- sr-users - kamailio.org

don't fragment flag in rtpengine request
by Juha Heinanen 27 Jun '15

27 Jun '15

i was wondering why sometimes requests to rtpengine don't get through and icmp unreachable destination error message is received. it turned out that in these error cases ip packet containing the rtpengine request has 0x02 (don't fragment) flag set even when the packet was bigger than 4 KB. in this case kamailio was running in an amazon container which may be doing something weird. is it somehow possible from the application (= kamailio) to control setting of that flag? -- juha

1 0

deb building system
by Victor Seva 26 Jun '15

26 Jun '15

Hello, I would like to inform that the deb nightly builds are going to be disabled. I'm having problems with the actual configuration and until I found the time to fix them the service will be out of order. No new debs will be produced until further notice. Cheers, Victor

2 1

limit calls to incoming only?
by Peter Villeneuve 26 Jun '15

26 Jun '15

Hi, Is it possible for Kamailio to restrict certain users (like in a group) to only receive calls but never dial out or send SIP Simple messages outbound, while allowing all other "regular" users to make outgoing calls and text messages as well as receive them? If so, which module do you guys recommend I look into? Cheers, Peter

2 3

fix_nated_contact NAT rtpproxy problems
by Vik Killa 25 Jun '15

25 Jun '15

Hello, We are trying to get Kamailio working as an SBC using rtpproxy(); The config seems to work except with some carriers, kamailio does not recognize the ACK because it does not match the contact header of the OK I've been trying to figure out how to get kamailio to send the OK back using the public IP address but I cannot figure out how. I've tried using various deployments of the nathelper module such as fix_nated_contact() set_contact_alias() No matter what though, Kamailio will not change the contact header it receives from FS before sending it back to the carriers. This has become frustrating as I read tutorials that showed this is possible, yet I am unable. Thank you. /V

1 0

"00" prefix record problem on ACC
by Yasin CANER 25 Jun '15

25 Jun '15

3 2

linking math.h while compiling a module
by Shouvanik Chakrabarti 25 Jun '15

25 Jun '15

Hello, I have developed a Kamailio module to handle group messaging and am using certain functions from the libm.so library. Compilation happens fine, but Kamailio subsequently fails to start because the math.h symbols are unlinked/undefined. Could someone please suggest a workaround for this. Thanks, Shouvanik.

2 4

How to disable retransmits via TCP connection? etc.
by Andrey Utkin 25 Jun '15

25 Jun '15

Hi! We meet such an issue with some kinds of client internet connections (e.g. reproduces always on local wifi, but doesn't repr. on 3g inet). We don't know what are exact network characteristics, but what happens from perspective of traffic on SIP server is the following: 1. the SIP client registers successfully to Kamailio (which is recent git master HEAD) with TCP transport for SIP; 2. another client calls this user; 3. Kamailio relays INVITE, it gets transmitted fully and ACK-ed on TCP level; 4. then, within 0.1-0.2 second, Kamailio fires first retransmit of that INVITE, which doesn't get fully transmitted (it is shown trunkated in sniffer output, no TCP ACK replies for it); 5. Kamailio tries to retransmit INVITE again, with the same result as in #5. 6. The caller gets SIP/2.0 408 Request Timeout. A piece of traffic from "ngrep -t -e -d any -W byline port 5060": https://gist.githubusercontent.com/krieger-od/219f9975e5efb980ff5b/raw/096a… The called side app is based on mobile Linphone app. Switching to UDP is not an option (in some networks SIP messages get delivered trunkated which breaks calls), we will check how it works with TLS transport a bit later when there's technical possibility. I have two questions: 1. How would I completely disable retransmissions to TCP connections? 2. Any ideas what can be the reason for this issue? Retransmissions by themselves? -- Andrey Utkin

4 11

Kamailio Server Maintenance - Wed-Thu Night
by Daniel-Constantin Mierla 25 Jun '15

