May 2016 - sr-users - kamailio.org

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

need help with RTPProxy in bridged mode
by Grant Bagdasarian 04 Apr '20

04 Apr '20

Hello, I'm using RTPproxy for the first time in bridged mode and I can't get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio. The setup is as following: Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch Kamailio is listening on two interfaces: 1) Private: 172.0.0.1 2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address) Freeswitch is listening on: 1) 172.0.0.2 Carrier is on: 1) 10.0.0.1 I've started an rtpproxy instance on the Kamailio box using: rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1 I've played around with rtpproxy_manage() and the various flags (ie, ei), but I can't get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier. It always sets it to its private address, instead of its public address. I'm using Kamailio 4.2 with sippy/rtpproxy 2.0. Could someone please point me into the right direction? Thanks! Grant

4 8

rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

5 10

fix_nated_contact and IPv6
by Sebastian Damm 27 Oct '17

27 Oct '17

Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: <sip:bob@192.168.8.132> outbound Contact: <sip:bob@1234:1234:0:1234:0:0:0:2> The correct way of transforming the Contact URI would be: Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]> This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian

4 4

NatHelper ignoring force_socket module parameter
by Asgaroth 24 Oct '17

24 Oct '17

Hi All, I have a strange issue, I have set the module parameter for nathelper's force_socket to a specfic ip/port, however, when I perform a sip trace I can see that all locally generated options messages are not sent from the socket defined in the modules parameters. I am not setting $fs anywhere in the script, so I am not over-writing it. Do any of the other module parameters have a bearing on if nathelper uses the force_socket parameter? My current nathelper settings are: modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "natping_interval", 20) modparam("nathelper", "natping_processes", 4) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:keepalive@domain.com") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", NAT_BFLAG) modparam("nathelper", "force_socket", "1.2.3.4:5060") I am using kamailio v4.3.1: # kamailio -V version: kamailio 4.3.1 (x86_64/linux) f38e67 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, 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: f38e67 compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7 Thanks

2 5

Kamailio IMS deployment
by Ramya Y 22 Oct '17

22 Oct '17

Hi, I am newbie to kamailio world and we would like to deploy Kamailio based IMS platform. Can any body share the link or Step by step procedure(user Guide) to deploy Kamailio IMS. Regards, Ramya

5 22

Kamailio 4.3.4: receive_msg(): no via found in reply
by Grant Bagdasarian 15 Aug '17

15 Aug '17

Hello, We're testing kamailio 4.3.4 with two kamailio's communicating with each other. One of them is a gateway with rtpengine at the carrier side, the other is a basic sip router. When calling a destination and cancelling the INVITE a 487 is received from the carrier and directly ACK'd by the first kamailio and in turn is forwarded to the second. But the 487 response received by the 2nd kamailio doesn't have a Via header which results in kamailio printing the error: receive_msg(): no via found in reply. Any ideas why the forwarded 487 doesn't contain a Via header and how to fix this? Note: IP's are changed. Carrier: 10.0.0.1 Kamailio 1: 10.14.0.1 Kamailio 2: 10.14.0.2 U 10.0.0.1:5060 -> 10.14.0.1:5060 SIP/2.0 487 Request Cancelled. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.0.0.1:5060 ACK sip:003112345678@10.0.0.1:5060 SIP/2.0. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. Max-Forwards: 67. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. From: "+31612345678"<sip:+31612345678@10.14.0.3:5060>;tag=109ac722. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. CSeq: 2 ACK. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.14.0.2:5060 SIP/2.0 487 Request Cancelled. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0.

4 16

Problem with t_relay() and VPN tunnel
by Carlos Ruiz Diaz 24 Jun '17

24 Jun '17

Hi, I have two SIP servers, *A* and *B*, connected each other though a OPENVPN tunnel. The server *B* needs to t_relay() every SIP message containing the method MESSAGE to the server *A* but these messages never reach destination. I have tested the tunnel connectivity and works fine. I wrote a Perl script (located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying to figure out what is happening but these messages are received by *A* and processed correctly but when *B* does the same from Kamailio, it is never received. Here is the route part of kamailio.cfg in *B*: Observation: ($rU == "1004") result is *true* * if(is_method("MESSAGE")) { if($rU == "1004") { xlog("L_INFO","En 1004"); rewritehost("10.8.0.1"); if (!t_relay()) xlog("L_INFO","MIO Error en t_relay"); t_reply("200", "Ok"); xlog("L_INFO","MIO despues rewrite"); exit; } .....* The perl script that WORKS: *$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2 From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796 To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>> Call-ID: 19722852989(a)192.168.1.2 CSeq: 15773 MESSAGE Contact: <sip:2002@**PublicIP-protected**:5060> Max-Forwards: 29 User-Agent: DBL Content-Type: text/plain Content-Length: 34 +595981[protected] hello from kamailio '; use IO::Socket; my $sock = IO::Socket::INET->new( Proto=>'udp', PeerHost=>'192.168.2.102', PeerPort=>'5060'); print "Sending msg $msg\n"; $sock->send($msg) or die "error sending $!\n"; *Please help! Thanks in advance. Carlos.

