Hi All
Last few days we have tried to setup Kamailio for IMS server setup.
We have not been successful.
Now we are planning to use Kamailio 4.2.0 code base (Latest) to setup
Kamailio IMS setup.
I am looking for a link/document, that gives proper step by step
instructions how to setup Kamailio as IMS servers (P-CSCF, I-CSCF and
S-CSCF).
Can somebody kindly provide the official link or any other working link,
that I can follow to setup P-CSCF, I-CSCF ,S-CSCF and HSS.
Many thanks
Austin
> Hi, we use Kazoo which loads acl and dial plan out a DB.
> We run in to smth that seems freeswitch default behavior so that's why I write here.
>
> We have a carrier that needs to register a extension to our platform as well.
> They use the same ip. And that creates a problem as when I place a call with the device using the extension it gets identified as carrier and the route is then not correct.
>
> The same happens the other way arround, if a call comes in via the carrier the ip is recognized as an extension ip and then it won't use the carrier route.
>
> I was wondering if I could use some flags or a way of changing the ip used or doing some magic to make it work. I would prefer not to use a proxy.
>
> The setup is kamailio>freeswitch
>
>
> Met vriendelijke groet, Martin
>
> Team snelgoedkoop
Hi guys,
I've an architecture with multiple Kamailio. I would like to implement some
check that allow me to verify if a server is online or not so for example i
can use msilo to save chat message and relay it when the server come's up.
Is it possible to use UAC module to send a REGISTER message from one server
to another so i can "dump" all message in very simple way? Is there any
example ?
Many thanks
-----
Marino Mileti
--
View this message in context: http://sip-router.1086192.n5.nabble.com/Kamilio-verify-online-servers-tp132…
Sent from the Users mailing list archive at Nabble.com.
The documentation for the htable module gives no guidance on whether
sht_lock()/unlock() should be used in all situations when writing or
reading from the htable. Should it be?
I was under the impression (not supported by anything said in the
documentation) that read and write access to the htable is fundamentally
thread-safe, i.e. it is safe to both read and write to the htable even
if the potential exists for another worker process to do the same
concurrently.
Is that not the case? If not, should there not be strong admonitions to
use sht_lock/sht_unlock around concurrent operations--and it sounds like
nearly all useful applications of the htable are essentially concurrent?
Thanks!
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Hi all,
i need some advice on how to solve a problem.
I've a scenario like following:
[SIP PBX] <---->CUCM (Call Manager v 10.5 )
User A: belong to CUCM
User B: belong to SIP PBX
User C: belong to SIP PBX
User A calls User B. After some conversation, User A wants to transfer call
to User C.
CUCM transfer the call with re-INVITE; in this case on SIP PBX 2 channel SIP
are used.
CUCM does not support transfer, on Trunk, using REFER method.
My idea is using Kamailio beetwen SIP-PBX and CUCM and handles call
generating REFER msg in some way.
Is this idea a good one?
Or there is a better way?
Regards
--
View this message in context: http://sip-router.1086192.n5.nabble.com/Generate-REFER-msg-tp132317.html
Sent from the Users mailing list archive at Nabble.com.
Hi all
I'm experiencing a problem while using a mobile client on a NGCP mr3.5.1 (based
on kamailio 4.1.6)
The problem is as follows:
A custom mobile client
B blink for linux
The path of a call from B to A would be this one:
B --> Kamailio LB --> Kamailio proxy --> Sems --> Kamailio LB --> A
Kamailio LB is a stateless proxy. Kamailio proxy is a stateful proxy and
registrar and sems acts as B2BUA.
Kamailio LB listens tcp,udp and tls to the outside and speaks udp to the
inside.
Kamailio proxy and sems only listen udp localhost (ports 5062 ans 5080).
