, nor disclose all or any part of the
> 
> contents to any other person. Thank you.*

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-- 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[SR-Users] TCP packet relay delayed (Kamailio 4.1.5)

[Andrew Chen]

Hi Kamailio community,

I need your urgent help regarding a situation where the Kamailio TCP packet
delayed in relaying SIP packets.  They delay between the time Kamailio logs
the relay to the time it left the system was ~20 seconds difference, which
is huge in SIP world.

Any ideas, thoughts or suggestions on how I can narrow down to the
problem?  We have about 30 cases of this today

Thanks in advance.

-- 
Andy Chen
Sr. Telephony Lead Engineer
achen@ fuze.com

-- 
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attachments may be confidential. If you are not an intended recipient, you

are hereby notified that any dissemination, distribution or copying of this

e-mail is strictly prohibited. If you have received this e-mail in error,

please notify the sender and permanently delete the e-mail and any

attachments immediately. You should not retain, copy or use this e-mail or

any attachment for any purpose, nor disclose all or any part of the

contents to any other person. Thank you.*
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Re: [SR-Users] Failure Route Issue

[Alex Balashov]

Hi Mack,

On Tue, Apr 17, 2018 at 03:03:37PM -0400, m...@dopensource.com wrote:

> Maybe, I’m not understanding.  How do I get Kamailio to send the ACK
> to the media server versus sending it to the SIP service.  It seems
> that Kamailio is remembering the negative ACK, which was already sent
> back to the SIP service and it’s sending it again versus sending the
> ACK to the Media Server.

For a negative ACK, the sequence of events should be something like:

UA 1   Kamailio  UA 2
=
--- INVITE --->

<- 100 Trying -
   -- INVITE -->

  <- 100 Trying -

  < 433 -

  - ACK >

 < 433 

  ACK >


Kamailio's transaction layer (TM) should absorb the negative ACK from UA
1 as part and parcel of t_check_trans().

This is all laid out in the stock config file boilerplate.

-- Alex

-- 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
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Re: [SR-Users] Failure Route Issue

[mack]

Thanks Alex…

Maybe, I’m not understanding.  How do I get Kamailio to send the ACK to the 
media server versus sending it to the SIP service.  It seems that Kamailio is 
remembering the negative ACK, which was already sent back to the SIP service 
and it’s sending it again versus sending the ACK to the Media Server.



> On Apr 17, 2018, at 2:26 PM, Alex Balashov  wrote:
> 
> Hi,
> 
> ACKs in response to negative replies are what are called hop-by-hop
> ACKs, and should be addressed to the next hop.
> 
> Also, it is not necessary to explicitly do an append_branch() in
> failure_route nowadays if manipulating the RURI. It'll be done for you.
> 
> -- Alex
> 
> On Tue, Apr 17, 2018 at 02:22:39PM -0400, m...@dopensource.com wrote:
> 
>> Hey All,
>> 
>> I have a weird issue.  I’m sending a SIP INVITE to a SIP Service that send’s 
>> me back a 433, which let’s me know that the number is invalid.  I then send 
>> an INVITE to a media server (FreeSwitch) to play a message that the number 
>> can’t be called, which looks to be working.  But, the final ACK from the SIP 
>> client is sent back to the SIP Service versus the media server.  The 
>> following diagram below depicts this: https://snag.gy/qMWfvk.jpg.  The code 
>> snippet in the failure_route for the the SIP Service has this:
>> 
>> if (t_check_status("403|433")) {
>>   if ($avp(s:teleblock_media_enabled) == "1") {
>>   # make sure media server can route back to kamailio
>>   record_route();
>>   revert_uri();
>>   $ru = "sip:" + $T_reply_code + $rU + "@" + 
>> $sel(cfg_get.teleblock.media_ip) + ":" + $sel(cfg_get.teleblock.media_port);
>>   append_branch();
>>   t_relay();
>>   exit;
>> 
>>   }
>> 
>> 
>> 
>> 
>> ___
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> 
> -- 
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> 
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Re: [SR-Users] Failure Route Issue

[Alex Balashov]

Hi,

ACKs in response to negative replies are what are called hop-by-hop
ACKs, and should be addressed to the next hop.