3 3

Missing accounting events in ACC table
by Muhammad Shahzad 23 May '17

23 May '17

Hi, About 3 weeks ago i upgraded one of my production server with latest stable kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about missing CDR events in ACC table. I observe following problems, 1. There are only BYE records in acc table, no record for INVITE or ACK. 2. In kamailio logs when ACK is received against 200 OK response for INVITE, i see following errors, -- ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse >From uri ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org' to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484' -- Of course all these errors are bogus, I have checked all headers in ACK (not just FROM header), they all seem perfectly fine and valid. 3. Then the dialog times out, -- WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags '1D3ECD34F5731AB845BA3064AC95BB2D' '7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4' -- 4. Any further sequential requests complain about "unable to find dialog", e.g. -- NOTICE: <script>: Sequencial 'BYE' request received from caller ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [ sip:00xxxxxxxxxx@sip.domain.com] WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog for BYE with route param '5ae1.d595' [7845:22877] -- 5. However the acc record for BYE is written to db and log file, -- NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain= sip.domain.com ;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy -- The same config was working fine with older version 4.2.0-97cab8. The kamailio config i am using is pretty much standard, -- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 ... modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # log to db modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "DBURL") ... request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); }; exit; }; #### only initial requests (no To tag) #### t_check_trans(); .... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog setflag(FLT_ACC); # do accounting setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(dlg_timeout) = 60; dlg_manage(); .... } -- Any ideas why its happening? Since it is 3 weeks old so may be problem has already been spotted and fixed by someone else. Otherwise let me know how can i provide more info to help fix this issue. Thank you.

3 11

How to make OutBound calls with kamailio
by sharad tyagi 23 May '17

23 May '17

Hi I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I have add and registered two IP phones with kamailio these phone able to call each other. How can i make outbound call with that I am using a asterisk server as PSTN Gateway. i am not famalier with kamailio script. Regards Sharad Tyagi SparkTG info Pvt Ltd

4 3

Cannot hear voice with symmetric NAT and STUN
by Khoa Pham 08 Feb '17

08 Feb '17

Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT, and RTP server is for symmetric NAT. Supposed A calls B. If A, B both use symmetric NAT and STUN, they cannot hear each other If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each other. Why is that? I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt for Prediction Failure, is that related to this problem ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

8 18

Using the auth_ephemeral module
by Hemanshu Vadehra 17 Jan '17

17 Jan '17

I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example? Regards, Hemanshu Vadehra hemanshu.v(a)directi.com

5 8

Siremis installation with postgresql.
by José Seabra 16 Dec '16

16 Dec '16

Hi there, I'm facing some issues trying to install siremis with postgresql. The file Config.xml has the correct configuration for postgresql driver but It seems that during installation phase siremis can't create database structure for postgresql. Anyone have installed siremis with postgresql, if yes can you share the steps needed for this installation. -- Cumprimentos José Seabra

4 6

Kamailio on Mac
by Carlos Ruiz Díaz 04 Aug '16

04 Aug '16

Hi all, I'm switching my development environment from Linux to Mac. Is there any tutorial on how to setup Kamailio on this OS? It may be an Unix flavor, but it's still quite different from Linux :). Thanks, -- Carlos http://caruizdiaz.com http://ngvoice.com +52 55 3048 3303

6 9

4.4.1: Only first UDP Listener is producing log messages
by Sebastian Damm 08 Jul '16

08 Jul '16

Hi, I upgraded one of our systems today to 4.4.1. And when checking whether everything is okay, I noticed, that only the first UDP listener (from kamcmd core.ps) is actually sending lines to the log. The only other logs I see are from usrloc timer worker. kamcmd> core.ps 10737 main process - attendant 10740 udp receiver child=0 sock=172.20.40.5:5060 10741 udp receiver child=1 sock=172.20.40.5:5060 10742 udp receiver child=2 sock=172.20.40.5:5060 10743 udp receiver child=3 sock=172.20.40.5:5060 We are sending quite a lot of information via xlog("L_NOTICE", "foo"); to syslog, but the only lines I see are from PID 10740. Kamailio works fine otherwise, handling all requests, the usrloc entries didn't drop, and customers don't complain. So I think it's only a logging issue. There are about 50k registrations on the machine and more than 10 Mbit of traffic. When I downgrade to 4.3.5, I get output from all listeners again. Has there been any change between 4.3 and 4.4 regarding logging? The upgrade document says nothing about this. Best Regards, Sebastian

2 4

S-CSCF (kamailio) crashes after deregistration of client
by Christoph Valentin 24 Jun '16

24 Jun '16

4 12

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 23 Jun '16

23 Jun '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0

3 5

question about RTPengine
by Dmitry 14 Jun '16

14 Jun '16

hello i read the documentation about RTPengine.and the documentation says that: If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer() what does it mean? actually - according to 3261 RTP media (both ways) may be turned on after ACK

3 5

4.4 compiler warnings on debian jessie
by Juha Heinanen 08 Jun '16

08 Jun '16

I'll come back to mem/tlsf_malloc.c issue. Camille asked if the compiler warnings on mem/tlsf_malloc.c were occurring on x86 32 bit machine. I now tried and, yes, they do not appear on i386 jessie, but not on amd64 jessie. So looks like this is the same issue that Ovidiu has reported. -- Juha