My problem comes when the ACK from B to A (to the INVITE-200) doesn't leave the
Kamailio LB to the proxy. The problem is that the kamailio proxy added itself
as sips in the record-route header:
U 2014/11/06 19:22:20.447717 127.0.0.1:5062 -> 127.0.0.1:5080
INVITE sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp SIP/2.0'
Record-Route:
<sips:127.0.0.1:5062;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw-->'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>'
Record-Route:
<sip:XX.XX.XX.XX;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>'
Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bKc36a.b42be7e4a4db27ac699a52fda7fb5fe0.0' Route:
<sip:lb@127.0.0.1;lr;received=sip:YY.YY.YY.YY:41956%3Btransport%3Dtls;socket=sip:XX.XX.XX.XX:5061>'
Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc36a.a58b9955ce93726b17c0240a555011f2.0'
Via: SIP/2.0/UDP
10.10.10.4:47879;received=ZZ.ZZ.ZZ.ZZ;rport=47879;branch=z9hG4bKPj6UmIovzyxDRCgcuz-VoXHEyJHRpA712x'
When the ACK arrives to the LB from B, it can't relay it to the Kamailio proxy:
NOTICE: <script>: New request on lb - M=ACK
R=sip:ngcp-lb@XX.XX.XX.XX;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31
NOTICE: <script>: Relaying request,
du='sips:127.0.0.1:5062;lr;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw--',
fs='udp:127.0.0.1:5060' - R=sip:127.0.0.1:5080;prxroute=1
WARNING:
<core> [forward.c:264]: get_send_socket2(): WARNING: get_send_socket:
protocol/port mismatch (forced udp:127.0.0.1:5060, to tls:127.0.0.1:5062)
I think the problem might be in the way the client registers? This is the
registration info where it uses sips in the contact:
contact: sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp
If I try the same calling blink, which has the following contact in the
registration it works without issues as the proxy doesn't insert sips in the
Record-Route header:
contact: sip:91807432@10.10.10.4:57160;transport=tls
Any thoughts?
cheers,
Jon
Hello,
Kamailio SIP Server v4.0.7 stable release is out.
This is a maintenance release of the 3rd old stable branch, 4.0, that
includes fixes since release of v4.0.6. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.0.x. Deployments running previous v4.0.x versions
are strongly recommended to be upgraded to v4.0.7, or even better, to
4.1.x or 4.2.x, as the focus from now on will be on maintenance of these
release series.
For more details about version 4.0.7 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2014/11/kamailio-v4-0-7-released/
Note: the latest stable branch is 4.2, at this moment with its latest
release v4.2.0. See more details about it at:
* http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
I am planning to release v4.0.7 based on latest source code in branch
4.0. This branch is going out of official maintenance, the current
latest stable branches being 4.2 and 4.1.
If you have any remarks regarding it, write back to mailing lists.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com
I am just wondering if anyone has seen this type of segfault on 4.2 ?
I'm trying to run kamailio as an scscf on centos 6.6 ?
Nov 6 14:17:20 scscf kernel: kamailio[6246]: segfault at 0 ip 00f3f425 sp
bfa0e9f4 error 6 in cdp.so[f08000+96000]
Hey Daniel,
Thanks for your fast reply!
I am using the 4.2.0 out of debian packages (not nightly). Will shift on
nightly and retest.
Not a big deal regarding 150 seconds, maybe it should be a good reason
for async reloads (reply with the delay number of seconds when reload
will happen). Reloads can be always implemented outside of kamailio but
since everybody would need it I assume a more centralized approach would
be useful.
Anyway, thanks again!
DanB
On 24.10.2014 12:00, sr-users-request(a)lists.sip-router.org wrote:
> Hello,
>
> are you using branch 4.2 or some tgz/package from 4.2.0 release? There
> was a fix related to such issue, so 4.2 branch must be used (or debian
> nightly builds for 4.2).
>
> I will have to check the code how flexible that is, but afaik, the limit
> on reload interval was thought to avoid sending again registration
> requests while retransmissions from previous one can occur.
>
> Cheers,
> Danel
