Daniel wrote:
please do the patch to store the new value in _tr_buffer and attach it
to mailing list or bug tracker. I haven't looked at code yet, but sounds
like there is indeed an issue. I will review the patch and apply it.
What was the conclusion about this? Was the patch ever submitted and
applied? If not, is the bug still there or has is been fixed by some
other means? I'm asking because this looked like a quite serious
issue.
-- Juha
Hi
we have developed the custom module where in we have created shared
memory area which is not used by any other module or process and we
locked it using locks as recommended. but because of locks the
kamailio is getting crashed every now and then . is it really
important to protect the shared memory area with locks. if yes
then how to avoid the crash ??
so to locate the crash we built kamailio wit h MM_DBG , and other
GCC debug flags . but we are getting error "/No symbol table info
available/" . how to locate the bugs. the kamailio is built wit h
1 MB of shared memory on MIPS based VOIP gateway which is having *
MB of flash and 16 MB or memory (RAM) .
thanks in advance .
Hi all,
I'm trying to use kamailio as a proxy between webrtc clients and an
openIMSCore. For this reason I'm currently using the dispatcher module to
distribute all the requests coming from the clients to the IMS core, and
enabled the websocket module for getting the SIP messages in. SO far it
works fine for the registration, but it doesn't for the INVITEs. I noticed
that kamailio receives the message but it doesn't deliver it to the
clients. Please find attached the log and the configuration file that I
used. I'm not an expert with the configuration, so any help or feedback
would be very appreciated. Thank you in advance.
Best regards,
Giuseppe Carella
I would like to share my experience with kamailio and other home pbx servers.
Kamailio on my kirkwood home router for my 6 SIP users is perhaps
overkill: I don't really need mysql and "scalability". But at last I
finally managed to make calling between registered users work stable.
My voip clients only work in all NAT scenarios if I work around some
bugs: to use csipsimple on android I had to change rtpproxy_manage()
to rtpproxy_manage("c") in kamailio's default config, so that problems
with conflicting c: entries in the SDP go away.
I propose kamailio could ship with a special example
kamailio-compatible.cfg that doesn't try to be RFC compliant, but
compatible to the most common voip clients. Right now the only thing I
would change for this is the option for rtpproxy_manage, but I'm sure
others will know more common quirks that could safely be enabled to
increase compatibility. I think this compatibility idea is what yate
sticks to for their defaults. In freeswitch you also have to do it all
manually, and it's much more work to figure things out in their
enormous config files.
The other SIP proxies I had tried before kamailio officially fit all
my requirements, including support for multihomed dynamic IPs, but
contrary to their claims it didn't work.
Yate was easy to set up, but the default dialplan is more confusing
than powerful and after having made everything work I realised yate
was clogging my CPU and RAM and after some time always randomly
stopped working. This is with only 2 users connected! It also wasn't
possible to fix NAT sdp while leaving the codecs section in the SDP
alone at the same time. I tried to debug the code, but the C++ was so
complex that I had to give up.
Freeswitch was much more difficult to setup, a multihomed setup with
dynamic IP was super buggy and it also didn't help that the
unintuitive configuration is all in complex unreadable XML
configuration files.
Kamailio and rtpproxy don't officially support dynamic IP address, but
I can just restart both each time my DSL provider forces me to a new
IP address. This happens automatically in the night and is no big
hassle really. The most simple, least-featureful solution works best
it seems.
Now the last problem I have with kamailio: I don't know how to connect
my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont).
I would like a simple way to do this, preferably without other
features that always seem to complicate the matters. Is there
something more lightweight and simple than asterisk, freeswitch and
yate, that people use successfully for this task together with
kamailio and rtpproxy?
u
Hi,
I am trying to add a log when receiving 1xx or 2xx reply message:
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
{
xlog("L_INFO","reply message: $T_reply_code ci:$ci");
route(NATMANAGE);
}
}
Adding the above line does not log the reply message number (180 or 183 or
200, ...).
Thanks,
R
Should UAC work with Sipgate and freevoipdeal (dellmont)?
Thanks for your answers everyone.
