Hello,
not so many fixes since the release of 5.3.1, but it looks enough to
release a new minor version from branch 5.3. I am planning to do in on
Thursday, Jan 9, 2020, if there are no new reports of relevant issues
that make us to reconsider and postpone.
Soon after we should probably do a release from branch 5.2 as well as
the last one from branch 5.1.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World …
[View More]Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
[View Less]
Greetings,
I'm testing some portability scenarios with carriers and i've ran into some
issues.
Take this example URI from ETSI TR 184 003 (
https://www.etsi.org/deliver/etsi_tr/184000_184099/184003/03.01.01_60/tr_18…
)
- Sip:+49234598765;npdi;rn=+49-D123@provider5.net;user=phone
In this case i need to parse user, user's parameters, domain and uri
parameters.
In order to that i'm using uri's transformations like {uri.user},
{uri.domain}, etc.
However, when "user=phone" is present, Kamailio'…
[View More]s parser works differently.
Without "user=phone" -> {uri.user} is equal to
"+49234598765;npdi;rn=+49-D123"
With "user=phone" -> {uri.user} is equal to "+49234598765" .
I really need to access "rn" and "npdi" parameters in those cases. How can
i do it when "user=phone" is present?
Is there a way to disable the "user=phone" exception for parsing?
Best Regards,
Duarte Rocha
[View Less]
Hi List,
History:
* In the past, I had deadlock which was, most probably, related to ssl1.1.
We have discussed this issue, and a fix is supposed to workaround the
issue that was detected.
* With latest 5.2.X, I have experienced ONCE a similar behavior with TCP
and TLS being mostly stuck. I have not been using this version much, but
the fix was supposed to be in the core of kamailio.
The status of the server this night:
* I'm today running version: kamailio 5.3.1 (x86_64/linux),
* Installed …
[View More]on stretch using http://deb.kamailio.org/kamailio53 repository.
* This versions use libssl1.1
* A user reported that he can't connect with TCP
* An average of 5000 IPs per 10 minutes are being banned by the pike module
(could be twice the same)
Yesterday/Today:
* at the end of the outage, I had 2479 IP in my ipban htable. (which is
equivalent to my statistics showing 2 bans/IP every 10 minutes = 5000)
* looking at my logs, it appears that most (ALL?) ip being banned... are my
regular users.
* looking at my logs, I can't understand why pike would block them.
This is a graph for statistics on my service for the last 24 hours:
https://www.antisip.com/sip-antisip-com-register/status2.html
Yesterday, at 22:18:39, kamailio started to BAN some IPs. 52 IPs were
banned in a period of 10 minutes. I can confirm this from my logs.
My pike configuration is this one:
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 64)
modparam("pike", "remove_latency", 4)
When detecting the issue, this morning, I typed:
$> sudo kamctl stats
$> sudo kamcmd htable.dump ipban
//FAILURE (answer too large...)
$> sudo kamctl trap
Then, I started an agent with TCP and it worked...???
Then, a few seconds, may be a minute after:
$> sudo kamcmd htable.dump ipban
//SUCCESS and shows 2479 banned ip.
and... everything is back to normal in a few minutes.
I haven't restarted kamailio, and all statistics are as expected, as usual.
Thus, it looks that " sudo kamctl trap" has triggered something. I already
experienced a similar behavior -when testing my ssl1.1 deadlock last year-.
2 questions:
1/ I beleive my "pike" configuration should not ban users. Is my pike
configuration wrong?
As an example, pike has banned an IP sending one message/second. I believe
my configuration should accept that?
2/ Could there still be a TLS issue with libssl1.1?
This is the result of the "kamctl trap":
https://sip.antisip.com/kamailio-pike-or-tls-issue-13-12-2019.kamctl-trap
Sorry for the long story & hoping to find a long term solution or at least
a workaround!
Regards
Aymeric
--
Antisip - http://www.antisip.com
[View Less]
Dear Kamailio team,
Greetings from Multiicon.
We setuping kamailio sip server and all thing are done but when we going to
add user we are getting Command not found.
Using below command. I also check the server status by using systemctl
status kamailio is showing active
Please advise on this.
Thanks & Regards,
VIVEK AGHERA
#technical department
IND : +91 63548 23958
<mailto:vivek@multiicon.in> vivek(a)multiicon.in
MULTIICON IDEOTECHNOLOGY …
[View More]PVT.LTD.
3rd Floor, Rumi Plaza, Airport Road, Rajkot - 360 001, Gujarat (INDIA)
Please consider the environment before printing this e-mail.
