Hello,
I want to run an SQL UPDATE command from my kamailio script. I am using
the command 'avp_db_query'.
How can I get the number of affected rows after the UPDATE runs?
David
hi
I have set up tls on kamailio successfully, but when I relay a TLS
client to an other proxy ($ru = "sip:" + $rU + "@" +
"127.0.0.1:5070;transport=udp";) via udp I get error messages.
>From the timing it seems like this warning always appears after I get
200OK from proxy, the Invite gets sent correctly, but kamailio seems
to fail while sending back the ACK.
18(11231) WARNING: <core> [forward.c:270]: WARNING: get_send_socket:
protocol/port mismatch (forced udp:77.180.86.63:5060, to
tls:127.0.0.1:5070)
TCP->UDP works.
TCP->TCP works.
UDP->UDP works.
UDP->TCP works.
TLS->TCP works.
TLS->UDP doesn't work.
Hello,
can someone please give me some pointers as to how to configure the proxy server to receive incoming calls from a PSTN line. I had not problem setting up extensions and making outgoing PSTN calls. I am not able to find any documents ( or I am not looking in the right place ) as to how to configure the system to receive incoming PSTN calls. The DID number I have from my provider gives me the 10 digit DID number and the IP address of their server. Thank you for the help.
Regards,
Arun
rtpproxy_offer and answer functions have "r" flag described as follows:
r - flags that IP address in SDP should be trusted. Without this flag,
rtpproxy ignores address in the SDP and uses source address of the SIP
message as media address which is passed to the RTP proxy.
how does rtpproxy "use" this ip address?
if sip message comes from behind nat, ip address in sdp is local
address, not the address where rtp packets come from. and if this
request has passed another (e.g. outbound) proxy before hitting the
current one, also source address of sip message is not the address where
rtp packets come from. either address thus seems to be useless for
rtpproxy.
-- juha
Hello everyone!
Recently I became interested in concept of putting couple of Asterisks
servers in cluster. So I tried to replicate this scenario in testing
environment. I was using database back-end for Asterisks settings and
statistics and Kamailio as a load balancer.
Obviously it did not worked [image: :cry:] . There is possible to make so
many mistakes. Now I'm trying to find some step by step tutorial, but
nothing yet.
If someone experienced this, please advice. And if someone have useful
links, please share.
Thanks a lot.
Hi,
Just a quick question, will the LCR module provide sufficient failover capacity? I have used the dispatcher module and like the way it marks a peer as down if it can't be contacted however I need to utilise the LCR functionality.
Maybe a combination of the both will be best?
Thanks,
Keith
Hi list,
I have a loadbalancer before sipproxies. I call t_reley() on the dispatcher to forward requests to destinations.
All goes well under low load.
But when load goes up to 500 requests/s, the loadbalancer still sends requests via only one tcp connection. This maxes out the tcp write queue.
Of course I can increase the 'tcp_conn_wq_max' value but it doesn't solve the root problem.
How do I make the loadbalancer start a new tcp connection to the destination when existing one becomes too busy? (Is it possible?)
Regards,
Allen
Hi
I am fine with registered users sending and receiving calls from/to PSTN.
Now I route registered users calls to/from PSTN fine
Kamailailio<----->freeswitch<------------>IVR (PSTN)
Best Practice for routing Trusted peers (inbound) to IVR/PSTN (SIP Trunking
Service)
Trusted peer1 -----> Kamailio -----> freeswitch ------------> IVR (PSTN)
Where do I/ how do I route calls from trusted peers not to users but down to
PSTN IVR?
Say for example Trusted Peer 1 sends to my Kamailio IP
sip:398998xxxx@xx.xx.xx.xx:5060
As I add a Trusted peer I would like to add a route and insert prefix for
when it routes to freeswitch --->
Do I use carrier route or LCR etc, I want to be able to add trusted peers
but add routes without editing the config file and restarting Kamailio.
The reason I ask is I'd prefer to do it correct method at the start as it's
a nightmare changing once I have lots of trusted peers.
