date:20170615

[SR-Users] SUBSCRIBE not work with public ip kamailio

2017-06-15 Thread Nguyen Tran Nhan

Hello,
I am using Kamailio with SEMS and rtpproxy for public ip address. I got
issue that SUBSCRIBE message does not work. SUBSCRIBE works well with
private ip address. Anyone face with this issue?

Thanks,
Nhan
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[SR-Users] re.subst does not work with $var?

2017-06-15 Thread Pranathi Venkatayogi

I am trying to replace "domain1" with "domain2" in the msrp(body).

Following stmt works where in I specify domain strings as literals.
var(modBody) = $(msrp(body){re.subst,/()(domain1)()/domain2/g});

I am having trouble making the same work with $var variables.
See below - this code does not do the replacement -

$var(domain1) = "()(domain1)()";
$var(domain2) = "domain2";
var(modBody2) = $(msrp(body){re.subst,/$var(domain1)/$var(domain2)/g});

Any clue/ideas or is it by design?

Thanks,
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com

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Re: [SR-Users] How to change MSRP from sip address?

2017-06-15 Thread Alex Balashov

Come on, man. You're going to e-mail the entire list + three random
individuals, just in case deterministic computers don't work
deterministically? 

We're all on the list, you know. :-)

> From: Pranathi Venkatayogi 
> To: "Kamailio (SER) - Users Mailing List" 
> CC: "abalas...@evaristesys.com" , 
> "mico...@gmail.com" , Mick McGrellis 
> 

-- 
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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] How to change MSRP from sip address?

2017-06-15 Thread Pranathi Venkatayogi

I need to replace the "From" address of MSRP message with something different.
I noticed that module does not allow one to set values from config file.

What does it take to enable this functionality? Does anyone have a prototype I 
can leverage?

Need to replace from domain with some other domain -

Given: MSRP body is - From: cust3 
#015#012To: 
#015#012DateTime: .

Wanted: MSRP body is - From: cust3 #015#012To: 
#015#012DateTime: .

Thanks,
Pranathi Venkatayogi
System Developer II
(520) 745-9447 x4466
www.cyracom.com

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Re: [SR-Users] Kamailio+Asterisk - Kamailio doesn't forward INVITE to 2nd Kamailio

2017-06-15 Thread Marko Tirs

Hello Daniel,
> yes, if you partitioned the users and you know by first digit where it should 
> be registered, then all is fine -- update > the r-uri and then you can relay 
> to the other server.

I do it (update the $ur with the right destination IP) for all calls (INVITEs). 
If a SIP client registered on Kamailio1 calls a client registered on Kamailio2 
then it functions. If an other client calls through an Asterisk (which is 
registered on Kamailio) then it doesn't function! Kamailio1 doesn't forward 
INVITE to Kamailio2, Asterisk gets 'Busy'!Please see my test cases bellow: 3 of 
them function, 1 of them doesn't function!

Asterisk obviously doesn't send some part of the INVITE message which is needed 
to call another Kamailio.

Can you give me some advise how to solve it? Which other variable should be set 
to relay the INVITE to the 2nd Kamailio?

Thank youRegardsMarko

  From: Daniel-Constantin Mierla 
 To: Marko Tirs ; Kamailio (SER) - Users Mailing List 

 Sent: Wednesday, June 14, 2017 10:43 AM
 Subject: Re: [SR-Users] Kamailio+Asterisk - Kamailio doesn't forward INVITE to 
2nd Kamailio

 Hello,

 On 13.06.17 14:35, Marko Tirs wrote:

  Hi all, 
  I have here: Kamailio 1: 192.168.0.11 Kamailio 2: 192.168.0.21 Asterisk 1: 
192.168.0.12 
  Asterisk1 is registered on Kamailio1 as User 100. Asterisk1 shouldn't be 
registered on Kamailio2.

  User 111 registered on Kamailio1 User 211 registered on Kamailio2 User 121 
registered on Asterisk1 
  Test cases: - 111 calls 211 - OK 211 calls 111 - OK 121 
calls 111 - OK 121 calls 211 - Kamailio1 doesn't forward INVITE to Kamailio2, 
Asterisk gets 'Busy' !? 

