Richard Fuchs writes:
> In the third case, the audio stream was setup as a=sendonly, explaining
> the one-way audio. Probably caused by Firefox not being able to access
> the playback device.
as i reported in another message, i have also lately noticed in my tests
that firefox sends invite with a
Hi there,
We haven't used gdb on Kamailio yet but some time soon we'll most likely
will. Obviously I'll post our findings.
Thanks again!
Cheers,
Kees
On 17-12-14 12:00, sr-users-requ...@lists.sip-router.org wrote:
> did you get a core dump file? If yes, send the full backtrace taken with
> gdb.
> "MS" == Marc Soda writes:
MS> I'm having a problem reassembling UDP packets on my Asterisk servers after
MS> passing through Kamailio
You could try having the kama->ast socket use tcp.
-JimC
--
James Cloos OpenPGP: 0x997A9F17ED7DAEA6
> "DM" == Daniel-Constantin Mierla writes:
DM> The question would be more specific to the error message printed from
DM> postgres client library:
DM> FATAL: no pg_hba.conf entry for host "129.240.1.1", user
DM> "foo_test_user", database " foo_test", SSL off
DM> Is it something that is docu
On 12/18/14 13:38, Andrey Utkin wrote:
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
>
>
>
Also I encounter such issue: even in the working scenario, the hangup
of one peer doesn't make the call end for another peer.
rtpengine: https://gist.github.com/krieger-od/1cfe84b53dc0d29cfb90
kamailio: https://gist.github.com/krieger-od/11c6bbf7dad15382e81b
ngrep: https://gist.github.com/krieger-o
2014-12-18 20:38 GMT+02:00 Andrey Utkin :
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
>
>
This works: call from sipml to linphone android:
rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
This doesn't work: few seconds after answer, there's
2014-12-18 20:09 GMT+02:00 Andrey Utkin :
> 2014-12-18 20:05 GMT+02:00 Richard Fuchs :
>> Amazon NAT is exactly why I've mentioned it, because on an Amazon
>> system, if you don't use the --interface option correctly
>> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
>> that
2014-12-18 20:05 GMT+02:00 Richard Fuchs :
> Amazon NAT is exactly why I've mentioned it, because on an Amazon
> system, if you don't use the --interface option correctly
> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
> that show in your log.
Thank you, will try with suc
On 12/18/14 12:55, Andrey Utkin wrote:
> 2014-12-18 19:30 GMT+02:00 Richard Fuchs :
>> Write error on RTP socket usually indicates an incorrect network setup,
>> for example trying to use a source address for IP packets which isn't
>> bound to any local network interface (especially if you're sitti
2014-12-18 19:30 GMT+02:00 Richard Fuchs :
> Write error on RTP socket usually indicates an incorrect network setup,
> for example trying to use a source address for IP packets which isn't
> bound to any local network interface (especially if you're sitting
> behind NAT), or local iptables rules re
On 12/18/14 12:11, Andrey Utkin wrote:
> Hi!
> I need to establish calls between WebRTC and usual SIP clients
> (exactly, sipml/jssip and linphone-android).
> I used configs from https://github.com/caruizdiaz/kamailio-ws and
> latest git master HEAD of both kamailio and
> rtpengine. I got calls fro
Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only wit
Hi Rob Moore,
Yes, I have intended to use TLS in client side to verify with server side.
I have tried to create cert files as :
Quick Certificate Howto
in http://kamailio.org/docs/modules/stable/modules/tls.html#tls.debugging
Then, I tried to use Blink phone to user crt file, But I see that I ca
Hi Thanh,
Are you intending to use Client certificates in your setup?
If not, the error “SSL3_GET_CLIENT_CERTIFICATE “ would lead me to believe that
your problem is modparam("tls", "require_certificate", 1) which when enabled
kamailio will require a certificate from the client.
I’m not an expe
Hi all,
I have tried several configure TLS in kamailio but no luck.
Please give me some suggestion that I can make it work correctly.
This is my configure in TLS module.
modparam("tls", "tls_method", "SSLv23")
modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem")
modparam("
Hi Carsten
>- Do you have the Dispatcher Activity detection turned on?