2 3

Wrong Publish status
by Aleš Šturm 06 Jun '16

06 Jun '16

Hello, Kamailio 4.3: I am testing Presence module with more complex scenario. I have 2 SIP clients (presentity) with equal identities (3918) and one watcher (2025). All my clients are Monster 0.9.25. My Kamailio Presence module is configured with "retrieve_order=1" (presentity records are retrieve by priority order). I have situation when wrong status is send to the watcher (2025). 1. First all three clients registers and publish its own status. Below is print from my MySQL database (presentity table) mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340687 | 1464337087 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 2. Then first 3918 client publishes his mood >11111<. The status in >Presentity< table is refreshed under the id=84480. NOTIFY is sent to 2025 with status with mood >11111<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 3. After that the second 3918 client publishes his mood >22222<. The status in >Presentity< table is refreshed under the right id=84481. NOTIFY is sent to 2025 with status with mood >22222<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 4. Then finally, the first 3918 changes his mood to "33333". The status in >Presentity< table is refreshed under the right id=84480. BUT, to 2025 is sent NOTIFY with mood "22222". "received_time" timestamp of status with mood "33333" is higher than timestamp of status with mood "22222". BUT the "id" attribute of "presentity" table of status with mood "33333" is smaller. I think that statuses with equal "priority" are retrieved by "id" and not by "received_time"!? mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340729 | 1464337129 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>33333</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | The Wireshark trace is attached. Can anyone help. Please? We have a big installation in the Gazprom's (Rusia) network, where Kamailio is used as a Presence server only. Thanks Ales

2 8

Kamailio 4.4.1 REGISTER Parse Error in PCSCF
by Mihail Dakov 02 Jun '16

02 Jun '16

Hi Folks, I have the strangest problem just after installing latest stable version. I had tried both ways using the repo (http://deb.kamailio.org/kamailio44) and downloading code with local compile (https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/) The initial scenario is to get M2M with commercial IP and subsequently use the Rx interface to do the same. We are blocked because we cannot even achieve clean register for the UEs. UE1(galaxy s4 zoiper) -------- Kamailio IMS aio | UE2(galaxy s4 zoiper) -------- | What happens is that the first register message triggers some quasi-endless loop in the PCSCF. Hence it cannot be forwarded to SCSCF, ICSCF and HSS. But this doesn't always happen and between all the parse problems there are some register messages which get passed to the scscf icscf and hss. Somehow the parser feeds itself from the previous parse run and thus loops quite intensively. The message I get is SIP/2.0 400 CSeq method does not match request method. I must say that reiterating the registration process sometimes i get the register and sometimes i don't. Has anyone encounter similar issue for similar use case and can share some thoughs about a fix? I attached relevant log file, pcaps and configuration files. Let me know if more config files are needed.

3 15

Nonce extract from SIP Meesage
by Sergio Serrano 02 Jun '16

02 Jun '16

Hi all, I just try integrate Kamailio with external software and We want taht kamailio doesn't authenticate users. I want that the autehtication process would be done by external software. I want To receive a REGISTER for a user. Then I send a 407 and then when I receive all paramters for autehtincation(nonce, response, digestURI, etc, etc etc) send that data to external software and receive if data is correct, then sent 200 OK and save location in external software. Bu t I can see any pseudovariable or enything else to obtain that dato to provide to external software. Anyone could think that this process could be done? Regards, Sergio

3 2

NSQ libevbuffsock dependency
by smititelu 02 Jun '16

02 Jun '16

Hi, Upon trying to compile latest Kamailio from sources I got an: " nsq.h:9:24: fatal error: evbuffsock.h: No such file or directory #include <evbuffsock.h> ^ " Looking on the new NSQ module documentation I see the libevbuffsock dependency. Is there other recommended way, to install the library besides compiling from sources at [1]? Thx, Stefan [1] https://github.com/mreiferson/libevbuffsock

2 2

sip trace
by Jeremy Betts 01 Jun '16

01 Jun '16

I have a new carrier that wants sip traces of certain scenarios. They also require IPSEC tunnelling for all SIP messages. So when I use tcpdump or any other similar tool I can see the incoming SIP messages but outgoing SIP messages are ESP encoded. I've played with the siptace module but can't seem to get it to log replies. Does anyone have any solutions?

2 2

Dynamic advertised ip for rtpengine/rtpproxy
by Serge Yuriev 01 Jun '16

01 Jun '16

Hi, We have several exits from our network and all of them natted. Could we set adv ip for rtpproxy/rtpengine from db in realtime or we must define them on daemon run time only? -- Wbr, Serge via mobile -- Serge via mobile

4 3

advertise xxx.xxx.xxx.xxx:5060 is weird
by Sébastien Brice 01 Jun '16

01 Jun '16

Hi guys, I have a local proxy running on private ip 192.168.3.97 and advertising the wan ip of my box (THEPUBLICIPOFMYBOX) this local Kamailio is a stateful proxy that TM REGISTERS and INVITES to a Kamailio registrar(PUBLICIPKAMAILIOREGISTRAR) (please have a look at the kamailio.cfg attached to this email.) I want to force the signalling and sessions through this local proxy (some kind of outbound proxy) I have jitsi and xlite behind (ip 192.168.3.39 and 192.168.3.16) that tries to call themselves. user agents keep retransmitting 200 OK because they never get the ACK following the 200 OK pickup. If i remove the advertise THEPUBLICIPOFMYBOX of the kamailio.cfg the three handshake is working ok and the call is no longer dropped after a few second! i have a true bidirectionnal session and that's working ok. The bad thing is RTP flows straight through router (thanks to rtpproxy i guess) but i want incoming and outgoing RTP traffic to go on central local node 192.168.3.97 How can i achieve that? thanks. PS: i am not using path header because i am not skilled enough and kamailio is really hard horse to train

