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37545 discussions

How install Kamailio with Asterisk
by ANTHONY HESNAUX 22 Jul '14

22 Jul '14

hello all, I would install kamailio with ASTERISK in realtime on CENTOS 6.5 x64. I followed the tutorial aspito: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but its not work. I have a problem with kamailio.cfg Its not possible to start kamailio When I loadmodule "uac.so" I want to make this configuration: SIP phone1 -> Register Kamailio -> ASTERISK Register SIP phone 2 -> Register Kamailio -> ASTERISK Register kamailio: 192.168.0.100 Port: 5060 ASTERISK: 192.168.0.100 Port 5080

3 6

Unknown caller gets online user's identity
by g.aloitus＠gmail.com 21 Jul '14

21 Jul '14

Hello, I'm using Kamailio version 4.1.4+precise (amd64). I have followed "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main difference in my setup compared to that one is that I continued use of Kamailio's database. The problem is as follows: I decided to put Kamailio and through it Asterisk reachable from internet. I have tried to configure Asterisk so that only calls of registered users would be possible, and they could only call to other registered users or conference rooms and echo test number. Then I took the following steps: I ensured that there was no online users with kamctl online. Then I launched MicroSIP (www.microsip.org), but I did not defined account, I simply set the protocol to tls and media encryption to mandatory, because I'm using these. I called to extension with xxx(a)my.public.ip.address (where xxx is extension) getting "unauthorized". And that was what I wanted. But if there is online users, calls go through, and incoming call is coming from Asterisk (in syslog I can find out that src_user=asterisk). Kamailio and Asterisk are listening the same IP address, but different port. I have refused connections to the Asterisk's port with iptables. I have defined my public IP address as domain in sip.conf. There is also other domain defined which corresponds to users' domain I am using in Kamailio's database. In kamailio.cfg there is if statement which prevents Kamailio not to be open relay: if (from_uri!=myself && uri!=myself) ... If I change this for example: if (from_uri!=myself || uri!=myself) I get what I want this time: no calls from outside, but I somewhat think that this is not a final solution. I have not found from log files such information which would have helped me. I have not yet investigated this problem so much that I could tell the logic behind the selection of online user's identity which is used. However, if I make a call to conference room I notice that Asterisk is thinking that one of online users has joined the conference. If I can recall correctly, I started with Kamailio version 3.2, and integrated it with Asterisk 11 (currently 11.10.2). Is there something which has changed in Kamailio, but what I have not changed in my setup which could explain this. Best, Teijo

6 18

kamcmd htable access problem, probably with int->string conversion
by Moritz Graf 21 Jul '14

21 Jul '14

Hi There, i implemented a mapping via htable. For debugging purpose I use kamcmd to see if the values are correctly filled. Doing so I discovered that there is a problem getting the values of the keys, where the name is a integer value. So it's the following behaviour: ##################################################################### root@kamailio-stage:~# kamcmd htable.dump maptodest { entry: 55 size: 1 slot: { item: { name: 1 value: 0000000000000000001 } } } { entry: 504 size: 1 slot: { item: { name: blub value: 0000000000000000001 } } } { entry: 686 size: 1 slot: { item: { name: 001 value: 0000000000000000001 } } } root@kamailio-stage:~# kamcmd htable.get maptodest blub { item: { name: blub value: 0000000000000000001 } } root@kamailio-stage:~# kamcmd htable.get maptodest 001 error: 500 - Not enough parameters (htable name and key name) ##################################################################### Is there a reasonable explanation why it works with blub, but not with 001? Why is he complaining about the number of parameters? Does anyone know a workarround? Thanks for any advise!! greeetz Moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg

2 3

pkg.stats
by Vitaliy Aleksandrov 21 Jul '14

21 Jul '14

Hi list, I have a problem with pkg.stats for tcp.main process on kamailio "4.1.2 73ea61-dirty". kamailio has 64M of private memory for each process (-M 64). pkg.stats for tcp.main process shows that is uses 75M and still has 62M free: { entry: 36 pid: 10032 rank: -4 used: 73109720 free: 65891192 real_used: 79146048 } root@webRTC:~# ps -p 10032 -o pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args PID TID S %CPU RSS SIZE SZ VSZ COMMAND 10032 10032 S 0 26420 67508 76821 307284 /.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64 -u kamailio -g kamailio This server handles 1000+ tcp/tls connections and works just fine for the last two months. That's why i think this is not a memory leak and just a pkg.stats problem. Does anybody else saw this or maybe somebody is already working to fix it ?

