Hello,
courtesy lod.com (sponsoring the hardware and bandwidth) and Fred Posner
helping with that, we got a server located in USA to use for the
project. Main purpose for now is using it as a RPM repository, available
as rpm.kamailio.org .
The server takes the packages from open suse build service, making a
local mirror. It is not yet automated, but should happen in the future
(anyone that can help here with some script, is more than welcome to
join the team) -- anyhow, we generate the rpms only at stable releases.
The mirror should be usable as an yum repository or individual package
download via http, more details at:
- http://rpm.kamailio.org
I am not an user of a distro using RPMs, therefore I am asking
interested people in the community to give it a try and report here
eventual issues.
Also, if you build packages for other distributions, we can mirror them
on the server -- get in touch to setup the process.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
Hi,
My proxy is bound to a.b.c.d:5060 and secondarily to e.f.g.h:5060.
Most traffic comes into a.b.c.d:5060 and is relayed out that interface.
Some initial INVITEs are routed out of e.f.g.h:5060, forced via $fs. The
problem is that the Record-Route header still reflects a.b.c.d:5060 and
that sequential messages from both ends end up entering and exiting
a.b.c.d:5060. Many endpoints don't like that; for instance, e2e ACKs
from the caller that come from a.b.c.d:5060 while the initial INVITE
came from e.f.g.h:5060 can be a problem in that they are not matched by
some UASs.
What is the best way to approach this? Is this something covered by
Kamailio's 'mhomed' core param?
Thanks,
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Please be kind to the English language:
http://www.entrepreneur.com/article/232906
Hello. I Use UAC module for register to my porviders (I have several
porviders at my server and). I successfully ring from provider to my
asterisk servers and route calls to users that registers at Kamailio.
So now I need route calls from users to my providers, But can not
understand how to send INVITE to provider?
Doesanyone has working example for it?
Hello Davy,
My constraint is that I cannot use TCP for this solution.
I am attaching my kamailio.cfg file, please please help me to resolve this
issue as soon as possible (it has already chewed many nights :( )
On Wed, Aug 13, 2014 at 8:50 PM, davy van de moere <
davy.van.de.moere(a)gmail.com> wrote:
> Hey Rahul,
>
> thank you for your pcap file. Your astpp is indeed not succeeding in
> getting the OK delivered correctly to your pjsua. Also those 400 Bad
> Session Description are no good.
>
> I do see as well fragmented packets passing by, perhaps an easy test would
> be to have everything be sent over TCP. Big sip packets tend to suffer over
> UDP networks. From your sip packets your setup looks quite standard,
> although with a lot of headers (which is fine), so I'ld give it a try with
> sending everything over tcp, t_relay_to_tcp should help you out!
>
> Good luck!
>
>
> --- In reply to ---
> I guess the attachment size was greater than 60Kb thereby getting
> quarentined.
> Re-sending it to you so that this issue could be resolved.
>
>
> 2014-08-12 14:42 GMT+02:00 davy <davy.van.de.moere(a)gmail.com>:
>
> If your Kamailio setup is close to vanilla, it should do it by default.
>> But Kamailio is a very powerfull tool, it can easily be setup to stop
>> passing over ACKs :)
>>
>> To attempt to answer what explicitly sends along the ACKs, that will most
>> likely be the t_relay function, together with the logic which went before
>> it…
>>
>> A good old tcpdump will most likely enlighten us.
>>
>> Op 12-aug.-2014, om 14:39 heeft Rahul MathuR <rahul.ultimate(a)gmail.com>
>> het volgende geschreven:
>>
>> Hello Davy,
>>
>> Thanks for writing back..
>>
>> Tonight I'll take the tcpdump on Kamailio box and share the file.
>>
>> Please note that Kamailio and Freeswitch are both on public IP & at
>> Freeswitch param, enable_timer=false is set.
>>
>> Is there any explicit way wherein ACKs can be transmitted to FS ?
>>
>>
>> Thanks in advance !
>>
>>
>> On Tue, Aug 12, 2014 at 1:15 PM, davy van de moere <
>> davy.van.de.moere(a)gmail.com> wrote:
>>
>>> Are you sure you're getting the ACK correctly to FS?
>>>
>>> FS typically has this behavior when it did not correctly receive a
>>> confirmation of an answer, and after 30 seconds disconnects, as for FS the
>>> call has failed.
>>>
>>> Do you have a trace of the packets?
>>>
>>> grtz,
>>> Davy Van De Moere
>>>
>>>
>>> 2014-08-12 13:37 GMT+02:00 Rahul MathuR <rahul.ultimate(a)gmail.com>:
>>>
>>>> Hello,
>>>>
>>>> I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and
>>>> Sip server FreeSwitch.
