date:20151204

Re: [SR-Users] Generating db-tables .html file

2015-12-04 Thread smititelu


On 03.12.2015 13:43, Daniel-Constantin Mierla wrote:

Looking quickly at, I noticed there is a page for this topic:
http://www.kamailio.org/wiki/devel/update-database-schema

It is missing the part for html doc, if you have time, would be good 
if you add your notes there.


Updated the current doku.

Thanks,
Stefan
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Re: [SR-Users] LCR module : re load_gws uri errors

2015-12-04 Thread Juha Heinanen

ycaner writes:

> Thanks for reply. before load_gws , setting ruri_user_avp to null solves
> request uri problems.

load_gws() tries to delete possibly existing avp values by this
statement:

delete_avp(gw_uri_avp_type|AVP_VAL_STR, gw_uri_avp);

obviously there is something wrong with if the avp needs to be deleted
by the script.

i'll try at some point try to figure out how to delete an avp based on
its name.

-- juha


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Re: [SR-Users] LCR module : re load_gws uri errors

2015-12-04 Thread Juha Heinanen

ycaner writes:

>   in load_gws function
>   there is a delete_avp for gw_uri_avp as you mentioned . but in my
>   view it needs a delete_avp for ruri_user_avp. 
>   Because it is reloading gws for different route.

I added deletion of possible old ruri_user_avp value when next_gw() is
called first time after load_gws().

It is currently in master branch only, but I can commit also to 4.3
branch if this fixes the issue.

-- Juha

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Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

2015-12-04 Thread smititelu


On 03.12.2015 12:30, Juha Heinanen wrote:

Richard Fuchs writes:


I have a prospective fix in a separate branch:
https://github.com/sipwise/rtpengine/tree/rfuchs/delete-branch

Can you please test it to see if it works for your use case, then I'll
merge into master.

Richard,

I had chance to test this branch and it worked as I expected.  With
earlier version these two offers:


Hello Juha,

If you have the scenario in place and some time for it, can you please 
try [1] for rtpengine module, kamailio side?


It is related to the pull-request for rtpengine hash table to keep the 
nodes selected for a call. We are not able to test it since we are not 
using rtpengine's via-branch feature. We are running it for ~ two weeks 
from now, in the test system, and didn't spotted problems so far 
(without via-branch).


Thank you,
1&1 Team

[1] https://github.com/smititelu/kamailio/commits/master

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Re: [SR-Users] LCR module : re load_gws uri errors

2015-12-04 Thread ycaner

in load_gws function
there is a delete_avp for gw_uri_avp as you mentioned . but in my
view it needs a delete_avp for ruri_user_avp.
Because it is reloading gws for different route.
Thanks.

04.12.2015 11:44 tarihinde Juha
Heinanen [via SIP Router] yazdı:

ycaner writes:

Thanks for reply. before load_gws , setting ruri_user_avp to
null solves

request uri problems.

load_gws() tries to delete possibly existing avp values by this

statement:

delete_avp(gw_uri_avp_type|AVP_VAL_STR, gw_uri_avp);

obviously there is something wrong with if the avp needs to be
deleted

by the script.

i'll try at some point try to figure out how to delete an avp
based on

its name.

-- juha

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Re: [SR-Users] LCR module : re load_gws uri errors

2015-12-04 Thread ycaner


  

  
  
Hello; 
  i commented your last commit. Sorry , i think i couldn't explain 
situation.
  Here is log for more understand. 
  
  
  - Here is first load_gws and next_gws() functions are 
  called- 
  6(10299) DEBUG: lcr [lcr_mod.c:1943]: load_gws(): load_gws(1,
  10218503027337, ) 
   6(10299) DEBUG: lcr [lcr_mod.c:2014]: load_gws(): added
  matched_gws[0]=[21, 4, 1, 6054284] 
   6(10299) DEBUG: lcr [lcr_mod.c:1887]: add_gws_into_avps(): added
  gw_uri_avp 21|sip:|0||0|3020380345||5080||;transport=udp|0
  with weight 6054284 
   6(10299) DEBUG: lcr [../../resolve.h:258]: str2ip(): str2ip:
  ERROR: too few dots in [3020380345] 
   6(10299) DEBUG: lcr [../../resolve.h:355]: str2ip6(): str2ip6:
  ERROR: too few colons in [3020380345] 
   6(10299) DEBUG: lcr [lcr_mod.c:2175]: generate_uris(): r_uri
  sip:10218503027337@ip:5080;transport=udp, dst_uri  
   6(10299) DEBUG: lcr [lcr_mod.c:2456]: next_gw(): added
  ruri_user_avp 10218503027337 
   6(10299) DEBUG: lcr [lcr_mod.c:2492]: next_gw(): added flags_avp
  0 
   6(10299) DEBUG: lcr [lcr_mod.c:2499]: next_gw(): added tag_avp
  0 
   this route/number gives busy response, i am gonna
  change route and number- 
  - Second loads_gws and next_gws functions are
  called.-- 
   8(10303) DEBUG: lcr [lcr_mod.c:1943]: load_gws(): load_gws(1,
  200015066109057, ) 
   8(10303) DEBUG: lcr [lcr_mod.c:2014]: load_gws(): added
  matched_gws[0]=[1, 5, 1, 2247144] 
   8(10303) DEBUG: lcr [lcr_mod.c:1887]: add_gws_into_avps(): added
  gw_uri_avp
  1|sip:|5|200010|2|3020380345||5080||;transport=udp|0 with
  weight 2247144 
   8(10303) DEBUG: lcr [../../resolve.h:258]: str2ip(): str2ip:
  ERROR: too few dots in [3020380345] 
   8(10303) DEBUG: lcr [../../resolve.h:355]: str2ip6(): str2ip6:
  ERROR: too few colons in [3020380345] 
   8(10303) DEBUG: lcr [lcr_mod.c:2175]: generate_uris(): r_uri
  sip:200010503027337@ip:5080;transport=udp, dst_uri
   
