- sr-users - kamailio.org

Roadmap for next major release v4.3.0
by Daniel-Constantin Mierla 22 Apr '15

22 Apr '15

Hello, it is time to nail down the roadmap to the next major release. We discussed during the last IRC devel meeting, proposing to get it out by beginning of June. Given we need to have at least one month of testing, I propose the next milestones: - freezing the development: Wednesday, April 22, 2015 - if testing goes smooth, then branching 4.3 after about one month: During the week starting May 18 - test more in beta phase, prepare packaging, etc. and release after 2-3 weeks: One of the days between June 4 and 11 Other suggestions or adjustments are welcome! Send them to mailing list to discuss further. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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Possible bug in TM module
by Mickael Marrache 22 Apr '15

22 Apr '15

Hi, I just got the following issue when stopping kamailio. kamailio[7279]: NOTICE: <core> [main.c:682]: handle_sigs(): Thank you for flying kamailio!!! kamailio[7279]: ERROR: ctl [ctl.c:369]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) kamailio[7279]: : <core> [mem/f_malloc.c:586]: fm_free(): BUG: fm_free: bad pointer 0x4f2dec6200000000 (out of memory block!), called from tm: h_table.c: free_cell(138) - aborting I always get the first ERROR when I stop kamailio, I need to look at the permissions. However, the next one is a bug. Thanks, Mickael

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Iterative DNS resolution by enum module ?
by jyaim 22 Apr '15

22 Apr '15

Hello everyone, Do you know if the kamailio enum module is able to process 'recursives' dns requests in case of CNAME or NS response, until a NAPTR answer is returned ? Thanks in advance! Jy

2 1

Problems with git.kamailio.org server?
by kai.ohnacker＠cbc.de 22 Apr '15

22 Apr '15

Hello, is there actually a problem with the git.kamailio.org server? If I try the command git clone --depth 1 git://git.kamailio.org/kamailio kamailio or git clone --depth 1 git://git.sip-router.org/kamailio kamailio I got the message "fatal: read error: connection reset by peer". Perhaps it is a problem of my setup? Best regards, Kai Ohnacker

2 3

siptrace for keep alive OPTIONS
by Koray Vatansever 22 Apr '15

22 Apr '15

Hi, It seems that siptrace module doesn't trace requests which are sent out by dialog module (i.e. keep-alive OPTIONS and dialog timeout BYE). Is there any easy way to do that? any suggestion?

1 0

rtpengine and security
by GG GG 21 Apr '15

21 Apr '15

Hello, what do you think about opening all RTP ports for rtpengine on Internet, is it a bad practice ? I wonder if it's possible to use rtpengine with all ports closed. Maybe someone could explain how rtpengine learn the source address when the SDP contains a local address. If your rtpengine server is under attack, could rtpengine choose the wrong ip source for RTP ? Thanks.

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UAC Module
by Ali Jibran 21 Apr '15

21 Apr '15

Hi all. I am using UAC module to establish two trunks to my provider. I can see that the trunks are registered successfully but the calls are coming only from the first one. The provider says server isn't responding on the second one. How is that possible when it says registered? Can somebody help me out here.

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ACK and BYE routing - wrong interface problem and TCP/UDP conversion troubles
by Dmitry Sytchev 21 Apr '15

21 Apr '15

Hi all! I'm still working on incorrect routing of ACKs/BYEs with certain conditions and asking for guru help. My network looks like this: 192.168.4.218 -> TCP -> NAT (1.2.2.94) -> (1.2.0.160) Kamailio (1.2.0.160) -> UDP-> backend servers (1.2.2.44). IP address 1.2.0.16 is a 1.2.0.160/32 on loopback interface lo, and it is announced via BGP to neighbours. There is also 1.2.3.46/30, which is used to maintain BGP session. Backends are Asterisk 11 machines with "outboundproxy" sip peer setting set to Kamailio address. Kamailio forwards registrations to Asterisk servers with authentication and caches successful attempts to its internal DB. Natted clients are managed adding 'received' param to reg contact. When backend calls client, Kamailio takes correct URI from 'received' and forwards request. There is a possibility Kamailio will have several IP addresses for outbound clients. To specify source address of forwarded packets I set $fs. For initial requests and responses all works fine, but there are troubles when dealing with ACKs and BYEs. In case I don't specify interfaces with 'listen' , Kamailio listens on 1.2.0.160 and 1.2.3.46. I use $fs to set 1.2.0.160 as outgoing interface, and it works for requests. But when Kamailio forwards ACKs and BYEs, it uses 3.46 instead of 1.2.0.160, especially in case of TCP client and UDP backend. If I set mhomed = 0 and specify Kamailio to listen only to 1.2.0.160, everything works fine, except UDP <-> TCP signalling conversion. When natted TCP client receives call from backend, it answers with 200 OK and ACK from backend is never back to client. At the same time UDP<->UDP works fine. I've noted that TCP and UDP ACKs differ in Record-route headers, in case of TCP there are two records of Kamailio UDP/TCP. I have enable_double_rr = 1. UDP ACK: ACK sip:0097242@192.168.4.218:5063 SIP/2.0. Via: SIP/2.0/UDP 1.2.2.44:5060;branch=z9hG4bK31d1a26c;rport. Route: <sip:1.2.0.160;lr=on;ftag=as7af34000>. Max-Forwards: 70. From: <sip:8121230000@1.2.2.44>;tag=as7af34000. To: <sip:0097242@192.168.4.218:5063;received=sip:1.2.2.94:64825>;tag=3122411720. Contact: <sip:8121230000@1.2.2.44:5060>. Call-ID: 56c5362d2250df6777e6665523964729@1.2.2.44:5060. CSeq: 102 ACK. User-Agent: Zero vPBX. Content-Length: 0. This ack forwarded to destination correctly (to external IP 1.2.2.94:64825). Actually, I don't understand why, because $du is null and no functions logged that rURI was modified... TCP ACK: ACK sip:0097242@192.168.4.218:5062;transport=TCP SIP/2.0 Via: SIP/2.0/UDP 1.2.2.44:5060;branch=z9hG4bK08f7b9f2;rport Route: <sip:1.2.0.160;r2=on;lr=on;ftag=as5cc06d60>,<sip:1.2.0.160;transport=tcp;r2=on;lr=on;ftag=as5cc06d60> Max-Forwards: 70 From: <sip:8121230000@1.2.2.44>;tag=as5cc06d60 To: <sip:0097242@192.168.4.218:5062;transport=TCP;received=sip:1.2.2.94:57735;transport=TCP>;tag=4089422268 Contact: <sip:8121230000@1.2.2.44:5060> Call-ID: 3a72968b552bdb8875e3215431f1522d@1.2.2.44:5060 CSeq: 102 ACK User-Agent: Zero vPBX Content-Length: 0 This ack doesn't get forwarded anywhere, although Kamailio says: DEBUG: <core> [forward.c:592]: forward_request(): Sending:#012ACK sip:0097242@192.168.4.218:5062;transport=TCP SIP/2.0#015#012Via: SIP/2.0/TCP 1.2.0.160;branch=z9hG4bK42a6.5e57399c6a2c1dae96f28c7381b6c64e.0#015#012Via: SIP/2.0/UDP 1.2.2.44:5060;received=1.2.2.44;branch=z9hG4bK26130626;rport=5060#015#012Max-Forwards: 69#015#012From: <sip:8121230000@1.2.2.44>;tag=as74c521ed#015#012To: <sip:0097242@192.168.4.218:5062;transport=TCP;received=sip:1.2.2.94:65535;transport=TCP>;tag=547751151#015#012Contact: <sip:8121230000@1.2.2.44:5060>#015#012Call-ID: 10dfa7d00a9848c411a28bb54edf0f14@1.2.2.44:5060#015#012CSeq: 102 ACK#015#012User-Agent: Zero vPBX#015#012Content-Length: 0#015#012#015#012. So I'm trying to understand - how Kamailio get correct destination address to send to in case of UDP <-> UDP ack, and why the same doesn't work in case of TCP? If I turn on mhomed = 1, it breaks forwarding of UDP ACKs too... Maybe it is because of 1.2.0.160/32 IP address it is placed on lo iface, or there should be some entries in IP routing table on Kamailio host allowing selection of 1.2.0.160? Thanks in advance! -- Best regards, Dmitry Sytchev, IT Engineer

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Re: [SR-Users] registrar: add optional check for local path during lookup()
by Charles Chance 21 Apr '15

21 Apr '15

Hi Volkan, It is in the master (development) branch, so will appear in the next major release (4.3). The code could not be backported to 4.2 because it was considered a new feature rather than a fix. Cheers, Charles P.S. please use mailing list in case the question is useful to others :-) On 21 April 2015 at 12:57, Volkan Hatem <volkan(a)hatem.net> wrote: > Hi Charles, > > I was hoping to see your change ( > http://git.kamailio.org/gitlist/index.php/kamailio/commits/635f23b12eff2431…) > in 4.2.4 > > I could not tell to which branch your code was submitted. Do you know when > it might find its way to an official release? > > Best, > -volkan > -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

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MOH SIP SERVERS
by Kevin LeClair 19 Apr '15

19 Apr '15

Greetings, I'm new to the VoIP / SIP world but have successfully provisioned my new Yealink T48G and have begun to setup some of the many features. Of course one of the coolest features I found was the Music On Hold (MOH) with a SIP server. I configured the T48G to your sip:music@iptel.org which worked awesomely! My question is: Can I setup a SIP server with you to upload a default mp3 for using with my own MOH complete with SIP server URI? OR are there other services that do this? I have some of my own servers but I'm guessing that many servers do not allow the SIP MOH functionality? Thanks so much for your time. MaineWWW - A Maine Website Design Company Otter Creek, Maine 207-669-4269 <http://www.mainewww.com/> mainewww.com "One Site for All Devices" <http://www.mainewww.com/> maine_www_logo_small2

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uri=myself question
by Andres 18 Apr '15

18 Apr '15

Hi, I have a question about about 'myself' usage. for example: if(uri==myself) { log("the request is for local processing\n"); }; I have always expected the above to hold true for any 'alias' definition in the config file. But I never really though about the domain module since I use the test "is_uri_host_local()" for that. So my question is, if we have an entry in the domain module will it resolve to true for the "(uri==myself)" test even if it's not an alias? Thanks, -- Technical Support http://www.cellroute.net

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Kamailio relays ACK of 200 OK to itself
by Dmitry Sytchev 17 Apr '15

17 Apr '15

Hi all! I can't make my Kamailio to correctly relay stateful ACKs for 200 OK. It takes URI from Route: header and trying relay it to itself instead of sending it to host sent 200 OK. I don't understand why it can't match transaction and relay to correct host. Seems BYEs aren't relayed correctly either... 192.168.4.218 -> NAT 1.2.2.94 -> Kamailo (1.2.3.46, 1.2.2.219) -> 172.19.126.105 192.168.4.218 sends INVITE to 1.2.3.45 which relays it to 172.19.126.105. After 172.19.126.105 answers with 200 OK, it never gets ACK for it back. Instead, ACK relayed by proxy to 1.2.3.46 (one of proxy addresses). Example of original ACK: ACK sip:89211234567@172.19.126.105:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.4.218:5062;branch=z9hG4bK1161188624. Route: <sip:1.2.3.46;lr=on;ftag=259196205;nat=yes>. From: "0099977" <sip:0099977@1.2.3.46>;tag=259196205. To: <sip:89211234567@1.2.3.46>;tag=as6bfd1fa1. Call-ID: 910089142(a)192.168.4.218. CSeq: 2 ACK. Contact: <sip:0099977@192.168.4.218:5062>. Max-Forwards: 70. User-Agent: Yealink SIP-T46G 28.71.14.4. Content-Length: 0. It sees that host is not local: check_self: host != me I think the problem is somewhere in my NAT handling script code, but seems I'm stuck and can't get what am I doing wrong, so I'll greatly appreciate any help. Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <ACK> Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: INFO: <script>: Nat detected Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: INFO: <script>: Set contact alias Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [dset.c:774]: uri_add_rcv_alias(): encoded <sip:0099977@192.168.4.218:5062> => [sip:0099977@192.168.4.218:5062;alias =1.2.2.94~58542~1] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: siputils [checks.c:106]: has_totag(): totag found Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: INFO: <script>: Request in dialog Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: rr [loose.c:90]: is_preloaded(): is_preloaded: No Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [172.19.126.105] == [127.0.0. 1] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==12 && [172.19.126.105] == [1.2. 2.219] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==11 && [172.19.126.105] == [1.2. 3.46] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [172.19.126.105] == [127.0.0. 1] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==12 && [172.19.126.105] == [1.2. 2.219] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==11 && [172.19.126.105] == [1.2. 3.46] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [forward.c:448]: check_self(): check_self: host != me Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: rr [loose.c:674]: after_strict(): Next hop: 'sip:1.2.3.46;lr=on;ftag=259196205;nat=yes' is loose router Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: rr [loose.c:724]: after_strict(): The last route URI: 'sip:1.2.3.46;lr=on;ftag=259196205;nat=yes' Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: outbound [ob_mod.c:496]: use_outbound(): Analysing ACK for outbound markers Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [1.2.3.46] == [127.0.0.1] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==12 && [1.2.3.46] == [1.2.2.2 19] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [1.2.3.46] == [1.2.3.4 6] Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: outbound [ob_mod.c:466]: use_outbound_non_reg(): outbound not used Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: siputils [checks.c:106]: has_totag(): totag found Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: siputils [checks.c:106]: has_totag(): totag found Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: NOTICE: <script>: REL: ACK sip:1.2.3.46;lr=on;ftag=259196205;nat=yes -> sip:1.2.3.46;lr=on;ftag=259196205;nat=yes Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=59 , global msg id=58 , T on entrance=0xffffffff Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=37956, isACK=1 Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: tm [t_funcs.c:312]: t_relay_to(): SER: forwarding ACK statelessly Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: 2132d578ec20fa9b478ada51e9f2965f Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [forward.c:211]: get_out_socket(): DEBUG: get_out_socket: socket determined: 0xf6e777ec Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(1.2.2.94, 192.168.4.218, 0) Apr 16 19:21:33 kam-sbc-1-01 /usr/sbin/kamailio[9178]: DEBUG: <core> [forward.c:592]: forward_request(): Sending:#012ACK sip:1.2.3.46;lr=on;ftag=259196205;nat=yes SIP/2.0#015#012Record-Route: <sip:1.2.3.46;lr=on;ftag=259196205>#015#012Via: SIP/2.0/UDP 1.2.3.46;branch=z9hG4bK4449.2132d578ec20fa9b478ada51e9f2965f.0#015#012Via: SIP/2.0/UDP 192.168.4.218:5062;rport=58542;received=1.2.2.94;branch=z9hG4bK1161188624#015#012From: "0099977" <sip:0099977@1.2.3.46>;tag=259196205#015#012To: <sip:89215572714@1.2.3.46>;tag=as6bfd1fa1#015#012Call-ID: 910089142(a)192.168.4.218#015#012CSeq: 2 ACK#015#012Contact: <sip:0099977@192.168.4.218:5062;alias=1.2.2.94~58542~1>#015#012Max-Forwards: 69#015#012User-Agent: Yealink SIP-T46G 28.71.14.4#015#012Content-Length: 0#015#012#015#012. -- Best regards, Dmitry Sytchev, IT Engineer

2 4

Loadbalancing proxy using DNS SRV instead of dispatcher
by Daniel Tryba 17 Apr '15

17 Apr '15

The idea is to use SRV records instead of a dispatcher by setting $rd to a DNS SRV enabled host (with all hosts having the same prio and weight). Problem is that when authentication is required the flow is as follows (with 2 hosts): 1: UAC INVITE/REGISTER -> LB -> SIP0 SIP0 401 with nonce1 challenge -> LB -> UAC UAC ACK -> LB -> SIP0 2: UAC INVITE/REGISTER with nonce1 response -> LB -> SIP1 SIP1 401 with nonce2 challenge -> LB -> UAC UACK ACK -> LB -> SIP1 3: UAC INVITE/REGISTER with nonce2 response -> LB -> SIP0 SIP0 401 with nonce1 challenge -> LB -> UAC UACK ACK -> LB -> SIP0 4: UAC either gives up or goes to 2 Using the dispatcher module this can be solved by using a hash over either from/to/request URI. I can't find any hints for forcing the dialog to 1 backend. BTW the idea is to keep the loadbalancing proxy as simple/stupid as possible (no db), the backends have a shared db.