>>>> Whenever I call directly from UAC to Sip server, the call gets
>>>> established for as long as I want, however when I use the proxy in between,
>>>> it gets disconnected within 30 seconds. It seems that FS sends a BYE within
>>>> 30 seconds.
>>>>
>>>> I would really appreciate if anybody please guide me where I am going
>>>> wrong in this case ?
>>>>
>>>> --
>>>> Warm Regds.
>>>> MathuRahul
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users(a)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(a)lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>>
>> --
>> Warm Regds.
>> MathuRahul
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users(a)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(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Warm Regds.
MathuRahul
--
Warm Regds.
MathuRahul
Hi List,
I am trying to keep a counter for number of messages received per source ip
in my kamailio script.
The basic challenge is to keep this data in a neat structure.
Ideally, I want some array with keys named after source IP addresses,
sth like number_of_messages["10.10.10.10"] = 23
Of course, this data structure shall be script-persistent (as opposed to
transaction persistent)
Is there any suggestion to do this? (keep a record of # of messages
received from each source IP address and log them)?
Consider the following snippet:
if (is_present_hf("Contact")) {
xlog("L_ALERT", "===== reply to SUBSCRIBE has Contact: $ct\n");
xlog("L_ALERT", "===== want to replace with $td\n");
xlog("L_ALERT", "===== regexp to use is /$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asterisk.bindport)/$td/\n");
if (subst_hf("Contact", "/127.0.0.1:5080/$td/", "a")) {
xlog("L_ALERT", "===== reply had Contact modified\n");
}
}
If I use the hardcoded regexp "/127.0.0.1:5080/$td/", subst_hf() replaces the Contact value correctly.
However, if I use "/$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asterisk.bindport)/$td/" (where asterisk.bindip and asterisk.bindport are the supposed values 127.0.0.1 and 5080), subst_hf() fails to replace the Contact value.
How do I make use of the existing configuration variables in order to substitute in the Contact header?
Hello,
is there a way to make SCSCF do forking based on SIP method ? The problem is that our implementation on app server is not fully done and we want INVITE to be forked the way it is but OPTIONS shouldn't be forked there since forking is happening twice in this case which results to storm of OPTIONS on the network.
thanks
--
Daniel Ciprus
Integration engineer
http://www.acision.com
9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com>
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somehow i am getting following error, if i start radius
Error: Errors reading dictionary: dict_init:
/etc/freeradius/dictionary.opensips[93]: dict_addattr: Duplicate attribute
name Event-Timestamp
If i remove it from /etc/freeradius/dictionary.opensips file it works! but
my it screwed my time in mysql table
ATTRIBUTE Event-Timestamp 230 string
Just run grep to find out but it is not anywhere define but still i am
getting error
root@sip:/etc/freeradius# grep "Event-Timestamp" * -r
dictionary.opensips:ATTRIBUTE Event-Timestamp 230 string
sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \
sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \
sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \
sql.conf: FROM_UNIXTIME(%{Event-Timestamp}), \
sql.conf: FROM_UNIXTIME(%{Event-Timestamp}), \
sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \
sql.conf: UNIX_TIMESTAMP('%S') - '%{Event-Timestamp}', \
Hello;
I am trying to build a prepaid accounting system. I use dialog module's
start/end event-route. When even_route[dialog:started] triggered, a
http_query works for the started call. When event_route[dialog:end] triggered , a
http_query works for call is ended. When http_query works in
event_route[dialog:started], i get some CRITICAL ERROR like
'Aug 12 12:16:09 /usr/local/sbin/kamailio[10151]: CRITICAL: dialog
[dlg_timer.c:205]: update_dlg_timer(): Trying to update a bogus dlg
tl=0x7f7634d925d8 tl->next=(nil) tl->prev=(nil)'
'Aug 12 12:16:09 /usr/local/sbin/kamailio[10151]: ERROR: dialog
[dlg_handlers.c:1263]: dlg_onroute(): failed to update dialog lifetime'
Http_query works very well. I can get all call information on Web
Servers but default_time out or other timeout isn't working.
After i commented out http_query in event_route[dialog:started] , it stopped giving
ERROR but i realized that dialog cannot uptade dialog hash map .I can see all finished call with "kamctl dialog show". After i closed
http_query in even_route[dialog:end] , all problems are solved.
I think a problem occured between http_query and dialog event_routes.It
cannot control dialog lifetime.Any idea about this problem or suggest an alternative way will be appreciated.My configuration is as follows.
Best Regards
Betül.
