I am installing KAMAILIO under DEBIAN. I have got it to work fine (TLS) until I upgraded my Comcast to business service to get static IP’s. Now it no longer works.
I am using the Cisco DPC3941 modem/router and am tring to connect an IPhone on WIFI to the PC SIP server on the LAN. Here are the log entries:
Feb 16 19:14:59 SIP-Server kamailio[2010]: NOTICE: <core> [main.c:738]: handle_sigs(): Thank you for flying kamailio!!!
Feb 16 19:14:59 SIP-Server kamailio[2019]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received
Feb 16 20:01:25 SIP-Server kamailio: INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression...
Feb 16 20:01:26 SIP-Server kamailio: INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected)
Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob
Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available
Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024
Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [init_socks.c:126]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2]
Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [ctl.c:283]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed
Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ctl (/usr/lib/x86_64-linux-gnu/kamailio/modules/ctl.so)
I am informed that SIP-ALG is disabled. Any ideas?
Hi ,
I am working kamailio 4.4.0 ,i want to remove my Contact header
completely.So I used remove_hf..but its now working ..Let me know the
solution
Am very thankful to you
Regards
Jefff
Hi there,
New user to Kamailio here. We currently have it up and running in a virtualized environment with 1 Kamailio sever, 1 Asterisk server and 1 MySQL server.
I'm currently writing install scripts to make deploying new nodes/servers easy and to keep settings the same across the board. I've chosed to load the db_cluster.so module in kamailio.cfg, as we will have 2x MySQL servers in master-master replication which will contain the 'kamailio' and 'asterisk' tables.
I've just hit a stumbling block - in `kamctlrc`, there is a field called `DBHOST=`. How can I reference my cluster here?
In kamailio.cfg, I simply define DBURL as "cluster//<clustername>". What is the syntax for 'DBHOST=' in 'kamctlrc'? Can I reference the cluster? Can I have 2 separate DBHOST= lines?
Looking for some guidance on this one.
Thanks,
Derek B.
I see there is a new SQLops function sql_query_async that has been added in 4.2 (I believe?).
Is there a good way to perform database insert using version 4.0?
Using mysql
Thanks!
-dan
Hi All,
I want to validate a IP address in kamailio for requests coming from
different IPs. I have browsed and found permissions module, however is
there a way where i can directly compare a source IP address to an address
stored in an avp and get a true or false value.
something like this
$si==192.168.2.0/24
Thanking You,
Sunil More
Hi All.
I need to use SIP-I with my upstream to set A number as unknown.
I was read about adding incapsulation ISUP using
if(has_body("application/sdp"))
{
set_body_multipart();
msg_apply_changes()
$var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c";
append_body_part("$var(acm)","application/isup;version=itu-t92+","signal;
handling=optional");
msg_apply_changes()
xlog("L_INFO", "ISUP Changes Applied Succesfully");
}
It is converts body to multipart and inserts ISUP
But it inserts is as text.
Is there any possibility to insert basic ISUP message to update it after using
sipt_destination($rU, 31, 4);
and
sipt_set_calling($fU, 4, 0, 3);
Thank you.
--
Best regards,
Sergey Basov e-mail: sergey.v.basov(a)gmail.com
this one (written by Daniel) http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
maybe a bit outdated but still consistent ?
the thing i am really stuck with (and concerning real-time) is that none of my extensions (from asterisk CLI) are online:
ns3325046*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime
102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT
103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
that drives asterisk crazy ! and logger reports: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) every time i place a call.
The tutorial written by daniel mention a channel configuration pretty minimal:
INSERT INTO sipusers (name, defaultuser, host, sippasswd, fromuser, fromdomain, mailbox)
VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102');
and since there's no context associated to the 102 extension i cant figure out where that channel enter the dialplan ? [public] [LocalSet] [default] ????
and a dialplan
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup
i am sorry to bother you with issues more asterisk oriented than kamailio.
By the way i took a good start with kamailio as it seems to work flawlessly on my system.
thx you.
On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote:
> Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration.
Which tutorial?
> the only thing i missed is asterisk behaviors'r regarding sip registration ?
That was a part of a tutorial I once saw. In essence asterisk uses the
kamailio database, UA registers on kamailio and is stored there,
asterisk sees the same data (realtime).
Sébastien BRICE VoIP, Support et Intégration
Hello. I am seeing the following error repeating again and again in the log
file:
/sbin/kamailio[14510]: ERROR: registrar [reply.c:199]: build_contact(): no
pkg memory left
I am using 64MB of SHM memory and 8MB of PKG memory.
What is the reason for this error? How to solve it?
Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration.
the only thing i missed is asterisk behaviors'r regarding sip registration ?
I tryed to place a call between 102 and 103 extensions and experimenting an issue
Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail !
-- Executing [103@public:1] Dial("SIP/102-00000001", "SIP/103") in new stack
[Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
ns3325046*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime
102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT
103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
apart that my sip.conf and extensions.conf are very minimal:
[LocalSets]
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,102,Voicemail(${EXTEN},b)
exten => _1XX,103,Hangup
[general]
context=LocalSets ; Default context for incoming calls. Defaults to 'default'
rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
i did INSERT TO the users in mysql tables (sipusers, sipregs and voicemail) and registering extensions from UA works ok (i am using jitsi)
Whats wrong with my setup ?
thank you