Also, it is not necessary to explicitly do an append_branch() in
failure_route nowadays if manipulating the RURI. It'll be done for you.

-- Alex

On Tue, Apr 17, 2018 at 02:22:39PM -0400, m...@dopensource.com wrote:

> Hey All,
> 
> I have a weird issue.  I’m sending a SIP INVITE to a SIP Service that send’s 
> me back a 433, which let’s me know that the number is invalid.  I then send 
> an INVITE to a media server (FreeSwitch) to play a message that the number 
> can’t be called, which looks to be working.  But, the final ACK from the SIP 
> client is sent back to the SIP Service versus the media server.  The 
> following diagram below depicts this: https://snag.gy/qMWfvk.jpg.  The code 
> snippet in the failure_route for the the SIP Service has this:
> 
>  if (t_check_status("403|433")) {
> if ($avp(s:teleblock_media_enabled) == "1") {
> # make sure media server can route back to kamailio
> record_route();
> revert_uri();
> $ru = "sip:" + $T_reply_code + $rU + "@" + 
> $sel(cfg_get.teleblock.media_ip) + ":" + $sel(cfg_get.teleblock.media_port);
> append_branch();
> t_relay();
> exit;
> 
> }
> 
> 
> 
> 
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[SR-Users] Failure Route Issue

[mack]

Hey All,

I have a weird issue.  I’m sending a SIP INVITE to a SIP Service that send’s me 
back a 433, which let’s me know that the number is invalid.  I then send an 
INVITE to a media server (FreeSwitch) to play a message that the number can’t 
be called, which looks to be working.  But, the final ACK from the SIP client 
is sent back to the SIP Service versus the media server.  The following diagram 
below depicts this: https://snag.gy/qMWfvk.jpg.  The code snippet in the 
failure_route for the the SIP Service has this:

 if (t_check_status("403|433")) {
if ($avp(s:teleblock_media_enabled) == "1") {
# make sure media server can route back to kamailio
record_route();
revert_uri();
$ru = "sip:" + $T_reply_code + $rU + "@" + 
$sel(cfg_get.teleblock.media_ip) + ":" + $sel(cfg_get.teleblock.media_port);
append_branch();
t_relay();
exit;

}




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Re: [SR-Users] CLID blacklist

[CELEA : Mathias]

Hello,
You can use diaplan module to acheive this.
Regards
Mathias

Le mar. 17 avril 2018 à 16:04, Nelson Migliaro 
 a écrit :

Hello,

I am looking for a module to block number based on CLID. If the 
module allows REG expresions it would really make my day.


I need to block some callers and I would like to use a module and 
database to avoid restarting kamailio service everytime a new number 
appears.


Thank you

Nelson.-




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[SR-Users] CLID blacklist

[Nelson Migliaro]

Hello,

I am looking for a module to block number based on CLID. If the module
allows REG expresions it would really make my day.

I need to block some callers and I would like to use a module and database
to avoid restarting kamailio service everytime a new number appears.

Thank you

Nelson.-
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Re: [SR-Users] Kamailio Websocket Questions

[Guillaume Bour]

Hello the list

Let me exhume this old message :)
I also encounter the same error message as Dimitry
When a webrtc device is registering itself, here what's output in the logs (the 
registration being successful or not):

   Apr 17 14:15:53 kamailio[11693]: 24(11720) ERROR:  
[core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad 
message (offset: 22)
   Apr 17 14:15:53 kamailio[11693]: 24(11720) ERROR:  
[core/parser/msg_parser.c:675]: parse_msg(): ERROR: parse_msg: 
message=

FYI, our kamailios are behind a HAProxy used for load-balancing

Dimitry, did you succeed to fix it in your setup ?

Kind Regards,
Guillaume Bour.