2 5

DHCP renewal conflicts with kamailio?
by Joonas Harjumäki 01 Jun '16

01 Jun '16

Hi! The attachment contains a log trace of kamailio 3.3.2. There is some errors in the log and I wonder if it is normal operation? The errors appeared in the middle of DHCP lease renewal and I wonder if the configuration (of dhclient/kamailio) could be improved to prevent such errors? Or to make sure kamailio retries udp_send sufficiently many times. The kamailio operated normally before renewal procedure began and after the procedure finished. Joonas Harjumäki

2 1

how to set kamctl to find ctl anf fifo that are not at /var/run
by Uri Shacked 01 Jun '16

01 Jun '16

Hi, I am using centos 7. By default on centos 7, the var/run file system is tmpfs. So, when kamailio runs with user and group other than root, it can not create the fifo file. I set the kamailio service and config files to work on usr/local/var/run/kamailio. Now, how do i set the kamctl to work with the fifo on /usr/local/var/run/kamailio ? Uri

3 3

Want dialog.default_timeout NOT to reset on subseq. req
by Andrey Utkin 01 Jun '16

01 Jun '16

Hi! I'd like to restrict max call duration. Currently we have something that partially works: dialog.default_timeout. But, as described in "How it works" section http://www.kamailio.org/docs/modules/devel/modules/dialog.html#idp2029360 "The dialog timeout is reset each time a sequential request is processed." So this option works for us when the useragents don't send sequential UPDATEs or reINVITEs. I am not sure if it is adequate to ask for this feature in dialog module. I am aware of possibility to terminate any call by management command, and of cnxcc module which has call duration control. I would love to find something that simple - a single option for max call duration, without external mechanism (cronjob etc.) or full-blown billing. Any suggestions are very welcome. Thanks in advance. Please reply to all recipients, I am not subscribed to this maillist.

2 1

513 Message too big for VIDEO calls
by Varghese Paul 31 May '16

31 May '16

Hi all, I am trying to setup WSS on opensips-2.2. I am getting "513 Message too big " when i am trying to make video calls. Audio calls and Registration is working fine.I tried to change the message_chunk_size in WSS module but still i am getting the message. Client : SIPML5 and JSSIP Opensip version : 2.2 I am attaching the sipml5 console logs. Please let me know is there any settings i can apply to resole this. Thanks in Advance Varghese Paul

2 1

Extending has_body() with additional mime types
by Giacomo Vacca 31 May '16

31 May '16

Hi all, apologies if this is an old topic but I've noticed that has_body() can provide false positives and would like to discuss a solution. As a practical example, has_body("application/vq-rtcpxr") was returning 1 for Content-Type "application/dialog-info+xml". The interpretation is that they are both unknown and so they match. Using 'if ($cT == "application/vq-rtcpxr)" provides a work around, but I wonder what's the suggested solution for this? a. Add the missing mime types (from RFC 6035 and 4235 respectively) to the list b. Instead of returning 1, return another code, e.g. 2, when the mime type is unknown c. Switch to string comparison when the mime type is unknown d. Something else? Thanks, Giacomo

2 1

Accept INVITE with transport=tcp
by Nahum Nir 31 May '16

31 May '16

Hi, I have a client that sends INVITE with transport=tcp. How can I make Kamailio to work with that? Thanks and regards, Nahum

2 1

Issue with LCR an failover routers
by Alberto Sagredo 30 May '16

30 May '16

Hi Im using LCR module and trying to handle 503/408 errors on provider to route to a second one. Actually im using if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Gateways no disponibles"); sl_send_reply("500", "Server Internal Error - No gateway"); exit; } else { xlog("GW Seleccionado '$avp(i:709)'\n"); xlog("Dominio de destino: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("No hay GW de Backup"); sl_send_reply("503", "Service not available - No gateway"); exit; } else { xlog("Enviado por primer GW\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); t_on_failure("FAILBACK"); } failure_route[FAILBACK] { if (t_is_canceled()) { exit; } if (t_check_status("408|50[34]")) { route(LCR_OUT_OP2); } } And using for OP2 same routing block with id=3... This second providers use AUTH so using: t_on_failure("TRUNKAUTH"); failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { if (to_gw("3")) { xlog("Autenticación "); $avp(auser) = "user"; $avp(apass) = "pass"; } uac_auth(); t_relay(); exit; } But getting this error: May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [t_fwd.c:1704]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [tm.c:1466]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Any idea how to resolve that?

