Hi all,
I'm working in a project for a service provider in which prepaid is an
essential requirement. They have about 12000 subscribers.
The core infrastructure will be Kamailio+RTPProxy while I have some
doubts about the prepaid feature. I'm thinking about a B2BUA (SEMS or
Freeswitch) that is called if the user belong to a "prepaid group" and
perform authorization and accounting; in this case the SIP flow could
be: Kamailio -> B2BUA(Prepaid) -> Kamailio. Is a good choice?
At the moment I cannot estimate the effort needed to develop this
section (base prepaid feature) of the project... Do you have some hints?
What are the prepaid solution implemented by you? Do you use your own
developed solution or do you think that is better to choose a commercial
solution (that provide also calling-card and other features)?
I have seen some commercial products (Portabilling, Jerasoft/Bilberry,
...); what do you think about?
Best Regards,
Hi,
i am using different modules like "mtree", "carrierroute", "drouting" and
so.
how can i tell how long did the reload takes?
how do i know if it fails?
is there a way to get a return code from kamailio that confirms that the
reload ended and successfuly?
BR,
Uri
Hello,
On 1/6/12 10:50 PM, Javier Vidal wrote:
> Hi Daniel, problem solved!!! thanks, was a bad REGISTER, i was wrong.
>
> Thank you!!!!
welcome, jut keep the mailing list always on cc, to help others to look
for solutions of similar cases.
Cheers,
Daniel
>
> El 06/01/2012, a las 19:34, Daniel-Constantin Mierla escribió:
>
>> Hello,
>>
>> On 1/6/12 6:47 PM, Javier Vidal wrote:
>>> Hi all,
>>>
>>> Two days ago, i had activated topoh module, and now in log file, when i receive a Options, appears this error
>>>
>>> /usr/sbin/kamailio[29936]: WARNING: topoh [th_msg.c:1063]: Invalid/Unparsed CSeq in message. Skipping.
>>>
>>> I dont' know solve it, only deactivating the topoh module!!
>> does the OPTIONS have the CSeq header? Can you paste here an ngrep trace of such OPTIONS?
>>
>> The warning message is printed from a function that tries to detect whether is a REGISTER or PUBLISH, which are requests that don't get masked since they end to a service.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi all,
Two days ago, i had activated topoh module, and now in log file, when i receive a Options, appears this error
/usr/sbin/kamailio[29936]: WARNING: topoh [th_msg.c:1063]: Invalid/Unparsed CSeq in message. Skipping.
I dont' know solve it, only deactivating the topoh module!!
Sorry for my english. ;)
Regards.
Javier Vidal
Dear Daniel,
Greetings and Happy New year to you all! I would like to thank you very much for your detailed and well structured tutorials on Kamailio. I am resending this email as I sent the earlier one before I was a member of the mailing list and I thought it may have been sent to the spam folder. If not then, I apologize for double posting.
I was trying to design a similar scenario like the one " Run your own SIP VoIP service on both IPv4 and IPv6 " as explained on Asipto web-page(http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). However I ran into difficulties with rtpproxy. The proxy is not routing media packets as it should. I followed the instruction regarding the kamailio.cfg file and edited the necessary lines according to my settings. I think my problem comes from setting up the rtpproxy.
1. I edited the /etc/default/rtpproxy file. i.e. I uncommented the line - CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock"
2. I started the rtpproxy with this command: rtpproxy -F -l ADDR_IPV4 -6 /ADDR_IPV6 . However, after this command the system returns the message : rtpproxy: setbindhost: No address associated with hostname
It displays the same message even when I replace ADDR_IPV4 and ADDR_IPV6 with IP addresses.
3. I have also tried using the udp socket:- CONTROL_SOCK=udp:127.0.0.1:22222, of course I have replaced the unix socket by this udp socket in the kamilio.cfg file.
So my question is, what did I do wrong with the settings that SIP messages are traversing normal from IPv4 client to IPv6 client and vice versa but media files(RTP packets) are not? The rtpproxy receives the media files from one client but fails to forward them to the other client.