25 Jun '15

Hello, during the night between Wed (June 24) and Thu (June 25), planned to start not early than 00:00 GMT+1, there will be some schedule maintenance work to the infrastructure that is hosting some of the kamailio.org servers. The main affected services will be: - main website (www.kamailio.org) - wiki portals - mailing lists (lists.sip-router.org) - git mirror (git.kamailio.org) It is expected to have short downtimes of few minutes. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio + RTPEngine + Asterisk
by Javier Aristizábal 24 Jun '15

24 Jun '15

Hello folks, I am trying to configure this on the same server: Kamailio v4.2 port 5060 RTPEngine v2.3 port 22222 Asterisk v11.18.0 port 5080 I am follow the Asipto tutorials, for the Kamailio and the real time configuration. Now, I can talk between extension and PSTN without issues. But the trouble is with the Status registration with Asterisk. And I need that Status OK because I am using queues in asterisk for call centre. I got this from with: sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101 (Unspecified) D Yes Yes 0 UNREACHABLE Cached RT the status is UNREACHABLE and the asterisk table sipregs won't update after each registration. To be sure, I tried first all config only with asterisk to check that is not a problem with MySQL or something. On my cfg file I have this for the asterisk registration: 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); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $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(); } Any clue? Thanks! -- Javier Aristizábal

2 16

Kamailio proxy at Amazon EC2 (behind NAT)
by Alexandru Covalschi 24 Jun '15

24 Jun '15

Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 15

Kannot start kamailio 4.3
by Yuriy Gorlichenko 23 Jun '15

23 Jun '15

Try to start kamailio on Ubuntu 14.04.02 Get this errors ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) my params for ctl # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 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, F_MALLOC, DBG_F_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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 1dbd53 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4

4 9

kamailio fails after start
by Yuriy Gorlichenko 22 Jun '15

22 Jun '15

Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual systemOpenVZ. after starting kamailio I see that it runs ok with kamailio start or kamctl start at ps -ax I see working processes. But after one minute of working kamailio fails with kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe I see nothing more at debug file. No errors. (I run with debug=4) my cfg parameters at start mhomed=1 memdbg=5 memlog=5 fork=yes children=6 version: kamailio 4.2.4 (x86_64/linux) ac3682 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, F_MALLOC, DBG_F_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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: ac3682 compiled on 04:27:04 Apr 10 2015 with gcc 4.8.2

2 2

Kamailio transformation
by Ali Taher 22 Jun '15

22 Jun '15

Hello, Suppose I have $var(a) = 1234#5678 and $var(b)=123455667 I need to get the number before '#' character if exists , meaning that from $var(a) I must get 1234# and from $var(b) I must get empty string. I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole string if # doesn't exists. There is a way to check if # exists by subtracting the length of the string before and after removing # character and check if length changes , but I want a cost-less method. Any help would be appreciated. Thanks Ali

3 2

Capture a Match from Regex comparison
by Sebastian Damm 22 Jun '15

22 Jun '15

Hi, we have some cases, where we get multiple PCMA lines with different numbers from a peer. One of our backend systems doesn't handle that correctly. So what I want to do is detect whether there is a PCMA line with a non-standard number in the SDP. This is how an SDP looks like: v=0 o=hiQ9200 2990220150519102450 1229717573 IN IP4 1.2.3.4 s=Phone Call c=IN IP4 1.2.3.4 t=0 0 m=audio 40174 RTP/AVP 8 0 18 96 97 13 100 98 99 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:0 PCMU/8000 a=fmtp:0 vad=no a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 G726-32/8000 a=rtpmap:97 AAL2-G726-32/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=rtpmap:98 PCMA/8000 a=gpmd:98 vbd=yes a=rtpmap:99 PCMU/8000 a=gpmd:99 vbd=yes a=sqn: 0 a=cdsc: 1 image udptl t38 a=sendrecv a=pmft: T38 a=ptime:20 I tried something like that (expecting that the number doesn't change): if (sdp_get_line_startswith("$avp(badcodec)", "a=rtpmap:98 PCMA")) { But this makes Kamailio drop the INVITE when arriving at the line in the config. I guess, "startswith" stops parsing the line at the colon. So I thought I'll make it more generically: sdp_get("$avp(sdp)"); if ($avp(sdp) =~ "a=rtpmap:([1-9][0-9][0-9]?) PCMA") { sdp_remove_codecs_by_id("$missingvariable"); } What I need is to capture the number in the a=rtpmap line so I can delete the codec by ID. But I couldn't find a way to capture the matches from my regex comparison. Am I missing something? (Other approaches for my problem would be welcome, too.) Thanks in advance. Best Regards, Sebastian

