Hey,
I’m noticing that the results from the websockets module when running # kamctl stats there are some odd results that may just be typos?
"websocket:ws_msrp_max_concurrent_connections = 0",
"websocket:ws_msrp_max_concurrent_connectons = 0”,
"websocket:ws_sip_max_concurrent_connections = 0",
"websocket:ws_sip_max_concurrent_connectons = 0",
Thanks!
~Noah
Hello all,
Kindly let us know if we can have a mechanism to have the voice stream in
using the SIP server completely encrypted as well as prove that it is
encrypted using any speific mechanism.
Thanks in advance
Vasan
Hi,
I'm trying to debug a Kamailio which runs out of memory after a while.
But following the hints at
https://www.kamailio.org/wiki/tutorials/troubleshooting/memory
doesn't help. I know it is the private memory that I'm missing. I
already increased the pkg size to 16M, but somehow that seems to be
not enough.
The Kamailio receives HEP messages, sends them to a lua script where
it is processed. Overall there are not more than 10 custom private
variables used.
In the log I'm seeing these messages:
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: app_lua
[app_lua_api.c:726]: app_lua_run_ex(): error executing: handle_packet
(err: 2)
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <script>:
SCRIPT: failed to execute lua function!
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core>
[core/mem/q_malloc.c:292]: qm_find_free():
qm_find_free(0x7f9744eea010, 232); Free fragment not found!
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core>
[core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f9744eea010, 232)
called from core: core/parser/msg_parser.c: get_
hdr_field(116), module: core; Free fragment not found!
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core>
[core/parser/msg_parser.c:118]: get_hdr_field(): out of memory
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core>
[core/parser/msg_parser.c:330]: parse_headers(): bad header field
[Via: SIP/2.0/WSS o6m]
Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: pv
[pv_core.c:647]: pv_get_callid(): cannot parse Call-Id header
kamcmd pkg.stats for this process gives me this output.
{
entry: 61
pid: 19134
rank: 61
used: 8748080
free: 4384
real_used: 16772832
total_size: 16777216
total_frags: 30
}
Now I tried to get a dump of what is in use, following the guidelines
on the wiki page mentioned above, but without success.
I can easily start and stop the traffic coming in to Kamailio (by
starting or stopping the captagent sending it there), but I can't get
any output. Here's what I did:
captagent stop
kamcmd cfg.set_now_int core memlog 1
kamcmd cfg.set_now_int core mem_dump_pkg 19134
captagent start ; sleep 1 ; captagent stop
(I started captagent so that the process actually had something to do
because that's what the wiki page says it needs for printing memory
information.)
During that second, there were quite a lot of packets running through
this process, but I did not get a memory dump. A shm_summary dump
however works as expected.
What am I missing? How do I get more information about what this
process is using its memory for?
TIA
Sebastian
Hi guys,
Any body know how to create pcap (with audio) file to use with sipp?
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="../scenarios/pcap/demo.cap"/>
</action>
</nop>
I tried with wireshark capturing rtp but I'm doing something wrong i think.
Thanks!
Diego
Hi,
I think this has been answered before, but I don't know the answer. Is
there a way to find out, how many workers are busy right now processing
packets? For a new setup I don't know whether I need 16 or 96 UDP
listeners, but I'd like to scale it correctly.
The setup just inspects HEP messages and sends some of them to a RabbitMQ.
So TM or SL stats probably won't help.
Best Regards
Sebastian
Hi All
Trying to set Kamailio as proxy in front of Asterisk.
When natted user registers is remembered in Kamailio via save("location").
Kamailio forwards Register to * with new contact field (pointing to Kamailio).
But when server originates Invite to user, kamailio transfer
request to "fake" address that was in original Register contact field
(getting it from lookup("location")).
How can i set Kamailio to send Invite to source ip and port of user?
Hi
Thanks in advance if anyone can point me in the correct direction .
I have kamailio running in front of some asterisk. SIP endpoint
register to their asterisk PBX via Kamailio dispatcher module. I'm running
rtpengine with a Wan and private interface to bridge audio stream between
these endpoints on the WAN to asterisk PBX running on LAN IP behind
Kamailio.
Calls from ext to ext work fine audio both directions , calls outbound to
PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio
both directions. But incoming calls via SIP provider no audio from
external caller to the asterisk ext neither asterisk to external caller
I reckon I have something wrong in my Kamailio.cfg . if I register an ext
direct to asterisk I get audio both ways on incoming calls. And rtp logs
I think my mistake in somewhere in the cfg below.
Do I need to handle invites from the backend asterisk servers different that
invites from sip endpoints?
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