Thank you
Tony Turner
Hi Klaus,
Thx for your suggestion. The workaround is working fine with
Pseudo-Variable $rb instead of $mb ("$var(sdp) = $(rb{s.striptail,2});")
However, the broken body is generated by Kamailio with the
filter_body() function. The server receives a multipart/mixed body and is
filtering the SDP part of it with following config (excerpt):
if (has_body("multipart/mixed")) {
if (filter_body("application/sdp")) {
remove_hf("Mime-Version");
remove_hf("Content-Type");
append_hf("Content-Type: application/sdp\r\n")
}
}
After filtering the SDP message, the message body terminates with double
CRLF, which was "space" between the individual parts / to the
uniqueBoundary. I tried before using "subst" with kamailio-like
parameters for removing the blank line:
if (subst('/(a=[a-z]+:.+)[\r\n]{4}$/\1\r\n/s')) {
and was wondering, why this does not work. The "search_body()" function
accepted the match pattern and so it could not be wrong at all:
if (search_body("a=[a-z]+:.+[\r\n]{4}$")) {
Should this work with Kamailio?
Regards,
Klaus
P.S. the body of the original SIP message looks as follows:
[...]
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: <length>
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 9550 3496 IN IP4 10.16.48.44
s=SIP Call
c=IN IP4 10.16.48.44
t=0 0
m=audio 16386 RTP/AVP 8 19
c=IN IP4 10.16.48.44
a=rtpmap:8 PCMA/8000
a=rtpmap:19 CN/8000
a=ptime:20
--uniqueBoundary
Content-Type: application/x-q931
Content-Disposition: signal;handling=optional
Content-Length: 48
0802037C052104030010231803210303
6C0C2103313730343530353335317D02
11017E0D0A2E0D0765007322143504
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional
IAM,
PRN,isdn*,,NET5*,
USI,rate,c,s,c,1
USI,lay1,alaw
TMR,00
CGN,04,,1,y,4,1704505351
UUS,0,1d4275050014804185000031362a640062f21001f62fbf
CPC,09
FCI,,,,,,,y,
GCI,67f9f408a73011e280056c205642ae68
--uniqueBoundary
-------- Original-Nachricht --------
Betreff:
Re: [SR-Users] 'subst' request for message body (SDP)
Datum:
Fri, 05 Jul 2013 17:01:56 +0200
Von:
Klaus Darilion
<klaus.mailinglists(a)pernau.at><mailto:klaus.mailinglists@pernau.at>
An:
Kamailio (SER) - Users Mailing List
<sr-users(a)lists.sip-router.org><mailto:sr-users@lists.sip-router.org>
Kopie (CC):
Klaus Feichtinger <klaus.lists(a)inode.at><mailto:klaus.lists@inode.at>
Is this broken body generated by Kamailio, or by some other entity?
If it is done by Kamailio with filter_body(), it should be fixed.
If done by some other entity, it should be fixed in the broken entity. I
have no idea why subst does not work, but as workaround you can try
something like:
$var(sdp) = $(mb{s.striptail,2});
set_body("$var(sdp)", "application/sdp");
regards
Klaus
On 05.07.2013 13:05, Klaus Feichtinger wrote:
> Hello,
>
> can anybody give me a hint, how I could delete the _empty_ (blank)
> last
> line of the message body / SDP (it consists of #012#015 only)? This
> blank
> line is the rest of the original MIME message, which was reduced to a
> standard message with content SDP only by kamailio-3.3.4, which is
> principally working fine.
>
> I´ve tried using the textops functions "subst" and "subst_body", but
> they
> do not behave as like as real SED. When I try this string manipulation
> with SED, it is working fine with [sed '${/^$/d}']. But the subst
> implementation does not support the additional parameters '${' / '}'.
>
> Does anybody know how to delete the last line of the whole message?
>
> regards,
> Klaus
>
> P.S. the end of the message looks like this:
>
> [...]#015#012a=rtpmap:8 PCMA/8000#015#012a=ptime:20#015#012#015#012
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> list
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