4 7

memcached module trimming the leading 8 char with innodb module
by Ethan Brooks 16 Apr '15

16 Apr '15

This is very simple Kamailio config and when i access Memcached directly everything seem to be fine but, when accessed via Kamailio he leading 8 char is omitted. I have included everything needed to test. kamailio mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" debug=0 fork=yes loadmodule "tm.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "xhttp.so" loadmodule "memcached.so" modparam("memcached", "servers", "localhost:11211") port=80 tcp_accept_no_cl=yes event_route[xhttp:request] { $var(number) = $hu; $var(number) = $(var(number){s.substr,2,0}); xhttp_reply("200", "OK", "text/html", "value: $mct(@@lnp.$var(number))"); } mysql: source /usr/share/mysql/innodb_memcached_config.sql install plugin daemon_memcached soname "libmemcached.so"; CREATE SCHEMA lnp; CREATE TABLE `lnp`.`lnp` ( `number` bigint(20) NOT NULL, `ported` bigint(20) DEFAULT NULL, PRIMARY KEY (`number`) ) ENGINE=InnoDB DEFAULT CHARSET=utf8; insert into lnp.lnp (number,ported) values (8055551212,8053214321); insert into innodb_memcache.containers (name,db_schema,db_table,key_columns,value_columns,flags,cas_column, expire_time_column,unique_idx_name_on_key) values ('lnp','lnp','lnp','number','ported', 0,0,0,'PRIMARY'); url: http://localhost/?8055551212 reponse: value: 21 memcached # nc localhost 11211 get @@lnp.8055551212 VALUE @@lnp.8055551212 0 10 8053214321 END

2 1

Kamailio + Mediaproxy - How to set up
by Filip Malenka 16 Apr '15

16 Apr '15

Hi all, I have searched for instructions for setting up Kamailio with mediaproxy instead of rtpproxy, but I can't find anything. I found only OpenSips pages and some other outdated examples but the config does not apply very much on kamailio.cfg. As much as I could find out is that Kamailio natively supports rtpproxy and OpenSips works with mediaproxy. Is there any solution for Kamailio with mediaproxy? Thank you! Filip -- Sent from my Android device with K-9 Mail. Please excuse my brevity.

3 5

SipML5-Kamailio+RtpEngine-ThirdpartySipServers- Not All SIP Servers are working
by Karthik Kondapaneni 16 Apr '15

16 Apr '15

Can someone help me please … I am using this setup . Sipml5 webrtc client(chrome) -----> Kamailio with RTP engine ------> ThridpartySip Server kamalio will act as mediator between webRTC client and NonWebRTC sip server . User will be registered on sip server and calls will be terminated to PSTN . This setup is working fine with one thirdparty sip server but not working with other SIP server . First server ( working ) is happily replying INVITE request and reponding with "TRYING " . But Second SIP server never reply to kamailio for INVITE requests and kamailio keep sending SDP and finally on SipML5 it shows "request timed out " Ports are open on Firewall . SIP Server 1: 192.168.102.237 SIP Server 2: 192.168.87.101 Kamailio IP : 172.17.185.46 Client PUblic ip : 79.207.241.94 1. Working Server Trace ---- INVITE sip:917638900890@192.168.102.237 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKJHBbZaGh9noFlUgS1VVr8DCImVbITu0q;rport=54273 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016"<sip:testcard@df7jal23ls0d.invalid ;alias=79.207.241.94~54273~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 INVITE Content-Type: application/sdp Content-Length: 819 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8650276303411124000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG m=audio 30246 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2398380477 cname:8WcQLQ8hOhk5MhWZ a=ssrc:2398380477 msid:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG 63b396a3-5126-405f-a389-75548cf0a634 a=ssrc:2398380477 mslabel:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG a=ssrc:2398380477 label:63b396a3-5126-405f-a389-75548cf0a634 a=sendrecv a=rtcp:30247 SIP/2.0 407 Proxy Authentication Required Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKJHBbZaGh9noFlUgS1VVr8DCImVbITu0q;rport=54273 From: "2016" <sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016" <sip:testcard@192.168.102.237:5060> Call-Id: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 INVITE Proxy-Authenticate: Digest realm="sip.dingdongbell.net", nonce="c2133e5e1e40805244c5f201cfbaee4d", algorithm=MD5 Content-Length: 0 ACK sip:917638900890@192.168.102.237 SIP/2.0 Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 ACK Content-Length: 0 Max-Forwards: 69 INVITE sip:917638900890@192.168.102.237 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK0929.61a39db61290521db7dc657d252c4182.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKnswxFdcreaCt0SMSRCqs2HPPOSN9adR6;rport=54273 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016"<sip:testcard@df7jal23ls0d.invalid ;alias=79.207.241.94~54273~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62566 INVITE Content-Type: application/sdp Content-Length: 819 Max-Forwards: 69 Proxy-Authorization: Digest username="testcard",realm="sip.dingdongbell.net ",nonce="c2133e5e1e40805244c5f201cfbaee4d",uri=" sip:917638900890@192.168.102.237 ",response="fb9c97e380c4231725d1119256f7f7d9",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8650276303411124000 2 IN IP4 172.17.185.46 s=DD- chrome t=0 0 a=msid-semantic: WMS tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG m=audio 30246 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2398380477 cname:8WcQLQ8hOhk5MhWZ a=ssrc:2398380477 msid:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG 63b396a3-5126-405f-a389-75548cf0a634 a=ssrc:2398380477 mslabel:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG a=ssrc:2398380477 label:63b396a3-5126-405f-a389-75548cf0a634 a=sendrecv a=rtcp:30247 SIP/2.0 100 Trying Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK0929.61a39db61290521db7dc657d252c4182.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKnswxFdcreaCt0SMSRCqs2HPPOSN9adR6;rport=54273 From: "2016" <sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: <sip:917638900890@192.168.102.237:5060> Call-Id: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62566 INVITE Content-Length: 0 2. Not working Server trace ( second server)-------------------- INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 Kamailio Config :----------------------------------- request_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); if (!is_method("REGISTER|INVITE|ACK|BYE|CANCEL|PRACK|MESSAGE|INFO|UPDATE")) { sl_send_reply("405", "Method not allowed"); exit; } # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route_preset("172.17.185.46:5090;nat=yes"); # handle registrations route(REGISTRAR); # if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; # } # if (!is_method("INVITE")) { # route(RELAY); # exit; # } route(RELAY); } route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); if(sdp_with_transport("RTP/SAVPF")) { rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force"); t_on_reply("REPLY_WS_TO_WS"); return; } # rtpengine_manage("froc+SP"); rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); # rtpengine_manage("froc-sp"); rtpengine_manage("force rtcp-mux-demux symmetric trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); # rtpengine_manage("co"); rtpengine_manage("replace-origin replace-session-connection"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } route[RELAY] { if (is_method("INVITE")) { route(SETUP_BY_TRANSPORT); if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; }

1 0

Kamailio load statistics
by Mickael Marrache 16 Apr '15

16 Apr '15

Hi, What are the statistics that should be used to monitor a Kamailio instance? I currently looked at shmem_real_used_size, shmem_used_size, dialogs_active_dialogs. However, I'm looking for other statistics that represent the load on the instance. Thanks, Mickael

1 0

rtpengine "SRT[C]P output wanted, but no crypto suite" messages
by Juha Heinanen 16 Apr '15

16 Apr '15

some calls that use rtpengine produce "SRT[C]P output wanted, but no crypto suite was negotiated" syslog messages. below is an example. the messages seem to appear after rtpengine has replied to offer request, but before it has got answer request that includes crypto suite negotiation in its sdp. the calls work fine and i was wondering if these syslog message are really needed? -- juha Apr 16 08:34:37 box /usr/bin/sip-proxy[18719]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1) Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Received command 'offer' from 127.0.0.1:46713 Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Dump for 'offer' from 127.0.0.1:46713: { "sdp": "v=0#015#012o=- 2909773022 619612917 IN IP4 192.168.43.192#015#012s=-#015#012c=IN IP4 192.168.43.192#015#012t=0 0#015#012a=tool:baresip 0.4.12#015#012m=audio 10022 RTP/SAVP 96 97 8 0 101#015#012b=AS:128#015#012a=rtpmap:96 speex/16000#015#012a=fmtp:96 mode="7";vbr=off;cng=on#015#012a=rtpmap:97 speex/8000#015#012a=fmtp:97 mode="7";vbr=off;cng=on#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=label:1#015#012a=rtcp-rsize#015#012a=ssrc:227522320 cname:sip:test@tutpro ... Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] ... .com#015#012a=rtcp-mux#015#012a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:lRfv9v0tfQAKXiJihqbgr7rNS5Gh9txvuqPv9Tt0#015#012a=ptime:20#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "call-id": "dd63b7bcd45655b8", "via-branch": "z9hG4bK83c7fd2c1c254f15", "received-from": [ "IP4", "94.100.20.99" ], "from-tag": "5408ab343077ba14", "command": "offer" } Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Creating new call Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Opened ports 8046..8047 for media relay Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Opened ports 8056..8057 for media relay Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Shutting down ICE agent (nothing to do) Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Replying to 'offer' from 127.0.0.1:46713 Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Response dump for 'offer' to 127.0.0.1:46713: { "sdp": "v=0#015#012o=- 2909773022 619612917 IN IP4 194.40.50.30#015#012s=-#015#012c=IN IP4 194.40.50.30#015#012t=0 0#015#012a=tool:baresip 0.4.12#015#012m=audio 8046 RTP/SAVP 96 97 8 0 101#015#012b=AS:128#015#012a=rtpmap:96 speex/16000#015#012a=fmtp:96 mode="7";vbr=off;cng=on#015#012a=rtpmap:97 speex/8000#015#012a=fmtp:97 mode="7";vbr=off;cng=on#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=label:1#015#012a=rtcp-rsize#015#012a=ssrc:227522320 cname:sip:test@tutpro.com#015#012a= ... Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] ... ptime:20#015#012a=sendrecv#015#012a=rtcp:8047#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:lRfv9v0tfQAKXiJihqbgr7rNS5Gh9txvuqPv9Tt0#015#012a=setup:actpass#015#012a=fingerprint:sha-1 F3:4F:0A:B8:75:BF:B6:29:45:8A:C2:74:8E:F9:10:F2:CB:10:50:37#015#012a=ice-ufrag:eUJUwfKq#015#012a=ice-pwd:bDuCQKxJTV5voKlVAarucAiqv1#015#012a=candidate:FvaQOQwZdXFAyxXW 1 UDP 2130706431 194.40.50.30 8046 typ host#015#012a=candidate:FvaQOQwZdXFAyxXW 2 UDP 2130706430 194.40.50.30 8047 typ host#015#012", "result": "ok" } Apr 16 08:34:37 box /usr/bin/sip-proxy[18719]: INFO: Routing INVITE <sip:jh@10.13.40.239:53269;transport=TLS;ob> to contact via <sip:94.100.20.99:53269;transport=TLS> Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8 port 8047] SRTCP output wanted, but no crypto suite was negotiated Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8 port 8046] SRTP output wanted, but no crypto suite was negotiated Apr 16 08:34:37 box /usr/bin/sip-proxy[18718]: INFO: ===== rtpengine_answer(ICE=force via-branch=2) Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Received command 'answer' from 127.0.0.1:51555 Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Dump for 'answer' from 127.0.0.1:51555: { "sdp": "v=0#015#012o=- 3638151276 3638151277 IN IP4 10.13.40.239#015#012s=pjmedia#015#012c=IN IP4 10.13.40.239#015#012t=0 0#015#012m=audio 4002 RTP/SAVP 96 101#015#012c=IN IP4 10.13.40.239#015#012a=rtcp:4003 IN IP4 10.13.40.239#015#012a=sendrecv#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=zrtp-hash:1.10 4261405ee55bb84370419b56f37c3966505a583ff6fee02e1c1b854cfccb5d99#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8/oEJd3pVqWg6tvAJkItNBxQ60sJS44mnwriZRrt#015#012a=rtpmap:96 speex/16000#015#012", "ICE": "forc ... Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] ... e", "call-id": "dd63b7bcd45655b8", "via-branch": "z9hG4bK83c7fd2c1c254f15", "received-from": [ "IP4", "94.100.20.99" ], "from-tag": "5408ab343077ba14", "to-tag": "Zi.x-8OwwXlfKNC-67q3D7M7PeMqbmGh", "command": "answer" } Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Replying to 'answer' from 127.0.0.1:51555 Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] Response dump for 'answer' to 127.0.0.1:51555: { "sdp": "v=0#015#012o=- 3638151276 3638151277 IN IP4 10.13.40.239#015#012s=pjmedia#015#012c=IN IP4 10.13.40.239#015#012t=0 0#015#012m=audio 8056 RTP/SAVP 96 101#015#012c=IN IP4 194.40.50.30#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=zrtp-hash:1.10 4261405ee55bb84370419b56f37c3966505a583ff6fee02e1c1b854cfccb5d99#015#012a=rtpmap:96 speex/16000#015#012a=sendrecv#015#012a=rtcp:8057#015#012a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:8/oEJd3pVqWg6tvAJkItNBxQ60sJS44mnwriZRrt#015#012a=candidate:FvaQOQwZdXFAyxX ... Apr 16 08:34:37 box rtpengine[31871]: [dd63b7bcd45655b8] ... W 1 UDP 2130706431 194.40.50.30 8056 typ host#015#012a=candidate:FvaQOQwZdXFAyxXW 2 UDP 2130706430 194.40.50.30 8057 typ host#015#012", "result": "ok" }

1 0

Re: [SR-Users] No CDR is written when dialog timeouts
by Mickael Marrache 15 Apr '15

15 Apr '15

I'm looking at the dlg_ontimeout function [dlg_handlers.c:1369] and I understand the code, the DLGCB_EXPIRED callbacks are not called when the send_bye parameter is set to 1. I'm looking at the block 1399-1408. From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Wednesday, April 15, 2015 3:53 PM To: sr-users(a)lists.sip-router.org Subject: RE: No CDR is written when dialog timeouts I also confirm the cdr_on_create() [acc_cdr.c:718] function is called since I can see the debug statement "dialog '%p' created!". From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Wednesday, April 15, 2015 3:48 PM To: sr-users(a)lists.sip-router.org Subject: No CDR is written when dialog timeouts Hi, I'm testing the dialog timeout scenario and I see that no CDR is written. This is my configuration: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_log_enable", 0) modparam("acc", "cdr_on_failed", 0) I can confirm the cdr_on_expired() [acc_cdr.c:684] function is not called since I don't see the debug statement "dialog '%p' expired!\n". (I precise that CDR generation works fine for regular hangup scenario.) Any idea what can cause the issue? Thanks, Mickael

3 8

Differentiating between regular hangup and dialog timeout
by Mickael Marrache 15 Apr '15

15 Apr '15

Hi, I'm looking for a way to differentiate between regular hangup and dialog timeout. In both case, the dialog:end event is called. Do you know which information can I use in the event route to differentiate between the two cases? Thanks, Mickael

2 3

Inconsistent dlg timer data warning at dialog timeout
by Mickael Marrache 15 Apr '15

15 Apr '15

Hi, I get the following warning when a dialog timeout occurs: kamailio[18889]: WARNING: dialog [dlg_req_within.c:214]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7f422986e448 [3547:6673] with clid 'mycvJWgmbmOSD5XQrcStKAKoTfW12XeR' and tags 'C5mlsaZCdTmybNSYgvWr75MBxWa5hVyA' 'L1ZzszpZ1MYO6qlQQ3FYYRWQYtF4F75r' Am I doing something wrong? Thanks, Mickael

3 6

Re: [SR-Users] No CDR is written when dialog timeouts
by Mickael Marrache 15 Apr '15

15 Apr '15

I also confirm the cdr_on_create() [acc_cdr.c:718] function is called since I can see the debug statement "dialog '%p' created!". From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Wednesday, April 15, 2015 3:48 PM To: sr-users(a)lists.sip-router.org Subject: No CDR is written when dialog timeouts Hi, I'm testing the dialog timeout scenario and I see that no CDR is written. This is my configuration: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_log_enable", 0) modparam("acc", "cdr_on_failed", 0) I can confirm the cdr_on_expired() [acc_cdr.c:684] function is not called since I don't see the debug statement "dialog '%p' expired!\n". (I precise that CDR generation works fine for regular hangup scenario.) Any idea what can cause the issue? Thanks, Mickael

1 0

No CDR is written when dialog timeouts
by Mickael Marrache 15 Apr '15

15 Apr '15

Hi, I'm testing the dialog timeout scenario and I see that no CDR is written. This is my configuration: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_log_enable", 0) modparam("acc", "cdr_on_failed", 0) I can confirm the cdr_on_expired() [acc_cdr.c:684] function is not called since I don't see the debug statement "dialog '%p' expired!\n". (I precise that CDR generation works fine for regular hangup scenario.) Any idea what can cause the issue? Thanks, Mickael

1 0

Shared location database used by independend kamailios
by Daniel Tryba 15 Apr '15

15 Apr '15

Is there an easy way to figure out on which server a uac is registered when using a shared database (modparam("usrloc", "db_mode", 3)). When uac1 is registered on server1 (dns srv loadbalancing) and uac2 is on server2. A call from uac1 to uac2 with a simple lookup("location") will result in server1 directly trying to connect to uac2. If this device is natted (or behind a stateful firewall) this will fail. In the location table the column socket contains the server which received the request. So if I know this value I can have server2 route the call to this value/server and have the call delivered to uac1 via an "existing connection". The idea was to use reg_fetch_contacts to fetch this/these socket values in $ulc and in case they are not local redirect/branch to those servers. But if the value of socket isn't local the $ulc for socket is null. I could use sqlops to fetch this manually, but is there an easier way I am missing?