   8(10303) DEBUG: lcr [lcr_mod.c:2492]: next_gw(): added flags_avp
  0 
   8(10303) DEBUG: lcr [lcr_mod.c:2499]: next_gw(): added tag_avp
  2 
  r_uri must be 200015066109057 but
  it gets first called r_uri because r_uri saved--- 
  
  
  Number 200015066109057 is
  stripped with five as underline. but lcr strips olds uri (10218503027337) 
as saved r_uri. 
  
  
  this route scripts a kind of that if pstn is busy , forward to
  call a cellphone. 
  
  

04.12.2015 13:27 tarihinde Juha
  Heinanen [via SIP Router] yazdı: 

 ycaner writes:
  
  
         in load_gws function
  
         there is a delete_avp for gw_uri_avp as you mentioned .
  but in my
  
         view it needs a delete_avp for ruri_user_avp. 
         Because it is reloading gws for different route.
  
  
  I added deletion of possible old ruri_user_avp value when
  next_gw() is
  
  called first time after load_gws().
  
  
  It is currently in master branch only, but I can commit also to
  4.3
  
  branch if this fixes the issue.
  
  
  -- Juha
  
  
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Re: [SR-Users] About Siremis Login Page Problem

2015-12-04 Thread Daniel-Constantin Mierla

Hello,

do you have the mod_redirect enabled for apache? Can you look at the
logs of apache and see if there is any error message?

Also, you can look at the logs of siremis in the folder siremis/logs/.

Cheers,
Daniel

On 03/12/15 12:49, Morshed wrote:
>
> Dear Concern,
>
> I have installed kamailo and siremis installed properly. But I cannot
> login to the siremis, the login page is not loaded successfully.
> Please have a look on the attached image and any suggestion from your end?
>
>  
>
> Regards,
>
>  
>
> Morshed Alam
>
> Support Engineer | IP Telephony
>
>  
>
> Link3 Technologies Ltd.
>
> Bulu Ocean Tower, 17th Floor
>
> 40, Kemal Ataturk Avenue, Banani,
>
> Dhaka 1213, Bangladesh.
>
> Tel: +88-02-9822288
>
> IP Phone : +88-09678997705
>
> Fax: +88-02-9821332
>
> Web: www.link3.net 
>
>  
>
>
>
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[SR-Users] issue with BYE

2015-12-04 Thread gerry kernan

Hi 
 
I'm looking for any pointers to resolve and issue I have .
When callee does a hang-up the call terminates on kamailio and asterisk but the 
caller's phone handset doesn't receive the BYE and doesn't hang-up.
If the caller does the hang up then the call terminates and the callee's 
handset hands up .
 
My setup is 
Kamailio WAN -> Kamailio Lan -> Asterisk LAN
 
I'm using Path and dispatcher modules to route the SIP traffic to the backup 
end asterisk server my the domain in the  SIP request. 
 
 
 
 
Gerry Kernan
 
 
Infinity IT   |   17 The Mall   |   Beacon Court   |   Sandyford   |   Dublin 
D18 E3C8   |   Ireland
Tel:  +353 - (0)1 - 293 0090   |   E-Mail:  gerry.ker...@infinityit.ie
 
Managed IT Services   Infinity IT - www.infinityit.ie
IP TelephonyAsterisk Consulting - www.asteriskconsulting.com
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Re: [SR-Users] If we print out $fu, $fU & $fd on onsend_route, it prints out incorrect values.

