Dear experts ,
I am using kamailio with rtp proxy module. I have 2 questions /issues .
1. When caller or callee ends the call the other end call is not
disocnnecting .
UA is pjsip based and behind NAT router. Present call flow is
pjsipUA (LAN_ip)-Router
Hello All,
I have installed kamailio with TLS module together with siremis on server
and all seems works fine except that sometimes appear in syslog file the
following errors:
/usr/sbin/kamailio[3266]: ERROR: core [tcp_read.c:289]: tcp_read_data():
error reading: Connection timed out
Take a look at :
modparam(registrar, max_contacts, x) to have up to x contacts for AOR
and in your REGISTRAR route if your save(location) has some extra parameters
(for example the 0x04 used to restrict one single record in loaction table
Messaggio originale
Da: ovoshl...@gmail.com
Hello,
I pushed a fix to kamailio master branch that should solve it -- last
commits to acc module. If you can give it a try and report results,
would be appreciated.
Cheers,
Daniel
On 03/10/14 17:00, Igor Potjevlesch wrote:
Hello Daniel,
I’m just seeing that 4.2 is scheduled for 15th
Do you refer to the http response only? Or to SIP as well?
Daniel
On 07/10/14 06:19, Gonzalo Gasca wrote:
Daniel,
I will re-write it in Kamailio, seems to be that during initial WS
negotiation (HTTP Connection Upgrade), Kamailio is already including
the Via header:
Via: SIP/2.0/TCP
Hello,
the errors can be because of various reasons such as:
- requiring a tls method not supported (tlsv1, ...)
- not having a common cypher
- requiring certificate, but the other party not providing one
- requiring a valid certificate, but the validation fails
Try to run with debug=3 and see
Next is the link to the tutorial detailing how to extend simires to
manage new database tables from web interface:
- http://kb.asipto.com/siremis:install40x:new-views
Cheers,
Daniel
On 02/10/14 18:30, cpcnetworking wrote:
I'm using Kamailio 4.15 with Siremis 4.1.0
both are great BTW!
I'd
You have to use event_route[tm:local-request] if you use uac module to
send out new REGISTER.
Cheers,
Daniel
On 03/10/14 18:32, Errol Samuels wrote:
I also tried your last suggestion at the point when the registeris
being forwarded to FS but it had no effect
# Forward REGISTER to Freeswitch
Hi Daniel,
I see the Via header in both initial Websocket upgrade response
(101) and in SIP 200 OK from Kamailio when Sipml5 client is
registering.
At SIP level including rport in initial REGISTER message from client
and getting a received field from Kamailio makes sense and I will
use your
Hello. We have multiple kamailio behind load balancer. We use UAC to send
REGISTER procedure to provider. Some providers drop wrong ip. So We have
loadbalanser with external IP and have kamailio servers with external IPs
too. So we need to send REGISTER packets to porovide with source IP of our
Thanks. I used some custom thiks to change $ru. Now I will delete it.
2014-10-07 0:33 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
are you using t_relay()? Parallel forking should be default with that
function.
Cheers,
Daniel
On 06/10/14 22:02, Yuriy Gorlichenko wrote:
Thanks for your input Daniel but pardon my ignorance as I am not 100% clear
where I need to add that event_route[tm:local-request].
Here is the point where the REGISTER is being sent to Freeswitch. So are
you able to provide an example of where I send the real User-Agent of the
device to
Thx, exactly what I was looking for!
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Hey Errol,
What I get from Daniel's email is that you only need to add this route in
your script and it will trigger itself automatically when uac_req_send()
function is executed.
event_route [tm:local-request] { # Handle locally generated requests
xlog(L_INFO, Routing locally generated
Hi Sammy,
Thanks for your response and the clarification.
What I am actually trying to do is to pass the real User-Agent info for
each device as the Registration is fowarded to Freeswitch so using your
code as an example instead of hardcoding the custom User-Agent: My Server
SIP Server which
No no, not like that, its a separate route and needs to be placed outside
the other route[] { ... }
On Tue, Oct 7, 2014 at 4:06 PM, Errol Samuels ewsamu...@gmail.com wrote:
Hi Sammy,
Thanks for your response and the clarification.
What I am actually trying to do is to pass the real
Richard,
After quite a bit of testing I can confirm there is no accidental success
here. With default settings the old RTP flow ceases 30 seconds after the
rejected re-invite.
- Jeff
On Mon, Sep 29, 2014 at 1:17 PM, Richard Fuchs rfu...@sipwise.com wrote:
On 09/25/14 12:05, Jeff Pyle
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