2013/5/15 u <ueberwachungsstaat(a)googlemail.com>:
> Should UAC work with Sipgate and freevoipdeal (dellmont)?
>
> Thanks for your answers everyone.
>
> 2013/5/15 Stefan Sayer <stefan.sayer(a)googlemail.com>:
>>
>> if you don't get it all (auth + registrations) working with kamailo,
>> you might try sems (iptel.org/sems) with sbc module (auth of outgoing
>> calls) and for registrations either reg_agent (cfg file) or
>> db_reg_agent (accounts from db) modules.
>>
>> load_plugins=uac_auth;registrar_client;reg_agent;sbc;xmlrpc2di
>> application=sbc
>>
>>
>> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_sbc.html
>> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_reg_agent.html
>> http://ftp.iptel.org/pub/sems/doc/current/AppDoc.html
>>
>> BR
>> Stefan
>>
>> o Edson - Lists on 05/15/2013 06:40 AM:
>>> Hi...
>>>
>>> Use UAC module to manage registrations and play a little with the
>>> config (INVITE section) to forward output calls correctly.
>>> ---
>>> Edson.
>>>
>>> Em 14/05/2013 11:27, u escreveu:
>>>> I would like to share my experience with kamailio and other home pbx
>>>> servers.
>>>>
>>>> Kamailio on my kirkwood home router for my 6 SIP users is perhaps
>>>> overkill: I don't really need mysql and "scalability". But at last I
>>>> finally managed to make calling between registered users work stable.
>>>> My voip clients only work in all NAT scenarios if I work around some
>>>> bugs: to use csipsimple on android I had to change rtpproxy_manage()
>>>> to rtpproxy_manage("c") in kamailio's default config, so that problems
>>>> with conflicting c: entries in the SDP go away.
>>>>
>>>> I propose kamailio could ship with a special example
>>>> kamailio-compatible.cfg that doesn't try to be RFC compliant, but
>>>> compatible to the most common voip clients. Right now the only thing I
>>>> would change for this is the option for rtpproxy_manage, but I'm sure
>>>> others will know more common quirks that could safely be enabled to
>>>> increase compatibility. I think this compatibility idea is what yate
>>>> sticks to for their defaults. In freeswitch you also have to do it all
>>>> manually, and it's much more work to figure things out in their
>>>> enormous config files.
>>>>
>>>> The other SIP proxies I had tried before kamailio officially fit all
>>>> my requirements, including support for multihomed dynamic IPs, but
>>>> contrary to their claims it didn't work.
>>>> Yate was easy to set up, but the default dialplan is more confusing
>>>> than powerful and after having made everything work I realised yate
>>>> was clogging my CPU and RAM and after some time always randomly
>>>> stopped working. This is with only 2 users connected! It also wasn't
>>>> possible to fix NAT sdp while leaving the codecs section in the SDP
>>>> alone at the same time. I tried to debug the code, but the C++ was so
>>>> complex that I had to give up.
>>>> Freeswitch was much more difficult to setup, a multihomed setup with
>>>> dynamic IP was super buggy and it also didn't help that the
>>>> unintuitive configuration is all in complex unreadable XML
>>>> configuration files.
>>>>
>>>> Kamailio and rtpproxy don't officially support dynamic IP address, but
>>>> I can just restart both each time my DSL provider forces me to a new
>>>> IP address. This happens automatically in the night and is no big
>>>> hassle really. The most simple, least-featureful solution works best
>>>> it seems.
>>>>
>>>> Now the last problem I have with kamailio: I don't know how to connect
>>>> my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont).
>>>> I would like a simple way to do this, preferably without other
>>>> features that always seem to complicate the matters. Is there
>>>> something more lightweight and simple than asterisk, freeswitch and
>>>> yate, that people use successfully for this task together with
>>>> kamailio and rtpproxy?
>>>>
>>>> u
>>>>
>>>> _______________________________________________
>>>> 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
>>>> .
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
I can't install siremis 3.0 for kaimailio after install kamailio, It take a
long long time to wait for and no end.
in CRIT.log of siremis/log, I get
'05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a
directory',''
Anyone can tell me how to do? Thank you all.