2 2

converting integer to string
by Henry Fernandes 30 May '16

30 May '16

In Kamailio 4.2, I’m having problems converting an AVP integer value into a string. My Kamailio config stores the return code from auth_check() and outputs it. $avp(auth_code) = auth_check("$fd", "subscriber", "1"); xlog("L_NOTICE", “response = $avp(auth_code)”); When I check the AVP against an integer, it behaves normally. However, I cannot convert it to a string. The return value -3 (invalid user) gets converted to the string “18446744073709551613” when outputting it. My hunch is that Kamailio is treating the AVP as an unsigned integer (rather than a signed integer) when it converts it to a string. If I try this with an AVP with a positive integer, it works as expected. Can someone tell me how to convert a negative integer into a string in Kamailio? -H

3 3

Variable in dialog event routes
by Minh Phan 30 May '16

30 May '16

Hi List, I'm trying to have an external application to write CDR to a database by doing this: event_route[dialog:start] { evapi_relay("{'cdr_id':* $dlg_var(cdr_id),* 'event': 'answered'}"); } event_route[dialog:end] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'ended'}"); } event_route[dialog:failed] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'failed'}"); } $dlg_var(cdr_id) is set inside a request route. However, in these event routes, its value is always null. Its seems that $dlg_var variables are not available in those event routes. What are the variables available in dialog event route? Any suggestion on how to achieve what I'm trying to do? Best regards, Minh

2 3

http_async_client segfault
by Jason Penton 30 May '16

30 May '16

Hi, Has anyone experienced segfault using http_async_query from http_async_client module? Here is BT: #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 53 iofgets_u.c: No such file or directory. (gdb) bt #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 #1 0x00007fca68f08afa in get_contents (stream=<optimized out>, len=1032, linebuf=0x7fca23ffe660 "127.0.0.1") at nss_files/files-XXX.c:201 #2 internal_getent (result=0x7fca23ffe5c0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, af=af@entry=0, flags=flags@entry=0) at nss_files/files-XXX.c:246 #3 0x00007fca68f09ca3 in _nss_files_gethostbyname4_r (name=name@entry=0x7fca2e790fa0 "localhost", pat=pat@entry=0x7fca23ffebe0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, ttlp=ttlp@entry=0x0) at nss_files/files-hosts.c:402 #4 0x00007fca79eb7757 in gaih_inet (name=<optimized out>, name@entry=0x7fca2e790fa0 "localhost", service=<optimized out>, req=req@entry=0x7fca341c7490, pai=pai@entry=0x7fca23ffed10, naddrs=naddrs@entry=0x7fca23ffed00) at ../sysdeps/posix/getaddrinfo.c:850 #5 0x00007fca79ebae0d in __GI_getaddrinfo (name=0x7fca2e790fa0 "localhost", service=0x7fca23ffeec0 "18000", hints=0x7fca341c7490, pai=0x7fca23ffee78) at ../sysdeps/posix/getaddrinfo.c:2406 #6 0x00007fca6e0f80f4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #7 0x00007fca6e104fa4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #8 0x00007fca6e10284b in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #9 0x00007fca723c9182 in start_thread (arg=0x7fca23fff700) at pthread_create.c:312 #10 0x00007fca79ee547d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 Camille, are you maintaining this module now?

2 2

Multiple async operations in single transaction fail (using http_async_client)
by Thom Seddon 30 May '16

30 May '16

Hi All, I've been doing some work this week with the new http_async_client module, it seems that with it's default behaviour it's not possible to execute another async query in the block immediately following the response, for example the following config: request_route { if (!t_newtran()) { sl_reply_error(); exit(); } http_async_query("http://localhost:8090/req_1", '{"user":"$fU"}', "REQ_1"); } route[REQ_1] { xlog("L_INFO", "route[REQ_1]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_1]: status $http_rs\n"); xlog("L_INFO", "route[REQ_1]: Sending REQ_2\n"); http_async_query("http://localhost:8090/req_2", '{"user":"$fU"}', "REQ_2"); } else { xlog("L_INFO", "route[REQ_1]: error $http_err)\n"); } } route[REQ_2] { xlog("L_INFO", "route[REQ_2]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_2]: status $http_rs\n"); } else { xlog("L_INFO", "route[REQ_2]: error $http_err)\n"); } } Generates the following error: 10(11832) INFO: <script>: route[REQ_1]: INIT 10(11832) INFO: <script>: route[REQ_1]: status 200 10(11832) INFO: <script>: route[REQ_1]: Sending REQ_2 10(11832) WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [53195:560306836] Looking at the source, http_async_client calls t_suspend, then goes away and performs the request in the background. On response it calls t_continue and then this block <https://github.com/kamailio/kamailio/blob/4.4/modules/tm/t_suspend.c#L270-L…> in t_continue will continue execution, the second query will cause http_async_client to call t_suspend whilst sill executing from the previous t_continue. Once they query is sent, the first t_continue will complete, removing the T_ASYNC_SUSPENDED flag. When the second query returns it doesn't find the flag and so fails as shown. I'm far from an expect on this so would really appreciate some pointers, it seems to fundamental to be a race condition in tm but the usage in http_async_client seems to match usage in other modules so I'm not sure on the best tactic. I've tried peppering append_branch and async_route call's in various places to no avail. Full debug log here: http://pastebin.com/raw/rFmm0fmf Any help appreciated, *Thom Seddon*

2 1

username of Contact header
by Al S 27 May '16

27 May '16

Hi, I am trying to read username portion of Contact header: My Contact header content has a URI value such as: sip:8131110000@10.10.10.10.:5060 and I am trying to read : 8131110000 I tried the following perl similar RE and it didn't work: ($var(main_number2)) = $ct =~ /sip:(.*)@/; Thanks,AS