Thanks for your help.
Maedot
Hi,I'm using the new kamailio 3.2 which has some changes in rtpproxy module. And i'm finding it hard to implement a certain configuration. Because of our network infrastructure i have to repackage rtp streams in one direction. Now in kamailio 3.1.5 i used to do this in this way:In main routeroute[RTPPROXY]{if(is_method("BYE"){ unforce_rtp_proxy();else if(is_method("INVITE")){ force_rtp_proxy();}in onreply_route force_rtp_proxy("z160"); Now in kamailio 3.2 the force_rtp_proxy() is replaced by rtpproxy_manage(). Here the script reads like:main routeroute[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return; rtpproxy_manage("z150"); if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
onreply_routethis block is called route[NATMANAGE]. Now if i put rtpproxy_manage("z150"), it tries to repackage in both direction which i do not want. So i need to write a separate block for onreply_route instead of calling route[NATMANAGE]. So please help me to write a block for onreply_route.Thanks in advance.
With Best Regards
Ariful Hossain Tuhin
email: 1. etothepowerpi(a)gmail.com 2. etothepowerpi(a)hotmail.com 3.etothepowerpi(a)yahoo.com
skype: freeburn1986
Hi,
can anyone see why this code wouldn't work? In the log is shows this
message after set_dlg_profile called - dialog [dlg_hash.c:523]: no
dialog callid='...' found
Dialogs are being saved to dialog table ok but nothing is in dialog_vars table.
modparam("dialog", "dlg_flag", DLGFLAG)
modparam("dialog", "hash_size", 128)
modparam("dialog", "default_timeout", 14400)
modparam("dialog", "dlg_match_mode", 0)
modparam("dialog", "table_name", "dialog")
modparam("dialog", "db_url", DBURL)
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_update_period", 10)
modparam("dialog", "profiles_with_value", "caller ; allcalls")
...
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
setflag(DLGFLAG); ## add to dialog list
$var(callcnt) = 0;
get_profile_size("caller", "$fu", "$var(callcnt)");
xlog("$fu has $var(callcnt) concurrent calls\n");
if ($var(callcnt) >= 2) {
sl_send_reply("503", "Simultaneous calls limit reached");
exit;
}
set_dlg_profile("caller","$fu");
}
--
Thanks
Ben
Hi,
we are using the PUA_XMPP and PUA modules from the master branch. When
the modules are started, everything are ok, the presence events from
XMPP are sent to kamailio SIP servers (PUBLISH/SUBSCRIBE) and cached
in the hash. But when there are several thousands records in the hash
tabel, the update_pua function called in the hashT_clean gives a lot
of "No memory left" error when the hashT_clean is waked up from the
time:
ERROR: pua [send_subscribe.c]: No memory left
ERROR: pua [pua.c]: while building tm dlg_t structure
The failed call is:
td = (dlg_t*)pkg_malloc(size);
if(td == NULL)
{
LM_ERR("No memory left\n");
return NULL;
}
in dlg_t* pua_build_dlg_t(ua_pres_t* presentity) function in
send_subscribe.c. The size is about 400 and something... It's
strange.....
Is it the memory leak in the PUA module?
I also try to increase the pkg_memory from 4MB default to 16MB, but it
doesn't help.
Any Idea?
Thanks in advanced
Laura
Hi
I do have 3 Kamailio servers, one for mobile phone registrations, one
for softphone registrations and one for SIP device registrations. Each
of those devices connects to it's perspective kamailio server
sip1.domain.comsip2.domain.comsip3.domain.com
All 3 Kamailio servers share the same database, and users can use
their kamailio user/pwd on any of the devices, now I want to use
encrypted passwords and remove clear text passwords from the database.
I did test with one server and all is fine,however if a user want to
register from the second kamailio server it does not work, basically
because the db domain entry from which the hash is created is
sip1.domain.com and stored in the db, while the user connects from to
sip2.domain.com this eventually generates a different hash.
Is there anyway to overcome this ? Can I exclude Domain from Hash
generation ? Any other option that allows me to do the above ?
Thanks