2015-12-04 Thread Daniel-Constantin Mierla

Hello,

in onsend_route, the 'standard' message variables still point to
incoming message structure. But you can see what is going to be sent out
with $snd(buf).

Cheers,
Daniel

On 04/12/15 14:28, Jurijs Ivolga wrote:
> Hi all,
>
> I opened a bug here:
>
> https://github.com/kamailio/kamailio/issues/430
>
> But then I was advised to write to mailing list.
>
> In nutshell:
>
> I'm running Kamailio 4.3.3.
>
> When I'm trying to update $ru, $fu & $fd in Kamailio conf:
> $rU = "1000";
> $fu = "sip:someth...@mydomain.com ";
> $fd = "newdomain.com ";
>
> And later I'm trying to print out $rU, $fu, $fU & $fd in onsend_route:
>
> |xlog("ALERT: From URI $fu \n"); xlog("ALERT: From domain $fd \n");
> xlog("ALERT: From user $fU \n"); xlog("ALERT: Request user $rU \n"); |
>
> Kamailio 4.3.3 prints out following:
> ALERT: From URI sip:2000@some_ip
> ALERT: From domain some_ip
> ALERT: From user 2000
> ALERT: Request user 1000
>
> But I believe it should:
>
> ALERT: From URI sip:someth...@mydomain.com 
> ALERT: From domain mydomain.com 
> ALERT: From user something
> ALERT: Request user 1000
>
> I'm using default Kamailio config with minor changes, you can check it
> in attachment.
>
> Is it proper behavior?
>
> Thank you!
>
> With kind regards,
>
> Jurijs
>
>
>
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[SR-Users] [permissions] allow_source_address_group() always return -1

2015-12-04 Thread Iñaki Baz Castillo

Hi all,

Kamailio 4.3 (Debian Wheezy package) with db_mysql and permissions
modules enabled.

Once started, this is the cached content of the "address" table:

~# kamctl mi "address_dump"
 22 <20, 99.23.86.78, 0> [TEST SERVER]

Within the script:

-
xlog("L_INFO", "request_route() [method:$rm, si:$si, sp:$sp]\n");

$var(group) = allow_source_address_group();
if ($var(group) != -1) {
  // ALWAYS HERE :(

  xlog("L_WARN", "invalid source IP [si:$si]\n");
}
-

Then I send an INVITE from such an allowed IP, and got:

INFO:

[SR-Users] If we print out $fu, $fU & $fd on onsend_route, it prints out incorrect values.

2015-12-04 Thread Jurijs Ivolga

Hi all,

I opened a bug here:

https://github.com/kamailio/kamailio/issues/430

But then I was advised to write to mailing list.

In nutshell:

I'm running Kamailio 4.3.3.

When I'm trying to update $ru, $fu & $fd in Kamailio conf:
$rU = "1000";
$fu = "sip:someth...@mydomain.com";
$fd = "newdomain.com";

And later I'm trying to print out $rU, $fu, $fU & $fd in onsend_route:

xlog("ALERT: From URI $fu \n");
xlog("ALERT: From domain $fd \n");
xlog("ALERT: From user $fU \n");
xlog("ALERT: Request user $rU \n");

Kamailio 4.3.3 prints out following:
ALERT: From URI sip:2000@some_ip
ALERT: From domain some_ip
ALERT: From user 2000
ALERT: Request user 1000

But I believe it should:

ALERT: From URI sip:someth...@mydomain.com
ALERT: From domain mydomain.com
ALERT: From user something
ALERT: Request user 1000

I'm using default Kamailio config with minor changes, you can check it in
attachment.

Is it proper behavior?

Thank you!

With kind regards,

Jurijs
#!KAMAILIO
#
# Kamailio (OpenSER) SIP Server v4.3 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: 
#
# Refer to the Core CookBook at http://www.kamailio.org/wiki/
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode: 
# - define WITH_DEBUG
#
# *** To enable mysql: 
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
#rtpproxy -l _your_public_ip_ -s udp:localhost:7722
# - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
#   block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT 
'';
  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT 
'';
#!endif

### Include Local Config If Exists #
import_file "kamailio-local.cfg"

### Defined Values #

# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#   as: auth_db, acc, usrloc, a.s.o.
#!ifndef DBURL
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1

Re: [SR-Users] [permissions] allow_source_address_group() always return -1

2015-12-04 Thread Iñaki Baz Castillo

Please forget me...

if ($var(group) != -1)

so yes, 20 != 1...

Sorry!