2 3

Kamailio with Blox SBC (opensips based) - Wrong CSeq
by kai.ohnacker＠cbc.de 22 Jun '15

22 Jun '15

Hello, my setup is a kamailio server as registrar and Blox as SBC. Blox is a freeware GUI with Opensips as framework. Kamailio is located in the private network and Blox operates with 2 NICs (private and public) Now the following problem: If I make a call from public (phonerlite) to private (mayah), the connection don’t reach the “framed” state and is “broken” after 30 sec. After evaluating the log files, I found that the CSeq from Kamailio is wrong. The invite comes from CSeq: 9 INVITE The reply is with CSeq 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK38ae.cf317344.0 Max-Forwards: 70 To: <sip:1001@192.168.1.3:8000>;tag=qTKGZvr0ItW From: "PhonerLite" <sip:1000@192.168.1.3:8000>;tag=1005520878 Call-ID: SBCbFgsQn5HcHpEUxJUVF8gQWVZREt6QwJvWVJkXCdAUzdkXl41CUJ4cEdWZV1NQ0tFYQ CSeq: 1 Contact: <sip:1001@192.168.0.33:5060> Record-Route: <sip:192.168.0.1;lr=on> User-Agent: MAYAH 4.9.12.0-2.1.0.45 Content-Length: 0 And here the information from the opensips log files: Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: no method found Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: bad cseq Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:get_hdr_field: bad cseq Perhaps someone of you knows this failure or can tell me how to solve this problem? Cheers, Kai

3 4

Redirect with path
by Kelvin Chua 22 Jun '15

22 Jun '15

on my registrar server: after calling lookup, and getting AOR with path, how do I send the path info back as a redirect? Kelvin Chua

2 2

event_route: what modules/functions are likely to work or not work
by Armen Babikyan 22 Jun '15

22 Jun '15

Hello, With respect to *_route blocks in kamailio configs, I gather that event_route is the newest type of route in the config vocabulary. However, many modules have functions that specifically state that they work within certain routes, like REQUEST_ROUTE and FAILURE_ROUTE, with the implication that they won't work in other routes. For example, there are some functions that "work" in EVENT_ROUTE as well, and that this could be a matter of either documentation not being updated yet, or funny/bad behavior I may encounter at some future point in time... Generally speaking, are there any general statements that can be made about what "common" modules/functions are definitely only available within request_route and onreply_routes? event_route blocks, in particular, elucidate many other use cases for several modules' functions that I'm sure the original authors didn't think of, and it would be great to have a high-level commentary/discussion like "module XYZ or modules that touch A, B, and C don't have a snowball's chance of working in an event_route". Comments are welcome. Thanks! Armen

2 1

Installation problem, kamailio 4.3.0 on ubuntu 14.04.02
by Edin Fejzić 22 Jun '15

22 Jun '15

I just installed kamailio 4.3.0 with apt-get. But it won't start. mysql and /etc/kamailio/kamctlrc done. Minimum change to kamailio.cfg Any idea? syslog output: /usr/sbin/kamailio[1291]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob /usr/sbin/kamailio[1291]: INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available /usr/sbin/kamailio[1291]: INFO: usrloc [hslot.c:51]: ul_init_locks(): locks array size 1024 /usr/sbin/kamailio[1291]: ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] /usr/sbin/kamailio[1291]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed /usr/sbin/kamailio[1291]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/lib/x86_64-linux-gnu/kamailio/modules/ctl.so) kernel: [ 13.978764] init: kamailio main process (1291) terminated with status 255

