Hello,
I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and Sip
server FreeSwitch.
Whenever I call directly from UAC to Sip server, the call gets established
for as long as I want, however when I use the proxy in between, it gets
disconnected within 30 seconds. It seems that FS sends a BYE within 30
seconds.
I would really appreciate if anybody please guide me where I am going wrong
in this case ?
--
Warm Regds.
MathuRahul
Hello,
Please let me know if there is any work around for this.
Details of the issue:
kamailio.cfg has this sequence in kamailio.cfg,
......
modparam("rtpproxy", "ice_candidate_priority_avp",
"$avp(ice_priority)" )
........
sdp_remove_line_by_prefix("a=candidate");
set_iattr("ice_priority", "1");
rtpproxy_manage("frwie");
I'm seeing an issue towards the bottom of the SDP message.
....
a=ssrc:738345272 mslabel:VPELrlXvdfGmtevlYLEeVm6zq4JHvI4g
a=ssrc:738345272 label:VPELrlXvdfGmtevlYLEeVm6zq4JHvI4ga0
a=ice-ufrag:YJuJWr9clIJHhENw
a=ice-pwd:3v1D1ShCg8xIlDZ9
a=candidate:1297581420 1 udp 659136 10.1.2.3 17534 typ host
generation 0
a=candidate:1297581420 2 udp 659136 10.1.2.3 17535 typ host
generation 0
a=ptime:20
\r\n
a=nortpproxy:yes\r\n
The call gets rejected by UA due to an invalid (empty) line in SDP.
Very old but somewhat similar issue:
http://lists.sip-router.org/pipermail/users/2008-September/019605.html
(Module authors, I'll be happy to code/test or assist in any way possible
if needed.)
Just to test it, I commented the following code snippet and the call
worked.
But that can't be the permanent fix.
Thank you,
- Deep N
Code snippet:rtp_proxy.c::force_rtp_proxy starting from line 2850
* if (proxied == 0 && nortpproxy_str.len) { cp =
pkg_malloc((nortpproxy_str.len + CRLF_LEN) * sizeof(char));
if (cp == NULL) { LM_ERR("out of pkg memory\n");
return -1; } anchor =
anchor_lump(msg, body.s + body.len - msg->buf, 0, 0); if
(anchor == NULL) { LM_ERR("anchor_lump failed\n");
pkg_free(cp); return -1;
} memcpy(cp, CRLF, CRLF_LEN);
memcpy(cp + CRLF_LEN, nortpproxy_str.s, nortpproxy_str.len);
if (insert_new_lump_after(anchor, cp, nortpproxy_str.len + CRLF_LEN, 0) ==
NULL) { LM_ERR("insert_new_lump_after failed\n");
pkg_free(cp); return -1;
} }*
Hello I try to use kamctl dispatcher reload cmd but in gives me an error
kamctl dispatcher reload
ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo
ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name",
"/tmp/kamailio_fifo")' in your config
ERROR: and also have loaded the mi_fifo module.
when i try to user kamcmd dispather.reload it gives me:
ERROR: connect_unix_sock: connect(/tmp/kamailio_ctl): No such file or
directory
At kamailio.cfg I defined modparams for ctl.so and enables ctl.so module too
modparam("ctl", "mode", 0600)
modparam("ctl", "user", "root")
modparam("ctl", "fifo", "fifo:/tmp/kamailio_fifo")
modparam("ctl", "fifo", "/tmp/kamailio_fifo2")
modparam("ctl", "fifo", "udp:*:2050") # fifo protocol over udp
modparam("ctl", "fifo", "tcp:*:2050") # fifo over tcp
At the kamctlrc I defined
FIFOPATH="/tmp/kamailio_fifo"
So I can not understand what is wrong with my FIFO file?
Hello,
I want to announce that a new developer got GIT write access to
repository: Jorj Bauer from University of Pennsylvania, USA - he is
mainly joining the efforts to maintain and develop further the SCA module:
* http://kamailio.org/docs/modules/stable/modules/sca.html
His git commit id is: jorj
My warm welcome and thanks for joining to help the project.
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
Greetings all.
I have following use case:
0) Peer A and B registers on kamailio via TLS.
1) Peer A establishes call to peer B.
2) Peer A changes network and call has to be refreshed. Peer A reregisters,
gathers ICE candidates, builds reINVITE and sends it to B. After peer A
changes network again, reregisters and waits for answer from old reINVITE.
3) Peer B receives reINVITE and sends answer.