2015-12-04 14:48 GMT+01:00 Iñaki Baz Castillo :
> 2015-12-04 14:38 GMT+01:00 Iñaki Baz Castillo :
>> -
>> $var(group) = allow_source_address_group();
>> if ($var(group) != -1) {
>>   // ALWAYS HERE :(
>>
>>   xlog("L_WARN", "invalid source IP [si:$si]\n");
>> }
>> -
>
> Interesting. If I add a log:
>
> -
> $var(group) = allow_source_address_group();
>
> xlog("L_INFO", "rc:$var(group)\n");
>
> if ($var(group) != -1) {
>   xlog("L_WARN", "invalid source IP [si:$si]\n");
> }
> -
>
> Then I get:
>
>   INFO:

[SR-Users] UAC dynamic reg_contact_addr()

2015-12-04 Thread Yuriy Gorlichenko

I have multiple ip addresses at my kamailio. I use uac module for
registration to sip providers. I have one provider but want to register
form different ip addresses used by my server. When register sends it takes
ip address form reg_contact_addr(). But if I want to register from another
interface with different ip i need to get to this parametr another ip
address. So As I understand it imposibleto do with avp. Does kamailio have
some methods to do this?
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Re: [SR-Users] [permissions] allow_source_address_group() always return -1

2015-12-04 Thread Iñaki Baz Castillo

2015-12-04 14:38 GMT+01:00 Iñaki Baz Castillo :
> -
> $var(group) = allow_source_address_group();
> if ($var(group) != -1) {
>   // ALWAYS HERE :(
>
>   xlog("L_WARN", "invalid source IP [si:$si]\n");
> }
> -

Interesting. If I add a log:

-
$var(group) = allow_source_address_group();

xlog("L_INFO", "rc:$var(group)\n");

if ($var(group) != -1) {
  xlog("L_WARN", "invalid source IP [si:$si]\n");
}
-

Then I get:

  INFO:

Re: [SR-Users] why is rtpenengine-delete deleting the whole call?

2015-12-04 Thread Richard Fuchs

On 12/03/2015 06:38 PM, Juha Heinanen wrote:

i noticed one more things during testing of rfuchs/delete-branch, which
i don't quite understand.

the test call is parallel forked to two destinations and offer (using
via-branch=1) is issued for both. thus the offers have the same params:

... "call-id": "138d9b5516741c30", "via-branch": "z9hG4bKe8998a18295855da", "received-from": [ "IP4", "192.98.102.10"
], "from-tag": "b5b2ffd6a2205b13", "command": "offer" }

..."call-id": "138d9b5516741c30", "via-branch": "z9hG4bKe8998a18295855da", "received-from": [ "IP4", "192.98.102.10" ],
"from-tag": "b5b2ffd6a2205b13", "command": "offer" }

That doesn't work. The point of the via-branch handling is to recognize
that this is a forked call/offer and so the via-branch should be
different in the two offers. Right now this only works if the call is
forked in a second SIP proxy instance daisy-chained before the one doing
the RTP proxy stuff, as Kamailio offers no mechanism to anticipate the
next outgoing via-branch value. Fixing this is on our to-do list.

question: how it is possible the call works, i.e., rtpengine still has
the call even when it was already deleted? does it remember that two
offers were made using same params and the call does not really get
deleted before it has received two deletes?

It still works if the answer comes in before the delete-delay triggers
actual deletion (which is the reason for delete-delay's existence). It's
also necessary that both offers were made with the same parameters, e.g.
not one with RTP and the second the SRTP (otherwise rtpengine could get
confused).

Cheers

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Re: [SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Alex Balashov


Doug,

If I understood your claim correctly, the INVITE request goes from 
Kamailio to Asterisk, and Asterisk returns a 401 challenge. This is 
because Asterisk doesn't trust Kamailio as an incoming source. It 
probably needs a [correct] sip.conf peer.


-- Alex

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303 Perimeter Center North, Suite 300
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Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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[SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Douglas Adami

Hello everyone,

I have Kamailio and Asterisk on the same machine with the same public IP.
When I try to forward calls to the asterisk (PSTN GW on port 5080), returns
the error 401, but does not reach the Asterisk any information. When I
switch to an IP out, it works.

Any idea?

Regards,
Doug
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Re: [SR-Users] Executing a C function from the Kamailio Script

2015-12-04 Thread Alex Balashov


Mack,

The language modules take advantage of the fact that those other 
languages (e.g. Lua, Python) are higher-level, _interpreted_ languages, 
and that their interpreters provide an API for embedding them into C 
programs. One cannot easily embed an on-the-fly C compiler into a C program.


Writing a Kamailio module is certainly the easiest way to call a custom 
C function from route script, and it's not terribly hard to get started. 
If your needs are very simple, you don't have to make extensive study of 
the module API. Just follow/modify this example module:


https://github.com/kamailio/kamailio/tree/master/modules/print

'print' is a stub that doesn't do anything, but provides a skeleton from 
which you can build your own module.


I suppose you could also embed your C function into a library that can 
be called from an interpreted language supported by Kamailio (Lua, 
Python, Perl, etc.), but that seems like it would be vastly more 
complicated than writing a Kamailio module.