4 9

DNS resolution and logical domain
by Mickael Marrache 15 Apr '15

15 Apr '15

Hi, I sometimes get the following errors: kamailio[16098]: ERROR: <core> [resolve.c:1693]: sip_hostport2su(): could not resolve hostname: "mycompany.com" kamailio[16098]: ERROR: tm [ut.h:319]: uri2dst2(): failed to resolve "mycompany.com" mycompany.com cannot be resolved using DNS (there is no corresponding DNS entry). I use this as a logical domain and I don't expect Kamailio to resolve it. Also, I set use_domain (usrloc module) to 0 so only the RURI username is used when searching locations. The flow is as follows: -the proxy receives the INVITE with sip: <sip:%3cuser%3e@myloadbalancerip:5060> <user>@myloadbalancerip:5060 which then forwards the INVITE to a proxy -the proxy looks for <user> internally and finds that this user has multiple subscribers (endpoints) -for each subscriber, a branch is appended using append_branch (or the RURI is replaced for the first subscriber) - the new RURI for each branch is sip: <sip:%3csubscriber%3e@mycompany.com> <subscriber>@mycompany.com -finally, lookup_branches is called to look for locations for every subscriber lookup_branches only uses the subscriber (RURI username) for searching locations since use_domain is set to 0. However, I sometimes get the errors I mentioned. Any idea why Kamailio sometimes tries to resolve the domain name? Thanks, Mickael

2 1

Error protoaddr2proxy
by Igor Potjevlesch 15 Apr '15

15 Apr '15

Hello, I got many errors like this one: ERROR: tm [tm.c:1192]: t_protoaddr2proxy(): tm: protoaddr2proxy: host name is empty It looks to be while I'm doing a sl_send_reply but I don't figure out why. Someone know what going wrong? Regards, Igor.

2 2

Carrierroute question
by Nelson Migliaro 14 Apr '15

14 Apr '15

Hello everyone, I have a SIP provider that gives me access to different routes over the same IP address but different prefixes. I have to send all calls to the same IP but I need to inlcude a prefix to allow vendor to route calls to different carriers. I will use carrierroute (cr) module and in case prefix 1 fails, I should send the call to prefix 2 and so on. The problem I have is cr module detects that this host (public vendor ip) was already used and I am getting this warning: NOTICE: carrierroute [cr_func.c:324]: cr_uri_already_used(): Candidate destination This machine use public DNS and I tried to set up 3 entries in /etc/hosts public_iP vendor_1 public_iP vendor_2 public_iP vendor_3 and instead of using the IP in rewrite_host I changed to the name: vendor_1, vendor_2. vendor_3 but I got error: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "vendor_1" Is there a better war to resolve this Thank you in advance

2 2

Error configuring kamailio with sql.
by Satish Verma 14 Apr '15

14 Apr '15

Respected sir, I want to configure SIP voip server using tutorial ( http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour). i have configured everything on my ubuntu server as given in tutorial. but i am getting error while running start command ( /etc/init.d/kamailio start) the error as given below. root@wifi:/etc/kamailio# /etc/init.d/kamailio start * Not starting Kamailio SIP Server: invalid configuration file! * * 0(32729) INFO: tls [tls_init.c:401]: init_tls_compression(): tls: init_tls: disabling compression... 0(32729) ERROR: <core> [cfg.y:3288]: yyparse(): cfg. parser: failed to find command force_rtp_proxy (params 0) 0(32729) : <core> [cfg.y:3428]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing loadmodule? 0(32729) ERROR: <core> [cfg.y:3288]: yyparse(): cfg. parser: failed to find command force_rtp_proxy (params 0) 0(32729) : <core> [cfg.y:3428]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing loadmodule? ERROR: bad config file (2 errors) i have tried running using default kamailio.cfg then it works fine. but i want to configure using script uploaded by u. please do the needful. i will be very thankful to you. With regards, Satish verma, Programming Assistant, UIET, Panjab University, Chandigarh. Mobile no +917307737732

3 2

Re: [SR-Users] CDR fields are all strings
by Mickael Marrache 14 Apr '15

14 Apr '15

I created a pull request that I tested using MongoDB. It works fine. Let me know if I can help more. Mickael From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Monday, April 13, 2015 5:13 PM To: miconda(a)gmail.com; 'Kamailio (SER) - Users Mailing List' Subject: RE: [SR-Users] CDR fields are all strings We are using MongoDB and all the values are inserted as strings. I also tried using the time_mode and time_format parameters, but it looks like they are only relevant for transactions accounting. Regarding the duration, it is set as a string for any backend. I think this behavior strongly limits us (and probably others) performing queries on CDRs (especially queries based on start and end time). Mickael From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel-Constantin Mierla Sent: Monday, April 13, 2015 3:03 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] CDR fields are all strings Hello, On 13/04/15 11:48, Mickael Marrache wrote: Hi, Why are all the CDR fields inserted as strings? For extra fields, I can understand it is easier to implement. However, for the built in fields (e.g. duration), we need to perform queries based on duration and time and it is not really possible using strings. Dialog variables hold only strings and those values are stored in variables, iirc -- that was the design behind the concept done by initial developer. Perhaps they can be easily updated to hold integers. Anyhow, iirc, at least with mysql it should work to insert any value enclosed in quotes, not matter the field type. For datetime that is acutually the default -- to use quotes, afaik. So if you change the types of columns, it should just work. We can push a patch if most of the people think is better to use different column types. Cheers, Daniel I see that the time field for failed transactions is inserted as a date. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

Playing with presence module and offline AoR
by Roberto Fichera 14 Apr '15

14 Apr '15

Hi All, I've started to play with presence module on kamailo 4.1.6, everything seems to work quite well up to now except one thing. I would like to notify the presence status of a NOT currently registered AoR as not online instead of not present. How can I do that? Thanks in advance, Roberto Fichera.

1 0

Timeout session Kamailio w/ rtpproxy v2
by Igor Potjevlesch 13 Apr '15

13 Apr '15

Hi everyone, I've sent my request on the rtpproxy google group without answer so try here. I'm using Kamailio 4.2.3 (and few older versions) with the new release of rtpproxy. I can't get the info of a session timeout (-n or -i options) from rtpproxy to my Kamailio proxy, using -n and the timeout socket. I tried: - Kamailio side: *modparam("rtpproxy", "timeout_socket", "unix:/var/run/rtpproxy_timeout.sock")* - rtpproxy side: *-n **unix:/var/run/rtpproxy_timeout.sock* and other configuration like tcp, or udp socket, but nothing is happening on the Kamailio part. The Kamailio rtpproxy man page give the example: *modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2 <http://127.0.0.1:8000/RPC2>")* Does it mean Kamailio only handles XMLRPC socket ? In this case, is rtpproxy able to handle XMLRPC (I remember a patch in an older rtpproxy version) In the case Kamailio could handle other socket types, which one could allow me to get a session timeout signal from rtpproxy in order to generate BYE message at the signalling side ? I could add that rtpproxy stops well RTP sessions that timed out. Regards, Igor Potjevlesc

2 1

Re: [SR-Users] Avoiding "script writer didn't release transaction" warnings
by Mickael Marrache 13 Apr '15

13 Apr '15

Any idea? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 2:08 PM To: sr-users(a)lists.sip-router.org Subject: RE: Avoiding "script writer didn't release transaction" warnings I think understand in which case it happens. When I receive a REGISTER, I create a transaction since I need to use the t_continue feature to wake up a suspended transaction. However, the save() command of the REGISTRAR module doesn't seem to call t_reply for replying the 200 (it looks like the SL module is used instead). Therefore, I get the warning every time it happens since the transaction is not released explicitly. Should I call t_release just before exiting the block handling the REGISTER? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 1:43 PM To: sr-users(a)lists.sip-router.org Subject: Avoiding "script writer didn't release transaction" warnings Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

2 6

Maximum feasible size of htable?
by Brooks Bridges 13 Apr '15

13 Apr '15

I have a small-ish project that could benefit from using an htable for storage of very basic data that doesn't change much, and I'd like to get an idea of what the group's experience is with storing larger amounts of data? Something in the 15-20GB range. Yes I could use Redis, but the idea here is to minimize the amount of dependencies and processing required. Thoughts? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 2

CDR fields are all strings
by Mickael Marrache 13 Apr '15

13 Apr '15

Hi, Why are all the CDR fields inserted as strings? For extra fields, I can understand it is easier to implement. However, for the built in fields (e.g. duration), we need to perform queries based on duration and time and it is not really possible using strings. I see that the time field for failed transactions is inserted as a date. Thanks, Mickael

2 2

Sipml5-Kamailio -RtpEngine -ThridpartySipServers- Not all sip servers working
by Karthik Kondapaneni 13 Apr '15

13 Apr '15

Dear experts , I am using this setup . Sipml5 webrtc client(chrome) -----> Kamailio with RTP engine ------> ThridpartySip Server kamalio will act as mediator between webRTC client and NonWebRTC sip server . User will be registered on sip server and calls will be terminated to PSTN . This setup is working fine with one thirdparty sip server but not working with other SIP server . First server ( working ) is happily replying INVITE request and reponding with "TRYING " . But Second SIP server never reply to kamailio for INVITE requests and kamailio keep sending SDP and finally on SipML5 it shows "request timed out " Ports are open on Firewall . SIP Server 1: 192.168.102.237 SIP Server 2: 192.168.87.101 Kamailio IP : 172.17.185.46 Client PUblic ip : 79.207.241.94 1. Working Server Trace ---- INVITE sip:917638900890@192.168.102.237 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKJHBbZaGh9noFlUgS1VVr8DCImVbITu0q;rport=54273 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016"<sip:testcard@df7jal23ls0d.invalid ;alias=79.207.241.94~54273~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 INVITE Content-Type: application/sdp Content-Length: 819 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8650276303411124000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG m=audio 30246 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2398380477 cname:8WcQLQ8hOhk5MhWZ a=ssrc:2398380477 msid:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG 63b396a3-5126-405f-a389-75548cf0a634 a=ssrc:2398380477 mslabel:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG a=ssrc:2398380477 label:63b396a3-5126-405f-a389-75548cf0a634 a=sendrecv a=rtcp:30247 SIP/2.0 407 Proxy Authentication Required Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKJHBbZaGh9noFlUgS1VVr8DCImVbITu0q;rport=54273 From: "2016" <sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016" <sip:testcard@192.168.102.237:5060> Call-Id: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 INVITE Proxy-Authenticate: Digest realm="sip.dingdongbell.net", nonce="c2133e5e1e40805244c5f201cfbaee4d", algorithm=MD5 Content-Length: 0 ACK sip:917638900890@192.168.102.237 SIP/2.0 Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 ACK Content-Length: 0 Max-Forwards: 69 INVITE sip:917638900890@192.168.102.237 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK0929.61a39db61290521db7dc657d252c4182.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKnswxFdcreaCt0SMSRCqs2HPPOSN9adR6;rport=54273 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016"<sip:testcard@df7jal23ls0d.invalid ;alias=79.207.241.94~54273~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62566 INVITE Content-Type: application/sdp Content-Length: 819 Max-Forwards: 69 Proxy-Authorization: Digest username="testcard",realm="sip.dingdongbell.net ",nonce="c2133e5e1e40805244c5f201cfbaee4d",uri=" sip:917638900890@192.168.102.237 ",response="fb9c97e380c4231725d1119256f7f7d9",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8650276303411124000 2 IN IP4 172.17.185.46 s=DD- chrome t=0 0 a=msid-semantic: WMS tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG m=audio 30246 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2398380477 cname:8WcQLQ8hOhk5MhWZ a=ssrc:2398380477 msid:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG 63b396a3-5126-405f-a389-75548cf0a634 a=ssrc:2398380477 mslabel:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG a=ssrc:2398380477 label:63b396a3-5126-405f-a389-75548cf0a634 a=sendrecv a=rtcp:30247 SIP/2.0 100 Trying Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK0929.61a39db61290521db7dc657d252c4182.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKnswxFdcreaCt0SMSRCqs2HPPOSN9adR6;rport=54273 From: "2016" <sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: <sip:917638900890@192.168.102.237:5060> Call-Id: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62566 INVITE Content-Length: 0 2. Not working Server trace ( second server)-------------------- INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 Kamailio Config :----------------------------------- request_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); if (!is_method("REGISTER|INVITE|ACK|BYE|CANCEL|PRACK|MESSAGE|INFO|UPDATE")) { sl_send_reply("405", "Method not allowed"); exit; } # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route_preset("172.17.185.46:5090;nat=yes"); # handle registrations route(REGISTRAR); # if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; # } # if (!is_method("INVITE")) { # route(RELAY); # exit; # } route(RELAY); } route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); if(sdp_with_transport("RTP/SAVPF")) { rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force"); t_on_reply("REPLY_WS_TO_WS"); return; } # rtpengine_manage("froc+SP"); rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); # rtpengine_manage("froc-sp"); rtpengine_manage("force rtcp-mux-demux symmetric trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); # rtpengine_manage("co"); rtpengine_manage("replace-origin replace-session-connection"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } route[RELAY] { if (is_method("INVITE")) { route(SETUP_BY_TRANSPORT); if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; }

1 0

End time in CDR
by Mickael Marrache 13 Apr '15

13 Apr '15

Hi, Very short question. Why the time at which the dialog ended is also included in the CDRs? There already are the start time and duration. Is it there to avoid end time computation? Mickael

1 0

Dialog based CDRs and failed dialogs
by Mickael Marrache 13 Apr '15

13 Apr '15

Hi, I'm using the ACC module to generate dialog based CDRs and I see that CDRs are generated even for failed calls. For example, a CDR is generated for cancelled calls. At first, I thought that dialog based CDRs are generated only for successfully established dialogs but it doesn't seem to be the case. Thanks, Mickael