#---------------- dialog params -------------
#!ifdef WITH_DIALOG
modparam("dialog", "enable_stats", 1)
modparam("dialog", "hash_size", 8192)
modparam("dialog", "rr_param", "did")
modparam("dialog", "dlg_flag",4)
modparam("dialog", "timeout_avp", "$avp(i:10)")
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 3600)
modparam("dialog", "detect_spirals", 1)
modparam("dialog", "profiles_with_value", "userid ; opkodu ; useropkodu ")
modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n")
modparam("dialog", "send_bye", 1)
#!endif
route[CHCK_DLG]{
xlog("L_INFO","DLG:AVP=> :>:$avp(i:77):$avp(op_kodu):$avp(i:69) ");
$dlg_ctx(timeout_bye)=1;
$avp(unique_id)=$sruid;
/* Time out suresi i:10 konusabilecegi sure saniye */
$avp(i:10)=0;
$avp(i:10)=(int)$avp(i:77);
#$avp(i:10)=10;
$dlg_ctx(timeout_route)=33;
$dlg_var(uniqueid)=$avp(unique_id);
$dlg_var(userid)=$avp(userid);
$dlg_var(opkodu)=$avp(op_kodu);
set_dlg_profile("userid","$avp(userid)");
set_dlg_profile("opkodu","$avp(op_kodu)");
set_dlg_profile("useropkodu","$avp(userid):$avp(op_kodu)");
if(get_profile_size("useropkodu","$avp(d_user_opkodu)")){
xlog("L_INFO","Userid_Size:$avp(d_user_opkodu) kk :
$avp(userid):$avp(op_kodu) ");
xlog("L_INFO","Userid_Limit:$sht(sayac=>$avp(userid):$avp(op_kodu))");
if(!($sht(sayac=>$avp(userid):$avp(op_kodu))>$avp(d_user_opkodu))){
sl_send_reply("403","Kapasite Asildi.");
exit;
}
}
dlg_manage();
return;
}
event_route[dialog:start]{
xlog("L_ALERT","START:CI:$dlg(callid):u_id:$dlg_var(uniqueid):U_id:$dlg_var(userid)
");
xlog("L_ALERT","START:lifetime:$avp(i:10):opkodu:$dlg_var(opkodu):$avp(aranan)"
);
#!ifdef WITH_UTILS
$var(http_res)=http_query("http://bla.com/somestuff.php?userid=$avp(userid)&aranan=$avp(aranan)&callid…","$var(result)");
if($var(http_res)=="200"){
json_get_field("$var(result)","sonuc","$var(snc)");
json_get_field("$var(result)","yorum","$var(yorum)");
json_get_field("$var(result)","yorum","$var(debug)");
if($var(snc)!=200){
xlog("L_ERR","START:CI:$ci:Sonuc:$var(snc):Yorum:$var(yorum):Debug:$var(debug)");
}else{
xlog("L_INFO","START:CI:$ci:Sonuc:$var(snc)");
}
}else{
xlog("L_ERR","START:CI:$ci:HTTP_RESULT:$var(http_res)");
}
#!endif
}
event_route[dialog:end]{
xlog("L_ALERT","END
::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::: :");
xlog("L_ALERT","END:CI:$ci:START_TIME:$dlg(start_ts):NOW:$TS");
$var(billsec)=$TS-$dlg(start_ts);
xlog("L_ALERT","END:SURE : $var(billsec): u_id:
$dlg_var(uniqueid):code:$T_reply_code:$DLG_status ");
xlog("L_ALERT","END:U_ID:$dlg_var(userid):O_Kodu:$dlg_var(opkodu):");
if($var(billsec)<0){
$var(billsec)=$DLG_lifetime;
xlog("L_ERR","START:CI:$ci:Billsec Negatif geldi :
lifetime alinacak:$DLG_lifetime");
}
#!ifdef WITH_UTILS
$var(http_res)=http_query("http://bla.com/somestuff.php?uniqueid=$dlg_var(uniqueid)&billsec=$var(bills…","$var(result)");
if($var(http_res)=="200"){
json_get_field("$var(result)","sonuc","$var(snc)");
json_get_field("$var(result)","yorum","$var(yorum)");
json_get_field("$var(result)","yorum","$var(debug)");
if($var(snc)!=200){
xlog("L_ERR","START:CI:$ci:Sonuc:$var(snc):Yorum:$var(yorum):Debug:$var(debug)");
}
}else{
xlog("L_ERR","START:CI:$ci:HTTP_RESULT:$var(http_res)");
}
#!endif
xlog("L_ALERT","END
::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::: :");
}