On Fri, Jun 03, 2016 at 11:11:58AM +, Nagorny, Dimitry wrote:
>Hi all,
> 
> 
> 
>Got INVITE resolved, but the Errors described below persist.
> 
> 
> 
> 
> 
>Best Regards
> 
>Dimitry Nagorny
> 
>Trainee
> 
> 
> 
>Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag
>von Nagorny, Dimitry
>Gesendet: Freitag, 3. Juni 2016 10:39
>An: Kamailio (SER) - Users Mailing List 
>Betreff: Re: [SR-Users] Kamailio Websocket Questions
> 
> 
> 
>Hello Daniel,
> 
> 
> 
>Thank you very much for your response. Here the portions for my websocket
>config. Of course all needed modules are loaded (one modparam for
>websocket ping) and I took the config all from
>http://kamailio.org/docs/modules/4.3.x/modules/websocket.html:
> 
>onreply_route[WS_REPLY] {
> 
>if (nat_uac_test(64)) {
> 
>   add_contact_alias();
> 
>}
> 
>if (proto==WS) {
> 
>   xlog("L_INFO", "Hit Reply from Websocket");
> 
>   rtpengine_answer("trust-address ");
> 
>}
> 
>}
> 
>event_route[xhttp:request] {
> 
>set_reply_close();
> 
>set_reply_no_connect();
> 
>   
> 
>if ($Rp != MY_INTERN_WSPORT){
> 
>   xlog("L_WARN", "HTTP request received on
>$Rp\n");
> 
>   xhttp_reply("403", "Forbidden", "", "");
> 
>   exit;  
> 
>}
> 
>if ($hdr(Upgrade)=~"websocket" &&
>$hdr(Connection)=~"Upgrade" && $rm=~"GET") {
> 
>   if ($hdr(Host) == $null ||
>!is_myself("sip:" + $hdr(Host))) {
> 
>   xlog("L_WARN", "Bad host
>$hdr(Host)\n");
> 
>   xhttp_reply("403",
>"Forbidden", "", "");
> 
>   exit;
> 
>   }
> 
>   if (ws_handle_handshake()) {
> 
>   exit;
> 
>   }
> 
>}
> 
>xhttp_reply("404", "Not Found", "", "");
> 
>}
> 
>event_route[websocket:closed] {
> 
>xlog("L_INFO", "WebSocket connection from $si:$sp has
>closed\n");
> 
>}
> 
> 
> 
>I have attached a debug log for the registration process over websocket,
>on line 49 you can see the error.
> 
> 
> 
>As for the INVITE I use now this:
> 
>if ($rU=~"^3$" && src_ip==$sel(cfg_get.pstn.gw_ip)) {
> 
>$ru = "sip:dnagorny@10.250.5.74";
> 
>uac_replace_to("sip:dnagorny@10.250.5.74");
> 
>rtpengine_offer("external internal trust-address");
> 
>route(RELAY);
> 
>}
> 
> 
> 
>Unfortunately I can see now that Kamailio tries to send it out on
>10.250.5.74:5060, which I am not even listening on. For the route(RELAY) I
>use this:
> 
>route[RELAY] {
> 
>if (is_method("INVITE")) {
> 
>   t_on_branch("MANAGE_BRANCH");
> 
>   if (proto==WS) {
> 
>   t_on_reply("WS_REPLY");
> 
>   } else {
> 
>   t_on_reply("MANAGE_REPLY");
> 
>   }
> 
>   t_on_failure("MANAGE_FAILURE");
> 
>}
> 
>if (is_method("CANCEL|BYE")) {
> 
>   rtpengine_delete();
> 
>}
> 
>   
> 
>if (!t_relay()) {
> 
>   sl_reply_error();
> 
>}
> 
>exit;
> 
>}
> 
> 
> 
>Is t_relay for Websocket the right choice? I somehow have my doubts but
>couldn’t find any suggestion in the documentation. 

[SR-Users] Kamailio support for IPv6 VoLTE Calls

[Sairam Subramanian]

Hi All,

Can anyone tell me which version of the Kamailio Open IMS supports IPv6
VoLTE Calls?