  Is the approach just to replace destination IP-address in $ru, depending from 
1st digit, right?  

 yes, if you partitioned the users and you know by first digit where it should 
be registered, then all is fine -- update the r-uri and then you can relay to 
the other server.

 Cheers,
 Daniel

  If not, what is the right approach to reach the remote users from Asterisk, 
which are registered on remote Kamailio and not on the local Kamailio? 
  Thank you Regards Marko

  My changes in kamailio.cfg in both kamailios (based on kamailio-basic.cfg) : 
--
 kamailio1.bindip = "192.168.0.11"
 kamailio2.bindip = "192.168.0.21"
  ... route {
      if (is_method("INVITE")) {
         if($rU=~"^1[0-9][0-9]$") {
             $ru = "sip:" + $rU + "@" + $sel(cfg_get.kamailio1.bindip) + 
":5060";
         }
         else if($rU=~"^2[0-9][0-9]$") {
             $ru = "sip:" + $rU + "@" + $sel(cfg_get.kamailio2.bindip) + 
":5060";
         } }
  ...

  sip.conf in Asterisk1: --
  register => 100:abc@192.168.0.124:5060/100

 [100]
 type=friend
 host=192.168.0.11
 secret=abc
 context=kamailio

 [121]
 type=friend
 secret=abc
 host=dynamic
 context=kamailio

  extensions.conf --- [kamailio]
 exten => _[1-4]XX,1,Dial(SIP/100/${EXTEN},30)
 exten => _[1-4]XX,n,Hangup()

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Re: [SR-Users] Problem changing the SDP origin after a failure route

2017-06-15 Thread Pascal Poudrier


Thank you John for your idea but it seem that when rtpproxy_manage change the SDP, it also change the RTP port that fit the rtpproxy session.So basically, I need to call rtpproxy_manage with the invite to change the SDP attribute but when falling back to the failure_route, I need to destroy the current rtpproxy session and create a new one base on the new parameters of the session (the changed user and destination ip) for the RTP stream.So to manage this correctly, I may need to destroy the first created rtpproxy session and create a new one with the new invite parameters.I will try to work on this.On June 14, 2017 at 5:19 PM John Petrini  wrote:You might try using the fix_nated_sdp function from the nathelper module to rewrite the c line in the SDP body.___John PetriniNOC Systems Administrator   //   CoreDial, LLC   //   coredial.com   // Hillcrest I, 751 Arbor Way, Suite 150, Blue Bell PA, 19422 P: 215.297.4400 x232   //   F: 215.297.4401   //   E: jpetr...@coredial.comInterested in sponsoring PartnerConnex 2017? Learn more.The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission,  dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer.On Wed, Jun 14, 2017 at 5:00 PM, Pascal Poudrier  wrote:Hi, first of all, here's the architecture of what I'm trying to do :Public client --> Kamailio and there is a private freeswitch behind  kamailio. I use rtpproxy to connect the client and the freeswitch that is used at media server.If a call is not answered from userA to userB and userB doesn't have a voicemail configured, I route the Invite to FreeSwitch and I change the destination user to 66 that take care of playing a nice message that the user doesn't have a voicemail.My problem is that the SDP Connection doesn't get updated to the right IP. It get changed to the external IP instead of the internal.Here's the original INVITE Packet :13:21:22.824290 IP 8.8.8.16.30893 > 8.8.8.18.5060: SIP: INVITE sip:1...@mydomain.com SIP/2.0 Eh@.?...x..mINVITE sip:1...@mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK-3e3b9e22 From: 102;tag=cacbdfac40ddd17ao0 To:  Remote-Party-ID: 102;screen=yes;party=calling Call-ID: 3ae5b914-591b8732@192.168.0.236 CSeq: 101 INVITE Max-Forwards: 70 Contact: 102 Expires: 240 User-Agent: Cisco/SPA122-1.3.5r(003) Content-Length: 263 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 16833207 16833207 IN IP4 192.168.0.236 s=- c=IN IP4 192.168.0.236 t=0 0 m=audio 16396 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecvNote that 192.168.0.x in the internal network of the clients and 8.8.8.16 is the public IP of the client and 8.8.8.18 is the kamailio ip.Here's the INVITE packet sent from kamailio to Freeswitch13:21:32.947196 IP 172.16.0.18.5060 > 172.16.0.19.5080: SIP: INVITE sip:666@172.16.0.19:5080 SIP/2.0 E... ...@.  ...e...e...P1INVITE sip:666@172.16.0.19:5080 SIP/2.0 Record-Route:  Test-Pascal: SDP avec freeswitch Via: SIP/2.0/UDP 172.16.0.18;branch=z9hG4bK31ea.e52b370f8248032680211fc7c253902a.1 Via: SIP/2.0/UDP 192.168.0.236:5060;rport=30893;received=8.8.8.16;branch=z9hG4bK-3e3b9e22 From: 102;tag=cacbdfac40ddd17ao0 To:  Remote-Party-ID: 102;screen=yes;party=calling Call-ID: 3ae5b914-591b8732@192.168.0.236 CSeq: 101 INVITE Max-Forwards: 69 Contact: 102 Expires: 240 User-Agent: Cisco/SPA122-1.3.5r(003) Content-Length: 279 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 16833207 16833207 IN IP4 8.8.8.18 s=- c=IN IP4 8.8.8.18 t=0 0 m=audio 46134 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=nortpproxy:yesNote that 172.16.0.18 is the internal IP of kamailio and 172.16.0.19 is the internal IP of FreeSwitchHere's the revealing par of kamailio configuration that us use when falling back to freeswitch :failure_route[TOVOICEMAIL] {$ru = "sip:666@" + $sel(cfg_get.freeswitch.bindip)+ ":" + $sel(cfg_get.freeswitch.bindport);$du = "sip:666@"+ $sel(cfg_get.freeswitch.bindip)+ ":" + $sel(cfg_get.freeswitch.bindport);force_send_socket(udp:172.16.0.18:5060);if (is_request()) {    xlog("L_INFO","$ft -- NATMANAGE -- Request");    if(has_totag()) {