2 1

Kamailio invite counter
by Ali Taher 19 Jun '15

19 Jun '15

Hello, I want to define a counter that increase in every invite. I tried the below : if (is_method("INVITE")) { $var(u)=$var(u)+1; xlog("L_INFO","Counter : $var(u) " ); } But the counter is reset to 1 randomly (even before restarting kamailio service) How can have a proper counter that count invite messages. Thanks Ali

4 7

Port 5061 issue
by Ankhit Vivekananda 19 Jun '15

19 Jun '15

Hello, I have configured kamailio 4.0.4 long back. Now, I wanted to enable the usage of port 5061 with TLS. By doing so, the client(video camera) can place outgoing calls fine. But while receiving an incoming call, the message does not get sent out to the other client. Hence, the call fails. I can place and receive calls on port 5060 successfully without any issues. Can anyone please tell me why port 5061 is giving me a problem. Also, as a test, I set up kamailio on another machine(version 4.1) and 5061 works fine in all directions. Its only on the other machine that I have problems with. Appreciate any help! Thank you, Ankhit

2 1

SIP-I trunking
by Jérôme Simionato 19 Jun '15

19 Jun '15

Hello, I'm planning to interconnect kamailio with a SIP-I provider (an i have no more information at this time from my telco). I read the documentation about the SIP-I module, who need to be completed to match our needs. basically, we need to read the location parameter for incoming call, and be able to set redirecting number for outgoing call Does anybody use this module for SIP-I ? Have you experiences of SIP-I trunking with telco ? And another question: when you make an ISUP (SS7) interconnection with a telco, you need to pass specific test to validate the correct implementation of the SS7 protocol (and it's national variant, eg. SPIROU in france). Is it the same thing on an SIP-I interconnection ? Any information about it will be appreciate, Cheers, Jérôme.

2 3

SIP Ping a device using Kamailio
by Nelson Migliaro 19 Jun '15

19 Jun '15

Hello, Is there a way to ping a device using Kamailio. In my scenario I need to access a Kamailio located in my DMZ over a firewall that is only open in LAN > DMZ direction. It is close in the other way round. In my current configuration I keep the link open using Asterisk qualify option in SIP. I would like to replace Astersik located in LAN with Kamailio thus I need to keep the link open. I know there is an option using dispatch but I would like to avoid using this module in order to keep it simple. Is there a "simple embedded function" in order to accomplish this. thank you

2 1

WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction
by Roberto Fichera 19 Jun '15

19 Jun '15

Hi All, I'm getting the warning in the subject after replacing a route() with a async_task_route(). In the called route() I'm doing some db query and send notify messages using the uac module. The async_task_route() is called once received a register message. Any idea what's is going on? Cheers, Roberto Fichera.

2 3

users with or without asterisk integration on the same kamailio database
by asterisk asterisk 19 Jun '15

19 Jun '15

I wish to know how I can have two kinds of users 1. some users with real time integration with asterisk 2. other users do not How can I achieve in the same kamailio instance?

3 3

Dispatcher and options_reply
by jay binks 19 Jun '15

19 Jun '15

Hello, So Im using dispatcher to poll SBC's behind my kamailio load balancer. What I need to do is get a count of how many "active" SBC's there are in the dispatcher module. ( how many SBC's are actively responding to their options ping ) so that I can then use this info when I reply to an OPTIONS on the load balancer. effectively , if there are no SBC's active, the LB needs to not reply to options. I COULD use ds_select_dst or something , but is that going to be overkill from a CPU perspective ? ( And I dont want ds_select_dst to do any modification to my otpions packets ) just wandering what everyone thinks the best way to achieve this would be. -- Sincerely Jay

2 1

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