Hello,
soon we should sketch the roadmap for the next major release 3.2. There
are some private git branches with lot of work on it, so we need to
synchronize a bit and decide about the proper time: before or after summer.
Several devels will be in Berlin for Linuxtag, but no so much
overlapping for a big face to face meeting, therefore might be better to
have a IRC chat conference sometime in the near future. Next week will
be fine for me, say Wednesaday till Friday, in the afternoon to allow
fair time of the day for US timezones.
If you have other proposals, speak up.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
Hello,
On 4/19/11 9:58 AM, Pavel Segeč wrote:
> Hi,
>
> may someone help me and recommend a software SIP client with a good IPv6
> support implemented? I tried Xlite, Bria, Linphone, SIP communicator and
> Jitsi but with a bad results. Each have some problem within pure IPv6
> environment.
the last sipit I attended (last year, though) was not rich at all in
IPv6 softphones. In the past there was a patched version of KPhone we
used at FhG Fokus for some tests, but I wouldn't call it production ready:
http://old.iptel.org/products/kphone/
From what was listed above, you touched most of the active softphones,
afaik ekiga,qutecom and twinkle do not have it, not sure about zoiper.
Cheers,
Daniel
I once wrote a SIP client based on pjsip:
http://www.ipcom.at/en/telephony/qjsimple/
AFAIK it works will using IPv6 when using UDP.
regards
Klaus
Am 19.04.2011 09:58, schrieb Pavel Segeč:
> Hi,
>
> may someone help me and recommend a software SIP client with a good IPv6
> support implemented? I tried Xlite, Bria, Linphone, SIP communicator and
> Jitsi but with a bad results. Each have some problem within pure IPv6
> environment.
>
> palo
>
>
> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6053
> (20110418) __________
>
> Tuto spravu preveril ESET NOD32 Antivirus.
>
> http://www.eset.sk
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
I'm trying to migrate ser-0.9.x to the current sr-3.1.x and right now trying
to figure out the acc functions/parameters. I used to use "acc_db_request" from
acc module, but it's not working using acc_db and acc_syslog in conjuntion.
acc_db has acc_db_request[0,1] but are not callable from the configuration
file, and acc_syslog has acc_db_request, but doesn't matter if i load this
module, the function isn't found...
loading modules under /usr/lib/siprouter/modules:/usr/lib/siprouter/modules_s
0(23455) : <core> [cfg.y:3412]: parse error in config file /tmp/ser.cfg,
line 358, column 74: unknown command, missing loadmodule?
line 358 is:
acc_db_request("480 Offline User", "missed_calls");
I would appreciate if you can guide me through the ACC documentation or
changes that it has suffered.. I tried to track the git log.. but many changes
since my old ser :(
Thank you
Claudio
pd: FYI i did this modifications to adjust my acc settings:
# Accounting module
+loadmodule "acc_syslog.so"
loadmodule "acc_db.so"
# ----------------- setting module-specific parameters ---------------
@@ -139,15 +140,13 @@
# number of flag, which will be used for accounting; if a message is
# labeled with this flag, its completion status will be reported
# modparam("acc", "log_flag", 1 )
-modparam("acc_db", "log_level", 1)
+modparam("acc_syslog", "log_level", 1)
# MySQL acc params
modparam("acc_db", "db_url", "mysql://ser:heslo@localhost/ser")
-modparam("acc_db", "db_flag", 2)
-modparam("acc_db", "db_missed_flag", 3)
+modparam("acc_db", "log_flag", 2)
+modparam("acc_db", "log_missed_flag", 3)
modparam("acc_db", "log_fmt", "miocfspD")
-# Acc time set to localtime (1)
-modparam("acc_db", "db_localtime", 1)
Hello,
What happened to checks.c file on uri_db module (modules_s)? Seem to be an
empty file now, only headers exist.
Is it the same to use modules_k/ version?
I'm currently at v3.1..
Regards,
Claudio
is there any means to access native tm stats from script? i know about
$stat pvs for accessing k versions of tm stats, but i want access to
native tm stats.
-- juha
Hey Guys,
I was recently playing with gateway-ing IPv4-IPv6 and hit the
following scenario:
* AOR having contacts on both ipv4 and ipv6 and I wanted to do
parallel forking.
RTPProxy bridging works without any issue on a normal setup, however
the problem shows up when needing to make calls toward rtpproxy to
return both sides of bridge or only one (ee and ie combinations). Did
any of you experiment with this scenario?
EG: Call comes from ipv4, you want to send it to both ipv4 and ipv6.
The branch route looks the right place to call rtpproxy but when
calling unforce_rtp_proxy() on CANCEL, will rtpproxy be aware about
which ports are we trying to cancel? One more issue would be with
re-invite which must go out with the same ip of rtpproxy as original
INVITE. Here we could store the bridge direction in some route
parameter but unfortunately adding route params is not possible in
branches.
So what do u think?
DanB
PS: The "normal" setup of forking calls only ipv4 or only ipv6 works
smooth, so support of ipv6 in kamailio or rtpproxy is not
questionable.
Hello.
I’m trying to get statistics to use charts in SIREMIS v2.0. I was able to
load the “Memory” statistics but I want to know if there is more information
to use from Kamailio, right now I have :
$stat(used_size)
$stat(real_used_size)
$stat(max_used_size)
$stat(free_size)
$stat(location-users)
$stat(location-contacts)
Where can I find the complete list of statistics to get from kamailio?.
Thanks in advance.
Ricardo Martinez.-
Hello all,
I'm having problems with attended transfer in certain scenarios. Let me first try to document the relevant parts of the setup:
There are two load-balanced Asterisk gateways connected to the old Nortel system. Currently, PSTN connections are done by Nortel. We then have two load-balanced kamailio servers. The phones are connected to these. There are several other servers not relevant to this question.
Blind transfer works in any scenario. I've been told that attended transfer works in any scenario if the original call is SIP to SIP. If the original call is from Nortel/PSTN to SIP, then attended transfer will usually not work.
Scenarios:
A call comes in from Nortel/PSTN (user A) through Nortel1 to a SIP device (user B). User B pushes the transfer button (puts current call on hold and makes a new call) and dials a Nortel / PSTN number to user C. This call goes out through the nortel2 server. Users B and C talk a short while before user B pushes the Transfer button again to connect users A and C. In this scenario, the REFER is forwarded by kamailio to nortel1, which replies "SIP/2.0 481 Call leg/transaction does not exist." The call to join is of course on the other asterisk (nortel2).
A similar scenario is where user B transfers to another SIP user. This call will only exist in one of the kamailio servers, and Asterisk will give the same response to the REFER.
A third scenario is where both calls are handled by the same Asterisk server. This scenario works.
I'm assuming I'll have to build REFER-handling logic into kamailio, but am unsure of how to proceed. Any suggestions?
Thanks in advance.
With kind regards,
Pan
Hello,
I am using Kamailio RLS and integrated XCAP Server. When I add a
contact with my client it updates my XCAP documents but there is no
additional SIP messaging between the client and Kamailio.
I have been assured (by the client author) that this is the correct
behaviour. However, having looked at both the XCAP Server and RLS
module code I cannot see anything that will either inform RLS that a
document has been updated or a timer to check for that.
Have I missed something obvious?
Thanks,
Peter
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