Thanks,

Sairam S
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Re: [SR-Users] Dialog module's keep-alive feature uses wrong CSeq?

[Daniel-Constantin Mierla]

Hello,

the keepalive is done via sort of a trick which needs to detect:

  - if one side is no longer connected to the network -- keepalive
results in retransmission timeout

  - if one side is no longer knowing the dialog - so call-id, to-tag and
from-tag values do not match -- keepalive results in 481 Call
Leg/Transaction Does Not Exist

Any other case is not relevant for the keepalive purpose of detecting if
a side is still in the call.

Therefore, CSeq is intentionally set to an lower value than expected by
each side, in order not to impact what is sent by each side.

If you want to do nat keepalive for the duration of the dialog, that can
be achieved by using nat_traversal. Or if you use a location service,
usrloc+nathelper does it for the registration session.

Cheers,
Daniel

On 17.04.18 08:25, Oded Arbel wrote:
>
>
> On Mon, Apr 16, 2018, 16:24 Dmitri Savolainen  > wrote:
>
> Ideally I think Kamailio should send correct (i.e increasing)
> CSeq numbers.
>
> in my mind it can't be increased by kamailio because of:
> 1. kamailio send OPTIONS with cseq+1
> 2. media server may send some  indialog reinvite with cseq+1 and
> then kamailio have to remember that OPTIONS and  translate
> reinvite to cseq+2.
>
>
> Fair point, statelessness is important.
>
>
> I don't know why for "ka-src" CSeq is 0 and for "ka-dst"  the one
> is  equal: so may be it is possible to be fixed 
>
>
> I'll investigate, under the assumption everyone agrees it's a bug.
>
>
>
>  (for example BYE) to be dropped by the firewall.
>
> this may be achieved by
> 1. usrloc pinging
>
>
> I'm not sure that is usrloc pinging, but if you mean nathelper's
> nat_ping, then that wouldn't work for me because it's only for MUAs
> that use REGISTER, which on my system does not always happen.
>
> 2. short re-registration period for endpoint (60 sec for ex)
>
>
> Aside from the fact that REGISTER is not guaranteed to ever happen,
> this is also a UA setting that can't be enforced by the server and has
> serious implications on battery life outside of a call.
>
> 3. let's media server send options by itself via kamailio
>
>
> This may or may not be possible on my setup, I'll have to investigate
> - thanks for the suggestion!
>
> -- 
>
> Oded Arbel
>
> oded.ar...@greenfieldtech.net 
> Greenfield Tech 
>
>
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-- 
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Re: [SR-Users] Dialog module's keep-alive feature uses wrong CSeq?

[Oded Arbel]

On Mon, Apr 16, 2018, 16:24 Dmitri Savolainen  wrote:

> Ideally I think Kamailio should send correct (i.e increasing) CSeq numbers.
>>
> in my mind it can't be increased by kamailio because of:
> 1. kamailio send OPTIONS with cseq+1
> 2. media server may send some  indialog reinvite with cseq+1 and then
> kamailio have to remember that OPTIONS and  translate reinvite to cseq+2.
>

Fair point, statelessness is important.


I don't know why for "ka-src" CSeq is 0 and for "ka-dst"  the one
> is  equal: so may be it is possible to be fixed
>

I'll investigate, under the assumption everyone agrees it's a bug.



>  (for example BYE) to be dropped by the firewall.
>
> this may be achieved by
> 1. usrloc pinging
>

I'm not sure that is usrloc pinging, but if you mean nathelper's nat_ping,
then that wouldn't work for me because it's only for MUAs that use
REGISTER, which on my system does not always happen.

2. short re-registration period for endpoint (60 sec for ex)
>

Aside from the fact that REGISTER is not guaranteed to ever happen, this is
also a UA setting that can't be enforced by the server and has serious
implications on battery life outside of a call.

3. let's media server send options by itself via kamailio
>

This may or may not be possible on my setup, I'll have to investigate -
thanks for the suggestion!

-- 

Oded Arbel
oded.ar...@greenfieldtech.net
[image: Greenfield Tech] 
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