Finally, are you one hundred percent sure that the functionality you 
need cannot be achieved any other way?


-- Alex

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Re: [SR-Users] issue with BYE

2015-12-04 Thread gerry kernan

Hi 
>From a trace I can see that the BYE was received from 10001 and sent to 
>asterisk(192.10.10.213) , but the kamailio(192.10.10.202) doesn't forward the 
>BYE back out to 10002@X.X.X.X
 
4   3.870042 X.X.X.X -> 212.126.39.60 SIP/SDP 1163 Request: INVITE 
sip:10...@sip.xyz.ie;user=phone |
  6   3.870688 192.10.10.202 -> 192.10.10.213 SIP/SDP 1445 Request: INVITE 
sip:10...@sip.xyz.ie;user=phone |
 11   3.873831 192.10.10.213 -> 192.10.10.202 SIP 703 Status: 401 Unauthorized |
 15   3.875223 212.126.39.60 -> X.X.X.X SIP 594 Status: 401 Unauthorized |
 17   3.894614 X.X.X.X -> 212.126.39.60 SIP 575 Request: ACK 
sip:10...@sip.xyz.ie;user=phone |
 19   3.895036 192.10.10.202 -> 192.10.10.213 SIP 857 Request: ACK 
sip:10...@sip.xyz.ie;user=phone |
 21   3.898506 X.X.X.X -> 212.126.39.60 SIP/SDP 1338 Request: INVITE 
sip:10...@sip.xyz.ie;user=phone |
 24   3.899177 192.10.10.202 -> 192.10.10.213 SIP/SDP 140 Request: INVITE 
sip:10...@sip.xyz.ie;user=phone |
 27   3.904115 192.10.10.213 -> 192.10.10.202 SIP 741 Status: 100 Trying |
 29   3.904322 212.126.39.60 -> X.X.X.X SIP 632 Status: 100 Trying |
 33   3.925201 192.10.10.213 -> 192.10.10.202 SIP/SDP 1018 Request: INVITE 
sip:10001@192.168.200.112:5062 |
 35   3.925485 192.10.10.213 -> 192.10.10.202 SIP 757 Status: 180 Ringing |
 36   3.92 192.10.10.202 -> 192.168.200.112 SIP/SDP 1156 Request: INVITE 
sip:10001@192.168.200.112:5062 |
 38   3.925686 212.126.39.60 -> X.X.X.X SIP 648 Status: 180 Ringing |
 41   3.945324 192.168.200.112 -> 192.10.10.202 SIP 607 Status: 100 Trying |
 43   3.945516 192.10.10.202 -> 192.10.10.213 SIP 521 Status: 100 Trying |
 45   3.988254 192.168.200.112 -> 192.10.10.202 SIP 644 Status: 180 Ringing |
 47   3.988485 192.10.10.202 -> 192.10.10.213 SIP 558 Status: 180 Ringing |
 49   4.064562 192.10.10.213 -> 192.10.10.202 SIP 757 Status: 180 Ringing |
 51   4.064845 212.126.39.60 -> X.X.X.X SIP 648 Status: 180 Ringing |
20  70   5.194932 192.168.200.112 -> 192.10.10.202 SIP/SDP 979 Status: 200 OK |
 74   5.195587 192.10.10.202 -> 192.10.10.213 SIP/SDP 893 Status: 200 OK |
 76   5.198424 192.10.10.213 -> 192.10.10.202 SIP 483 Request: ACK 
sip:10001@192.168.200.112:5062 |
 78   5.198750 192.10.10.202 -> 192.168.200.112 SIP 621 Request: ACK 
sip:10001@192.168.200.112:5062 |
 80   5.199155 192.10.10.213 -> 192.10.10.202 SIP/SDP 1117 Status: 200 OK |
 84   5.199750 212.126.39.60 -> X.X.X.X SIP/SDP 1008 Status: 200 OK |
 86   5.224751 X.X.X.X -> 212.126.39.60 SIP 645 Request: ACK 
sip:10001@192.10.10.213:5060 |
 88   5.225136 192.10.10.202 -> 192.10.10.213 SIP 854 Request: ACK 
sip:10001@192.10.10.213:5060 |
28 112   7.960369 192.168.200.112 -> 192.10.10.202 SIP 548 Request: BYE 
sip:10002@192.10.10.213:5060 |
114   7.960889 192.10.10.202 -> 192.10.10.213 SIP 716 Request: BYE 
sip:10002@192.10.10.213:5060 |
116   7.963035 192.10.10.213 -> 192.10.10.202 SIP 655 Status: 200 OK |
120   7.963501 192.10.10.202 -> 192.168.200.112 SIP 546 Status: 200 OK |
122   8.038144 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
124   8.137929 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
126   8.338033 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
35 140   8.737860 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
36 149   9.537899 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
37 157  11.137935 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
38 169  14.337813 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE 
sip:10002@X.X.X.X:16082 |
39 201  22.814799 X.X.X.X -> 212.126.39.60 SIP 722 Request: BYE 
sip:10001@192.10.10.213:5060 |
203  22.815254 192.10.10.202 -> 192.10.10.213 SIP 931 Request: BYE 
sip:10001@192.10.10.213:5060 |
 