Hi,
When the Server is running it is crashing occasionally.
(gdb) bac
#0 0x000000000043359e in ?? ()
#1 0x0000000000000000 in ?? ()
(gdb) file /usr/local/sbin/kamailio
Reading symbols from /usr/local/sbin/kamailio...done.
(gdb) bac
#0 free_lump_list (lump_list=<value optimized out>) at data_lump.c:504
#1 del_nonshm_lump (lump_list=<value optimized out>) at data_lump.c:661
#2 0x00007f571ff6cd9c in ?? ()
#3 0x00000000008db480 in mem_pool ()
#4 0x00007f571ff6c7e1 in ?? ()
#5 0x000000000139086e in ?? ()
#6 0x00000039294546e1 in ?? ()
#7 0x000000000140c7d0 in ?? ()
#8 0x00007f561f7acb80 in ?? ()
#9 0x00007f561f3fa6e0 in ?? ()
#10 0x00007f561f3fa6e0 in ?? ()
#11 0x0000000000000001 in ?? ()
#12 0x0000000000000000 in ?? ()
What could be going wrong here?
Krish Kura
using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind
NAT registered via UDP I get no voice.
The e72 strangely sends a single udp packet from a wrong port (49152)
before the rtp stream should start.
This quirk of the e72 doesn't seem to work well with rtpproxy if the
following analysis is true:
rtpproxy detects that single UDP packet from the wrong port and so we
think that is where everything else will also come from and stop
listening on other ports. we then also answer on that wrong port.
Although all subsequent incoming packets arrive from the expected
(49172) port sent also in the sdp and to the right one we had sent in
the sdp earlier we never receive them, because we still listen on that
wrong port with that one bogus packet.
tcplog, rtp should be running, but no voice:
...
00:02:13.829682 IP p5795BC1A.dip0.t-ipconnect.de.49172 >
koln-5d81d2a9.pool.mediaWays.net.49184: UDP, length 73
00:02:13.847063 IP koln-5d81d2a9.pool.mediaWays.net.49184 >
p5795BC1A.dip0.t-ipconnect.de.49152: UDP, length 73
...
/usr/sbin/rtpproxy -l 93.129.210.169 -s
unix:/var/run/rtpproxy/rtpproxy.sock -u kamailio kamailio -p
/var/run/rtpproxy/rtpproxy.pid
...
DBUG:handle_command: received command "Uc100,8,0,98
_39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 192.168.1.56 40000
o5r7rmu5p1hc7vci0591;1"
INFO:handle_command: new session _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB, tag
o5r7rmu5p1hc7vci0591;1 requested, type strong
INFO:handle_command: new session on a port 49184 created, tag
o5r7rmu5p1hc7vci0591;1
INFO:handle_command: pre-filling caller's address with 192.168.1.56:40000
DBUG:doreply: sending reply "49184 93.129.210.169
"
DBUG:handle_command: received command "Lc100,98
_39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 87.149.188.26 49172
o5r7rmu5p1hc7vci0591;1 5qt1pav3
INFO:handle_command: lookup on ports 49184/49196, session timer restarted
INFO:handle_command: pre-filling callee's address with 87.149.188.26:49172
DBUG:doreply: sending reply "49196 93.129.210.169
"
INFO:rxmit_packets: callee's address filled in: 87.149.188.26:49152 (RTP)
DBUG:handle_command: received command "D
_39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 5qt1pav3dsafun5p47o30tj2od1urm6v
o5r7rmu5p1hc7vci0591"
INFO:handle_delete: forcefully deleting session 1 on ports 49184/49196
INFO:remove_session: RTP stats: 1 in from callee, 3465 in from caller,
3466 relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 49184/49196 is cleaned up
DBUG:doreply: sending reply "0
what do you guys think? it only occured with kamailio+rtpproxy so far.
any insight about why the e72 sends such packet? it doesn't happen
when i use sip over tcp. is my analysis right, is it a bug in
rtpproxy?
greetings
hiro