2 3

Msilo not delivering message (status 408)
by Peter Villeneuve 21 Jul '14

21 Jul '14

Hi, This is most likely a stupid mistake on my route config, but msilo correctly stores messages for offline users, but when they come back online, the message never arrives. Looking through the debug logs indeed there is an error, with status 408, which suggests kamaiio can't find a route to the user that just logged. Before I spend hours looking through all the routing blocks, perhaps someone more exprienced in kamailio can see the problem right away and offer a solution? Thanks DEBUG: msilo [msilo.c:1148]: m_dump(): msg [1-12] for: sip:1010@my.domain.com DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e3040, 9, -1, 0x10) fd_no=2 called DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f61ced1b1b0, state 1, fd=9, id=1 DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f61ced1c4f8 DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f61ced1b1b0, 1 from 0 DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89ec40, 26, 2, 0x7f61ced1b1b0), fd_no=21 DEBUG: <core> [tcp_main.c:3515]: handle_tcp_child(): handle_tcp_child: CONN_RELEASE 0x7f61ced1b1b0 refcnt= 1 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced1d868, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): *completed with status 408* [mid: 140058058335472/13] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <13> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:13 fl:8 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced5f098, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): completed with status 408 [mid: 140058058329184/12] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <12> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:12 fl:8

4 19

please help with Msilo config
by Miguel Rios 19 Jul '14

19 Jul '14

Hi list, I'm a newbie when it comes to kamailio, although I have a fair amount of SIP experience. I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with Msilo support. I just used the default kamailio.cfg file (changing obviously the relevant parameters for my setup) and have not touched the routing blocks. I'm very confused about how routing works, and I don't have a background in C. I've read the wiki up and down and have looked at inumerous tutorials spread out on the internet (most of which just add to my confusion because they seem out of date), but I still don't get it. I managed to setup a working kamailio server where local users can call each other fine. Now when I try to add the msilo module, and specially the routing example from MSILO Module is when things get tricky. I'm sure my error is a fairly basic one and has to do with the routing logic and syntax. Could some kind soul please share a copy of the whole ####### Routing Logic ######## on downwards for a basic no frills kamailio with msilo setup? Thanks, Miguel MSILO Module 3. Parameters 3.1. db_url (string) Database URL. Default value is “mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the “db_url” parameter View on kamailio.org Preview by Yahoo

3 3

Fwd: Msilo not delivering message (status 408)
by Peter Villeneuve 18 Jul '14

18 Jul '14

Thanks Daniel. UDP is the same story as TCP and TLS, unfortunately. Looking through the logs, I see strange errors in parsing the contact header when msilo prepares to send the notification back to 1000. This is for the notification message that should have been sent back to 1000 after he sent a message to 1010 (offline). I also see that kamailio reports that 1000's location was not found, even though he's obviously registered. The more I play around with this, the more confused I become. Do you think this contact parsing error could be the root of the problems? Cheers, Peter Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:1000@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK90dc.68116a97000000000000000000000000.0>; state=16 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=10 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:1000@my.domain.com] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [< sip:1000@my.domain.com>#015#012] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=25 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-bb2e Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: *ERROR: <core> [parser/contact/contact.c:175]: skip_name(): skip_name(): Error in contact, scheme separator not found* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/contact.c:202]: parse_contacts(): parse_contacts(): Error while skipping name part Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: nathelper [nhelpr_funcs.c:300]: get_contact_uri(): failed to parse Contact body Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: !------------------------------------>WE HAVE A NEW MESSAGE Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: registrar [lookup.c:181]: *lookup(): contact for [1000] found by address* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [79.my.server.IP1] == [127.0.0.1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==14 && [79.my.server.IP1] == [79.my.server.IP1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: ------------------MESSAGE PROCESSING: *location not found*, storing offline message ID: (sip:1000@85.xx.xx.xx:59287;transport=TLS) Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:605]: m_store(): ------------ start ------------ Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:648]: m_store(): using user id [sip:1000@85.xx.xx.xx:59287;transport=TLS] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:732]: m_store(): 'From' header: <sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:849]: m_store(): message stored. T:<sip:1000@my.domain.com> F:< sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:858]: m_store(): sending info message. Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:899]: m_store(): contact header found On Thu, Jul 17, 2014 at 11:01 PM, Daniel-Constantin Mierla < miconda(a)gmail.com> wrote: > You should not call m_store() if src_ip==myself -- in this way you avoid > to store notification messages that cannot be delivered. > > Why notifications are not delivered has to be troubleshooted there. Have > you tried also with UDP? > > Cheers, > Daniel > >