Gerry Kernan
 
 
Infinity IT   |   17 The Mall   |   Beacon Court   |   Sandyford   |   Dublin 
D18 E3C8   |   Ireland
Tel:  +353 - (0)1 - 293 0090   |   E-Mail:  gerry.ker...@infinityit.ie
 
Managed IT Services   Infinity IT - www.infinityit.ie
IP TelephonyAsterisk Consulting - www.asteriskconsulting.com
Contact CentreTotal Interact - www.totalinteract.com
 
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
gerry kernan
Sent: Friday 4 December 2015 14:51
To: sr-users@lists.sip-router.org
Subject: [SR-Users] issue with BYE
 
Hi 
 
I'm looking for any pointers to resolve and issue I have .
When callee does a hang-up the call terminates on kamailio and asterisk but the 
caller's phone handset doesn't receive the BYE and doesn't hang-up.
If the caller does the hang up then the call terminates and the callee's 
handset hands up .
 
My setup is 
Kamailio WAN -> Kamailio Lan -> Asterisk LAN
 
I'm using Path and dispatcher modules to route the SIP traffic to the backup 
end asterisk server my the domain in the  SIP request. 
 
 
 
 
Gerry Kernan
 
 
Infinity IT   |   17 The Mall

Re: [SR-Users] issue with BYE

2015-12-04 Thread Alex Balashov


Hello Gerry,

What is the exact topology of the call flow? i.e. where are the 
endpoints situated in relation to Kamailio and Asterisk, and how are 
Kamailio and Asterisk situated in relation to each other?


Second, can you provide your Kamailio configuration and also a packet 
capture illustrating the misrouting of the BYE? You can send it 
privately if you're not keen to post it to the list.


Cheers,

-- Alex

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303 Perimeter Center North, Suite 300
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Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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Re: [SR-Users] issue with BYE

2015-12-04 Thread Alex Balashov


Gerry,

Is there any way you could provide this capture with the full SIP 
payload, not just the request line of every message?


Thanks,

-- Alex

On 12/04/2015 12:25 PM, gerry kernan wrote:


Hi

 From a trace I can see that the BYE was received from 10001 and sent to
asterisk(192.10.10.213) , but the kamailio(192.10.10.202) doesn’t
forward the BYE back out to 10002@X.X.X.X 

4   3.870042 X.X.X.X -> 212.126.39.60 SIP/SDP 1163 Request: INVITE
sip:10...@sip.xyz.ie;user=phone |

   6   3.870688 192.10.10.202 -> 192.10.10.213 SIP/SDP 1445 Request:
INVITE sip:10...@sip.xyz.ie;user=phone |

11   3.873831 192.10.10.213 -> 192.10.10.202 SIP 703 Status: 401
Unauthorized |

15   3.875223 212.126.39.60 -> X.X.X.X SIP 594 Status: 401 Unauthorized |

17   3.894614 X.X.X.X -> 212.126.39.60 SIP 575 Request: ACK
sip:10...@sip.xyz.ie;user=phone |

19   3.895036 192.10.10.202 -> 192.10.10.213 SIP 857 Request: ACK
sip:10...@sip.xyz.ie;user=phone |

21   3.898506 X.X.X.X -> 212.126.39.60 SIP/SDP 1338 Request: INVITE
sip:10...@sip.xyz.ie;user=phone |

24   3.899177 192.10.10.202 -> 192.10.10.213 SIP/SDP 140 Request: INVITE
sip:10...@sip.xyz.ie;user=phone |

27   3.904115 192.10.10.213 -> 192.10.10.202 SIP 741 Status: 100 Trying |

29   3.904322 212.126.39.60 -> X.X.X.X SIP 632 Status: 100 Trying |

33   3.925201 192.10.10.213 -> 192.10.10.202 SIP/SDP 1018 Request:
INVITE sip:10001@192.168.200.112:5062 |

35   3.925485 192.10.10.213 -> 192.10.10.202 SIP 757 Status: 180 Ringing |

36   3.92 192.10.10.202 -> 192.168.200.112 SIP/SDP 1156 Request:
INVITE sip:10001@192.168.200.112:5062 |

