Any recommendation web interface for gathering statistics other than SIREMIS? I installed SIREMIS but unable to resolve login problem and my request for help on ASIPTO didn't get any response also unable to find siremis mailing list. Your help is greatly appreciated.
Thanks
Abdulmailk
hello folks,
i'm really new with kamailio (a couple fo days) and I need some
suggestion/samples/idea/HELP to realize my environment. I've been starting
to read the docs on website but there are a lot and i don't know where
start and which kind of modules need to use
I''d like to realize a trunk from kamailio and asterisk. Each one has their
own sip users. kamailio sip users are some inside network and some outside
network (behind nat)
Could someone be son kind to show me the right way to configure kamailio
for incoming/ougoing calls from/to asterisk and for using sip peers (behind
nat and not)?. Also same samples are really preciated.
Thanks a lot in advance
Kind regards
Lorenzo
--
"Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non
conta un cazzo, 1941 ... sono anche un autore"
Hi guys,
Is there a code somewhere in the project i can reuse to implement
milliseconds on $Ts? for example introduce a "$Tms" or a "$TmS" to refer to
a pv that stores milliseconds instead of seconds?
Kelvin Chua
rr module has check_route_param(re) function that can be used to check
if local route header has a param that matches re. however, there does
not exist a function to find out what is the value of a given rr param.
for example, if route header has param ;param=foo, how to extract value
foo of param?
there is @rr.uri.params select that together with {param.value,name}
transformation might be able to do the job?
-- juha
On the previous issue (disjoint networks), I eventually settled for binding Asterisk on localhost:5080, and using rtpproxy to route media to the networks as required. The only snag is that I had to write a script to rewrite the Kamailio configuration in
order to take current IPs into account.
Now, for the same project. What I want now is to be able to express multiple domains with an Asterisk backend. That is: given pbx.company1.com and pbx.company2.com, two DNS addresses that point to the same server, I want to have separate accounts
bob(a)pbx.company1.com and bob(a)pbx.company2.com, with separate registration and accounting, without collisions.
I know about the Kamailio "domain" module, and I have enabled WITH_MULTIDOMAIN and filled the kamailio.domain table for it to work. Now the authentication is taking domains into account.
Next, I want to express two non-colliding registrations for Asterisk. What comes to mind is username rewrite - Kamailio sees bob(a)company1.com, and forwards a REGISTER to Asterisk using bob_company1_com as the SIP username. I am looking right now at the
WITH_ASTERISK block that uses the uac module:
# Forward REGISTER to Asterisk
route[REGFWD] {
if(!is_method("REGISTER"))
{
return;
}
$var(rip) = $sel(cfg_get.asterisk.bindip);
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport);
$uac_req(furi)="sip:" + $au + "@" + $var(rip) + ":5060";
$uac_req(turi)="sip:" + $au + "@" + $var(rip) + ":5060";
$uac_req(hdrs)="Contact: <sip:" + $au + "@"
+ $sel(cfg_get.kamailio.bindip)
+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n";
else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";
uac_req_send();
}
#!endif
The above is the non-encoding version. What is the best way to modify it to do a domain encoding in the username? Or, if a better solution exists, what is it?
Hi,
Has anybody seen this :
Mar 2 17:25:16 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2011]: generate_res_buf_from_sip_res(): out of mem
Mar 2 17:25:16 kamailio kam-pcscf[20741]: ERROR: tm [t_reply.c:1943]: relay_reply(): ERROR: relay_reply: no mem for outbound reply buffer
followed by:
Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2011]: generate_res_buf_from_sip_res(): out of mem
Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: tm [t_reply.c:1943]: relay_reply(): ERROR: relay_reply: no mem for outbound reply buffer
Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2168]: build_res_buf_from_sip_req(): ERROR: build_res_buf_from_sip_req: out of memory ; needs 386
Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/contact.c:194]: parse_contacts(): parse_contacts(): No memory left
Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts
Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing
However, seems like we still had enough mem:
[root@kamailio kamailio]# vmstat 1
procs -----------memory---------- ---swap-- -----io---- --system-- -----cpu-----
r b swpd free buff cache si so bi bo in cs us sy id wa st
3 0 0 9526348 230308 13282472 0 0 0 4 0 1 6 0 94 0 0
1 0 0 9526216 230308 13282524 0 0 0 0 1313 84424 17 8 75 0 0
1 0 0 9526092 230308 13282616 0 0 0 0 1482 86882 16 9 75 0 0
[root@kamailio monit.d]# kamailio -v
version: kamailio 4.2.0-dev0 (x86_64/linux)
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 17:42:42 Dec 18 2013 with gcc 4.4.7
Unfortunately I had to reload p-cscf to get service working again.
--
Daniel Ciprus
Integration engineer
http://www.acision.com
9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com>
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Hello,
I am using kamailio 4.1.1, I am wondering if kamailio supports call forwarding , three-way call, and call transfer natively? Does it expect the endpoints to do the RTP mixing?
Thanks
Absul
Hi,
I´m having problems with kamailio as LCR .
I have a route with 14 different gateways, each one with priority 1 and all
with the same weight.
If one gateway is full, busy, down or answers with any failed response to
LCR , I jump to the next gateway in the list. I´m presenting the problem
that its stops jumping after 12 gateways and returns the message 500
Forking capacity exceeded (12/SL). I need to increase this limit , so all
the gateways in the list give the call a try.
Regards,
Camila
*Camila Troncoso Solar **|* Ingeniero de Desarollo
+56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net
Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com
<http://www.redvoiss.com>*