The problem is this answer never gets to peer B. Kamailio cannot route
answer as old TLS connection is closed and new one did not established
before reregistering.
Is there way to solve or avoid this situation?
Thank you :)
Hi Guys,
I have used the http_query function in the past, in conjunction with the json module, on version 3.3 and unbuntu.
I am now using version 4.1 on Centos 6.5, and I am using the http_query, which I'm aware records in the results the first line of the response.
Can you select the result to be recorded depending on the line in the response, for example if a number is returned only in line 4, can this be extracted from a 200 ok response using this function?
Or is this then a need to use JSON module, and if so which repository for centos 6.5 contains it! :)
Sorry for the question!
Thanks
Jon
Hello. I try to integrate dispathcer module (with db table dispatcher) to
my Kamailio server.
I have problem with changing IP at db table, but when I calling - call goes
to old IP that alread deleted from db.
I tested my table with query that selects destination address from table,
and it replyes to me correct address.
I use round-robin ("4") algorythm
modparam("dispatcher", "db_url",DBURL)
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "setid_col", "setid")
modparam("dispatcher", "destination_col", "destination")
modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "flags", 3)
modparam("dispatcher", "dst_avp", "$avp(i:271)")
modparam("dispatcher", "grp_avp", "$avp(i:272)")
modparam("dispatcher", "cnt_avp", "$avp(i:273)")
modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.0.2")
modparam("dispatcher", "ds_ping_interval",15)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_ping_reply_codes",
"class=2;code=403;code=404;code=484;class=3")
How to disable this cache?
Hi list,
Is there a way to survey the message buffer (iei the queue where arriving
sip messages wait to be processed by worker threads) in Kamailio?
Is there a way to check the number of messages in this buffer?
Additionally, is it possible to survey the number of messages by method
(INVITE, BYE, etc.)?
I know that the ratelimit module (
http://kamailio.org/docs/modules/stable/modules/ratelimit.html) shall
access such buffers, but I do not see any variable accessible in the config
offered by that module.
Am I missing sth or is there a way to do so?
Thanks
Alireza
Hello everybody,
I am new on this environment ( Kamailio). I try to install(settle)
Kamailio 4.1.4, on an O.S Suse Sles11 SP3, all the compilation took
place, with the exception of the installation (make install), where I
obtain the following error:
make[2]: Nothing to be done for `install-if-newer'.
touch /usr/local/lib64/kamailio/modules/userblacklist.so
install -m 755 userblacklist.so /usr/local/lib64/kamailio/modules
make[2]: `libkmi.so.1.0' is up to date.
make[2]: `libsrdb1.so.1.0' is up to date.
make[2]: `libkcore.so.1.0' is up to date.
make[2]: `libsrutils.so.1.0' is up to date.
LD (gcc) [M usrloc.so] usrloc.so
make[2]: Nothing to be done for `install-if-newer'.
make[2]: Nothing to be done for `install-if-newer'.
make[2]: Nothing to be done for `install-if-newer'.
make[2]: Nothing to be done for `install-if-newer'.
touch /usr/local/lib64/kamailio/modules/usrloc.so
install -m 755 usrloc.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xhttp.so
install -m 755 xhttp.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so
install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xlog.so
install -m 755 xlog.so /usr/local/lib64/kamailio/modules
touch /usr/local/lib64/kamailio/modules/xprint.so
install -m 755 xprint.so /usr/local/lib64/kamailio/modules
/bin/sh: -c: line 18: syntax error near unexpected token `fi'
*/bin/sh: -c: line 18: ` fi ; \'
make: *** [install-cfg] Error 2 *
Could you help me please to solve this error,
Best Regards,
Djamel.
I have this config:
WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio -> WebRTC sipml5
When placing a communication between two sipml5 points I get this errors:
ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su:
could not resolve hostname: "df7jal23ls0d.invalid"
ERROR: <core> [forward.c:532]: forward_request(): ERROR: forward_request:
bad host name df7jal23ls0d.invalid, dropping packet
ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used:
Unresolvable destination (478/SL)
If I place a comm between WebRTC sipml5 -> Kamailio -> Asterisk ->
Softphone (connected to Asterisk)
Everything works fine
I'm running RTPEngine as RTP Proxy for DTLS/SRTP issues
SIP INVITE signalling Ringing 180 message and 200 OK gets done fine
Any ideas where this issue might come from? Looks something related with
media traffic (RTP).
Kind regards!