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42 participants
37519 discussions

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/

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