2 4

Re: [SR-Users] MULTIDOMAIN Environment
by Bruno Emer 13 Apr '15

13 Apr '15

Hello Daniel! Thanks for your help again!! I placed the instruction just after the "authentication" and now it is working as I want! Bruno Emer 2015-04-09 15:33 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a scenario here where I need to put more than two companies into > the same Kamailio server and to accomplish this I'm using a multi-domain > configuration. > > For now, I am able to create users in different domains and register all > of them. The users are able to place and receive internal calls (as I am > using this just to get internal communications working), so I am ok with > this part of my configuration. > > The problem that I'm facing now is that users from domain "A" are able to > place calls to users from domain "B". I need to deny calls between > different domains, and I don't know the best way to achieve this. > > Here are my domains create (output from "kamctl domain show"): > > domain:: enterprise.com > did:: enterprise.com > domain:: enterprise2.com > did:: enterprise2.com > > > And here are my users created: > > username: bob > domain: enterprise.com > > username: alice > domain: enterprise2.com > > > So, can you please help me to get this? > > > Here is my .cfg file: > > > #!KAMAILIO > #!define WITH_PGSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > ##!define WITH_NAT > ##!define WITH_DEBUG > #!define WITH_MULTIDOMAIN > ##!define WITH_SIPTRACE > #!define WITH_SQLOPS > #!define WITH_XMLRPC > ##!define WITH_603 > #!define WITH_TLS > # > > > #!substdef "!MY_WS_PORT!80!g" > #!substdef "!MY_WSS_PORT!81!g" > #!substdef "!MY_WS_ADDR!eth1:MY_WS_PORT!g" > #!substdef "!MY_WSS_ADDR!tls:eth1:MY_WSS_PORT!g" > > #!define WITH_WEBSOCKETS > > > #!ifdef ACCDB_COMMENT > ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; > ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default > ''; > ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL > DEFAULT ''; > #!endif > > ####### Include Local Config If Exists ######### > import_file "kamailio-local.cfg" > > ####### Defined Values ######### > > # *** Value defines - IDs used later in config > #!ifdef WITH_PGSQL > # - database URL - used to connect to database server by modules such > # as: auth_db, acc, usrloc, a.s.o. > #!define DBURL "postgres://kamailio:kamailiorw@localhost/kamailio" > #!endif > #!ifdef WITH_MULTIDOMAIN > # - the value for 'use_domain' parameters > #!define MULTIDOMAIN 1 > #!else > #!define MULTIDOMAIN 0 > #!endif > > # - flags > # FLT_ - per transaction (message) flags > # FLB_ - per branch flags > #!define FLT_ACC 1 > #!define FLT_ACCMISSED 2 > #!define FLT_ACCFAILED 3 > #!define FLT_NATS 5 > > #!define FLB_NATB 6 > #!define FLB_NATSIPPING 7 > > ####### Global Parameters ######### > > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > #!ifdef WITH_DEBUG > debug=4 > log_stderror=yes > #!else > debug=2 > log_stderror=no > #!endif > > > memdbg=5 > memlog=5 > > log_facility=LOG_LOCAL0 > > fork=yes > children=4 > > /* uncomment the next line to disable TCP (default on) */ > disable_tcp=no > tcp_accept_no_cl=yes > > /* uncomment the next line to disable the auto discovery of local aliases > based on reverse DNS on IPs (default on) */ > #auto_aliases=no > > /* add local domain aliases */ > #alias="sip.mydomain.com" > > /* uncomment and configure the following line if you want Kamailio to > bind on a specific interface/port/proto (default bind on all available) > */ > #listen=udp:10.0.0.10:5060 > > /* port to listen to > * - can be specified more than once if needed to listen on many ports */ > port=5060 > > #!ifdef WITH_TLS > enable_tls=yes > #!endif > > # life time of TCP connection when there is no traffic > # - a bit higher than registration expires to cope with UA behind NAT > tcp_connection_lifetime=3605 > tcp_rd_buf_size=6144 > > listen=eth1 > #!ifdef WITH_WEBSOCKETS > listen=MY_WS_ADDR > #!ifdef WITH_TLS > listen=eth1 > #!endif > #!endif > > ####### Custom Parameters ######### > > # These parameters can be modified runtime via RPC interface > # - see the documentation of 'cfg_rpc' module. > # > # Format: group.id = value 'desc' description > # Access: $sel(cfg_get.group.id) or @cfg_get.group.id > # > > #!ifdef WITH_PSTN > # PSTN GW Routing > # > # - pstn.gw_ip: valid IP or hostname as string value, example: > # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" > # > # - by default is empty to avoid misrouting > pstn.gw_ip = "" desc "PSTN GW Address" > pstn.gw_port = "" desc "PSTN GW Port" > #!endif > > #!ifdef WITH_VOICEMAIL > # VoiceMail Routing on offline, busy or no answer > # > # - by default Voicemail server IP is empty to avoid misrouting > voicemail.srv_ip = "" desc "VoiceMail IP Address" > voicemail.srv_port = "5060" desc "VoiceMail Port" > #!endif > > ####### Modules Section ######## > > # set paths to location of modules (to sources or installation folders) > #!ifdef WITH_SRCPATH > mpath="modules/" > #!else > mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" > #!endif > > #!ifdef WITH_PGSQL > loadmodule "db_postgres.so" > #!endif > > loadmodule "mi_fifo.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "mi_rpc.so" > loadmodule "acc.so" > > #!ifdef WITH_AUTH > loadmodule "auth.so" > loadmodule "auth_db.so" > #!ifdef WITH_IPAUTH > loadmodule "permissions.so" > #!endif > #!endif > > #!ifdef WITH_ALIASDB > loadmodule "alias_db.so" > #!endif > > #!ifdef WITH_SPEEDDIAL > loadmodule "speeddial.so" > #!endif > > #!ifdef WITH_MULTIDOMAIN > loadmodule "domain.so" > #!endif > > #!ifdef WITH_PRESENCE > loadmodule "presence.so" > loadmodule "presence_xml.so" > #!endif > > #!ifdef WITH_NAT > loadmodule "nathelper.so" > loadmodule "rtpproxy.so" > #!endif > > #!ifdef WITH_TLS > loadmodule "tls.so" > #!endif > > #!ifdef WITH_ANTIFLOOD > loadmodule "htable.so" > loadmodule "pike.so" > #!endif > > #!ifdef WITH_XMLRPC > loadmodule "xmlrpc.so" > #!endif > > #!ifdef WITH_DEBUG > loadmodule "debugger.so" > #!endif > > #!ifdef WITH_SIPTRACE > loadmodule "siptrace.so" > #!endif > > #!ifdef WITH_SQLOPS > loadmodule "sqlops.so" > #!endif > > #!ifdef WITH_WEBSOCKETS > loadmodule "xhttp.so" > loadmodule "websocket.so" > loadmodule "nathelper.so" > #!endif > > # ----------------- setting module-specific parameters --------------- > > # ---- sip_trace params ---- > #!ifdef WITH_SIPTRACE > modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost > /kamailio") > modparam("siptrace", "trace_on", 1) > modparam("siptrace", "trace_flag", 22) > modparam("siptrace", "trace_sl_acks", 0) > #!endif > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- tm params ----- > # auto-discard branches from previous serial forking leg > modparam("tm", "failure_reply_mode", 3) > # default retransmission timeout: 30sec > modparam("tm", "fr_timer", 30000) > # default invite retransmission timeout after 1xx: 120sec > modparam("tm", "fr_inv_timer", 120000) > > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > > > # ----- registrar params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts per AOR */ > #modparam("registrar", "max_contacts", 10) > # max value for expires of registrations > modparam("registrar", "max_expires", 300) > modparam("registrar", "min_expires", 30) > # set it to 1 to enable GRUU > modparam("registrar", "gruu_enabled", 0) > > > # ----- acc params ----- > /* what special events should be accounted ? */ > modparam("acc", "early_media", 0) > modparam("acc", "report_ack", 0) > modparam("acc", "report_cancels", 0) > /* by default ww do not adjust the direct of the sequential requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "log_flag", FLT_ACC) > modparam("acc", "log_missed_flag", FLT_ACCMISSED) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;src_ip=$si;" > "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) > /* enhanced DB accounting */ > #!ifdef WITH_ACCDB > modparam("acc", "db_flag", FLT_ACC) > modparam("acc", "db_missed_flag", FLT_ACCMISSED) > modparam("acc", "db_url", DBURL) > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;src_ip=$si;" > "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > #!endif > > > # ----- usrloc params ----- > /* enable DB persistency for location entries */ > #!ifdef WITH_USRLOCDB > modparam("usrloc", "db_url", DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- auth_db params ----- > #!ifdef WITH_AUTH > modparam("auth_db", "db_url", DBURL) > modparam("auth_db", "calculate_ha1", 0) > #modparam("auth_db", "password_column", "password") > modparam("auth_db", "load_credentials", "") > modparam("auth_db", "use_domain", MULTIDOMAIN) > #modparam("auth_db", "use_domain", 1) > > # ----- permissions params ----- > #!ifdef WITH_IPAUTH > modparam("permissions", "db_url", DBURL) > modparam("permissions", "db_mode", 1) > #!endif > > #!endif > > > # ----- alias_db params ----- > #!ifdef WITH_ALIASDB > modparam("alias_db", "db_url", DBURL) > modparam("alias_db", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- speeddial params ----- > #!ifdef WITH_SPEEDDIAL > modparam("speeddial", "db_url", DBURL) > modparam("speeddial", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- domain params ----- > #!ifdef WITH_MULTIDOMAIN > modparam("domain", "db_url", DBURL) > # register callback to match myself condition with domains list > modparam("domain", "register_myself", 1) > #!endif > > > #!ifdef WITH_PRESENCE > # ----- presence params ----- > modparam("presence", "db_url", DBURL) > > # ----- presence_xml params ----- > modparam("presence_xml", "db_url", DBURL) > modparam("presence_xml", "force_active", 1) > #!endif > > > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > > > #!ifdef WITH_TLS > # ----- tls params ----- > modparam("tls", "connection_timeout", 60) > modparam("tls", "tls_debug", 10) > modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") > #modparam("tls", "verify_certificate", 1) > #modparam("tls", "require_certificate", 0) > #!endif > > #!ifdef WITH_ANTIFLOOD > # ----- pike params ----- > modparam("pike", "sampling_time_unit", 2) > modparam("pike", "reqs_density_per_unit", 16) > modparam("pike", "remove_latency", 4) > > # ----- htable params ----- > # ip ban htable with autoexpire after 5 minutes > modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") > #!endif > > #!ifdef WITH_XMLRPC > # ----- xmlrpc params ----- > modparam("xmlrpc", "route", "XMLRPC"); > modparam("xmlrpc", "url_match", "^/RPC") > #!endif > > #!ifdef WITH_DEBUG > # ----- debugger params ----- > modparam("debugger", "cfgtrace", 1) > #!endif > > > #!ifdef WITH_WEBSOCKETS > # ----- nathelper params ----- > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > # Note: leaving NAT pings turned off here as nathelper is _only_ being > used for > # WebSocket connections. NAT pings are not needed as WebSockets have > # their own keep-alives. > #!endif > > > > ####### Routing Logic ######## > > # Main SIP request routing logic > # - processing of any incoming SIP request starts with this route > # - note: this is the same as route { ... } > request_route { > > route(REQINIT); > > > #!ifdef WITH_SIPTRACE > if($hdr(X-Omnz-SipTrace) !=$null) > { > sip_trace(); > setflag(22); > } > #!endif > > > #!ifdef WITH_WEBSOCKETS > if (nat_uac_test(64)) { > # Do NAT traversal stuff for requests from a WebSocket > # connection - even if it is not behind a NAT! > # This won't be needed in the future if Kamailio and the > # WebSocket client support Outbound and Path. > force_rport(); > if (is_method("REGISTER")) > fix_nated_register(); > else { > if (!add_contact_alias()) { > xlog("L_ERR", "Error aliasing contact > <$ct>\n"); > sl_send_reply("400", "Bad Request"); > exit; > } > } > } > #!endif > > > # NAT detection > route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are > routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) > record_route(); > > # account only INVITEs > if (is_method("INVITE")) > { > setflag(FLT_ACC); # do accounting > } > > # dispatch requests to foreign domains > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > route(PRESENCE); > > > # handle registrations > route(REGISTRAR); > > if ($rU==$null) > { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > > # dispatch destinations to PSTN > route(PSTN); > > # user location service > route(LOCATION); > } > > > route[RELAY] { > > # enable additional event routes for forwarded requests > # - serial forking, RTP relaying handling, a.s.o. > > > #!ifdef WITH_603 > if (is_method("INVITE")) { > if($hdr(X-Omz-int-Id) ==$null) > { > # request with no Username in RURI > sl_send_reply("603","Declined"); > exit; > } > } > #!endif > > > > if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { > if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); > } > if (is_method("INVITE|SUBSCRIBE|UPDATE")) { > if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); > } > if (is_method("INVITE")) { > if(!t_is_set("failure_route")) > t_on_failure("MANAGE_FAILURE"); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > > # Per SIP request initial checks > route[REQINIT] { > #!ifdef WITH_ANTIFLOOD > # flood dection from same IP and traffic ban for a while > # be sure you exclude checking trusted peers, such as pstn gateways > # - local host excluded (e.g., loop to self) > if(src_ip!=myself) > { > if($sht(ipban=>$si)!=$null) > { > # ip is already blocked > xdbg("request from blocked IP - $rm from $fu > (IP:$si:$sp)\n"); > exit; > } > if (!pike_check_req()) > { > xlog("L_ALERT","ALERT: pike blocking $rm from $fu > (IP:$si:$sp)\n"); > $sht(ipban=>$si) = 1; > exit; > } > } > #!endif > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > > if(!sanity_check("1511", "7")) > { > xlog("Malformed SIP message from $si:$sp\n"); > exit; > } > } > > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > > > #!ifdef WITH_WEBSOCKETS > if ($du == "") { > if (!handle_ruri_alias()) { > xlog("L_ERR", "Bad alias <$ru>\n"); > sl_send_reply("400", "Bad > Request"); > exit; > } > } > #!endif > > > > route(DLGURI); > if (is_method("BYE")) { > setflag(FLT_ACC); # do accounting ... > setflag(FLT_ACCFAILED); # ... even if the > transaction fails > } > else if ( is_method("ACK") ) { > # ACK is forwarded statelessy > route(NATMANAGE); > } > else if ( is_method("NOTIFY") ) { > # Add Record-Route for in-dialog NOTIFY as > per RFC 6665. > record_route(); > } > route(RELAY); > } else { > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(PRESENCE); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > route(RELAY); > exit; > } else { > # ACK without matching transaction > ... ignore and discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > } > > # Handle SIP registrations > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > if (!save("location")) > sl_reply_error(); > > exit; > } > } > > # USER location service > route[LOCATION] { > > #!ifdef WITH_SPEEDDIAL > # search for short dialing - 2-digit extension > if($rU=~"^[0-9][0-9]$") > if(sd_lookup("speed_dial")) > route(SIPOUT); > #!endif > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > if(alias_db_lookup("dbaliases")) > route(SIPOUT); > #!endif > > $avp(oexten) = $rU; > if (!lookup("location")) { > $var(rc) = $rc; > route(TOVOICEMAIL); > t_newtran(); > switch ($var(rc)) { > case -1: > case -3: > send_reply("404", "Not Found"); > exit; > case -2: > send_reply("405", "Method Not Allowed"); > exit; > } > } > > # when routing via usrloc, log the missed calls also > if (is_method("INVITE")) > { > setflag(FLT_ACCMISSED); > } > > route(RELAY); > exit; > } > > # Presence server route > route[PRESENCE] { > if(!is_method("PUBLISH|SUBSCRIBE")) > return; > > #!ifdef WITH_PRESENCE > if (!t_newtran()) > { > sl_reply_error(); > exit; > }; > > if(is_method("PUBLISH")) > { > handle_publish(); > t_release(); > } > else > if( is_method("SUBSCRIBE")) > { > handle_subscribe(); > t_release(); > } > exit; > #!endif > > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==$null) > { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > > # Authentication route > route[AUTH] { > #!ifdef WITH_AUTH > > #!ifdef WITH_IPAUTH > if((!is_method("REGISTER")) && allow_source_address()) > { > # source IP allowed > return; > } > #!endif > > if (is_method("REGISTER") || from_uri==myself) > { > # authenticate requests > if (!auth_check("$fd", "subscriber", "1")) { > auth_challenge("$fd", "0"); > exit; > } > # user authenticated - remove auth header > if(!is_method("REGISTER|PUBLISH")) > consume_credentials(); > } > # if caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (from_uri!=myself && uri!=myself) > { > sl_send_reply("403","Not relaying"); > exit; > } > > #!endif > return; > } > > # Caller NAT detection route > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > add_contact_alias(); > } > setflag(FLT_NATS); > } > #!endif > return; > } > > # RTPProxy control > route[NATMANAGE] { > #!ifdef WITH_NAT > if (is_request()) { > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > } > } > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) > return; > > rtpproxy_manage("co"); > > if (is_request()) { > if (!has_totag()) { > if(t_is_branch_route()) { > add_rr_param(";nat=yes"); > } > } > } > if (is_reply()) { > if(isbflagset(FLB_NATB)) { > add_contact_alias(); > } > } > #!endif > return; > } > > # URI update for dialog requests > route[DLGURI] { > #!ifdef WITH_NAT > if(!isdsturiset()) { > handle_ruri_alias(); > } > #!endif > return; > } > > # Routing to foreign domains > route[SIPOUT] { > if (!uri==myself) > { > append_hf("P-hint: outbound\r\n"); > route(RELAY); > } > } > > # PSTN GW routing > route[PSTN] { > #!ifdef WITH_PSTN > # check if PSTN GW IP is defined > if (strempty($sel(cfg_get.pstn.gw_ip))) { > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not > defined\n"); > return; > } > > # route to PSTN dialed numbers starting with '+' or '00' > # (international format) > # - update the condition to match your dialing rules for PSTN > routing > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > return; > > # only local users allowed to call > if(from_uri!=myself) { > sl_send_reply("403", "Not Allowed"); > exit; > } > > if (strempty($sel(cfg_get.pstn.gw_port))) { > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > } else { > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" > + $sel(cfg_get.pstn.gw_port); > } > > route(RELAY); > exit; > #!endif > > return; > } > > # XMLRPC routing > #!ifdef WITH_XMLRPC > route[XMLRPC] { > # allow XMLRPC from localhost > if ((method=="POST" || method=="GET") > && (src_ip==127.0.0.1)) { > # close connection only for xmlrpclib user agents (there > is a bug in > # xmlrpclib: it waits for EOF before interpreting the > response). > if ($hdr(User-Agent) =~ "xmlrpclib") > set_reply_close(); > set_reply_no_connect(); > dispatch_rpc(); > exit; > } > send_reply("403", "Forbidden"); > exit; > } > #!endif > > # route to voicemail server > route[TOVOICEMAIL] { > #!ifdef WITH_VOICEMAIL > if(!is_method("INVITE")) > return; > > # check if VoiceMail server IP is defined > if (strempty($sel(cfg_get.voicemail.srv_ip))) { > xlog("SCRIPT: VoiceMail rotuing enabled but IP not > defined\n"); > return; > } > if($avp(oexten)==$null) > return; > > $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) > + ":" + $sel(cfg_get.voicemail.srv_port); > route(RELAY); > exit; > #!endif > > return; > } > > # manage outgoing branches > branch_route[MANAGE_BRANCH] { > xdbg("new branch [$T_branch_idx] to $ru\n"); > route(NATMANAGE); > } > > # manage incoming replies > onreply_route[MANAGE_REPLY] { > xdbg("incoming reply\n"); > if(status=~"[12][0-9][0-9]") > route(NATMANAGE); > } > > # manage failure routing cases > failure_route[MANAGE_FAILURE] { > route(NATMANAGE); > > if (t_is_canceled()) { > exit; > } > > #!ifdef WITH_BLOCK3XX > # block call redirect based on 3xx replies. > if (t_check_status("3[0-9][0-9]")) { > t_reply("404","Not found"); > exit; > } > #!endif > > #!ifdef WITH_VOICEMAIL > # serial forking > # - route to voicemail on busy or no answer (timeout) > if (t_check_status("486|408")) { > $du = $null; > route(TOVOICEMAIL); > exit; > } > #!endif > } > > > #!ifdef WITH_WEBSOCKETS > onreply_route { > if (nat_uac_test(64)) { > # Do NAT traversal stuff for replies to a WebSocket > connection > # - even if it is not behind a NAT! > # This won't be needed in the future if Kamailio and the > # WebSocket client support Outbound and Path. > add_contact_alias(); > } > } > > event_route[xhttp:request] { > set_reply_close(); > set_reply_no_connect(); > > if ($Rp != MY_WS_PORT > > #!ifdef WITH_TLS > && $Rp != MY_WSS_PORT > #!endif > > ) { > xlog("L_WARN", "HTTP request received on $Rp\n"); > xhttp_reply("403", "Forbidden", "", ""); > exit; > } > > xlog("L_DBG", "HTTP Request Received\n"); > > if ($hdr(Upgrade)=~"websocket" > && $hdr(Connection)=~"Upgrade" > && $rm=~"GET") { > xlog("L_DBG", "WebSocket\n"); > xlog("L_DBG", " Host: $hdr(Host)\n"); > xlog("L_DBG", " Origin: $hdr(Origin)\n"); > > if ($hdr(Host) == $null ) { > xlog("L_WARN", "Bad host $hdr(Host)\n"); > xhttp_reply("403", "Forbidden", "", ""); > exit; > } > > # Optional... validate Origin > # Optional... perform HTTP authentication > > # ws_handle_handshake() exits (no further configuration > file > # processing of the request) when complete. > if (ws_handle_handshake()) > { > # Optional... cache some information abou the > # successful connection > exit; > } > } > > xhttp_reply("404", "Not found", "", ""); > } > > event_route[websocket:closed] { > xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); > } > #!endif > > > >