Re: [SR-Users] best practice to re-install rtpengine after jessie to stretch upgrade?

2017-06-15 Thread Daniel Tryba

On Thu, Jun 15, 2017 at 09:33:29AM +0300, Juha Heinanen wrote:
> I made a test upgrade from Debian Jessie to Stretch while rtpengine
> 5.1.1 was running on Jessie.  What is the best practice to get
> xt_RTPENGINE module installed in the new Stretch linux kernel, since it
> does not happen automatically.
> 
> Should I reinstall rtpengine* packages or is there some better way?

Running 5.0.x on Stretch a collegue fixed this by:
* adding SYSTEMCTL_SKIP_REDIRECT=yes to
  /etc/default/ngcp-rtpengine-daemon to just use the
  supplied init.d script
* adding the -w flag to iptables to avoid concurrency problems
  at boot time



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Re: [SR-Users] two way routing

2017-06-15 Thread Daniel Tryba

On Wed, Jun 14, 2017 at 05:28:35PM -0400, Jean Cérien wrote:
> I have the registration working, but when my kamailio receives an INVITE, I
> am a bit lost regarding what to do. in my route(AUTH), I am trying to check
> uac_reg_status, but no success.
 
> and the invite arriving
> INVITE sip:1100@192.168.2.228 SIP/2.0
> 
> Any help will be greatly appreciated

Either accept all INVITEs from the IPs you are registering, use
something like db_aliases to map 1100 to your known username or change
the setup of the PBX you are registering to to not use the dialed nummer
in the R-URI but the username (it is an asterisk so it is the difference
between Dial(SIP/${EXTEN}@username) and Dial(SIP/username))

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Re: [SR-Users] Does Kamailio send BYE if opposite side lost connection?

2017-06-15 Thread Abdoul Osséni

opposite site ie caller or callee ?

here the extract of the dialog documentation:
dlg_set_property(attr)

Set a dialog property - an attribute that enable/disable various behaviours
(e.g., sending keep alive requests).

Meaning of the parameters is as follows:

   -

   *attr* - name of property. It can be:
   - 'ka-src' - send keep alive OPTION requests to caller
  - 'ka-dst' - send keep alive OPTION requests to callee
  - 'timeout-noreset' - don't reset timeout on in-dialog messages
  reception


If keep alive is enabled for a dialog, the module will send SIP OPTIONS
requests with CSeq lower or equal than last request within dialog, with the
scope of detecting if the destination is still in the call. If the keep
alive request results in a local timeout or '481 Call Leg/Transaction Does
Not Exist', then the dialog is ended from the server.