2 1

4.4: Kamailio crashes at start - sometimes
by Sebastian Damm 27 May '16

27 May '16

Hi, we have a machine running 16 Kamailio instances, and while upgrading to 4.4.1 (from 4.3.5), 8 of them wouldn't start. When downgrading to 4.3.5, they all start again. All of them have pretty identical configuration files, except IPs, ports and some code enabled or disabled via defines. After comparing a working and non-working configuration and adjusting setting by setting, we finally ended up with a working configuration. The difference is, that it won't start when a part of the code DOES NOT get included. If it gets included, it will start. This is the mentioned part of our main route: #!ifdef ENABLE_INV_RATELIMIT # Check for INVITE limit if (is_method("INVITE") && $au == $null && !($ua =~ "sipgate") ) { $var(invcount) = $shtcn(invcount=>%~$fU); xlog("L_INFO", "INVITE Requests from $fU in last 30 seconds: $var(invcount)\n"); if ($var(invcount) < 12) { $var(uniqcid) = $ci + $Ts + $ft; $var(tkey) = $fU + '-' + $(var(uniqcid){s.md5}{s.substr,0,10}); $sht(invcount=>$(var(tkey))) = 1; $var(uniqcid) = $null; $var(tkey) = $null; } if ($var(invcount) > 10) { if ($var(invcount) == 11 ) { xlog("L_NOTICE", "User $fU ($var(domain2use)) over ratelimit for new calls, rejecting.\n"); } # Enable this only after evaluating the impact! append_to_reply("Retry-After: 30\r\n"); sl_send_reply("503", "Call Rate Limit Exceeded"); exit; } } #!endif If we put this line at the top of the configuration file, everything works: #!define ENABLE_INV_RATELIMIT If we delete this line, startup does not work. It just sits in ps for one minute without forking, and then gets terminated. We enabled a bit of debugging, and this is apparently the error causing Kamailio to shutdown: May 25 14:50:15 kammel /usr/sbin/kamailio[24989]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 53: nathelper May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0x9f0428 and 0xa34428 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0x9f0240, 82, 1, (nil)), fd_no=0 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Bad file descriptor [9] May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1747]: tcp_receive_loop(): failed to add tcp main socket to the fd list May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1815]: tcp_receive_loop(): exiting... I have no idea, how this part of the code could lead to this error, but it is reproducable, that at least on this system setting or disabling this define fixes or breaks the startup. Does anybody have an idea, what's happening there? Best Regards, Sebastian

2 4

Multihomed kamailio - one IP for each account
by Attila Megyeri 27 May '16

27 May '16

Hi Folks, I am looking for a solution to do the following, basically an SBC-like solution. I have an internal asterisk system that should make outbound calls towards a provider, but that provider authenticates the customer based on IP address. And I have many such customers. The provider is not willing to make changes to his practice... Asterisk is bound to a single IP, so there is not much I can do. I was thinking about creating several outbound accounts for asterisk, and then a multi-homed Kamailio would do the following: If request comes from user1, Kamailio would send the request to the provider from IP1. If request comes from user2, Kamailio would send the request to the provider from IP2. And so on, for as many IPs as many customers I want to handle. Is there a more elegant way to do this? If not, how should I implement this? Thanks in advance, Attila

2 4

Kamailio Via: header issue
by Laura 27 May '16

27 May '16

Good morning list, I need to ask you a question about a problem we are fighting on our VoIP platform based on Kamailio 4.3.5. Our platform is made by more nodes over Europe countries, and we are encountering a problem with the size of the SIP package which, exceeds the physiological limit of about 1350 bytes. The real problem is caused from Via: Header (inside the SIP packet) which is added to each element of the system for transit on the call. >From what we have analyzed, the problem is not so much due to the number of entered Via: records but, by the fact that they contain a branch parameter extremely long. Here the call flow example from CARRIER A —> Kamailio1 —> Kamailio2 —> CARRIER B Extract from SIP message sent from CARRIER A to K1 INVITE sip:999912341234@192.168.158.42 SIP/2.0. Via: SIP/2.0/UDP 192.168.220.141:5060;branch=z9hG4bK49e4301c;rport. Extract from SIP message sent from K1 to K2 INVITE sip:999912341234@192.168.127.244 SIP/2.0. Record-Route: <sip:192.168.158.42;lr;did=194.a0f1;nat=yes>. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bK0635.7c1e97c2d5c01ea98e8d0e7fa23a3822.0. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.cuOBV4OBVLMGZfWxF-W.y6Mx1LgRWIC9gIgx4fzxj7MBP7MBVAOBF4M.1jmxuqC93X3heroEWvH9vsCFN43qd4zRj4Mlyf3l1LNSQLpx4uMx3A. Extract from SIP message sent from K2 to CARRIER B INVITE sip:999912341234@192.168.65.248 SIP/2.0. Record-Route: <sip:192.168.127.244;lr;did=194.5122;nat=yes>. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bK0635.f843472880b380acde3a33732975fde9.0. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1LNSQLpx4uMx3AzuaVHRaYpB1JNEt736cQkBIvalaJmly6Mlj7W6Mfg.qw3leqWRMAMRKrz.rIzSPAg.pE3.Vl3.MZMBV7My**. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1JNEt736cQkBIvalaJmwWLORv4mK0Hot3w.jpam6t4kRWWOEWu.4eRWFQGKqfWauYEKwrSOKN7k.gWkxtMgue9mjMsg4IhkxaEkhrbW4uGjxpWPIg0.4eSWF47MRY1kUKfPlqlRxWvg9e5aKp6mxI6N4WS.BPlNRP4kIebWFudkR3loxtM.IWa.U0ZWUKWkxW0. Is there anyone here with the idea how to solve this problem ? Best regards