[View Less]
Hello,
I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine.
So far, everything is working fine, I'm able to register an extension and
make a call, but for some reason, when i'm trying to call a WebRTC
extension from any SIP Extension Kamailio is sending INVITE, WebRTC
extension is sending back 200 OK, and then Kamailio is trying to send an
ACK through UDP protocol, and not through wss, as it's supposed to do. This
is how invite is looking:
INVITE sip:nl7oe4ss@…
[View More]vjbh7r4im6j7.invalid;transport=wss SIP/2.0
Record-Route: <sip:my-company.net
;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Via: SIP/2.0/WSS 123.123.123.123:10443
;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0
Via: SIP/2.0/UDP 192.168.50.237:5060
;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060
Max-Forwards: 70
From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c
To: <sip:15@192.168.50.210:5060>
Contact: <sip:11@192.168.50.237:5060>
Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net
CSeq: 102 INVITE
User-Agent: Proxy
Date: Wed, 03 Apr 2019 17:11:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Alert-Info:
Content-Type: application/sdp
Content-Length: 596
Server: SIP Proxy
and then WebRTC app is replying with 200 OK:
SIP/2.0 200 OK
Record-Route: <sip:my-company.net
;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Via: SIP/2.0/WSS 123.123.123.123:10443
;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0
Via: SIP/2.0/UDP 192.168.50.237:5060
;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060
To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6
From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c
Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net
CSeq: 102 INVITE
Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Proxy-WEBRTC
Content-Type: application/sdp
Content-Length: 901
and finally, Kamailio is trying to send this ack through UDP protocol:
ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0
Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport
Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes>
Max-Forwards: 70
From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c
To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6
Contact: <sip:11@192.168.50.237:5060>
Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net
CSeq: 102 ACK
User-Agent: Proxy
Content-Length: 0
If i'm trying to force it through TLS, i'm receiving error:
get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443,
to udp:22.22.22.22:23317)
Can someone point me in the right direction, please?
Thank you.
[View Less]
Hi,
Am using Kamailio 5.1.9 version
My Setup : client1 -> kamailio server 1 ( IP : 10.211.160.172) ---->
kamailio server 2( IP : 10.211.160.176) -> client2
I have a scenario where kamailio server 1 has to initiate an outgoing tls
connection to kamailio server 2, i have set the server_name and server_id
in the client profile in tls.cfg like below on kamailio server 1
[client:default]
verify_certificate = no
require_certificate = no
server_name = mahesh.client.com
[client:10.211.…
[View More]160.172:5061]
method = TLSv1+
verify_certificate = yes
require_certificate = yes
private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key
certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt
ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt
cipher_list = RSA
verify_depth = 9
server_name = btip.176.com
server_id = btip.176.com
And in sar.cfg
$xavp(tls=>server_name)="btip.176.com";
$xavp(tls=>server_id)="btip.176.com";
$du = "sip:10.211.160.176:5061;transport=tls";
....
t_relay();
What i observe is that , when client hello is sent by 10.211.160.172 to
10.211.160.176, i dont see Extension server_name being sent. Am i missing
anything. Please help !
[View Less]
Hi All!
I have moved from kamailio 5.2.X to 5.3.X and I have now an error
with calling "sr.err" from my lua script:
sr.err("ERROR: param $tU [" .. tu .. "]\n")
app_lua [app_lua_api.c:626]: app_lua_runstring(): error from Lua:
/etc/kamailio/lua/my-script.lua:69: attempt to index global 'sr' (a nil
value)
I do understand that I should add this to continue with old way:
loadmodule "app_lua_sr.so"
modparam("app_lua_sr", "register", "sl")
However, I would prefer to fix my script …
[View More]to use the NEW way but can't find
it.
Thanks a lot.
Aymeric
--
Antisip - http://www.antisip.com
[View Less]
Dear all,
Not a direct question on kamailio, but on rtpengine, so feel free to ignore
:)
I have a farm of rtpengine servers , supporting our set of kamailio's.
On recent decent hardware (hp dl360 gen9/10 servers), I typically push my
systems upto 2500 channels per machine. Then I tend to see the load of the
machine go up in a logarithmic manner. E.g. a load of 3 supports 2500
channels, but when its at 3000 channels, load can be at 8. etc.
I have the kernel module running, and I have a good …
[View More]impression it's being
used too. No transcoding is going on, no recording, ... I've tried on
different OS'es, different network cards, and I always end up at that 2500
channels +-.