If 'timeout-noreset' is set, dialog timeout won't be reset upon reception
of in-dialog messages (default behavior).

This function can be used from ANY_ROUTE.

Regards

2017-06-15 9:41 GMT+02:00 Nguyen Tran Nhan :

> Thank Abdoul,
> I use your suggest config and kamailio start now but there is no message
> when opposite site lost connection.
>
> Thanks,
> Nhan
>
> On Thu, Jun 15, 2017 at 2:07 PM, Abdoul Osséni 
> wrote:
>
>> Hi,
>>
>> I think, you can use the following config.
>>
>> .
>> .
>> .
>> #!define FLT_DLG 9
>> .
>> .
>> .
>> loadmodule "dialog.so"
>> .
>> .
>> .
>> # - dialog params -
>> modparam("dialog", "enable_stats", 1)
>> modparam("dialog", "dlg_flag", FLT_DLG)
>> modparam("dialog", "send_bye", 1)
>> modparam("dialog", "ka_timer", 5)
>> modparam("dialog", "ka_interval", 30)
>>
>> .
>> .
>> .
>> route[RELAY] {
>> .
>> .
>> .
>> if (is_method("INVITE")) {
>> setflag(FLT_DLG);
>> dlg_set_property("ka-src");
>> dlg_set_property("timeout-noreset");
>> }
>>
>> dlg_manage();
>> .
>> .
>> .
>>
>> Regards
>>
>> Abdoul.
>>
>>
>>
>>
>> 2017-06-15 9:00 GMT+02:00 Nguyen Tran Nhan :
>>
>>> Thanks, I am able to install dialog module. Just add dlg_flag config for
>>> starting kamailio. But I think this flag is deprecated. It is using
>>> dlg_manage() now?
>>>
>>> Thanks,
>>> Nhan
>>>
>>> On Wed, Jun 14, 2017 at 9:29 AM, Nguyen Tran Nhan >> > wrote:
>>>
 About installation dialog module, I add load dialog.so and it require
 to load outbound.so. After that it need to enable stcp. I think there is no
 stcp.so module? enable_stcp=1 does not work for me.

 Thanks,
 Nhan

 On Tue, Jun 13, 2017 at 8:59 AM, Nguyen Tran Nhan <
 nguyent...@nccsoft.vn> wrote:

> Thanks Daniel,
> I am using rtpproxy, so I think the patches is good for me. Have you
> got any information about patch for rtpproxy?
>
> Thanks,
> Nhàn
>
> On Mon, Jun 12, 2017 at 10:19 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> two options coming in my mind:
>>
>> 1) use dialog module and enable keepalive within the dialog
>> 2) if you use an rtp relay (for nat traversal, etc), then thee were
>> some patches that rtpproxy is sending an xmlrpc command to kamailio, 
>> which
>> via dialog module can send a bye. Not sure what is the current status 
>> with
>> that patch or if rtpengine has anything similar.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>> On 10.06.17 06:10, Nguyen Tran Nhan wrote:
>>
>> Hi all,
>> I am working in basic call. I got common cause that opposite side
>> (callee) lost connection and caller still hangging in the call. Is there
>> any message to server to caller in this case?
>>
>> Thanks,
>> Nhan
>>
>>
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>>
>>
>> --
>> Daniel-Constantin Mierlawww.twitter.com/miconda -- 
>> www.linkedin.com/in/miconda
>> Kamailio Advanced Training - www.asipto.com
>> Kamailio World Conference - www.kamailioworld.com
>>
>>
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>

>>>
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>>>
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>>
>>
>> --
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>> Ingénieur Réseaux et systèmes chez THALES
>> Co-Fondateur de ON SERVICES
>> Tél : +33 601 135 167
>>
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>>

Re: [SR-Users] Kazoo module json library kamailio 4.4.1

2017-06-15 Thread Grant Bagdasarian

Hi Daniel and Luis,

I've installed the json-c library, but it still isn't able to compile the kazoo 
module.