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Re: [SR-Users] Welcome to the "Users" mailing list
by John Jhj 13 Apr '15

13 Apr '15

Hello, We have setup Kamailio SIP Server v4.1.7 at latest our Rackspace server (ubuntu 14.10) Chat works fine on all cases and calls are working for Local network only( I think its when it doesn't uses rtpproxy). Over Internet( on real ips rather than local one), we are able to make calls(audio/video) and we can see that Our packets are being transmitted properly to the server, but we are unable to get those packets to client side back from the server. So, Client-to-Server : works but, Server-to-Client : it doesn't works FYI, We have confirmed that firewalll is inactive on our server and required ports are open. We have tried with and without tls (self signed certificates only). The SIP client we used to test the environment is JITSI. We are getting following error : Apr 10 10:46:39 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:39 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:2636]: force_rtp_proxy(): no available proxies Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25641]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25641]: ERROR: rtpproxy [rtpproxy.c:2636]: force_rtp_proxy(): no available proxies Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25641]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1428662804;method=INVITE;from_tag=978a791a;to_tag=3458055147;call_id=6c08683633dd586d0e634f5e2bbfa99c@0 :0:0:0:0:0:0:0;code=200;reason=OK;src_user=user1;src_domain= mobileappplatform.com ;src_ip=202.157.76.21;dst_ouser=user4;dst_user=user4;dst_domain=192.168.5.115 Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25643]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25643]: ERROR: rtpproxy [rtpproxy.c:2636]: force_rtp_proxy(): no available proxies Apr 10 10:46:48 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:48 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:1960]: unforce_rtp_proxy(): no available proxies Apr 10 10:46:49 telezang-app /usr/local/sbin/kamailio[25644]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1428662809;method=BYE;from_tag=978a791a;to_tag=3458055147;call_id=6c08683633dd586d0e634f5e2bbfa99c@0 :0:0:0:0:0:0:0;code=200;reason=OK;src_user=user1;src_domain= mobileappplatform.com ;src_ip=202.157.76.21;dst_ouser=user4;dst_user=user4;dst_domain=202.157.76.21 Thanks On Fri, Apr 10, 2015 at 4:38 PM, <users-request(a)lists.opensips.org> wrote: > Welcome to the Users(a)lists.opensips.org mailing list! > > To post to this list, send your email to: > > users(a)lists.opensips.org > > General information about the mailing list is at: > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > If you ever want to unsubscribe or change your options (eg, switch to > or from digest mode, change your password, etc.), visit your > subscription page at: > > > http://lists.opensips.org/cgi-bin/mailman/options/users/followtest111%40gma… > > > You can also make such adjustments via email by sending a message to: > > Users-request(a)lists.opensips.org > > with the word `help' in the subject or body (don't include the > quotes), and you will get back a message with instructions. > > You must know your password to change your options (including changing > the password, itself) or to unsubscribe. It is: > > 123456 > > Normally, Mailman will remind you of your lists.opensips.org mailing > list passwords once every month, although you can disable this if you > prefer. This reminder will also include instructions on how to > unsubscribe or change your account options. There is also a button on > your options page that will email your current password to you. >

2 1

TLS & log files
by Tributh 13 Apr '15

13 Apr '15

Hi, is there any way to configure the tls curve like secp384r1 in the config? I use actually version 4.2.4 On the other side i can't see any connection Information in the logs. I would like to see something like: "Connected with TLSv1.1 using cipher ECDHE-RSA-AES256-GCM-SHA384 and curve secp384r1" With this logs the variety of the offered ciphers can be shortened to the most secure and only needed by the clients. Regard T.Tributh

2 1

Kamailio LDAP authentication
by Marek Moravcik 11 Apr '15

11 Apr '15

Hello, I'd like to authenticate Kamailio users in LDAP. But it looks like Kamailio need to download password from LDAP and authenticate user on it's own. Is there any possibility to send password to LDAP and let LDAP to say, if the user can be sign in? Thanks Marek Moravčík

2 1

In acc for 5xx calls result in "Unknown column 'time_attr' in 'field list'"
by Ethan Brooks 10 Apr '15

10 Apr '15

I am seeing a very strange error saying that time_attr is an unknown column and the insert is failing. What is strange is that it is in the acc table and is being populated for all successful calls. It seem to only happen for 5xx unsuccessful calls and the INVITE is not recorded. 4xx response are recorded. *config* modparam("acc", "time_mode", 1); modparam("acc", "failed_transaction_flag", 3) *syslog* Apr 9 20:40:52 a-deploy-callprocessing3 /usr/sbin/kamailio[9732]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'time_attr' in 'field list' Apr 9 20:40:52 a-deploy-callprocessing3 /usr/sbin/kamailio[9732]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Apr 9 20:40:52 a-deploy-callprocessing3 /usr/sbin/kamailio[9732]: ERROR: acc [acc.c:504]: acc_db_request(): failed to insert into database *Database Fields:* `time_attr` int(11) DEFAULT NULL, `time_exten` int(11) DEFAULT NULL, *kamailio -v* version: kamailio 4.2.3 (x86_64/linux) 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: unknown compiled on 12:08:27 Feb 16 2015 with gcc 4.9.1

2 2

weight of rtpengine rtp proxy?
by Juha Heinanen 10 Apr '15

10 Apr '15

rtpengine readme tells in section 2: The balancing inside a set is done automatically by the module based on the weight of each RTP proxy from the set. i have not found in rtpengine_sock module parameter description, how is the weight of a rtp proxy is specified specified. it contains an example: modparam("rtpengine", "rtpengine_sock", "1 == udp:localhost:12221 udp:localhost:12222") how can i tell what the weight of each of two proxies is? -- juha

3 4

Re: [SR-Users] fifo error and kamailio wont start
by Bruno Emer 10 Apr '15

10 Apr '15

Ok Joli! Here is the way I do this. At the begining of your kamailio.cfg, just after "#!KAMAILIO" insert this define: #!define WITH_DEBUG In your global parameters, you must now define what happens if "WITH_DEBUG" is configurated, so add these lines: #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif On Modules Selection, load the module debbuger.so: #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif And set the module specific parameters: #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif Maybe some of these informations can be there, but I am not sure about the default configuration file. After these changes, restart your kamailio and you should see a lot of information in your screen. Then, just try to find the error there. I just asked something here in this forum as well, and posted my .cfg file there. If you want to check the places where I put these lines, take a look at the topic "MULTIDOMAIN Environment". After discover the error and finish your troubleshooting, you can just comment the line #!define WITH_DEBUG putting one more # at the beginning of it and you will not see the debug information anymore. *Bruno Emer*

1 0

Re: [SR-Users] fifo error and kamailio wont start
by Bruno Emer 10 Apr '15

10 Apr '15

Hello Joli! To see what is going on, is a good idea to set debug=4 so you can see exactly what is going on. If you don't know how to do this, please let me know and I can try to help you. Bruno Emer

2 1

fifo error and kamailio wont start
by Joli Martinez 10 Apr '15

10 Apr '15

Hello, I am installing the latest version of Kamailio on CentOS 7 from Git. Everything compiled correctly, but when I try to start it I am getting a fifo error and Kamailio wont start [error] kamailio: : <core> [cfg.y:3428]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 296, column 39: Can't initialize shared memory kamailio: ERROR: mi_fifo [mi_fifo.c:271]: mi_destroy(): FIFO stat failed: Bad address kamailio: Starting kamailio: kamailio.service: control process exited, code=exited status=1 systemd: Failed to start LSB: Kamailio (OpenSER) - the Open Source SIP Server. systemd: Unit kamailio.service entered failed state. [/error] any suggestions on how to fix this?

2 2

Keeping track of routing/dialog attribute value pairs
by Dirk Teurlings - SIGNET B.V. 10 Apr '15

10 Apr '15

Hello, Throughout the routes of kamailio we use multiple variables. These consist of $var, $avp, $dlg_var and probably more. What's the policy for these variables when used. When, if even, are they being cleaned up after a call has been terminated. Is this something we need to be doing ourselves with functions like avp_delete("foo")? Many thanks! Dirk

4 4

MULTIDOMAIN Environment
by Bruno Emer 10 Apr '15

10 Apr '15

Hello all! I have a scenario here where I need to put more than two companies into the same Kamailio server and to accomplish this I'm using a multi-domain configuration. For now, I am able to create users in different domains and register all of them. The users are able to place and receive internal calls (as I am using this just to get internal communications working), so I am ok with this part of my configuration. The problem that I'm facing now is that users from domain "A" are able to place calls to users from domain "B". I need to deny calls between different domains, and I don't know the best way to achieve this. Here are my domains create (output from "kamctl domain show"): domain:: enterprise.com did:: enterprise.com domain:: enterprise2.com did:: enterprise2.com And here are my users created: username: bob domain: enterprise.com username: alice domain: enterprise2.com So, can you please help me to get this? Here is my .cfg file: #!KAMAILIO #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB ##!define WITH_NAT ##!define WITH_DEBUG #!define WITH_MULTIDOMAIN ##!define WITH_SIPTRACE #!define WITH_SQLOPS #!define WITH_XMLRPC ##!define WITH_603 #!define WITH_TLS # #!substdef "!MY_WS_PORT!80!g" #!substdef "!MY_WSS_PORT!81!g" #!substdef "!MY_WS_ADDR!eth1:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:eth1:MY_WSS_PORT!g" #!define WITH_WEBSOCKETS #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_PGSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "postgres://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=no tcp_accept_no_cl=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 tcp_rd_buf_size=6144 listen=eth1 #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=eth1 #!endif #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_SIPTRACE loadmodule "siptrace.so" #!endif #!ifdef WITH_SQLOPS loadmodule "sqlops.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" #!endif # ----------------- setting module-specific parameters --------------- # ---- sip_trace params ---- #!ifdef WITH_SIPTRACE modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost /kamailio") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_sl_acks", 0) #!endif # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 300) modparam("registrar", "min_expires", 30) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 0) #modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #modparam("auth_db", "use_domain", 1) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "connection_timeout", 60) modparam("tls", "tls_debug", 10) modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #modparam("tls", "verify_certificate", 1) #modparam("tls", "require_certificate", 0) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { route(REQINIT); #!ifdef WITH_SIPTRACE if($hdr(X-Omnz-SipTrace) !=$null) { sip_trace(); setflag(22); } #!endif #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) fix_nated_register(); else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. #!ifdef WITH_603 if (is_method("INVITE")) { if($hdr(X-Omz-int-Id) ==$null) { # request with no Username in RURI sl_send_reply("603","Declined"); exit; } } #!endif if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_WEBSOCKETS onreply_route { if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { xlog("L_DBG", "WebSocket\n"); xlog("L_DBG", " Host: $hdr(Host)\n"); xlog("L_DBG", " Origin: $hdr(Origin)\n"); if ($hdr(Host) == $null ) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information abou the # successful connection exit; } } xhttp_reply("404", "Not found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif

2 1

The caller doesn't receive reject message when the callee has more one register location
by xuefeng zhang 10 Apr '15

10 Apr '15

Dear Sir/Madame, I find out a caller doesn't receive reject message when a callee has been registered more times with different device then keep only one callee's device is available.I use the linphone's app to test it,It always reproduce on 4.2.4 stable version.The linphone's app reject a new call that will send 603 message,but the caller can't receive the 603 message.The caller will be continue at calling state. So I add a source code "*should_relay=branch" at 1384 line number in t_reply.c file. Hope you can change it. Thanks! Xuefeng Zhang

2 3

apache 2.4 wstunnel
by Juha Heinanen 10 Apr '15

10 Apr '15

apache 2.4 has capability to tunnel websocket requests: https://httpd.apache.org/docs/2.4/mod/mod_proxy_wstunnel.html the tunnel also automatically upgrades the connection to websocket connection meaning that ws_handle_handshake() is not needed. has anyone managed to configure kamailio to handle sip requests via apache wstunnel? -- juha

4 9

Cannot register Aastra 6757i
by David Matichuk 10 Apr '15

10 Apr '15

I set up a new account and I am unable to register my phone. It shows "no Service" I'm using an Aastra 6757i. David

1 0

DTLS not working with kamailio
by renuka ghate 09 Apr '15

09 Apr '15

I have configured the kamailio server for TLS.. but my soft phone supports only DTLS, how do i configure kamailio to support DTLS. regards, Renuka

4 3

Re: [SR-Users] Avoiding "script writer didn't release transaction" warnings
by Mickael Marrache 09 Apr '15

09 Apr '15

I think understand in which case it happens. When I receive a REGISTER, I create a transaction since I need to use the t_continue feature to wake up a suspended transaction. However, the save() command of the REGISTRAR module doesn't seem to call t_reply for replying the 200 (it looks like the SL module is used instead). Therefore, I get the warning every time it happens since the transaction is not released explicitly. Should I call t_release just before exiting the block handling the REGISTER? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 1:43 PM To: sr-users(a)lists.sip-router.org Subject: Avoiding "script writer didn't release transaction" warnings Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

1 0

Avoiding "script writer didn't release transaction" warnings
by Mickael Marrache 09 Apr '15

09 Apr '15

Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

1 0

Work with radius
by Никитенко Виталий 09 Apr '15

09 Apr '15

Hi. I have scheme: User -> Kamailio -> Asterisk -> PSTN | V Radius Server At the beginning dialog (INVITE) kamailio send to radius server START packet which indicates Called, Calling, Accounting Session ID, connect_time, setup_time. When the terminating call (BYE) kamailio forms STOP package. How can I get on kamailio Called, Calling, connect_time that I have in kamailio for START packet if I only match Accounting Session ID? Thank.