Is that what I can expect from rtpengine? Or am I really overlooking
something? What numbers do you push out of your systems?
kind regards and happy 2020!
[View Less]
I have a Debian server running Kamailio and i have all my users regiatering
and talking to each other. I do create/remove users from Siremis. I have
followed the guide from here:
https://www.powerpbx.org/content/kamailio-v5-siremis-v5-debian-v9-mariadb-a…
I would like to add freeswitch to the same server and add conference,
voicemail, supplementary services for my users.
I have given it a try to install freeswitch and tried to configure it
accordingly to
https://kb.asipto.com/freeswitch:…
[View More]kamailio-3.1.x-freeswitch-1.0.6d-sbc
But.now the Siremis gui shows a lot of errors and i cannot login any more.
Since it is a Virtual machine i reverted it back and it works without
freeswitch.
How could i integrate freeswitch to my system please?
[View Less]
Dear Users
Can you please name some of VoLTE/IMS Clients (Like IMSDroid for Android) for Windows/Linux/MAC to test IMS Network. IMSdroid available clients on the internet are not working with android v8,9 and 10. or if someone can guide me to create a working APK from IMSDroid available code on Github<https://github.com/DoubangoTelecom/imsdroid>.
Regards
Hamid
Hi All,
Am using Kamailio 5.1.9 version.
*Below is my tls.cfg*
[server:default]
method = TLSv1+
verify_certificate = no
require_certificate = no
private_key = server.key
certificate = server.crt
ca_list = bundle.crt
cipher_list = RSA
verify_depth = 9
[client:default]
verify_certificate = no
require_certificate = no
[server:10.211.160.172:5061]
method = TLSv1+
verify_certificate = yes
require_certificate = yes
private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key
…
[View More]certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt
ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt
cipher_list = RSA
verify_depth = 9
server_name = btip.172.com
[server:10.211.160.172:5061]
method = TLSv1+
verify_certificate = yes
require_certificate = yes
private_key = /root/mahesh_openssl/profile1/ctip_172_server_private.key
certificate = /root/mahesh_openssl/profile1/ctip_172_server_public.crt
ca_list = /root/mahesh_openssl/profile1/ctip_ca_public.crt
cipher_list = RSA
verify_depth = 9
server_name = ctip.172.com
My Kamailio server ip is 10.211.160.172
i)When i initiate a tls connection from remote server(which is also a
kamailio server) say 10.211.160.176 to 10.211.160.172
In the client hello am setting sni name as btip.172.com => so on
10.211.160.172 side it is picking up the server profile with serve_name
btip.172.com for the tls handshake.*// Working as expected*
ii)When i initiate a tls connection from another remote server(Which is
also a kamailio server) say 10.211.160.163 to 10.211.160.172
In the client hello am setting sni name as ctip.172.com => so on
10.211.160.172 side it is picking up the server profile with serve_name
ctip.172.com for the tls handshake.*// Working as expected*
iii)When i initiate a tls connection from another remote server(Which is
also a kamailio server) say 10.211.160.175 to 10.211.160.172
In the client hello am NOT setting sni name => so on 10.211.160.172 side
should it pick up the server default profile or the first profile to which
IP and port matches ?
what i observe from logs is that it is picking up the server profile with
server_name ctip.172.com for the tls handshake.
I had a look at the code in function tls_lookup_cfg, from the debug
prints i understand it is trying to match profile for IP and port
if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip))* // IP and
port matched*
{
if(sname && sname->len>0) *//Incoming Client hello dint have
sname, so it will hit the else part*
{
if(p->server_name.s && p->server_name.len==sname->len
&& strncasecmp(p->server_name.s, sname->s, sname->len)==0)
{
LM_DBG("socket+server_name based TLS server domain
found\n");
return p;
}
}
else
{
return p; *// so it is returning the first profile to which IP and
port matched.*
}
}
Am i missing anything or is this a bug ? if in the clienthello there is no
sni , what needs to be done to make use of the default profile for the tls
handshake ? Or is this something fixed in latest.
I just Tried and Modified the code as below, after which it is giving the
server default profile when no sni in Incoming Client Hello.
if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip))
{
if(sname && sname->len>0)
{
if(p->server_name.s && p->server_name.len==sname->len
&& strncasecmp(p->server_name.s, sname->s, sname->len)==0)
{
LM_DBG("socket+server_name based TLS server domain
found\n");
return p;
}
}
else
{
if( (type & TLS_DOMAIN_SRV) && (p->server_name.s) )
{
LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__);
return cfg->srv_default;
}
else
{
LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__);
return p;
}
}
}
Regards,
Mahesh.B
[View Less]
Good morning,
I am a telecommunication engineer from the National Establishment of Aerial
Navigation (Algeria). First of all i would like to thank you for making the
SIP sever Kamailio available.