38   3.925686 212.126.39.60 -> X.X.X.X SIP 648 Status: 180 Ringing |

41   3.945324 192.168.200.112 -> 192.10.10.202 SIP 607 Status: 100 Trying |

43   3.945516 192.10.10.202 -> 192.10.10.213 SIP 521 Status: 100 Trying |

45   3.988254 192.168.200.112 -> 192.10.10.202 SIP 644 Status: 180 Ringing |

47   3.988485 192.10.10.202 -> 192.10.10.213 SIP 558 Status: 180 Ringing |

49   4.064562 192.10.10.213 -> 192.10.10.202 SIP 757 Status: 180 Ringing |

51   4.064845 212.126.39.60 -> X.X.X.X SIP 648 Status: 180 Ringing |

20  70   5.194932 192.168.200.112 -> 192.10.10.202 SIP/SDP 979 Status:
200 OK |

74   5.195587 192.10.10.202 -> 192.10.10.213 SIP/SDP 893 Status: 200 OK |

76   5.198424 192.10.10.213 -> 192.10.10.202 SIP 483 Request: ACK
sip:10001@192.168.200.112:5062 |

78   5.198750 192.10.10.202 -> 192.168.200.112 SIP 621 Request: ACK
sip:10001@192.168.200.112:5062 |

80   5.199155 192.10.10.213 -> 192.10.10.202 SIP/SDP 1117 Status: 200 OK |

84   5.199750 212.126.39.60 -> X.X.X.X SIP/SDP 1008 Status: 200 OK |

86   5.224751 X.X.X.X -> 212.126.39.60 SIP 645 Request: ACK
sip:10001@192.10.10.213:5060 |

88   5.225136 192.10.10.202 -> 192.10.10.213 SIP 854 Request: ACK
sip:10001@192.10.10.213:5060 |

28 112   7.960369 192.168.200.112 -> 192.10.10.202 SIP 548 Request: BYE
sip:10002@192.10.10.213:5060 |

114   7.960889 192.10.10.202 -> 192.10.10.213 SIP 716 Request: BYE
sip:10002@192.10.10.213:5060 |

116   7.963035 192.10.10.213 -> 192.10.10.202 SIP 655 Status: 200 OK |

120   7.963501 192.10.10.202 -> 192.168.200.112 SIP 546 Status: 200 OK |

122   8.038144 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

124   8.137929 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

126   8.338033 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

35 140   8.737860 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

36 149   9.537899 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

37 157  11.137935 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

38 169  14.337813 192.10.10.213 -> 192.10.10.202 SIP 697 Request: BYE
sip:10002@X.X.X.X:16082 |

39 201  22.814799 X.X.X.X -> 212.126.39.60 SIP 722 Request: BYE
sip:10001@192.10.10.213:5060 |

203  22.815254 192.10.10.202 -> 192.10.10.213 SIP 931 Request: BYE
sip:10001@192.10.10.213:5060 |

*Gerry Kernan*

cid:image001.jpg@01D105A5.2701B0E0

*Infinity IT   |   17 The Mall   |   Beacon Court   |   Sandyford   |
Dublin D18 E3C8   |   Ireland*

*Tel:  +353 - (0)1 - 293 0090   |   E-Mail: *gerry.ker...@infinityit.ie
**

**

*Managed IT Services__Infinity IT*- www.infinityit.ie


*IP Telephony__Asterisk Consulting*– www.asteriskconsulting.com


*Contact Centre__Total Interact*– www.totalinteract.com


*From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
Behalf Of *gerry kernan
*Sent:* Friday 4 December 2015 14:51
*To:* sr-users@lists.sip-router.org
*Subject:* [SR-Users] issue with BYE

Hi

I’m looking for any pointers to resolve and issue I have .

When callee does a hang-up the call terminates on kamailio and asterisk
but the caller’s phone handset doesn’t receive the BYE and doesn’t

[SR-Users] Executing a C function from the Kamailio Script

2015-12-04 Thread Mack Hendricks

We have some logic in a C function that we would like to execute from the 
Kamailio script.  I see there are modules for calling functions in other 
languages.  But, I don’t see anything for C.  Is creating a module the only way 
to leverage this C function?

Thanks,

-Mack
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Re: [SR-Users] UAC dynamic reg_contact_addr()

2015-12-04 Thread Yuriy Gorlichenko

do I need to recompile kamailio with

make EXTRA_DEFS="-DWITH_EVENT_LOCAL_REQUEST" cfg

?
 Because at my installation adding  event_route[tm:local-request]
fives me syntax error.


