Dera Sir
In the SER based Switch , Billing is possible?
If possible , what are the option possible like cdr , monthly trafic , termination status etc....
Waiting for your reply
Thanks and Regards
Parikhita
Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/
Hi everybody,
May be this is a stupid question but I want to know what is the
difference between kamailio-1.x.0-notls_src.tar.gz and
kamailio-1.x.0-tls_src.tar.gz
I guess it is tls support but INSTALL file of notls version talks
about compile with TLS support, that is why I am asking...
...
- compile with TLS or SCTP support
make TLS=1
make SCTP=1
Thanks
Luciano
Hello,
I have to relay SIP requests to a special "redundant" destination (= FQDN - e.g. "test.kamailio.loc") that consists of 3 hosts with different priorities.
After startup Kamailio creates a DNS-SRV request to the DNS server and receives a response with following answer (e.g.):
_sip._udp.test.kamailio.loc: type SRV, class IN, priority 10, weight 100, port 5060, target host1.test.kamailio.loc
_sip._udp.test.kamailio.loc: type SRV, class IN, priority 20, weight 90, port 5060, target host2.test.kamailio.loc
_sip._udp.test.kamailio.loc: type SRV, class IN, priority 30, weight 80, port 5060, target host3.test.kamailio.loc
In worst case - if the prime and secondary host (host1 and host2) are unavailable - I expect that kamailio tries relaying the request to the third host in the list. But it doesn't. It makes retransmission of the original invite to the second host and does not try reaching the third host.
For tuning the switchover to the alternative target(2) I have set the module parameter "fr_timer" for TM to 3 seconds (modparam("tm", "fr_timer", 3)).
Is this behaviour of Kamailio-1.4.3 and Kamailio-1.5.0 (I've tested it with both versions) as expected (= limited to support only two answers in a DNS SRV reply)? Does anybody have experience with this scenario? Can anybody give me a hint?
regards,
Klaus F.
--
Computer Bild Tarifsieger! GMX FreeDSL - Telefonanschluss + DSL
für nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a
Hi,
I'm trying to use *_dialoginfo modules but I'm facing to a
missunderstanding, or a bug.
In fact, the presence server is working and I'm also able to subscribe to
dialog event.
But when pua_dialoginfo uses pua to send the Publish, I don't know where it
going...
I even not sure publishes are really sent.
This the log :
DBG:dialog:run_create_callbacks: dialog=0xb59d30b0
DBG:pua_dialoginfo:__dialog_created: new INVITE dialog created:
from=sip:alice@domain
DBG:core:grep_aliases: comparing host [0:domain:0] with us [0:domain:0]
DBG:core:grep_aliases: match found for: [0:domain:0]
DBG:core:check_self: host == me
DBG:pua_dialoginfo:build_dialoginfo: new_body: <?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="full" entity="sip:alice@domain"> <dialog
id="MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q."
call-id="MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q."
direction="initiator"> <state>Trying</state> <remote>
<identity>sip:bob@domain</identity> <target uri="sip:bob@domain"/>
</remote> <local> <identity>sip:alice@domain</identity>
<target uri="sip:alice@domain"/> </local> </dialog> </dialog-info>
DBG:pua_dialoginfo:dialog_publish: publish uri= sip:alice@domain
DBG:pua_dialoginfo:print_publ: publ:
DBG:pua_dialoginfo:print_publ: uri= sip:alice@domain
DBG:pua_dialoginfo:print_publ: id=
DIALOG_PUBLISH.MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q.
DBG:pua_dialoginfo:print_publ: expires= 0
DBG:pua:send_publish: pres_uri=sip:alice@domain
DBG:pua:search_htable: core_hash= 118
DBG:pua:search_htable: record not found
DBG:pua:send_publish: insert type
DBG:pua:send_publish: UPDATE_TYPE and no record found
DBG:pua:send_publish: request for a publish with expires 0 and no record
found
DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout
DBG:tm:run_reqin_callbacks: trans=0xb59d3270, callback type 1, id 0 entered
Regards,
Eric.
Ha ha thanks guys. Who wants the 25cents?
Ok i dont have an issue with Asterisk/Trixbox as i've been using that for years. I also dont have an issue with Kamailio installation. The issue i have though is that being a newbie to Kamailio i am struggling to understand and come up with a cfg file to allow the following.
1.To allow Kamailio to accept calls from a specific asterisk/trixbox (authentication would be best).
2.To forward all calls from Kamailio to the Sip provider (how do i set that up? No auth required for sip provider).
3.To apply rtp proxy or nathelper if required.
4.To send cdrs to a mysql database.
I've had several attempts at writing stuff to the route part of the cfg file but it always fails to load. Is the best method to find the failure to use /var/log/messages?
Thats about it and oh my amazon purchase of "Building Telphony systems with Openser" book turned up today so hopefully reading that might make a difference.
Taff..
--- On Thu, 5/3/09, olivier.taylor(a)gmail.com <olivier.taylor(a)gmail.com> wrote:
> From: olivier.taylor(a)gmail.com <olivier.taylor(a)gmail.com>
> Subject: Re: [Kamailio-Users] Newb doco request
> To: "Henning Westerholt" <henning.westerholt(a)1und1.de>
> Cc: users(a)lists.kamailio.org
> Date: Thursday, 5 March, 2009, 3:36 PM
>
>
>
>
>
>
> Pesos, of course :)
>
>
>
> Olivier
>
>
>
> Henning Westerholt a écrit :
>
> On Thursday 05 March 2009, olivier taylor wrote:
>
>
> 25 cents bounty, just to understand what's
> his problem ;)
>
>
>
> euro- or dollar-cent? ;-)
>
> Henning
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Hello,
I need to do arithmetic operations.
I could do some test with having looks at :
http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#arithmetic_operati
ons
Simple opérations like $var(a)=1+1 works.
But, when we have :
$var(n)=0;
while ($var(j)<$dbr(nb_max=>rows)){
$var(n) = $var(n) + $dbr(nb_max=>[0,$var(j)]);
xlog("L_NOTICE", "log5 $var(n) + $dbr(nb_max=>[0,$var(j)])
\n"); ##à give us log5 0+10
where is the problem?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
IIRC some devels in the past asked for conversion from svn to gitin
kamailio (openser), but at that time SF didn't offer it.
Andrei started the discussion of how to maintain the parallel source
trees for the two projects. We should get to a decision soon.
Cheers,
Daniel
On 03/02/2009 03:01 PM, Jan Janak wrote:
> FYI:
>
> http://apps.sourceforge.net/trac/sitedocs/wiki/Git
>
> Jan.
>
> _______________________________________________
> sr-dev mailing list
> sr-dev(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
>
--
Daniel-Constantin Mierla
http://www.asipto.com
Hi,
I'm after some useful doco to allow me to setup trunks from Trixbox to Kamailio and push all calls to a sip provider. Would possibly need rtpproxy or nathelper as well.
Can anyone point me in the right direction as i've been combing the web for days and unable to find anything with good explanations and basic enough for a Kamailio beginner.
Thanks,
Taff.