Hello,
I'm running kamailio 5.1.4 and running into a issue with htable. I use
it to store some dialog data so i can use it in a http route. When i use
the callid as a key it creates a issue for callid's starting with eq.
Here some example code that i use to reproduce the issue:
modparam("htable", "htable", "ht_test=>size=16;autoexpire=10")
$var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" +
$Ts;
xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key
eq1234");
$var(htid) = "eq1234";
$sht(ht_test=>$var(htid)) = $var(htable-val);
$var(htable-val) = $null;
if ($shtcn(ht_test=>$var(htid)) > 0)
{
xlog("L_WARN", "HT_TEST: Key eq1234 found!");
} else {
xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!");
}
$var(htid) = $null;
Results in :
Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>:
HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234
Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>:
HT_TEST: Key eq1234 NOT found!
When i prefix eq with a 1 it works fine:
modparam("htable", "htable", "ht_test=>size=16;autoexpire=10")
$var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" +
$Ts;
xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key
eq1234");
$var(htid) = "1eq1234";
$sht(ht_test=>$var(htid)) = $var(htable-val);
$var(htable-val) = $null;
if ($shtcn(ht_test=>$var(htid)) > 0)
{
xlog("L_WARN", "HT_TEST: Key eq1234 found!");
} else {
xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!");
}
$var(htid) = $null;
Results in:
Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>:
HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234
Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>:
HT_TEST: Key eq1234 found!
Am i running into a bug? Or are there some restrictions on the key's i
can use? I'm still going trough the logs to check if there are more
callid's that have a issue with matching.
Thanks,
Jan Hazenberg
Hi,
I'm using Kamailio 4.2.0 and I'm developing a SIP-based application.
Today I made a test connecting two devices directly to an S7 sharing the
Internet, I usually use a VPN even if they are both connected to the same
network because in the future they will be in different networks.Without
the VPN calls were not received.
This has never happened to me, I have repeated tests and in an S6 it has
also happened to me.
So I have checked my Kamailio server and I have detected that when
executing: kamctl ul show,
the received field did not appear:
With VPN:
Contact:: sip:pruebaT@47.60.42.40:10924;transport=udp Q=
Expires:: 1992
Callid:: iwNoYb~QR-
Cseq:: 21
User-agent:: Unknown (belle-sip/1.6.1)
Received:: sip:47.60.42.40:10924
State:: CS_NEW
Flags:: 0
Cflag:: 64
Socket:: udp:10.0.0.4:5060
Methods:: 4294967295
Ruid:: uloc-5a9d9097-9fa-431c
Instance::
<urn:uuid:8b50dde3-d07f-4752-9a24-7b8ddf219a99>
Reg-Id:: 0
Last-Keepalive:: 1539614732
Last-Modified:: 1539614732
Without VPN:
Contact:: sip:pruebaT@37.10.156.189:51163;transport=udp Q=
Expires:: 1996
Callid:: webjvHycBs
Cseq:: 21
User-agent:: Unknown (belle-sip/1.6.1)
State:: CS_NEW
Flags:: 0
Cflag:: 0
Socket:: udp:10.0.0.4:5060
Methods:: 4294967295
Ruid:: uloc-5a9d9097-9fc-1feb
Instance::
<urn:uuid:89306a81-e2d1-4b7a-8206-107e3cbc0688>
Reg-Id:: 0
Last-Keepalive:: 1539614531
Last-Modified:: 1539614531
Would someone know why it happened? Thanks
--
José Antonio Gutiérrez Delgado
Responsable técnico
Oficina: (+34) 923040031
<http://www.arsoft-company.com/>
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tarea de todos.*
I am fresher and want to learn about Kamailio Soft switch,
Anyone can provide me link or document from where i can learn.
I will be thankful for such help,
Thanks & Regards,
Shailendra Kumar
Hi all,
I have an issue with SDP from Kamailio side in the following scenario: simple call from A to B.
When B answers a call and sends '200 OK' to A, A delays ACK for a while, so '200 OK' re-transmission happens.
After the fourth try to re-send '200 OK' towards A, Kamailio changes SDP (specifically connection information 'c', IP address become address of A instead of proxy IP as it was in the previous attempts). At some point A sends ACK and call is established, but last received '200 OK' has invalid SDP and it prevents successful media establishment.
I wonder if there any re-transmission counter or whatever what can affect this scenario and change SDP?