1 0

[OT] LVUG, the London VoIP User Group
by Richard Brady 18 Jul '14

18 Jul '14

Hi folks LVUG is a monthly get-together in London to discuss open-source VoIP technology, including Kamailio. If you're based in or near London please join us this coming Tuesday for the third LVUG which will be held in Old Street. The format is a couple short talks followed by beers at the venue or in a nearby pub. Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/ Hope to see you there! Richard -- Richard Brady E: rnbrady(a)gmail.com

1 0

Kamailio presence (NAT, XCAP)
by Radu Vatav 18 Jul '14

18 Jul '14

Hello, I'm just trying to use SIP as a replacement for Skype, nothing more than that. All is working except of presence. Using Jitsi 2.4.4997 (Windows x64) from home behind NAT and kamailio 4.1.4+wheezy on a rented server somewhere. My biggest problem is that I don't know exactly how it is supposed to work. Ok, this is what I understood so far: the clients connect via SIP to the server and can call each other, but for some reason the server can't notify them via SIP of presence changes when they are behind NAT. So the clients also connect to the built in xcap-server, register themselves as presence "watchers" and the xcap-server notifies them via udp/tcp of presence changes of the other users which are watched. All is very nice but it's not working and I don't know where to look for the problem. Which are exactly the steps in communication and what data exactly is there sent which way (when working correctly, which it doesn't) - no clue. I somehow put the configuration together from the 2-3 examples I found on the internet, I don't know if it is ok to attach something to this email, so here it is: https://vatav.cc/kam/kam_cfg.7z Here is the network activity on the server: https://vatav.cc/kam/kam_ngrep.7z So me ("radu") and "alice" are home behind the NAT, one is online and wants to see how the other one goes online and offline. I don't know what exactly to look for, but it doesn't look to me like there is anything sent to one user when the other one goes on-/off-line (there is a lot of "proxy authentication required" which drives me nuts, again I don't know if this is correct or not; Jitsi seems to be connected to the XCAP server, if I type in a wrong address for "XCAP server" Jitsi complains it can't connect). Some other things: - "kamctl online" shows almost real-time who is logged-in, so this works - siremis shows a bunch of users in "watchers list", but none in "active watchers list" - actually when a user goes online it contains the contacts of that user, but only briefly, when I refresh the view, the list is empty again. I don't know if this is how it is supposed to work, anyway, Jitsi doesn't show any change of the status of the user who changed status at all. Any help would be appreciated. It's sad something this simple as I need (basic Skype replacement) is so difficult to get working. Maybe this is why there is relatively low interest in SIP and everybody stays with MS/Google/... because "there isn't something else". Best regards, Radu Vatav

2 1

db_postgres Error
by Rene Montilva 18 Jul '14

18 Jul '14

Hi list i'm config SQLOps Module with postgresql db module or is postgresql driver? this is my kamailio config: #!ifndef DBURL #!define DBURL "postgres://user:ouser@x.x.x.x/kamailio" #!endif sql_query("ca", "select * from xxxxx","ra"); if($dbr(ra=>rows)> 0){ xlog("L_INFO", "saldo:$dbr(ra=>[0,0]) y seg:$dbr(ra=>[0,1]) "); $var(credit) = $dbr(ra=>[0,0]); # 30$ of credit $var(cost_per_sec) = $dbr(ra=>[0,1]); # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second sql_result_free ("ra"); if($var(cost_per_sec) > $var(credit)) { sl_send_reply("402","payment required"); exit; } } i got this warning and would like why this happen: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [SSL SYSCALL error: Connection timed out#012] WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [SSL SYSCALL error: EOF detected#012]

2 1

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