But actually i have got issues during the installation process. I was using
the steps described on the link below :
https://www.kamailio.org/wiki/install/devel/git
The issue was encountered at the step #make all, an error was found, and
here is what was displayed on the terminal:
*…
[View More]linux-vd3w:/usr/local/src/kamailio-devel/kamailio # make allmake -C src/
all make[1]: Entering directory
'/usr/local/src/kamailio-devel/kamailio/src'generating core/autover.h ...CC
(gcc) [kamailio] core/ver.oLD (gcc) [kamailio]
kamailiowhich: no mysql_config in
(/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no
mysql_config5 in
(/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no
mariadb_config in
(/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no
mysql_config in
(/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no
mysql_config5 in
(/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no
mariadb_config in
(/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)CC (gcc) [M
db_mysql.so] my_res.oIn file included from
my_res.c:24:0:my_cmd.h:27:10: fatal error: mysql.h: No such file or
directory #include <mysql.h> ^~~~~~~~~compilation
terminated.make[2]: *** [../../Makefile.rules:100: my_res.o] Error
1make[1]: *** [Makefile:511: modules] Error 1make[1]: Leaving directory
'/usr/local/src/kamailio-devel/kamailio/src'make: *** [Makefile:30: all]
Error 2linux-vd3w:/usr/local/src/kamailio-devel/kamailio #*
Actually my knowledge on linux (opensuse leap) that i am using is very
limited. Hopefully i will be able to use the server once installed and it
will be used for ip recording tests purpose.
Thank you for reading this email.
Good bye.
[View Less]
I want proxy MSRP data on kamailio for SIP call (not websocket).
To do this need update connection information for "message" media in SDP.
How i can properly do it only for one media (other medias will be proxed
using rtpproxy or rtpengine ).
Looks as fix_nated_sdp fix for all medias in SDP.
Sergey
Hi,
First off please forgive my lack of knowledge on how TCP works.
We are using Kamailio 5.0.7 and we have an issue where clients are
connecting via TCP and their NAT devices are closing up. Because of this we
want to send TCP keep alives every so often.
1) When restarting Kamailio it's sends a RST. Is this Kamailio sending it
out or is it linux sending it when the application is killed? The issue we
have is if say we need to do a restart 2-3 times (yes we should normally do
that) then we …
[View More]end up with 3x registrations in the db (since when using db
mode if the connection goes away it won't remove the reg from the db (as
per -
https://kamailio.org/docs/modules/5.0.x/modules/usrloc.html#usrloc.p.handle…
)
2) I haven testing with the following settings.
a. tcp_crlf_ping=yes
b. tcp_keepcnt = 3
c. tcp_keepidle = 5
With the above I see the TCP keep alives coming in every 75 seconds. If I
tcp_keepintvl = 10 then I see TCP keep alives going out from Kamailio to
the phone. As per
https://www.kamailio.org/wiki/cookbooks/5.0.x/core#tcp_keepintvl it says
"Time interval between keepalive probes, when the previous probe failed".
Looking at my captures Kamilio sends out the TCP keep alive and gets it
back. Is Kamailio not seeing it? With the above it seems to be working the
way I want it but I want to make sure that I am doing it right.
3) For the devs on here how hard would it be to implement handle_lost_tcp
for DB-Only?
TIA and a happy new year to all.
Regards,
Dovid
[View Less]
Hi,
I wants to remove media type "image" from SDP. For that I have tried to
use KSR.sdpops.remove_media but it is not working. Not working in the sense
that there is no effect of that function on SDP. I am using lua. Kindly
find more details below.