Roman Romanchenko
VoIP devel
Sipwise GmbH
Hi,
I forget, can $dlg_var(...) variables nowadays be reconstituted, along
with dialog state in general, using database backing?
What about profile membership?
Last time I brushed with this topic was probably in 4.x days, and as I
remember the general idea was that dialog state could be recovered on
restart, but without dialog variables or profile counts. Is this
accurate? Is it still true? Or has the amount of state serialised into
database persistence increased since then?
Thanks much!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hi ppl,
I have this problem with call transfer, may be someone can help.
The phone to the far right is registered with the Registrar to the far left
using two PATH headers (trespassing two proxy ports, 5070 then 5060).
As you can see in the graph below, after receiving the UPDATE request,
Kamailio relays it further from port 5060, I expect it to be from 5070 just
like the dialog forming INVITE and the CANCEL afterwards.
[image: image.png]
The UPDATE has a to-tag, but unlike the original INVITE - it has no Route
header!???
route[*WITHINDLG*] {
if (!has_totag()) return;
if (loose_route()) {
route(DLGURI);
if (is_method("BYE")) {
...
}
else if ( is_method("ACK") ) {
route(NATMANAGE);
}
else if ( is_method("NOTIFY") ) {
record_route();
}
route(RELAY);
exit;
}
if ( is_method("ACK") ) {
...
}
# handle UPDATE method for in-dialog requests
if (is_method("*UPDATE*")) {
route(DLGURI);
record_route();
route(RELAY);
}
}
Thanks in advance.
Greetings,
in order to upgrade kamailio server from 3.0.3 to 5.1.4 version, I am
conducting test calls and it is not completelly successful. The main
components I have are:
1- Media Gateway (Asterisk) : 192.168.226.133
2- Signalling Gateway (Kamailio 5.1.4) 192.168.226.132
3- Global Dispatcher (Kamailio 3.0.3 in front of the Signalling Gateway.) :
192.168.226.130
Both caller and callee have the dispatcher's IP as a proxy and their
registration requests is passed to be registered at the Signalling Gateway
(Kamailio 5.1.4) and saved on the MySQL server. The caller reaches the
callee properly, who in its turn answeres the call and sends the OK msg to
the caller who gets it. Unfortunately, when the caller sends the ACK for
this INVITE to the callee, the ACK does not reach the callee because it is
trapped by the dispatcher at the last stage because the uri was dropped and
the Route header was used instead. This behaviour is abnormal since as I
know the Route header should be converted into a Via header and the uri
stays untouchable.
The ACK was looped into the dispatcher itself .i.e. the dispatcher sent it
to itself adding always the via header ""Via: SIP/2.0/UDP
192.168.226.130;branch=0"" endlessly.
Once again, here is an example of the ACK before the dispatcher and how the
dispatcher forwards the ACK afterwards.
the supposed path is as following:
caller --> GDP--> SGW --> MGW--> SGW --> GDP --> callee
Although the GDP (before the calle after the SGW) received the uri well
from SGW inside the ACK as it appears below:
ACK sip:yyyyyyyyyyyy@192.168.226.130:5060;rinstance=1608f7c153f40093;transport=UDP
SIP/2.0
Record-Route: <sip:192.168.226.132;lr=on;ftag=as3b989c5f>
Via: SIP/2.0/UDP
192.168.226.132;branch=z9hG4bK136f.1e8349c235fff01ee1849f3456f8868a.0
Via: SIP/2.0/UDP 192.168.226.133:5060;rport=5060;branch=z9hG4bK34871fe9
Route: <sip:192.168.226.130;lr;ftag=as3b989c5f;did=d5b.87890dc6>
Max-Forwards: 69
From: "xxxxxxxxxxxxxx" <sip:xxxxxxxxxxxx@domain.com>;tag=as3b989c5f
To: <sip:yyyyyyyyyyyyy@192.168.226.132:5060>;tag=1af2c424
Contact: <sip:xxxxxxxxx@192.168.226.133:5060>
Call-ID: 7ed2ab533d4c43086746d0e620bdff06(a)domain.com
CSeq: 102 ACK
User-Agent: MGW
Content-Length: 0
The global dipstacher (GDP) has used the Route header (which has no R-URI
yyyyyyyyy) instead of the uri:
ACK sip:192.168.226.130;lr;ftag=as3b989c5f;did=d5b.87890dc6 SIP/2.0
Record-Route: <sip:192.168.226.132;lr=on;ftag=as3b989c5f>
Via: SIP/2.0/UDP 192.168.226.130;branch=0
Via: SIP/2.0/UDP
192.168.226.132;rport=5060;branch=z9hG4bK136f.1e8349c235fff01ee1849f3456f8868a.0
Via: SIP/2.0/UDP 192.168.226.133:5060;rport=5060;branch=z9hG4bK34871fe9
Max-Forwards: 68
From: "xxxxxxxxxx" <sip:xxxxxxxxxxx@domain.com>;tag=as3b989c5f
To: <sip:yyyyyyyyy@192.168.226.132:5060>;tag=1af2c424
Contact: <sip:xxxxxxx@192.168.226.132:5060>
Call-ID: 7ed2ab533d4c43086746d0e620bdff06(a)domain.com
CSeq: 102 ACK
User-Agent: MGW
Content-Length: 0
Thanks in advance for your contribution
Abdulaziz
Hi Experts.