>Can you send me your config, so i can verify a few things?
My dispatcher configuration is:
# - dispatcher params -
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "force_dst", 2)
modparam("d
Marc Soda writes:
> How can I force Kamailio to use TCP for SIP when relaying the call? I
> haven't found much info on it.
for example, you can assign to $du a sip uri that includes ;transport=tcp
param.
-- juha
___
SIP Express Router (SER) and Kamai
Content-Length: 1901.
So trimming up the headers isn't going to get me anywhere... I'm not
comfortable enough with WebRTC to know what to trim out of the SDP, either.
How can I force Kamailio to use TCP for SIP when relaying the call? I
haven't found much info on it.
Marc
On Thu, Dec 18, 2014
Hi,
quick questions:
- Do you have the Dispatcher Activity detection turned on?
- Do you do something like "ds_mark_dst()" in your failure route?
- Can you send me your config, so i can verify a few things?
Thanks,
Carsten
2014-12-18 15:14 GMT+01:00 José Seabra :
> Hello Daniel,
> Thank you fo
Hello Daniel,
Thank you for your reply,
Please check out the attachment with kamailio logs in mode debug 3, i
didn't find any errors, the only thingh that is confusing me is this line:
/usr/local/sbin/kamailio[13690]: DEBUG: dispatcher [dispatch.c:1745]:
ds_select_dst_limit(): alg hash [0]
I don'
Hello,
a short note to announce for everyone interested that submissions of
presentation proposals is now open for Kamailio World 2015 (Berlin, May
27-29). More details at:
-
http://conference.kamailio.com/k03/2014/12/kamailio-world-2015-call-for-speakers/
Cheers,
Daniel
--
Daniel-Constantin
Hello,
I asked in the first email - do you get error messages in syslog? If
not, then you have to run with debug=3 and look at the log messages to
see if you get further hints.
Cheers,
Daniel
On 18/12/14 12:29, José Seabra wrote:
> Hello Daniel,
> Do you need more information from my kamailio se
Hello Daniel,
Do you need more information from my kamailio settup, in order to try
understand why this issue happens?
Thank you
BR
José Seabra
2014-12-16 12:16 GMT+00:00 José Seabra :
>
> Update:
> My last email has the wrong example, the correct code is:
>
> if(!ds_select_dst("9", "4"))
> {
>
No no you've to edit you scripts...
For examplein a failure_route:
if (t_check_status("408")) {
#!ifdef WITH_ACCOUNTING
setflag(FLT_ACCFAILED);
#!endif
}
In this way you can "log" a call missed for timeout.
Massimo Varriale (IPZeta) wrote
>
Hi Marino,
now the parameter is set to number 2
#!define FLT_ACCMISSED 2
Should raised to 3 or more?
Thanks
Il giorno 18/dic/2014, alle ore 11:02, Marino Mileti ha scritto:
> Hi Massimo
>
> you've to modify also your scripts and set flag FLT_ACCMISSED when you want
> to log the missed or
On 18/12/14 02:58, Marc Soda wrote:
> So gzcompress is no good with Asterisk then? Is that meant to be used
> only with another Kamailio proxy?
Apparently Apple Facetime is using this kind of compression (as it was
reported on a blog and triggered the implementation in Kamailio), but
one cannot
On 18 Dec 2014, at 03:01, Alex Balashov wrote:
> Indeed, gzcompress is a Kamailio-only concept.
In fact we got it from Facetime...
SHould be easy to implement in Asterisk too.
/O
>
> On 17 December 2014 20:58:22 GMT-05:00, Marc Soda wrote:
> So gzcompress is no good with Asterisk then? Is t
Hi Massimo
you've to modify also your scripts and set flag FLT_ACCMISSED when you want
to log the missed or failed call
Marino
-
Marino Mileti
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Kamailio-CDRs-tp11p12.html
Sent from the Users mailing list archi
Hi Guys!
I'm trying to store CDRs for Kamailio calls.
Following Siremis tutorial (http://siremis.asipto.com/install-accounting/) I'm
able to gather Succesfull Calls and using the MySQL stored procedure I can have
a CDR table.
But what I can see now is that only outbound calls with a duration a
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