Hello,
I see the tsilo module works only with registered contacts using location
table. Is there a possibility that the ts_append function can actually take
the destination URI as an argument and append branch to respective
destination?
Basically I want to avoid maintaining location table. In a typical mobile
use case, where the incoming call is suspended and a push notification is
generated to the called party; the called party generates a REGISTER
request with expires header as 0; kamailio just extracts the contact header
and sends the call.
Problem comes in when the same user receives a PUSH on another device; I
cannot figure a neat way to append the same call on the new device.
Is there a better alternative to achieve this without looking up the
location table. Thanks for any help.
- Jayesh
Hello All,
I’m trying to interop with a provider whose ACK during an inbound call seems to be misrouted.
During the INVITE, OK, ACK handshake, my understanding (which matches the examples in RFC 3665) is that the R-URI in the ACK should be the Contact specified in the 200 OK.
I have also found multiple discussions that imply that this is necessary. However, I can’t find a convincing normative reference for this... Is there a reference (ideally in RFC 3261) that makes this clear?
Cheers,
Dave.
Hi,
I'm using kamailio 5.0.0 and trying to use core commands through xmlrpc
interface but the requests are failing towards Kamailio.
I have compiled and enabled the xmlrpc module in config and I have this
on my kamailio.cfg:
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is a bug in
# xmlrpclib: it waits for EOF before interpreting the response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
I have tried to test this by using curl and calling "core.uptime" method
in xml body (core.uptime works when executed through kamcmd commandline
tool).
The curl is called from the same machine that I have Kamailio installed
on:
"curl -X POST -H 'Host: 127.0.0.1:5060' -H 'User-Agent: xmlrpcclient'
-H 'Content-Type: text/xml' -d '<?xml
version='1.0'?><methodCall><methodName>core.uptime</methodName><params></params></methodCall>'
http://127.0.0.1:5060"
This fails and I only get: "curl: (7) Failed to connect to 127.0.0.1
port 5060: Connection refused"
network traffic shows me this.. firewall is disabled completely when
doing the test:
1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 74 55630→5060 [SYN] Seq=0
Win=43690 Len=0 MSS=65495 SACK_PERM=1 TSval=1347930477 TSecr=0 WS=64
2 0.000032 127.0.0.1 -> 127.0.0.1 TCP 54 5060→55630 [RST, ACK]
Seq=1 Ack=1 Win=0 Len=0
Can anyone help me to get to the right direction with this?
Cheers,
--Olli
Hello List,
I'm having an issue with rtpengine changing the source port of the RTP
stream after receiving a 183 from the carrier. Though RTP engine continues
to send RTP to the proper carrier IP and port the carrier drops the inbound
audio once the source port changes.
Can anyone tell me what might trigger this type of behavior from rtpengine
and if it's expected?
Thank You,
___
John Petrini
Hi,
About 3 weeks ago i upgraded one of my production server with latest stable
kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about
missing CDR events in ACC table. I observe following problems,
1. There are only BYE records in acc table, no record for INVITE or ACK.
2. In kamailio logs when ACK is received against 200 OK response for
INVITE, i see following errors,
--
ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse
>From uri
ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI
NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org'
to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484'
--
Of course all these errors are bogus, I have checked all headers in ACK
(not just FROM header), they all seem perfectly fine and valid.
3. Then the dialog times out,
--
WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg
with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags
'1D3ECD34F5731AB845BA3064AC95BB2D'
'7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4'
--
4. Any further sequential requests complain about "unable to find dialog",
e.g.
--
NOTICE: <script>: Sequencial 'BYE' request received from caller
ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [
sip:00xxxxxxxxxx@sip.domain.com]
WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog
for BYE with route param '5ae1.d595' [7845:22877]
--
5. However the acc record for BYE is written to db and log file,
--
NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered:
timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain=
sip.domain.com
;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy
--
The same config was working fine with older version 4.2.0-97cab8. The
kamailio config i am using is pretty much standard,
--
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_DLG 4
...
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 1)
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
# log to db
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", "DBURL")
...
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
t_relay();
};
exit;
};
#### only initial requests (no To tag) ####
t_check_trans();
....
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_DLG); # create dialog
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCFAILED); # ... even if the transaction fails
$avp(dlg_timeout) = 60;
dlg_manage();
....
}
--
Any ideas why its happening? Since it is 3 weeks old so may be problem has
already been spotted and fixed by someone else. Otherwise let me know how
can i provide more info to help fix this issue.
Thank you.
Hi
I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I
have add and registered two IP phones with kamailio these phone able to
call each other.
How can i make outbound call with that I am using a asterisk server as PSTN
Gateway. i am not famalier with kamailio script.
Regards
Sharad Tyagi
SparkTG info Pvt Ltd
hi all!
i'm trying to use pua_dialoginfo module (current master) with
modparam("pua_dialoginfo", "use_pubruri_avps", 1)
so for $avp(puburis_caller) = "sip:111111@mydomain";
i have PUBLISH with xml "entity" from "From:" field, not from my avp.
PUBLISH sip:111111@mydomain SIP/2.0
Via: SIP/2.0/UDP 213.170.66.150:5071
;branch=z9hG4bK9a08.f20b7367000000000000000000000000.0
To: <sip:111111@mydomain>
From: <sip:111111@mydomain>;tag=b5f1e2ebdfeaa37cea72edd75fe37385-0dfc
CSeq: 10 PUBLISH
Call-ID: 6d71159d7ab5b9ff-23010@mydomain
Content-Length: 260
User-Agent: Dev_Kamailio_UA
Max-Forwards: 70
Event: dialog
Expires: 41
Content-Type: application/dialog-info+xml
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="full" entity="sip:003@mydomain">
<dialog id="yciKcJRNdO1HRjFg0qOoPUe4V3IhbqIW" direction="initiator">
<state>Trying</state>
</dialog>
</dialog-info>
is it ok?
according to doc:
> If use_pubruri_avps is enabled, PUBLISH-requests reporting
> dialog-information about the caller (entity=caller) are sent using the
> value of the specified avp as R-Uri.
>
--
Savolainen Dmitri
Hello everyone
I need to do a lookup for an existing user using the last N digit of its
phone number. For example, if a subscriber has a phone number equal to
"+391234567890" I would like to call him using the full phone number or
also the last 8 digits "34567890".
There is a simple way to do this using kamailio 5.0.1?
--
Emanuele Gambaro
---
email: emanuele.gambaro(a)pynlab.com
skype: sarbyn_work
OpenPGP Key: https://goo.gl/fdeVnI
Hi,
I notice Siremis is not usable for me as it is very limited in the user management interface.
I will suggest the developpers to include a USER management GUI visual interfase, as nowadays i use command line to do the following by putty tool:
a) "Create Users"
kamctl add <user> <password>
b) "Delete Users"
kamctl rm <user>
c) "Show users in database"
kamctl db show subscriber
d) "Show online users"
kamctl online
Adding this on Siremis will help a lot for daily use as a Manager Account.
Someone knows the Siremis author to add this suggestion?
or someone has already coded a VISUAL GUI interfase for the above and could share it?
Regards,
Javier