3 8

rtpproxy module question
by gmele 27 May '16

27 May '16

Hello, we are using an RTP Proxy from rtpproxy.org as media relay to establish communication between our mobile phones. Of course, we are using the kamailio rtpproxy module to modify the SDP payload and control the proxy. In our Kamailio configuration, we have 1 kamailio configured as Proxy and one kamailio configured as Registrar. So calls go through the Proxy and then to the Registrar who will update the SDP header and select an available rtp proxy. We have noticed that sometimes, the rtp udp flow between the phones isn't routed properly by the rtpproxies, ending in the communication drop (all the SIP nego is working well, and the SDP are correctly patched with the rtp proxy address and port). Analyzing the RTP proxy packets, we have found that the Kamailio registrar gives the Kamailio proxy ip address in the RTP proxy create session command, but keeps the original sdp port. command looks like this: Uc96,101 DC -PbO~Wnm <proxy_ip> <port from original sdp phone packet> PZU5OITCW;1 We are using rtpproxy_manage() without any flags. It seems to us that this ip and port are used as default forward route as long as the callee hasnt connected to the rtpproxy. *Is it correct? * If its true, Its seems to us that this cant work as we are mixing the proxy sip address with a udp port open on the phone ? *Is our analysis correct ?* Can we use some option in rtpproxy_manage to replace the proxy ip by the phone ip as seen in the via route ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.h… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio not sending reply
by Jeremy Betts 26 May '16

26 May '16

Hello, I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider. I've included traces and configuration excerpts below. Any help is much appreciated! Problem Call (No Reply Sent): U 63.79.178.192:5060 -> 184.171.164.100:5060 INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. Record-Route: <sip:63.79.178.192;lr>. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 INVITE. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. Max-Forwards: 27. m: <sip:199.173.94.144:5060;transport=UDP>. k: 100rel, resource-priority, replaces. c: application/sdp. l: 235. P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone>. Privacy: none. . v=0. o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. s=-. p=+1 6135555555. c=IN IP4 199.173.68.106. t=0 0. m=audio 55380 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 63.79.178.192:5060 -> 184.171.164.100:5060 ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 ACK. l: 0. Max-Forwards: 27. . Successful Call (Reply Sent): # U 64.136.173.31:5060 -> 184.171.164.100:5066 INVITE sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. Record-Route: <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. To: <sip:7146466334@184.171.164.100>. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Contact: <sip:17143258018@64.136.173.31:5060>. Supported: timer. Session-Expires: 1800;refresher=uac. Min-SE: 90. P-Asserted-Identity: <sip:17143258018@192.168.20.76>. Privacy: none. Expires: 120. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 274. . v=0. o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. s=Session Controller. c=IN IP4 69.85.185.142. t=0 0. m=audio 37970 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 184.171.164.100:5066 -> 64.136.173.31:5060 SIP/2.0 403 DID Lookup Failed. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Server: x-Freevoice SIP Proxy 4.21. Content-Length: 0. . # U 64.136.173.31:5060 -> 184.171.164.100:5066 ACK sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 ACK. Max-Forwards: 70. Content-Length: 0. Kamailio Config: request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { setflag(FLT_ACCFAILED); if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(is_method("INVITE")||is_method("MESSAGE")){ route(ORIGINATE); #route(FROMAST); } route[ORIGINATE] { sl_send_reply("403", "DID Lookup Failed"); exit; } Jeremy Betts <https://www.youtube.com/user/FreevoiceLLC> <https://www.youtube.com/user/FreevoiceLLC>

2 2

webrtc tlsv1 alert access denied
by Arsen 26 May '16

26 May '16

Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried sipml5 and tryit.jssip.net same issue with both clients, also it seems I have these errors only when I use firefox, when I use chrome it even doesn't show me an error.. Any ideas? Thanks! -- Regards, Arsen.

2 3

SRV and dialog stickiness
by Alex Balashov 26 May '16

26 May '16

Hello, Sorry if this is a tired, worn question, but I've not dealt much with Kamailio's SRV support before: If a registrant has a contact binding whose domain component is subject to an SRV lookup with load-balanced or weighted entries, how does one solve the problem of ensuring that subsequent in-dialog requests go to the same host as the initial INVITE? Does Kamailio offer some facility for doing this? Is it somehow accommodated by SIP? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

rtpengine and SAVPF
by Ján Füri 26 May '16

26 May '16

Hi all, Im struggling with incoming calls to webrtc clients. I am trying to change INVITE's SDP using rtpengine from RTP/AVP or RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well. But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as well, rtpengine creates incorrect SDP. It includes twice m=audio RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds it by 488 Not Acceptable Here. Example of my incorrect INVITE : Content-Type: application/sdp Content-Disposition: session Content-Length: 1393 v=0 o=SBC 1464238673 1464238674 IN IP4 <MyPublicIP> s=SBC c=IN IP4 <MyPublicIP> t=0 0 m=audio 30454 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30455 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5X2wgU83bIwrwkHSymjnw48SiMGa1n+eIl50yr99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:iglHZqry a=ice-pwd:eyVXbFNoviYYlu5uuRZlnixFwQ a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30454 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30455 typ host m=audio 30484 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30485 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ozDJ3G/wmaYedQbcTafbhaTIt6raIJa6ugrLUC99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:gM1viWqu a=ice-pwd:zdS4tP1Mj4hLTe6huBWYrhtI1s a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30484 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30485 typ host My setup : incoming call -> MediaServer (freeswitch) -> SIP & Websocket proxy (Kamailio + Rtpengine) -> webRTC clients Is there anybody who could help me with this issue or explain me how to remove the second line of m=audio RTP/SAVPF ? -- Jan