2015-12-05 0:04 GMT+03:00 Daniel-Constantin Mierla :

> Hello,
>
> On 04/12/15 14:13, Yuriy Gorlichenko wrote:
> > I have multiple ip addresses at my kamailio. I use uac module for
> > registration to sip providers. I have one provider but want to
> > register form different ip addresses used by my server. When register
> > sends it takes ip address form reg_contact_addr(). But if I want to
> > register from another interface with different ip i need to get to
> > this parametr another ip address. So As I understand it imposibleto do
> > with avp. Does kamailio have some methods to do this?
> >
> perhaps makes sense to add some extra columns to be able to specify the
> local socket and contact addresses.
>
> For now, I expect to work by using event_route[tm:local-request] where
> you force the socket with $fs.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> http://miconda.eu
>
>
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Re: [SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Alex Balashov


When you say gateway, you mean Asterisk?

Do you have a capture illustrating this flow?

On 12/04/2015 01:13 PM, Douglas Adami wrote:


Yes, I send in public_same_ip:5080 and did not get anything at the gateway.

Cheers,
Doug

2015-12-04 18:05 GMT+00:00 Alex Balashov >:

You mean that you forward the request to localhost:5080, in essence,
and it does not receive it?

Are you forwarding to localhost:5080 or public_ip:5080? That may
traverse different interfaces. And by the same token, do local
firewall and/or forwarding policies allow for this?

Cheers,

-- Alex


On 12/04/2015 01:02 PM, Douglas Adami wrote:

Alex,

I use Asterisk only as gateway (PSTN). The gateway, when the
same IP,
does not receive the INVITE, only when it is in another machine.

Thanks for your response.

Doug


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--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920  (toll-free) /
+1-678-954-0671  (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Douglas Adami

Alex,

I use Asterisk only as gateway (PSTN). The gateway, when the same IP, does
not receive the INVITE, only when it is in another machine.

Thanks for your response.

Doug
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Re: [SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Alex Balashov

You mean that you forward the request to localhost:5080, in essence, and 
it does not receive it?


Are you forwarding to localhost:5080 or public_ip:5080? That may 
traverse different interfaces. And by the same token, do local firewall 
and/or forwarding policies allow for this?


Cheers,

-- Alex

On 12/04/2015 01:02 PM, Douglas Adami wrote:


Alex,

I use Asterisk only as gateway (PSTN). The gateway, when the same IP,
does not receive the INVITE, only when it is in another machine.

Thanks for your response.

Doug


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--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Douglas Adami

Yes, I send in public_same_ip:5080 and did not get anything at the gateway.

Cheers,
Doug

2015-12-04 18:05 GMT+00:00 Alex Balashov :

> You mean that you forward the request to localhost:5080, in essence, and
> it does not receive it?
>
> Are you forwarding to localhost:5080 or public_ip:5080? That may traverse
> different interfaces. And by the same token, do local firewall and/or
> forwarding policies allow for this?
>
> Cheers,
>
> -- Alex
>
>
> On 12/04/2015 01:02 PM, Douglas Adami wrote:
>
> Alex,
>>
>> I use Asterisk only as gateway (PSTN). The gateway, when the same IP,
>> does not receive the INVITE, only when it is in another machine.
>>
>> Thanks for your response.
>>
>> Doug
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] Kamailio + Asterisk, same machine, same public IP.

2015-12-04 Thread Douglas Adami

Yes, gateway = asterisk. No, I will capture and then send.

Thank you for your help.

2015-12-04 18:17 GMT+00:00 Alex Balashov :

> When you say gateway, you mean Asterisk?
>
> Do you have a capture illustrating this flow?
>
> On 12/04/2015 01:13 PM, Douglas Adami wrote:
>
> Yes, I send in public_same_ip:5080 and did not get anything at the gateway.
>>
>> Cheers,
>> Doug
>>
>> 2015-12-04 18:05 GMT+00:00 Alex Balashov > >:
>>
>> You mean that you forward the request to localhost:5080, in essence,
>> and it does not receive it?
>>
>> Are you forwarding to localhost:5080 or public_ip:5080? That may
>> traverse different interfaces. And by the same token, do local
>> firewall and/or forwarding policies allow for this?
>>
>> Cheers,
>>
>> -- Alex
>>
>>
>> On 12/04/2015 01:02 PM, Douglas Adami wrote:
>>
>> Alex,
>>
>> I use Asterisk only as gateway (PSTN). The gateway, when the
>> same IP,
>> does not receive the INVITE, only when it is in another machine.
>>
>> Thanks for your response.
>>
>> Doug
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>> mailing list
>> sr-users@lists.sip-router.org > sr-users@lists.sip-router.org>
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> --
>> Alex Balashov | Principal | Evariste Systems LLC
>> 303 Perimeter Center North, Suite 300
>> Atlanta, GA 30346
>> United States
>>
>> Tel: +1-800-250-5920  (toll-free) /
>> +1-678-954-0671  (direct)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>> list
>> sr-users@lists.sip-router.org 
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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