Hello,
it looks that many branches still exist after being merged, discovered
by running git-sweep. The list is next.
If anyone from devs or users spots a branch that was not merged and
needs to be reviewed, let me know. In several days I plan to remove them
to keep the git repo a bit more sane.
Cheers,
Daniel
$ git-sweep preview
Fetching from the remote
These branches have been merged into master:
andrei/blst_send_flags
andrei/cancel_reason
andrei/cdefs2doc
andrei/counters
andrei/fixups
andrei/mod_f_params
andrei/mod_register
andrei/module_interface
andrei/path
andrei/pointer_alias_warnings
andrei/raw_sock
andrei/rpc_async
andrei/rve_f_params
andrei/script_vars
andrei/send_flags
andrei/shm_early_init
andrei/switch
andrei/tcp_tls_changes
andrei/to_parser_fix
andrei/type_conversion
co/crypto_name_collision
co/tcp_closed_event_enhancements
curl
cvs-head
ez/newstuff
henning/trie
henning/trie_mods
janakj/missing_imports
kamailio_modules
lazedo-patch-1
lazedo-patch-2
lazedo/presence_improvements
lazedo/presence_xml_fix_dummy
luismartingil/msrp_crash
mariusbucur/dmq
mariuszbihlei/dnssec
misi/ua-profile
sd_journal_send_xavp
ser_core_cvs
ser_modules
tirpi/cfg_framework_multivalue
tirpi/script_callbacks
tmp/build_local
tmp/build_request
tmp/core_events
tmp/hpw_curl_improvements
tmp/k3.0_sr_backports
tmp/ruri_branch
tsearle-futex-try
tteras/db_sqlite
--
Daniel-Constantin Mierla
http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
Hello,
I'm trying to understand why a RE-INVITE is still discard by Kamailio
(4.2.6). The RE-INVITE has the last Contact URI as Request-URI which is not
Kamailio himself. But there are the route headers set.
What are the other checks done by the function "loose_route()"?
Regards,
Igor.
Hello everyone.
I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 503 error (please look at my attached kamailio.cfg)
I use "ds_probing_mode", 1 and "ds_ping_reply_codes", "class=2;class=3;class=4" to OPTIONS ping the two asterisk instances but they do appear down both all the time.
That is very annoying none of the INVITE go through Asterisk, they all are stopped because of my failure_route that sets no destination available
ERROR: <script>: Destination down: INVITE sip:9980048587420715@XXX.XXX.XXX.XX (sip:ipofAsterisk1:5060)
ERROR: <script>: Destination down: INVITE sip:9980048587420715@YY.YY.YYY.YY (sip:ipofAsterisk1:5060)
ERROR: <script>: Destination down: INVITE sip:000442030044700@XXX.XX.X.XXX (sip:ipofAsterisk2:5060)
ERROR: <script>: Destination down: INVITE sip:000442030044700@YY.YYY.YY.YYY (sip:ipofAsterisk2:5060)
NOTICE: <script>: Media server sip:172.16.0.202:5060 failed to answer, selecting other one!
NOTICE: <script>: No destination, sent 404
I don't know what's wrong with my setup, could someone give an eye to my kamailio.cfg and head me on the right track ?
Thanks a lot.
Seb
When placing a call and checking for an answering machine using mod_com_amd, it always sees *silent_state*. Strangely, when I stream audio TO the session, mod_com_amd registers this audio and responds accordingly. So, it seems that somehow mod_com_amd is listening to the wrong end of the session.
Any help would be greatly appreciated.
Best regards,
Michael Jepson
Hi all,
What's the most efficient way to check if an xavp is set?
I have tried-
is_avp_set() from avpops module
which resulted in-
ERROR: avpops [avpops.c:915]: fixup_is_avp_set(): bad attribute name
<$xavp(tm_contacts)>
and
if ($xavp(tm_contacts)) {}
which resulted in -
WARNING: <core> [rvalue.c:1007]: rval_get_int(): automatic string to int
conversion for "<<xavp:0x7feac74cfcb8>>" failed
To give a little background, I am using the UAC_redirect module
and t_next_contacts() in conjunction with 302 redirects .However if there
is only a single contact, $xavp(tm_contacts) is not created and
t_next_contacts() fails, hence why I'd like to check if it exists first.
Thanks
In kamailio server register is done with 200ok response and now i send
invite but kamailio not give replay 200ok
What i need to do configure of kamaulio file or not
And kamailio.log file show some error like
Parse_first_line():parse_first_line: bad message
And
Parse_msg:message=<INVITE ....to, from,via all filed.......>
And
Receive _msg(): CORE PARSING OF SIP MESSAGE FAILED
I feel like I have to be missing something obvious here, but I can't for the life of me find anything. Is there not a basic random number/string generation function in core or any of the modules anywhere? I find that incredibly unlikely that such a simple function that could have so many uses wouldn't exist, but I just can't find anything...
Any pointers?
Brooks Bridges | Sr. Voice Services Engineer
O1 Communications
5190 Golden Foothill Pkwy
El Dorado Hills, CA 95762
office: 916.235.2097 | main: 888.444.1111, Option 2
email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>
Hello,
I'm trying to setup latest Siremis from GIT and stuck on DB stage.
ERROR: SQLSTATE[08006] [7] FATAL: password authentication failed for
user "www-data" FATAL: password authentication failed for user "www-data"
Why installer tries to use web server user instead of supplied?
DB is PGSQL at localhost.
Kam DB created successfully via kamdbctl.
--
Serge S. Yuriev
Lead VoIP engineer
Hello,
I'm using sipcapture module in Kamailio to capture sip packets and save them
into mysql database.
Everything is working as expected and I got the packets registered in the
database , yet I'm wondering if there is a way to open these packets
directly in Wireshark , or do I need an intermediate step to convert these
packets to pcap format (if possible) in order to be read by Wireshark.
Any help or hint would be appreciated.
Regards,
Ali