I did some tests with DIALPLAN module and I have some doubts about how to
define the regular expressions in the database.
For example, using the start character '^' (beginning of the string), it
looks not working:
kamcmd> dialplan.dump 1
{
DPID: 1
ENTRIES: {
ENTRY: {
PRIO: 1
MATCHOP: 2
MATCHEXP: ^(\+39)(.*)$
MATCHLEN: 0
SUBSTEXP: ^(\+39)(.*)$
REPLEXP: \2
ATTRS: 1
}
}
}
kamcmd> dialplan.translate 1 "+39123456"
error: 500 - No translation
Removing it, it works... but not as desired:
kamcmd> dialplan.dump 1
{
DPID: 1
ENTRIES: {
ENTRY: {
PRIO: 1
MATCHOP: 1
MATCHEXP: (\+39)(.*)$
MATCHLEN: 0
SUBSTEXP: (\+39)(.*)$
REPLEXP: \2
ATTRS: 1
}
}
}
kamcmd> dialplan.translate 1 "+39123456"
{
Output: 123456"
Attributes: 1
}
kamcmd> dialplan.translate 1 "xxxxxxx+39123456"
{
Output: 123456"
Attributes: 1
}
Can someone help me understand how the regular expression should be used in
DIALPLAN, please?
Br. Stefano
Hi experts.
I intended ATTRIBUTES is a value that is returned by dialplan's
dp_translate as the result as shown by kamcmd command:
kamcmd> dialplan.translate 2 s:+3912345
{
Output: 3912345
Attributes: test
}
Is my understanding correct?
I tried many ways, including using AVP and XAVP for the result, but no
success:
dp_translate("2", "$rU/$xavp(dp_result)");
pv_xavp_print();
DEBUG: dialplan [dialplan.c:222]: dp_get_ivalue(): dpid is 2 from constant
argument
DEBUG: dialplan [dialplan.c:243]: dp_get_svalue(): searching 7
DEBUG: dialplan [dialplan.c:331]: dp_translate_f(): input is +3912345
DEBUG: dialplan [dp_repl.c:592]: translate(): regex operator testing over
[+3912345]
DEBUG: dialplan [dp_repl.c:662]: translate(): found a matching rule
0x7f56aef690d8: pr 1, match_exp ^\+39.....
DEBUG: dialplan [dp_repl.c:481]: rule_translate(): copying match <3912345>
token size 2
DEBUG: dialplan [dialplan.c:340]: dp_translate_f(): input +3912345 with
dpid 2 => output 3912345
INFO: <core> [xavp.c:494]: xavp_print_list_content(): +++++ start XAVP
list: 0x7f56aeffaaa8 (level=0)
INFO: <core> [xavp.c:498]: xavp_print_list_content(): *** XAVP name:
dp_result
INFO: <core> [xavp.c:499]: xavp_print_list_content(): XAVP id:
3539756229
INFO: <core> [xavp.c:500]: xavp_print_list_content(): XAVP value type:
2
INFO: <core> [xavp.c:509]: xavp_print_list_content(): XAVP value:
3912345
INFO: <core> [xavp.c:531]: xavp_print_list_content(): ----- end XAVP list:
0x7f56aeffaaa8 (level=0)
If possible, how can I get both OUTPUT and ATTRIBUTES as result?
Stefano