#locate json.h
/usr/include/json-c/json.h
/usr/local/src/kamailio-4.3/kamailio/lib/srutils/srjson.h
/usr/local/src/kamailio-4.3/kamailio/modules/kazoo/kz_json.h
/usr/local/src/kamailio-4.4/kamailio/lib/srutils/srjson.h
/usr/local/src/kamailio-4.4/kamailio/modules/cfgt/cfgt_json.h
/usr/local/src/kamailio-4.4/kamailio/modules/debugger/debugger_json.h
/usr/local/src/kamailio-4.4/kamailio/modules/kazoo/kz_json.h
/usr/local/src/kamailio-4.4.1/kamailio/lib/srutils/srjson.h
/usr/local/src/kamailio-4.4.1/kamailio/modules/cfgt/cfgt_json.h
/usr/local/src/kamailio-4.4.1/kamailio/modules/debugger/debugger_json.h
/usr/local/src/kamailio-4.4.1/kamailio/modules/kazoo/kz_json.h

Any ideas? Perhaps the wrong location?

Regards,

Grant

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Luis 
Azedo
Sent: woensdag 14 juni 2017 11:19
To: Kamailio (SER) - Users Mailing List ; 
mico...@gmail.com
Subject: Re: [SR-Users] Kazoo module json library kamailio 4.4.1

Hi,

Daniel is correct (as usual).

Best

From: sr-users 
>
 on behalf of Daniel-Constantin Mierla 
>
Sent: Wednesday, June 14, 2017 9:49:46 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kazoo module json library kamailio 4.4.1

Hello,

from the include header file name, it seems to be json-c project. But maybe 
someone more familiar with the module can confirm better.

Cheers,
Daniel

On 13.06.17 09:57, Grant Bagdasarian wrote:
Hi,

Could someone please tell me which json library I have to install in order for 
the kazoo module to compile?
I've tried many different json library, none of them work.
It keeps complaining:

make all
CC (gcc) [M kazoo.so]   kz_trans.o
In file included from kz_trans.c:53:0:
kz_json.h:12:18: fatal error: json.h: No such file or directory
#include 
  ^
compilation terminated.
../../Makefile.rules:97: recipe for target 'kz_trans.o' failed
make[1]: *** [kz_trans.o] Error 1
Makefile:511: recipe for target 'modules' failed
make: *** [modules] Error 1

There is another thread regarding this same issue, but no solution was provided 
by the creator of the thread.
Any ideas?

Regards,

Grant

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[SR-Users] Topos: contact header is not updated avec RE-INVITE

2017-06-15 Thread Abdoul Osséni

Hello,

I have an issue when loading topos module on Kamailio 5.0.2 version.

root@proxy:/home/tcpdump# kamailio -V
version: kamailio 5.0.2 (x86_64/linux)



The call flow is: uac --> kamailio --> Asterisk

1) Invite from uac
2) Kamailio forward the invite to asterisk
3) asterisk send 200 OK to kamailio
4) kamailio send 200 OK to uac -> in the conctact header, i can see the IP
address of kamailio
5) uac send ACK to kamailio and kamailio forward the ACK to Asterisk
6) uac send re-invite: "media change" to kamailio.
7) kamailio forward the re-invite to asterisk and asterisk send 200 OK to
kamailio
8) kamailio send 200 ok to uac --> but the contact header contains IP of
Asterisk and not kamailio: is it normal?

Thank for you help.

Regards
-- 
Abdoul OSSENI
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Re: [SR-Users] Does Kamailio send BYE if opposite side lost connection?

2017-06-15 Thread Abdoul Osséni

Hi,

I think, you can use the following config.

.
.
.
#!define FLT_DLG 9
.
.
.
loadmodule "dialog.so"
.
.
.
# - dialog params -
modparam("dialog", "enable_stats", 1)
modparam("dialog", "dlg_flag", FLT_DLG)
modparam("dialog", "send_bye", 1)
modparam("dialog", "ka_timer", 5)
modparam("dialog", "ka_interval", 30)

.
.
.
route[RELAY] {
.
.
.
if (is_method("INVITE")) {
setflag(FLT_DLG);
dlg_set_property("ka-src");
dlg_set_property("timeout-noreset");
}

dlg_manage();
.
.
.

Regards

Abdoul.