*Kamailio version*: kamailio 5.2.5 (x86_64/linux) 62d35f
*SDP to process:*
INVITE sip:xxxxxxxxxx@xxxxxxxxx.com:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-60823-1-0
From: xxxxxxxxxx <sip: xxxxxxxxxx@ xxxxxxxxxx.com:5060…
[View More]>;tag=xxxxx
To: xxxxxxxxxx <sip: xxxxxxxxxx@ xxxxxxxxxx.com:5060>
Call-ID: 1-60823(a)172.16.19.64
CSeq: 1 INVITE
Contact: sip:xx@xxxxxxx:5060
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 337
v=0
o=zt 53655765 2353687637 IN IP4 xxx.xxx.xxx.xxx
s=-
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=image xxxx RTP/AVP udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
m=audio xxxx RTP/AVP 0
a=rtpmap:0 PCMU/8000
*Code snippet:*
KSR.sdpops.remove_media("image")
logger.log("info", "SDP:" .. headers.get("$sdp(body)"))
*Log snippet:*
https://pastebin.com/5JZmQUfq
One more thing I would like to mention that if I do not use dialog module
then the function KSR.sdpops.remove_media works. But I can not avoid using
dialog module.
Thanks in advance.
Mitesh
[View Less]
Hi all
I'm trying to set a dialog timeout when a call is established:
event_route[dialog:start] {
if(dlg_set_timeout("10")) {
xinfo("10 secs");
}
}
But when the time expired I obtain this error:
Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: CRITICAL: dialog
[dlg_timer.c:199]: update_dlg_timer(): Trying to update a bogus dlg
tl=0x93947b30 tl->next=(nil) tl->prev=(nil)
Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: ERROR: dialog
[dlg_hash.c:1267]: update_dlg_timeout(…
[View More]): failed to update dialog lifetime
# kamailio -v
version: kamailio 5.3.1 (i386/linux)
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE,
USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,
TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024,
BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled with gcc 6.3.0
Any idea?
Thanks and Happy New Year
[View Less]
Dear friends,
I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”).
Unfortunately I got debug message like this:
…
[View More] ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy
ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it
ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping
And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR?
------------------------------------
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网络与交换技术国家重点实验室
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Hi,
I am using rtjson module to do some call routing, as part of that I've added an additional section to the document to include authentication information.
I am using the uac_auth function from within failure_route to send authentication on to the uri. This works fine... however.
When the initial invite is sent, the To/From headers are updated as per the rtjson document. This isn't the case however when the uac_auth function re-sends the invite with the digest header. The original to/from …
[View More]are being sent.
I am using the dialog module
restore_mode = auto
restore_dlg = 1
I assume there is something I need to call, but none of the RTJSON functions seem to be available within failure_route. Any pointers as to where I am going wrong would be greatly appreciated.
Thanks in advance.
Ben
failure_route[TRUNKAUTH] {
if (t_is_canceled()) {
exit;
}
if(t_check_status("401|407")) {
$avp(auser) = $avp(s:authUser);
$avp(apass) = $avp(s:authPass);
uac_auth();
t_relay();
exit;
}
}
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Is it possible to use a table other than 'version' a the version table? i.e. if I'm in a shared DB, and there's an existing table named 'version' with completely different schema, is there a way I can create a table like 'kamailio_version' with the correct schema and data and use it instead?
-- Ben Kaufman
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Greetings,
How sould a call duration be calculated?
Let's say the call creation on Kamailio as a proxy has those steps :
1 - Invite is received
2 - Provisional responses
3 - 200 OK is received
4 - ACK to 200 OK is received
5 - BYE is received
6 - 200 OK to BYE is received
Should the duration begin to count on step 3 (200-Ok Received) or 4 (ACK
received) ?
Should it end on the step 5 or 6?
ACC module tells me that start time for CDR can either be on dialog
creation or confirmation -
https:/…
[View More]/www.kamailio.org/docs/modules/5.2.x/modules/acc.html#acc.p.cdr_star…
- What is the dialog creation ? Is it when the INVITE is received ?
- According to DIALOG module, a confirmed dialog can be "Waiting for ACK"
or "Active Call". Which one is used by acc module?
Best Regards,
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Another year getting to its end, the season holidays are ahead,
therefore it is no better time than now to express my thanks and
greetings to all the friends, developers, supporting companies and
community members that made 2019 an outstanding year for Kamailio project.
Enjoy the holidays! Merry Christmas!
Daniel
* Santa is flying Kamailio! *
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - April 27-29, 2020, …
[View More]in Berlin -- www.kamailioworld.com
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Hi all.
I have Kamailio with 2 addr:
kamailio_external_ip - for client interconnect
kamailio_internal_ip - for local media server interconnect
Like this:
client - kamailio_external_ip - kamailio_internal_ip - media_server
"topos" module hide sip headers for both "client" and "media_server" side.
Is it true: with topos i can't hide headers only for the "client" side and
show them for the "media_server"? I want full header set visible for the
"media_server", but only kamailio_external_ip …
[View More]visible for the "client".
--
Savolainen Dmitri
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