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Sequential calls messes up presence with pua-dialog info
by Ali Pey 25 May '16

25 May '16

Hello, We are using pua-dialoginfo and presence modules for BLF. When the first call comes in or goes out, every thing works great; however, a second call to the same key/extension (e.g a call waiting which is not answered) would turn off the key even though the original call/dialog is still active. Has anyone had experienced such problem? How can I fix this? Any constructive suggestion would be appreciated. Regards, Ali Pey

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nh_reload_rtpp not available
by david 25 May '16

25 May '16

dear all i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo exported function i'm trying to setup the dburl for the rtpengine sets using db_text but when i do kamctl fifo which i see all the commands of the module except the reload_rtpp nh_enable_rtpp nh_show_rtpp nh_ping_rtpp nh_show_hash_total /usr/local/kamailio/sbin/kamctl fifo nh_reload_rtpp 500 command 'nh_reload_rtpp' not available these are the config lines i have for that #!define DBURLRTP "text:///var/log/rtpengine_conf" loadmodule "db_text.so" modparam("rtpengine", "db_url", DBURLRTP) modparam("rtpengine", "table_name", "rtpproxy") and the database is like root@test:/var/log/rtpengine_conf# ls rtpproxy version root@test:/var/log/rtpengine_conf# cat version table_name(str) table_version(int) rtpproxy:1 root@test:/var/log/rtpengine_conf# cat rtpproxy set_id(int) url(str) weight(int) disable(int) 1:udp\:x.x.x.x\:x:100:0 2:udp\:x.x.x.x\:x:100:0 is there something i missed here? do i must use mysql or postgres drivers for this? (i use a test machine to old to install a driver of those with kam 4.4 maybe?) best regards david

1 1

doubt using rate limit
by david escartin 25 May '16

25 May '16

Hello Daniel i had the call rate for some minutes actually. maybe 4-5min and all the time the behaviour was like that. it's true that in that kamailio there wa no any other traffic, only those test calls i set with a sipp to simulate the load best regards david Hello, for how long you got this 20req/second? The module is not counting like continuous time, but more like traffic on each second, iirc. Cheers, Daniel On 23/05/16 11:24, david escartin wrote: > Hello all > > i have a quick question about using module ratelimit on kamailio 4.2 > or 4.4. > on tests made before running it on production, i set the RED > algorithm with a limit of 10 calls per second, but i saw the call > limitation was not done until we got more than 20 calls per second or > so on the queue. we didnt get any INVITE discarded. > after surpassing around 20 INV/sec, it's true the algorithm starts to > work fine and only 10INV/sec are passed more or less > > on production i have no easy way to check this behaviour, besides it's > a feature barely used. > > so i only want to know if this behaviour is known, or if i'm missing > soemthing > > my config is like this > > modparam("ratelimit", "timer_interval", 5) > modparam("ratelimit", "queue", "4:INVITE") > modparam("ratelimit", "pipe", "4:RED:10") > > route[RATE_LIMIT] { > $var(limitation)="4"; > if (!rl_check("$var(limitation)")) { > send_reply("505","Limiting"); > xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, > rejecting with 505\n"); > route(CLEAR); > exit; > } > } > > > thanks a lot and regards > david escartin > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio for sip routing architecture
by Loic Chabert 25 May '16

25 May '16

Hello Kamailio users, I'm working on a sip infrastructure, with opensource technologies, and of course Kamailio will run on my new infrastructure :) Kamailio will be my central routing engine, between SBC (running on asterisk), and somes boxes (asterisk too) where customers will be registred. Please find below my future infrastructure: ---------- -------------- -------- ---------------- | SBC1 | - - - - - | - - - -| kamailio1 |- - - - - | - - - - - | CB1 |-----------------| SIPPHONE | ---------- | -------------- | -------- ---------------- | | ---------- | | -------- | SBC2 | - - - - - | | - - - - - | CB2 | ---------- | | -------- | -------------- | ---------- | - - - -| kamailio2 |- - - - -| -------- | SBCn |- - - - - | -------------- | - - - - - | CBn | ---------- | | -------- Kamailio high availabality will be done with loopback address and BGP in two datacenters. SIP carriers will be connected to SBC and customer to one CB. I have made some tests with kamailio, and i use LCR to route call from CB to SBC, using wheigt and prorities to send call to carriers. Then, when a call arrived from carriers, i use DNS NATPER lookup to find the right CB box. CB HA is done using DNS SRV. So user is registred at one location, and only one. No location sharing beetwen boxes. I have some question concerning my architecture, and technologies used to routing call. Is LCR is the best option for routing call between multiple carriers ? DNS NAPTER is a modern solution to find an endpoint on this architecture ? Existing a better way to route call with kamailio, for exemple rt-json and curl ? In my point of view, kamailio should be a routing engine, using internal technologies, to maximise performances. Less work with kamailio and external app, best performances in sip routing. Is it the good vision ? Thanks in advance.

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