2017-06-15 9:00 GMT+02:00 Nguyen Tran Nhan :

> Thanks, I am able to install dialog module. Just add dlg_flag config for
> starting kamailio. But I think this flag is deprecated. It is using
> dlg_manage() now?
>
> Thanks,
> Nhan
>
> On Wed, Jun 14, 2017 at 9:29 AM, Nguyen Tran Nhan 
> wrote:
>
>> About installation dialog module, I add load dialog.so and it require to
>> load outbound.so. After that it need to enable stcp. I think there is no
>> stcp.so module? enable_stcp=1 does not work for me.
>>
>> Thanks,
>> Nhan
>>
>> On Tue, Jun 13, 2017 at 8:59 AM, Nguyen Tran Nhan 
>> wrote:
>>
>>> Thanks Daniel,
>>> I am using rtpproxy, so I think the patches is good for me. Have you got
>>> any information about patch for rtpproxy?
>>>
>>> Thanks,
>>> Nhàn
>>>
>>> On Mon, Jun 12, 2017 at 10:19 PM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
 Hello,

 two options coming in my mind:

 1) use dialog module and enable keepalive within the dialog
 2) if you use an rtp relay (for nat traversal, etc), then thee were
 some patches that rtpproxy is sending an xmlrpc command to kamailio, which
 via dialog module can send a bye. Not sure what is the current status with
 that patch or if rtpengine has anything similar.

 Cheers,
 Daniel



 On 10.06.17 06:10, Nguyen Tran Nhan wrote:

 Hi all,
 I am working in basic call. I got common cause that opposite side
 (callee) lost connection and caller still hangging in the call. Is there
 any message to server to caller in this case?

 Thanks,
 Nhan


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 www.linkedin.com/in/miconda
 Kamailio Advanced Training - www.asipto.com
 Kamailio World Conference - www.kamailioworld.com


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>>>
>>
>
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>
>


-- 
Abdoul OSSENI
Ingénieur Réseaux et systèmes chez THALES
Co-Fondateur de ON SERVICES
Tél : +33 601 135 167
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Re: [SR-Users] Does Kamailio send BYE if opposite side lost connection?

2017-06-15 Thread Nguyen Tran Nhan

Thanks, I am able to install dialog module. Just add dlg_flag config for
starting kamailio. But I think this flag is deprecated. It is using
dlg_manage() now?

Thanks,
Nhan

On Wed, Jun 14, 2017 at 9:29 AM, Nguyen Tran Nhan 
wrote:

> About installation dialog module, I add load dialog.so and it require to
> load outbound.so. After that it need to enable stcp. I think there is no
> stcp.so module? enable_stcp=1 does not work for me.
>
> Thanks,
> Nhan
>
> On Tue, Jun 13, 2017 at 8:59 AM, Nguyen Tran Nhan 
> wrote:
>
>> Thanks Daniel,
>> I am using rtpproxy, so I think the patches is good for me. Have you got
>> any information about patch for rtpproxy?
>>
>> Thanks,
>> Nhàn
>>
>> On Mon, Jun 12, 2017 at 10:19 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> two options coming in my mind:
>>>
>>> 1) use dialog module and enable keepalive within the dialog
>>> 2) if you use an rtp relay (for nat traversal, etc), then thee were some
>>> patches that rtpproxy is sending an xmlrpc command to kamailio, which via
>>> dialog module can send a bye. Not sure what is the current status with that
>>> patch or if rtpengine has anything similar.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>> On 10.06.17 06:10, Nguyen Tran Nhan wrote:
>>>
>>> Hi all,
>>> I am working in basic call. I got common cause that opposite side
>>> (callee) lost connection and caller still hangging in the call. Is there
>>> any message to server to caller in this case?
>>>
>>> Thanks,
>>> Nhan
>>>
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing 
>>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierlawww.twitter.com/miconda -- 
>>> www.linkedin.com/in/miconda
>>> Kamailio Advanced Training - www.asipto.com
>>> Kamailio World Conference - www.kamailioworld.com
>>>
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing List
>>> sr-users@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>
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[SR-Users] best practice to re-install rtpengine after jessie to stretch upgrade?

2017-06-15 Thread Juha Heinanen

I made a test upgrade from Debian Jessie to Stretch while rtpengine
5.1.1 was running on Jessie.  What is the best practice to get
xt_RTPENGINE module installed in the new Stretch linux kernel, since it
does not happen automatically.

Should I reinstall rtpengine* packages or is there some better way?

-- Juha

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