3 2

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 08 Apr '15

08 Apr '15

Hi, all I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio severs share the the same database with db_mode=3, and no registration replication. Use pjsua2 as SIP client for testing. The test setup is as follows: kamailio server 1(k1): 10.0.1.30:5061 <http://10.0.1.30:5061/> kamailio server 2(k2): 10.0.1.32:5061 <http://10.0.1.32:5061/> sip client c1: sip:16317@10.0.1.30 <mailto:sip%3A16317@10.0.1.30>, client ip: 10.0.1.254 sip client c2: sip:72316@10.0.1.30 <mailto:sip%3A72316@10.0.1.30>, client ip: 10.0.1.254 c1 is registered with k1. c2 is registered with k2. When c1 calls c2, the call flow is as follows: INVITE: c1------>k1------>k2------>c2 200 OK: c1<------k1<------k2<------c2 So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error. The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself instead of using the existing TLS connection to c1 to forward the BYE. BYE: c1 X k1<------k2<------c2 | |------->Connection refused/500 Here is the kamailio log on k1: Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 10.0.1.30:55999 <http://10.0.1.30:55999/> failed (RST) Connection refused Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 10.0.1.30:55999 <http://10.0.1.30:55999/>: connect & send for 0x7fc96a68a1a0 failed: Connection refused (111) This is the routing logic for BYE in kamailio.cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... If I add location based routing for BYE from peer kamailio, then the BYE seems to be forwarded correctly: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails # If BYE coming from Kamailio peer, route BYE by location $var(peerlist)=$sel(cfg_get.trusted.peers); $var(i) = 0; while($var(i)<$(var(peerlist){param.count})) { xlog("L_DBG", "=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n"); if(src_ip==$(var(peerlist){param.valueat,$var(i)})) { lookup("location"); xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n"); break; } $var(i) = $var(i) + 1; } } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Thanks

3 5

rtpstat counters
by Andres 08 Apr '15

08 Apr '15

In the documentation at: http://kamailio.org/docs/modules/4.2.x/modules/rtpproxy.html I see a variable called $rtpstat which "Returns the RTP-Statistics from the RTP-Proxy. The RTP-Statistics from the RTP-Proxy are provided as a string and it does contain several packet-counters." But I cannot find out where exactly it describes the packet counters. For example the xlog entry: log("L_ALERT", "STATS-rtpproxy,$rtpstat\n"); Will yield an output like: STATS-rtpproxy,60 596 660 1256 0#012 I need to find the documentation for those counters. Any ideas? Thanks, -- Technical Support http://www.cellroute.net

1 0

Variable interpolation in xavp identifiers
by Alex Balashov 08 Apr '15

08 Apr '15

Hello, There seems to be an implementational difference in interpolation of PVs in AVP and XAVP identifiers and/or field names. The pseudovariables documentation[1] on classical AVPs says: The 'id' can be: ... pseudo variable - if value of pv is integer, id is integer, if string, id is string So, for example, this composition of variables works: $var(f) = 'donkey'; $avp($var(f) = 'eehaw'; xlog("L_INFO", "Donkey says: $avp(donkey)\n"); It correctly prints: Donkey says: eehaw Making some assumptions, I tried this, and it did not work similarly as expected: $var(f) = 'donkey'; $xavp(r=>$var(f)) = 'eehaw'; xlog("L_INFO", "Donkey says: $xavp(r=>donkey)\n"); This prints: Donkey says: <null> And a PV dump confirms that the PV value is not interpolated: Mar 31 18:26:15 localhost /usr/local/sbin/kamailio[2399]: INFO: debugger [debugger_api.c:1799]: dbg_dump_json(): {"$xavp(r)":[{"$var(f)":["eehaw"]}]} Interpolation of variables does not work for either field or stem names. Is this a design principle of XAVPs, or an oversight? Thanks, -- Alex [1] http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#avp_id_-_avps -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

BYE not forwarded
by Grant Bagdasarian 08 Apr '15

08 Apr '15

Hello, I have the following setup: UAC (Bria) ------ Kamailio -------- Freeswitch Kamailio is used to authenticate the client using basic authentication. Upon authentication the INVITE is sent to Freeswitch for further handling. Kamailio stays in the path using record-route. At this point, the authentication and calling FS works fine. For now, FS just answers, plays a file and hangsup the call. Answering the call and playing a file works fine. When FS hangs up the call, the BYE message is sent back to Kamailio. The problem is the BYE message from FS contains a Route header which points to Kamailio again. Which causes Kamailio to send the BYE message back to itself. Kamailio then replies with a 404. The BYE is never sent to the UAC. When I remove the record_route() for INVITEs, the call works fine, but Kamailio isn't kept in the signaling path. The BYE is sent directly to the UAC. T 10.0.5.52:54822 -> 10.0.5.50:5060 [AP] BYE sip:grant@10.0.0.40:5060;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 10.0.5.52;rport;branch=z9hG4bK99X8a4Q8BFy1a. Route: <sip:10.0.5.50;transport=tcp;lr=on>. Max-Forwards: 70. From: <sip:123456789@10.0.0.40>;tag=aey7rKe7DmQte. To: <sip:grant@10.0.0.40>;tag=dbc9c27b. Call-ID: NmQ4YjYyMDFkNmMwZjg0ZmU5NjQ3ODhjYzU0MDBmNGI. CSeq: 73909300 BYE. User-Agent: CM SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Here is the WITHINDLG route: route[WITHINDLG] { xlog("L_INFO", "[R-ROUTE-WITHINDLG:$ci] !>" "Route:WITHINDLG;UnixTime:$TV(un);RequestURI:$rU;RequestMethod:$rm;FromTag:$ft;ToTag:$tt;CallID:$ci;From:$fU;To:$rU\r\n"); if (!has_totag()) { return; } # Sequential request within a dialog should take the path determined by record-routing if (loose_route()) { route(RELAY); exit; } else { if (is_method("ACK")) { if (t_check_trans()) { # No Loose-Route, but stateful ACK. Must be ACK after 487 route(RELAY); exit; } else { # ACK without matching transaction exit; } } sl_send_reply("404","Not here"); } exit; } I'm using TCP as the transport. Kamailio: 4.2.3 Freeswitch: 1.5.15 I've attached a SIP trace from the point of view of Kamailio. UAC: 10.0.0.40 Kamailio: 10.0.5.50 Freeswitch: 10.0.5.52 What is going on here? The whole Record-Route thing confuses me a lot. Regards, Grant

3 5

Fritzbox issues kamailio 4.0.7 regarding stale nonce?
by Daniel Tryba 08 Apr '15

08 Apr '15

Did anyone notice issues with Fritzbox 7390 devices? When upgrading from 4.0.1 to 4.0.7 some of these devices fail to register after some time. They keep trying to reuse an old nonce and ignore the new one in the 401 response to the REGISTER. It will try 11 times to register with the same cseq, nonce and nc, then it will try to register (11 times) with cseq+1, nc+1 and still the old nonce. So www_auth fails with -4: stale nonce. I could find anything in kamailio auth module to accept this behavior. -- Telefoon: 088 0100 700 Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024

2 4

Kamailio V 2.4.2 Dialog_ng crash and IMS_USRLOC_SCSCF doc
by Yasin CANER 08 Apr '15

08 Apr '15

3 6

Re: [SR-Users] sr-users Digest, Vol 119, Issue 4
by Yufei Tao 08 Apr '15

08 Apr '15

Hi Daniel Thanks for the reply! Here are things we use: * keepalived for HA * Ubuntu 12.04 There seems to be problems when Kamailio operates on both UDP and TCP at the same time. But I'll upgrade Kamailio from v4.0.0. to v4.0.7 first and see what we get, since there are a few bug fixes to do with TCP between these two versions. Cheers, Yufei On 4 April 2015 at 11:00, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: SIP messages over UDP with sizes over MTU (Yufei Tao) > 2. Re: SIP messages over UDP with sizes over MTU > (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 3 Apr 2015 12:04:49 +0100 > From: Yufei Tao <yufei.tao(a)gmail.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU > Message-ID: > < > CAJwP0iTBGk8S91CheS+xAON2fJgyKZH9HFcwTbvcLygmzLNd+g(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Daniel > > We don't have pike module enabled. > > When the problem occurred on an VIP, we observed these: > * Kamailio stopped responding to any messages that were sent to the VIP, > not just OPTIONS > * The OPTIONS messages are not big. But the other SIP messages, e.g. some > of the INVITE/OK that came from some SBCs can be big > * netstat showed the Recv-Q on that VIP had a lot of bytes accumulated, > while Kamailio was not seeing/reading them > * Kamailio responded fine on its real IP, when I sent OPTIONS pings to it > using sipsak > * After restarting Kamailio it started working. But after 1 week or so the > same problem happened again > > Since this only happened after running for 1 week or so, we didn't have any > traces to show what exactly happened at the particular time when it > happened. It is possible some SIP messages may have come in fragmented and > some are just too big, depending on the route they came from etc. So I was > wonder if it was possible that the UDP receive buffer was filled somehow > with messed-up messages. Is there anyway to check this? Any suggestions > where I should start looking please? Or is it generally a bad idea to use > UDP when there are messages that may be too big, either fragmented or not? > > Since the it is running in the production environment, I'd like to get some > confidence that a Kamailio upgrade will fix the problem first before I > change anything there. > > Cheers, > Yufei > > > > Date: Wed, 01 Apr 2015 16:27:44 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org > > > Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU > Message-ID: <551C0060.3010002(a)gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > first, not related to the topic you reported, I would recommend at least > upgrading to latest 4.0.x, there were many fixes from v4.0.0 and there > are no changes to be done to config or database -- simply deploy latest > 4.0.x and restart. > > Back to this topic. Is all the other traffic handled apart of big OPTIONS? > > Do you have pike module enabled? If yes, can you double check and be > sure that the SBC is not blacklisted (traffic from it should not be > handled via pike_check_req()). > > Cheers, > Daniel > > On 01/04/15 15:42, Yufei Tao wrote: > > Hi > > > > We've got Kamailio (v4.0.0) connected to some SBC, which sends SIP > > traffic and periodic OPTIONS pings to Kamailio's VIP. Kamailio > > responds to the OPTIONS pings with OK, i.e. in the main route block: > > if (is_method("OPTIONS")) > > { > > sl_send_reply("200","OK"); > > exit; > > } > > > > All works fine for a week or so, then Kamailio stopped responding to > > the OPTIONS pings on the VIP it listens to. But it still respond to > > OPTIONS pings that are sent to its real IP. The real IP is not used > > for receiving/sending any traffic while only the VIP is. So it seems > > that Kamailio is still working, but maybe having problems with the > > receiving buffer for the VIP? > > > > We do see that some SIP messages sent to Kamailio's VIP are too big > > (sometimes over 1500 bytes). My question is, in this case, what would > > be expected to happen? Is it possible somehow the receiving buffer for > > the VIP got messed up by the big UDP messages? Any one seen similar > > problems? What is the suggested solution? > > > > We're considering moving to TCP. But since this is production > > environment, I want to get some confidence that the problem we saw was > > likely to have been caused by the UDP message being too large. > > > > Cheers, > > Yufei >

1 0

File tranfere with Kamailio server(postgres)
by Vivek Bohra 08 Apr '15

08 Apr '15

Hi all, I want to know that file transfere service implementation by webrtc client api can be implemented with kamailio server. Best Regards vivek

2 4

Re: [SR-Users] Reg. invalid cseq error. (Daniel-Constantin Mierla)
by Badri Ranganathan 08 Apr '15

08 Apr '15

Thanks a lot Daniel. It was indeed the Cseq coming out of order from my box. Your comment helped me in identifying the problem. Regards, Badri. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: 03 April 2015 11:00 To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 119, Issue 3 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Make deb (Daniel-Constantin Mierla) 2. Re: [sr-dev] Announcement: Kamailio is now systemd-rtc-server (Alex Balashov) 3. Re: Kamailio coredump when "INVITE" enter in failure route (Jos? Seabra) 4. Re: Kamailio coredump when "INVITE" enter in failure route (Daniel-Constantin Mierla) 5. Re: Upgrade from Sip Express Router 0.9.6 (Andres) 6. Dialog based CDRs and failed dialogs (Mickael Marrache) 7. Re: Upgrade from Sip Express Router 0.9.6 (Daniel-Constantin Mierla) 8. Re: Dialog based CDRs and failed dialogs (Daniel-Constantin Mierla) 9. Re: Upgrade from Sip Express Router 0.9.6 (Andres) 10. Re: Dialog based CDRs and failed dialogs (Mickael Marrache) 11. Kamailio v4.2.4 Released (Daniel-Constantin Mierla) 12. Reg. invalid cseq error. (Badri Ranganathan) 13. Re: Reg. invalid cseq error. (Daniel-Constantin Mierla) 14. Re: Dialog based CDRs and failed dialogs (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Thu, 02 Apr 2015 12:07:06 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: jay binks <jaybinks(a)gmail.com> Cc: "Kamailio $SER$ - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Make deb Message-ID: <551D14CA.4030201(a)gmail.com> Content-Type: text/plain; charset="utf-8" Do you see db_cassandra being compiled in the logs printed by 'make deb'? Cheers, Daniel On 02/04/15 03:37, jay binks wrote: > I have managed to confirm that "make deb" does not actually build > db_cassandra but it does make a deb for it :) ( has changelog etc but > no db_cassandra.so file ) > > why does it not build the .so when calling make deb ?? > can someone point me in the direction to debug that part. > > doesnt seem to be related to the debian rules or control file .. > > is there something like modules.lst for when using "make deb" ? > > > On 2 April 2015 at 00:24, Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> wrote: > > The spec files to build debian packages are kept inside > 'pkg/kamailio/deb/' directory, groupped per distribution name > inside specific sub-directories (using debian and ubuntu version > names -- the one 'debian' is the generic one, but for a specific > version you should choose the one matching the name). > > Then I am not that familiar with debian spec files to be able to > advice by heart, however you should find easily on the web many > sources of information about the files inside those folder and how > to build deb packages. > > Say you want to build for debian wheezy: > > - edit the files inside pkg/kamailio/deb/wheezy/ to fit your needs > - in the root directory with kamailio sources do: > > ln -s pkg/kamailio/deb/wheezy debian > make deb > > And you should get the packaging process started. > > Cheers, > Daniel > > > On 01/04/15 14:46, jay binks wrote: >> Sorry, this confused me a little.. >> can you give an example of what you mean ? >> >> On 1 April 2015 at 22:40, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> you have to edit spec files and remove the other modules in >> the folder matching your debian distribution inside >> 'pkg/kamailio/deb/'. This is the only way I know to change >> the packaging content for debs. Then make a symlink inside >> root folder of kamailio server named 'debian' to the >> respective folder inside 'pkg/kamailio/deb/' and run 'make deb'. >> >> Cheers, >> Daniel >> >> >> On 01/04/15 13:50, jay binks wrote: >>> is there any way to make deb packages for only 1 of the >>> modules ?? >>> currently it takes a long time to run "make deb" >>> >>> im simply trying to build and package db_cassandra ( need >>> the package for my environment ) >>> >>> As a side note, is there any chance db_cassandra could be >>> added to the kamailio deb packages ? >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda >> Kamailio World Conference, May 27-29, 2015 >> Berlin, Germany - http://www.kamailioworld.com >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> -- >> Sincerely >> >> Jay > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda > Kamailio World Conference, May 27-29, 2015 > Berlin, Germany - http://www.kamailioworld.com > > > > > -- > Sincerely > > Jay -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

