Hi all,
I have successfully setup a Kamailio IMS environment using just a single
machine running three kamailio instances. I am using kamailio 4.2.4
compiling it myself. I followed this guide:
http://nil.uniza.sk/instant-messaging/simple/configuring-im-and-presence-ka…
to setup a presence AS using the scscf instance. I used kamdbctl script
to create the presence tables with one inconsistence with the guide. I
had to create a db in which I created a couple of columns (table_name,
table_version) to satisfy the kamdbctl script as it was failing when
running "kamdbctl presence".
At the moment the problem is that my kamailio instance is failing with
the following error: scscf/scscf[5256]: ERROR: <core> [db.c:422]:
db_table_version(): invalid type (3) or nul (0) version columns for
presentity
...
and then some time later:
scscf/scscf[5256]: ERROR: <core> [db.c:440]: db_check_table_version():
querying version for table presentity
scscf/scscf[5256]: ERROR: presence [presence.c:322]: mod_init(): error
during table version check
scscf/scscf[5256]: ERROR: <core> [sr_module.c:968]: init_mod(): Error
while initializing module presence
(/home/ng40/kamailio-4.2.4/modules/presence/presence.so)
scscf/scscf[5256]: DEBUG: cdp_avp [mod.c:224]: cdp_avp_destroy():
Destroying module cdp_avp
I suspect that the problem relays on creating manually the db where I
inserted the two columns mentioned above (table_name, table_version) so
that I could execute "kamdbctl presence". Could someone assist me into
solving this issue?
br,
md.
Hi all
Experiencing a commonly reported issue where calls drop out after 30 seconds or so. Mainly because the provider hangs up after not recognising/receiving ACK in response to 200 OK.
Unfortunately (or maybe fortunately), I haven't had much experience with Enswitch so was hoping someone in the community might help guide me as to which rules Enswitch might be using to match ACKs to calls in progress. Maybe there is another avenue I should be investigating.
Here's a sample of the 200 OK and ACK that repeats.
13:44:04.155646 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1058
E..>.M..?..Ug.v..........*J.SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 172.21.0.226;rport=5060;branch=z9hG4bKfe94.efbf7fbcaf8bd15243a61fdc9d6d1e78.0^M
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK65f00a0c;rport=5080^M
Record-Route: <sip:PROVIDERIP;lr=on>^M
Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes>^M
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes>^M
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M
CSeq: 103 INVITE^M
Server: Enswitch^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces^M
Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060>^M
Content-Type: application/sdp^M
Content-Length: 286^M
^M
v=0^M
o=root 2110894460 2110894461 IN IP4 PROVIDERMEDIAIP^M
s=Asterisk PBX 11.3.0^M
c=IN IP4 PROVIDERMEDIAIP^M
t=0 0^M
m=audio 15594 RTP/AVP 0 8 3 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=sendrecv^M
13:44:04.164519 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525
E..)!A..@..v....g.v.......T.ACK sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0^M
Via: SIP/2.0/UDP 172.21.0.226;branch=z9hG4bKfe94.472e9fc0479de79b4f176cc9585d8880.0^M
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK752b5264;rport=5080^M
Route: <sip:PROVIDERIP;lr=on>^M
Max-Forwards: 69^M
From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M
To: <sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M
Contact: <sip:PROVIDERUSER@127.0.0.1:5080>^M
Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M
CSeq: 103 ACK^M
User-Agent: Elastix 3.0^M
Content-Length: 0^M
Hello there,
I'm trying to configure Kamailio 4.2.4 to authenticate client REGISTERs against a test FreeRADIUS on the localhost, however I have yet to succeed.
My installation includes Kamailio 4.2.4, radiusclient 1.1.6, FreeRADIUS 2.2.5 and all of them on a Debian 8.0 machine.
I believe I have got all the configurations correct in kamailio.cfg, radiusclient.conf, radius servers file, and freeradius server since the communication between the components are ok.
My problem is that when I use a SIP Client such as 3CX SoftPhone, FreeRADIUS gives me an answer included in SIPClient-LOCAL-RADIUS-NOK.txt. I have isolated the problem being in the Digest-Response part since I have ran some other tests that I have included in the zip file attached, but can't figure out why this shouldn't work. From what I have figured out, my SIP Client does not calculate the correct Digest-Response and I have seen this in all the clients I have tested so far.
For example, my client sends "5f8dfe8aa4e13994b4ceade49ee0d3d4" although it's supposed to send "528f22527e9cf003df29ede9c243e615".
I would really appreciate if you would point out what I did wrong in setting things up. I have done so much googling but have not found anything useful.
Regards,
Maziar Sanaii Ashtiani
Hi,
Hope you are doing fine. Can you please assist me how can I change the IP for Kamailio. Right now its up on this IP:
loading modules under /usr/lib/kamailio/modulesListing onudp: 98.212.180.96 [98.212.180.96] :5060tcp: 192.168.18.99 [192.168.18.99] :8888
I want to change both upd and tcp. Please advise.
ThanksMudasar
Hi
I have two boxes with internal and external interfaces. Each has Kamailio at the public interface and asterisk on the internal one.
Making calls within a box is fine but would like to load balance using the other asterisk box.
However using dispatcher I have noticed packets are not being routed from Kamailo to the second asterisk.
Packets from Kamailo are originating from the external interface and sent to the second asterisk box using private ip address. Off course this is not routable. I have tried masquerading but no joy. Do I need to change kamailio configuration?
I want to have a set of asterisk boxes behind Kamailio using dispatcher. I'm I missing a trick?
Eric
Hello,
From the documentation for t_lookup_request()[1], it is not clear
whether it creates a new transaction if one does not exist:
"Checks if a transaction exists. Returns a positive value
if so, negative otherwise. Most likely you will not want to
use it, as a typical application of a look-up is to introduce
a new transaction if none was found. However this is safely
(atomically) done using t_newtran."
This language is quite unclear in the following respects:
1) Why would I not want to use it?
2) Does the "typical application" mean that t_lookup_request() _will_
create a new transaction if one does not exist?
3) Do I have some means of choosing whether to invoke t_lookup_request()
in a "typical" or "atypical" application?
4) This statement "this is safely (atomically) done using t_newtran"
implies that creating a new transaction using t_lookup_request() (which
also implies that yes, it does create a new transaction) is "unsafe" and
"non-atomic". Why is this?
5) Does this mean that the effect here would be to create two transactions?
if(!t_lookup_request())
t_newtran();
-- Alex
[1]
http://kamailio.org/docs/modules/4.2.x/modules/tm.html#tm.f.t_lookup_request
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
there exists rpc command htable.delete that can be used to delete a key
from htable. is there a way to delete a key in the config file like
assigning $null to it or do i need to use sht_rm_name_re function that
feels like an overkill?
-- juha