Websockets, set_contact_alias(), & IPv6
by Anthony Messina 08 Apr '15

08 Apr '15

I've been working on Kamailio websocket integration and I believe I'm having issues with the IPv6 address representation in the Contact header's alias parameter. After Googling, it appears after https://github.com/kamailio/kamailio/commit/814c08f3 the IPv6 contact is represented in brackets []. However, it when using websockets, and the alias= parameter uses brackets [] around and IPv6 address, there are message parsing issues. With a header such as the following Contact: <sip:wstest1@example.com;gr=urn:uuid:26140e27-0ab7-4e65-98e3-3d0909b1434e;alias=[2001:db8:0:1]~48768~6> Asterisk 13.2.0 will give the following error: pjsip:0 <?>: sip_transport. Error processing 1855 bytes packet from UDP 10.77.79.3:5060 : PJSIP syntax error exception when parsing 'Request Line' header on line 12 col 129: And when sipjs, or jssip are used with either Firefox or Chrome, they send garbage in the ACK request URI: Kamailio logs something like the following and the ACK cannot be processed: WARNING: sanity [sanity.c:236]: check_ruri_scheme(): failed to parse request uri [�a�{1me▒s�na@50�9��1.�8:�2v0;i��a��=11>7�n7x>10O�2v0~1] Is it proper to have [] brackets around the IPv6 alias address in the Contact header? Does the value need to be quoted? If I force the browser to use IPv4, without changing anything else, both jssip and sipjs work perfectly in Firefox and Chrome. -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 5

Re: [SR-Users] [RTPproxy] Re: [OpenSIPS-Users] Announcing rtpproxy v2.0.0
by Maxim Sobolev 08 Apr '15

08 Apr '15

Pull requests are welcome, guys. We are hitting 30k rtp sessions on one of our largest clusters. Lot of changes are coming into git repo soon to make rtp_cluster component run smoothly under such conditions. On Mar 16, 2015 8:56 PM, "John Mathew" <john.mathew(a)divoxmedia.com> wrote: > Yes > > On Tuesday, 17 March 2015, Zheng Frank <zhengyumingapple(a)gmail.com> wrote: > >> Do you mean ROHC ? >> >> 2015-03-14 12:39 GMT+08:00 Maxim Sobolev <sobomax(a)sippysoft.com>: >> >>> Do you have any particular RFC in mind? >>> On Mar 12, 2015 10:28 AM, "John Mathew" <john.mathew(a)divoxmedia.com> >>> wrote: >>> >>>> Hi, >>>> >>>> Maxim, >>>> Is there any plans for rtp header compression in future. I can't see >>>> anything in the change log for 2.0.0 >>>> >>>> >>>> On Tuesday, 10 March 2015, Maxim Sobolev <sobomax(a)sippysoft.com> wrote: >>>> >>>>> Hi All, >>>>> >>>>> I'm happy to announce that we have released rtpproxy v2.0.0. >>>>> >>>>> You can review the release notes here: >>>>> https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 >>>>> >>>>> -sobomax >>>>> >>>>> >>>> >>>> -- >>>> Sent from iPhone 6 >>>> >>>> -- >>>> You received this message because you are subscribed to the Google >>>> Groups "rtpproxy" group. >>>> To unsubscribe from this group and stop receiving emails from it, send >>>> an email to rtpproxy+unsubscribe(a)googlegroups.com. >>>> To post to this group, send email to rtpproxy(a)googlegroups.com. >>>> To view this discussion on the web visit >>>> https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B… >>>> <https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B…> >>>> . >>>> For more options, visit https://groups.google.com/d/optout. >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> > > -- > Sent from iPhone 6 > > -- > You received this message because you are subscribed to the Google Groups > "rtpproxy" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to rtpproxy+unsubscribe(a)googlegroups.com. > To post to this group, send email to rtpproxy(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpWvmaQnE8qM8ibb10d%3DNj9… > <https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpWvmaQnE8qM8ibb10d%3DNj9…> > . > For more options, visit https://groups.google.com/d/optout. >

1 0

Reg. invalid cseq error.
by Badri Ranganathan 07 Apr '15

07 Apr '15

Hi all, I make calls towards the Kamailio from my call generator and I get a message saying "invalid cseq error" sometimes at the 1005th registration or sometimes at the 1013thregistration, but the cseq number seems to be fine. Any idea about why this would happen? Thanks, Badri.

3 2

Upgrade from Sip Express Router 0.9.6
by Andres 07 Apr '15

07 Apr '15

How would one even approach the daunting task of upgrading a fully operational Sip Express Router 0.9.6 installation with thousands of users that has been running unmodified for over 10 years. I would like some pointers on how to approach the mysql database migration from the old schema to the Kamailio 4.2 schema. I sure hope I don't have to start from scratch. Maybe somebody has already done an upgrade script? Regarding the config file, I will certainly have to rewrite it but I was wondering if there was a list of deprecated commands or syntaxes that would no longer be valid and their equivalent replacement. Thanks, -- Technical Support http://www.cellroute.net

3 6

Problem unsubcribing list
by JIvars 07 Apr '15

07 Apr '15

Hello i want to unsubscribe from sr-users(a)lists.sip-router.org but it is impossible, i have tried with: http://lists.sip-router.org/cgi-bin/mailman/options/sr-users But the email never arrives, could you help me please Best regards.

3 2

Regarding multi user chat in Kamailio IMS server
by MadhuManjushaAripirala 07 Apr '15

07 Apr '15

Hello all, I have RCS based client. I would like to implement MUC(multi chat user) in existing Kamailio IMS Server. Please help me to input the procedure.Is it really possible to do it ? Please share me URL for any existing "C" source code which I can directly imbibe into Kamailio IMS serve -- View this message in context: http://sip-router.1086192.n5.nabble.com/Regarding-multi-user-chat-in-Kamail… Sent from the Users mailing list archive at Nabble.com.

2 4

flag f in search_hf() is not working for me
by schwatzi＠web.de 07 Apr '15

07 Apr '15

Hi. I think there is a bug in function search_hf_f() in textops.c Here is what I do if ( method == SUBSCRIBE ) { if ( !search_hf("Event", "call-completion", "f") ) { sl_send_reply("489","Bad Event"); exit; } } But here is a SUBSCRIBE that do pass the proxy. SUBSCRIBE sip:101@192.168.1.50 SIP/2.0 Via: SIP/2.0/UDP 77.10.66.72:59793;branch=z9hG4bKPj.O5IXVtDsImPNfCGnDkvj91INsEfvZsm Max-Forwards: 70 From: <sip:101@192.168.1.50>;tag=8-cXkrmwx5WGSIAYgYaXW0oTTsEAEM1S To: <sip:101@192.168.1.50> Contact: <sip:101@192.168.1.8:59793;ob> Call-ID: O12dPIx7Wiih2b.o36UFmMdXr6RE-8Jl CSeq: 24359 SUBSCRIBE Route: <sip:192.168.1.50;transport=udp;lr> Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: CSipSimple_mako-21/r2457 Content-Length: 0 As you can see the Event header doesn't contain "call-completition". Looking at function search_hf_f() i saw that if nothing is found value -1 will be returned. if there is the match then 1 is returned. (see line 2779 and 2797) But in line 2782 the function will return 1 instead of -1 index 50a6f07..a945937 100644 --- a/modules/textops/textops.c +++ b/modules/textops/textops.c @@ -2779,7 +2779,7 @@ static int search_hf_f(struct sip_msg* msg, char* str_hf, char* re, char *flags) return 1; } else { if(flags!=NULL && *flags=='f') - return 1; + return -1; } } else { hfl = hf; I am no programmer so it's possible that I misunderstood something. greetz Paolo

2 1

GSM SMS service
by Raj Roy Ghandhi 07 Apr '15

07 Apr '15

Hi Friends, I need to send text to GSM phones from SIP client. SIPSimple text -> GSM SMS Does anybody know good SMS gateway service that I can use for this purpose. Best Regards, Roy Gandhi.

2 1

Regarding IMS/RCS complaint clients for Kamailio IMS/RCS server
by MadhuManjushaAripirala 07 Apr '15

07 Apr '15

Hello, Please let me know 1.What are the IMS and RCS based clients available to test on Kamailio IMS/RCS server? 2.On browsing for Kamailio Installation,I found below 3 different links,please suggest me which one to be followed. https://loadmultiplier.com/node/76 http://www.kamailio.org/wiki/tutorials/ims/installation-howto http://kamailio.org/wiki/install/4.2.x/git 3.From the first link above,I see there are separate steps mentioned to install CSCF nodes and HSS but where I find no information for setting up RCS AS(application server) module.Please let me know on this. Thanks & Regards, Manjusha. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Regarding-IMS-RCS-complaint-clients… Sent from the Users mailing list archive at Nabble.com.

3 6

Fwd: INVITE routing issue when using Private IP (RFC1918) network
by pslack14 06 Apr '15

06 Apr '15

Hello, I have a Kamailio IMS instance setup, using different hosts for each IMS function (P-CSCF,S-CSCF,I-CSCF etc) and am able to register clients successfully. However, when Client-A tries to call Client-B (both registered to the same S-CSCF), the call is failing with the "403 Forbidden - You must register first with a S-CSCF" error. . All hosts & clients are running on the same subnet using RFC1918 IP addresses, there is no NAT involved. I've seen the same issue on both 4.2 and Master branches. I have the nathelper 'nat_uac_test' parameter set to 18. Looking at the debug prints and SIP traces, I can see that the 403 Error is being generated by the P-CSCF when it receives the INVITE from the S-CSCF (for forwarding to Client-B). Digging deeper into the debug logs, it looks like this is a result of a failed contact match - I've experimented with different values for "hashing_type", which do change the routing logic but still I see the 403 Error. Here's 2 excerpts from the debug prints for different values of hashing_type : *modparam("ims_usrloc_pcscf", ""hashing_type",hashing_type", 0) :* 2(19636) DEBUG: ims_registrar_pcscf [service_routes.c:183]: getContactP(): > Searching in usrloc for 10.133.202.17:5060 (Proto 1) 2(19636) DEBUG: ims_usrloc_pcscf [udomain.c:618]: get_pcontact_by_src(): > Trying to find contact by src with URI: [sip:*@10.133.202.17:5060] 2(19636) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): > Searching for contact in P-CSCF usrloc [sip:*@10.133.202.17:5060] 2(19636) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning > hash slot: [1670705485] 2(19636) DEBUG: ims_registrar_pcscf [service_routes.c:186]: getContactP(): > No entry in usrloc for 10.133.202.17:5060 (Proto 1) found! 2(19636) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): > Searching for contact in P-CSCF usrloc [sip:john@10.133.202.61:64019 > ;transport=udp] 2(19636) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning > hash slot: [397265208] 2(19636) DEBUG: ims_registrar_pcscf [service_routes.c:119]: > checkcontact(): Port 64019 (search 5060), Proto 1 (search 1), reg_state > registered (search registered) 2(19636) ERROR: *** cfgtrace:request_route=[Orig_Initial] > c=[/etc/kamailio/kamailio.cfg] l=926 a=26 n=send_reply *modparam("ims_usrloc_pcscf", ""hashing_type",hashing_type", 2) :* 1(19401) DEBUG: ims_registrar_pcscf [service_routes.c:183]: getContactP(): >> Searching in usrloc for 10.133.202.17:5060 (Proto 1) > > 1(19401) DEBUG: ims_usrloc_pcscf [udomain.c:618]: get_pcontact_by_src(): >> Trying to find contact by src with URI: [sip:*@10.133.202.17:5060] > > 1(19401) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): >> Searching for contact in P-CSCF usrloc [sip:*@10.133.202.17:5060] > > 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:97]: >> get_alias_host_from_contact(): no params > > 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:183]: get_aor_hash(): using >> host for hash [10.133.202.17] > > 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): >> Returning hash slot: [-1564914093] > > 1(19401) DEBUG: ims_registrar_pcscf [service_routes.c:186]: >> getContactP(): No entry in usrloc for 10.133.202.17:5060 (Proto 1) found! > > 1(19401) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): >> Searching for contact in P-CSCF usrloc [sip:john@10.133.202.61:64013 >> ;transport=udp] > > 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:176]: get_aor_hash(): Looks >> like this contact is natted - contact URI: [10.133.202.61] but came from >> received_host: [10.133.202.17] so will use received_host for hash > > 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): >> Returning hash slot: [-1564914093] > > 1(19401) ERROR: *** cfgtrace:request_route=[Orig_Initial] >> c=[/etc/kamailio/kamailio.cfg] l=926 a=26 n=send_reply > > Can anyone offer any hints as to how to resolve this? I am attaching a SIP call flow - I can supply a complete Kamailio debug trace if needed. Thanks Phill

1 0

Ip Address
by badican c-tin 05 Apr '15

05 Apr '15

HelloI have a little project and I need to find the ip address from my SIP phone. How can I find my ip addres of the phone?Have you a command or a service whith this functions? Thanks for help !!!Best regards,Constantin

1 0

Re: [SR-Users] SIP messages over UDP with sizes over MTU
by Yufei Tao 03 Apr '15

03 Apr '15

Hi Daniel We don't have pike module enabled. When the problem occurred on an VIP, we observed these: * Kamailio stopped responding to any messages that were sent to the VIP, not just OPTIONS * The OPTIONS messages are not big. But the other SIP messages, e.g. some of the INVITE/OK that came from some SBCs can be big * netstat showed the Recv-Q on that VIP had a lot of bytes accumulated, while Kamailio was not seeing/reading them * Kamailio responded fine on its real IP, when I sent OPTIONS pings to it using sipsak * After restarting Kamailio it started working. But after 1 week or so the same problem happened again Since this only happened after running for 1 week or so, we didn't have any traces to show what exactly happened at the particular time when it happened. It is possible some SIP messages may have come in fragmented and some are just too big, depending on the route they came from etc. So I was wonder if it was possible that the UDP receive buffer was filled somehow with messed-up messages. Is there anyway to check this? Any suggestions where I should start looking please? Or is it generally a bad idea to use UDP when there are messages that may be too big, either fragmented or not? Since the it is running in the production environment, I'd like to get some confidence that a Kamailio upgrade will fix the problem first before I change anything there. Cheers, Yufei Date: Wed, 01 Apr 2015 16:27:44 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU Message-ID: <551C0060.3010002(a)gmail.com> Content-Type: text/plain; charset="windows-1252" Hello, first, not related to the topic you reported, I would recommend at least upgrading to latest 4.0.x, there were many fixes from v4.0.0 and there are no changes to be done to config or database -- simply deploy latest 4.0.x and restart. Back to this topic. Is all the other traffic handled apart of big OPTIONS? Do you have pike module enabled? If yes, can you double check and be sure that the SBC is not blacklisted (traffic from it should not be handled via pike_check_req()). Cheers, Daniel On 01/04/15 15:42, Yufei Tao wrote: > Hi > > We've got Kamailio (v4.0.0) connected to some SBC, which sends SIP > traffic and periodic OPTIONS pings to Kamailio's VIP. Kamailio > responds to the OPTIONS pings with OK, i.e. in the main route block: > if (is_method("OPTIONS")) > { > sl_send_reply("200","OK"); > exit; > } > > All works fine for a week or so, then Kamailio stopped responding to > the OPTIONS pings on the VIP it listens to. But it still respond to > OPTIONS pings that are sent to its real IP. The real IP is not used > for receiving/sending any traffic while only the VIP is. So it seems > that Kamailio is still working, but maybe having problems with the > receiving buffer for the VIP? > > We do see that some SIP messages sent to Kamailio's VIP are too big > (sometimes over 1500 bytes). My question is, in this case, what would > be expected to happen? Is it possible somehow the receiving buffer for > the VIP got messed up by the big UDP messages? Any one seen similar > problems? What is the suggested solution? > > We're considering moving to TCP. But since this is production > environment, I want to get some confidence that the problem we saw was > likely to have been caused by the UDP message being too large. > > Cheers, > Yufei

2 1

Kamailio v4.2.4 Released
by Daniel-Constantin Mierla 02 Apr '15

02 Apr '15

Hello, Kamailio SIP Server v4.2.4 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.3. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.4. For more details about version 4.2.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/04/kamailio-v4-2-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Kamailio coredump when "INVITE" enter in failure route
by José Seabra 02 Apr '15

02 Apr '15

Hi there, I have noticed that my kamailio proxy, version 4.2.2 is originating core dumps. This happens when Kamailio receives an INVITE in failure route, then execute the function async_sleep(3), after that changes request uri, then it stops with following error message shown below: <core> [mem/f_malloc.c:586]: fm_free(): BUG: fm_free: bad pointer 0x7f8d26605ed8 (out of memory block!), called from <core>: parser/msg_parser.c: set_path_vector(822) - aborting Please see the attachment that contains the " gdb bt full" from core.dump originated Thank you for your attention Best Regards -- Cumprimentos José Seabra

2 4

rtpengine documentation error for forced rtpproxying?
by Daniel Tryba 02 Apr '15

02 Apr '15

I'm looking to replace rtpproxy with rtpengine. All calls have to be forced thru rtpengine and according to the documentation the equivalent of rtpproxy_manage("f"); is rtpengine_manage("force"); But rtpenginge doesn't recognize it as a valid flag. Using the legacy rtpproxy mod it still works though. Is it a documentation failure or is a specific version of rtpengine required for forced? Apr 1 17:54:28 pisco rtpengine[3432]: Got valid command from 127.0.0.1:43193: offer - { "sdp": "v=0#015#012o=tryba 8000 8001 I N IP4 10.0.3.169#015#012s=SIP Call#015#012c=IN IP4 10.0.3.169#015#012t=0 0#015#012m=audio 5016 RTP/AVP 8 0 9 2 3 18 4 99 101 13 #015#012a=sendrecv#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:2 G726-3 2/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:99 iLBC/8000#015#012 a=fmtp:99 mode=20#015#012a=ptime:20#015#012a=rtpmap:101 telephone- event/8000#015#012a=fmtp:101 0-11#015#012", "flags": [ "force " ], "call-id": "5 ... Apr 1 17:54:28 pisco rtpengine[3432]: ... f13ffb48e98404a(a)10.0.3.169", "received-from": [ "IP4", "109.235.34.226" ], "from-tag ": "3819edfdf2df044a", "command": "offer" } Apr 1 17:54:28 pisco rtpengine[3432]: Unknown flag encountered: 'force' -- Telefoon: 088 0100 700 Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024

2 4

Planning to release Kamailio v4.2.4
by Daniel-Constantin Mierla 02 Apr '15

02 Apr '15

Hello, I am considering to release a new maintenance version from latest stable branch - to be v4.2.4 - next week, most likely on Thursday, April 2. If there is anything important to get in and not discussed yet, bring the topic on development mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 1

Recommendation for toggling MSILO m_store auto-notification reply on/off
by Anthony Messina 02 Apr '15

02 Apr '15

Using Kamailio 4.2-dev and MSILO, is it possible to "toggle" the auto- notification reply MESSAGE using something similar to the following where "$var(msilo_reply)" is emtpy at startup (or are there suggestions for a better method): modparam("msilo", "from_address", "$var(msilo_reply)") ... if(CONDITION WHERE I WANT THE AUTO-REPLY RETURNS POSITIVE) { $var(msilo_reply)="$rU(a)example.net"; } m_store($rU) ... My use case is that I would like the auto-notification reply to occur in some instances, but not others. I also do not want to create a loop where the auto-notification replies are also stored. I was originally using a simple test: if(src_ip!=myself), but I have begun to use the IMC module as well as the EXEC module and these MESSAGEs appear to originate from "myself." Looking at the MSILO documentation I see there is "extra_hdrs(string)," but this appears to be designed to add headers to dumped MESSAGEs. I was hoping there would be a way to add a custom header to the auto-notification, on which I could filter/drop generated replies earlier in the route, but it doesn't appear that this exists. My preference for the best-case scenario would be to control the generation of the auto-notification reply in the first place rather than having to add a control to detect the reply and drop it. Thanks in advance for any recommendations. -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 3

SIP messages over UDP with sizes over MTU
by Yufei Tao 01 Apr '15

01 Apr '15

Hi We've got Kamailio (v4.0.0) connected to some SBC, which sends SIP traffic and periodic OPTIONS pings to Kamailio's VIP. Kamailio responds to the OPTIONS pings with OK, i.e. in the main route block: if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } All works fine for a week or so, then Kamailio stopped responding to the OPTIONS pings on the VIP it listens to. But it still respond to OPTIONS pings that are sent to its real IP. The real IP is not used for receiving/sending any traffic while only the VIP is. So it seems that Kamailio is still working, but maybe having problems with the receiving buffer for the VIP? We do see that some SIP messages sent to Kamailio's VIP are too big (sometimes over 1500 bytes). My question is, in this case, what would be expected to happen? Is it possible somehow the receiving buffer for the VIP got messed up by the big UDP messages? Any one seen similar problems? What is the suggested solution? We're considering moving to TCP. But since this is production environment, I want to get some confidence that the problem we saw was likely to have been caused by the UDP message being too large. Cheers, Yufei

2 1

SIP Authentication Best Practices
by Grant Bagdasarian 01 Apr '15

01 Apr '15

Hello, I'm implementing basic authentication using the auth and auth_db module, and I was wondering which domain to use when calling the auth_check function? Should I use the From domain or the To domain? Or both? The default configuration file deployed with kamailio uses $fd, but a code sample in the htable module page uses $td (dictionary attack sample). Aren't both the From and To header spoofable and not really used in SIP message routing?

1 0

jsonrpc_exec crashes kamailio
by Grant Bagdasarian 01 Apr '15

01 Apr '15

Hello, Kamailio crashes when I try to execute the following jsonrpc_exec functions: jsonrpc_exec('{"jsonrpc": "2.0", "method": "domain.reload", "id": 0}'); #domain module jsonrpc_exec('{"jsonrpc": "2.0", "method": "addressReload", "id": 0}'); #permissions module This one works fine: jsonrpc_exec('{"jsonrpc": "2.0", "method": "htable.reload", "params": ["tablename"], "id": 0}'); Any ideas? Running on Debian 7.8 and Kamailio 4.2.3. Backtrace: *** glibc detected *** /usr/local/sbin/kamailio: double free or corruption (fasttop): 0x0000000001f1ff00 *** ======= Backtrace: ========= /lib/x86_64-linux-gnu/libc.so.6(+0x75be6)[0x7f7009d50be6] /lib/x86_64-linux-gnu/libc.so.6(cfree+0x6c)[0x7f7009d5598c] /usr/local/lib64/kamailio/modules/jsonrpc-s.so(+0x2834)[0x7f7003e12834] /usr/local/lib64/kamailio/modules/jsonrpc-s.so(+0xb59a)[0x7f7003e1b59a] /usr/local/sbin/kamailio(do_action+0x749a)[0x41fc24] /usr/local/sbin/kamailio(run_actions+0x643)[0x42bc9a] /usr/local/sbin/kamailio(do_action+0x737e)[0x41fb08] /usr/local/sbin/kamailio(run_actions+0x643)[0x42bc9a] /usr/local/sbin/kamailio(do_action+0x10073)[0x4287fd] /usr/local/sbin/kamailio(run_actions+0x643)[0x42bc9a] /usr/local/sbin/kamailio(do_action+0x737e)[0x41fb08] /usr/local/sbin/kamailio(run_actions+0x643)[0x42bc9a] /usr/local/sbin/kamailio(do_action+0x737e)[0x41fb08] /usr/local/sbin/kamailio(run_actions+0x643)[0x42bc9a] /usr/local/sbin/kamailio(do_action+0x408d)[0x41c817] /usr/local/sbin/kamailio(run_actions+0x643)[0x42bc9a] /usr/local/sbin/kamailio(run_top_route+0x98)[0x42c3dd] /usr/local/sbin/kamailio(receive_msg+0x12fc)[0x50acec] /usr/local/sbin/kamailio(receive_tcp_msg+0x47a)[0x5f356a] /usr/local/sbin/kamailio(tcp_read_req+0x1798)[0x5f4d09] /usr/local/sbin/kamailio[0x5f7d19] /usr/local/sbin/kamailio[0x5eb830] /usr/local/sbin/kamailio(tcp_receive_loop+0x362)[0x5f8d6d] /usr/local/sbin/kamailio(tcp_init_children+0x7e2)[0x5e2453] /usr/local/sbin/kamailio(main_loop+0x390b)[0x4a8e33] /usr/local/sbin/kamailio(main+0x47cc)[0x4ae0fb] /lib/x86_64-linux-gnu/libc.so.6(__libc_start_main+0xfd)[0x7f7009cf9ead] /usr/local/sbin/kamailio[0x417db9] ======= Memory map: ======== 00400000-007c3000 r-xp 00000000 08:01 803340 /usr/local/sbin/kamailio 009c2000-009cc000 rw-p 003c2000 08:01 803340 /usr/local/sbin/kamailio 009cc000-00a89000 rw-p 00000000 00:00 0 01e84000-01f0a000 rw-p 00000000 00:00 0 [heap] 01f0a000-01f47000 rw-p 00000000 00:00 0 [heap] 01f47000-01f6d000 rw-p 00000000 00:00 0 [heap] 7f6ff8000000-7f6ff8021000 rw-p 00000000 00:00 0 7f6ff8021000-7f6ffc000000 ---p 00000000 00:00 0 7f6ffdbc6000-7f6ffdbdb000 r-xp 00000000 08:01 913924 /lib/x86_64-linux-gnu/libgcc_s.so.1 7f6ffdbdb000-7f6ffdddb000 ---p 00015000 08:01 913924 /lib/x86_64-linux-gnu/libgcc_s.so.1 7f6ffdddb000-7f6ffdddc000 rw-p 00015000 08:01 913924 /lib/x86_64-linux-gnu/libgcc_s.so.1 7f6ffdddc000-7f6ffddde000 r-xp 00000000 08:01 784463 /usr/lib/x86_64-linux-gnu/gconv/CP1252.so 7f6ffddde000-7f6ffdfdd000 ---p 00002000 08:01 784463 /usr/lib/x86_64-linux-gnu/gconv/CP1252.so 7f6ffdfdd000-7f6ffdfde000 r--p 00001000 08:01 784463 /usr/lib/x86_64-linux-gnu/gconv/CP1252.so 7f6ffdfde000-7f6ffdfdf000 rw-p 00002000 08:01 784463 /usr/lib/x86_64-linux-gnu/gconv/CP1252.so 7f6ffdfdf000-7f6ffdfe2000 r-xp 00000000 08:01 914284 /lib/x86_64-linux-gnu/libgpg-error.so.0.8.0 7f6ffdfe2000-7f6ffe1e1000 ---p 00003000 08:01 914284 /lib/x86_64-linux-gnu/libgpg-error.so.0.8.0 7f6ffe1e1000-7f6ffe1e2000 rw-p 00002000 08:01 914284 /lib/x86_64-linux-gnu/libgpg-error.so.0.8.0 7f6ffe1e2000-7f6ffe1f3000 r-xp 00000000 08:01 787833 /usr/lib/x86_64-linux-gnu/libp11-kit.so.0.0.0 7f6ffe1f3000-7f6ffe3f2000 ---p 00011000 08:01 787833 /usr/lib/x86_64-linux-gnu/libp11-kit.so.0.0.0 7f6ffe3f2000-7f6ffe3f3000 r--p 00010000 08:01 787833 /usr/lib/x86_64-linux-gnu/libp11-kit.so.0.0.0 7f6ffe3f3000-7f6ffe3f4000 rw-p 00011000 08:01 787833 /usr/lib/x86_64-linux-gnu/libp11-kit.so.0.0.0 7f6ffe3f4000-7f6ffe40a000 r-xp 00000000 08:01 914012 /lib/x86_64-linux-gnu/libz.so.1.2.7 7f6ffe40a000-7f6ffe609000 ---p 00016000 08:01 914012 /lib/x86_64-linux-gnu/libz.so.1.2.7 7f6ffe609000-7f6ffe60a000 r--p 00015000 08:01 914012 /lib/x86_64-linux-gnu/libz.so.1.2.7 7f6ffe60a000-7f6ffe60b000 rw-p 00016000 08:01 914012 /lib/x86_64-linux-gnu/libz.so.1.2.7 7f6ffe60b000-7f6ffe686000 r-xp 00000000 08:01 913957 /lib/x86_64-linux-gnu/libgcrypt.so.11.7.0 7f6ffe686000-7f6ffe886000 ---p 0007b000 08:01 913957 /lib/x86_64-linux-gnu/libgcrypt.so.11.7.0 7f6ffe886000-7f6ffe887000 r--p 0007b000 08:01 913957 /lib/x86_64-linux-gnu/libgcrypt.so.11.7.0 7f6ffe887000-7f6ffe88a000 rw-p 0007c000 08:01 913957 /lib/x86_64-linux-gnu/libgcrypt.so.11.7.0 7f6ffe88a000-7f6ffe89a000 r-xp 00000000 08:01 787835 /usr/lib/x86_64-linux-gnu/libtasn1.so.3.1.16 7f6ffe89a000-7f6ffea99000 ---p 00010000 08:01 787835 /usr/lib/x86_64-linux-gnu/libtasn1.so.3.1.16 7f6ffea99000-7f6ffea9a000 r--p 0000f000 08:01 787835 /usr/lib/x86_64-linux-gnu/libtasn1.so.3.1.16 7f6ffea9a000-7f6ffea9b000 rw-p 00010000 08:01 787835 /usr/lib/x86_64-linux-gnu/libtasn1.so.3.1.16 7f6ffea9b000-7f6ffeaa2000 r-xp 00000000 08:01 918716 /lib/x86_64-linux-gnu/librt-2.13.so 7f6ffeaa2000-7f6ffeca1000 ---p 00007000 08:01 918716 /lib/x86_64-linux-gnu/librt-2.13.so 7f6ffeca1000-7f6ffeca2000 r--p 00006000 08:01 918716 /lib/x86_64-linux-gnu/librt-2.13.so 7f6ffeca2000-7f6ffeca3000 rw-p 00007000 08:01 918716 /lib/x86_64-linux-gnu/librt-2.13.so 7f6ffeca3000-7f6ffed5c000 r-xp 00000000 08:01 784529 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.4 7f6ffed5c000-7f6ffef5b000 ---p 000b9000 08:01 784529 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.4 7f6ffef5b000-7f6ffef61000 r--p 000b8000 08:01 784529 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.4 7f6ffef61000-7f6ffef63000 rw-p 000be000 08:01 784529 /usr/lib/x86_64-linux-gnu/libgnutls.so.26.22.4 7f6ffef63000-7f6ffef74000 r-xp 00000000 08:01 801540 /usr/lib/x86_64-linux-gnu/libodbcinst.so.1.0.0 7f6ffef74000-7f6fff174000 ---p 00011000 08:01 801540 /usr/lib/x86_64-linux-gnu/libodbcinst.so.1.0.0 7f6fff174000-7f6fff175000 rw-p 00011000 08:01 801540 /usr/lib/x86_64-linux-gnu/libodbcinst.so.1.0.0 7f6fff175000-7f6fff1d2000 r-xp 00000000 08:01 803365 /usr/lib/x86_64-linux-gnu/odbc/libtdsodbc.so 7f6fff1d2000-7f6fff3d2000 ---p 0005d000 08:01 803365 /usr/lib/x86_64-linux-gnu/odbc/libtdsodbc.so 7f6fff3d2000-7f6fff3d5000 r--p 0005d000 08:01 803365 /usr/lib/x86_64-linux-gnu/odbc/libtdsodbc.so 7f6fff3d5000-7f6fff3d6000 rw-p 00060000 08:01 803365 /usr/lib/x86_64-linux-gnu/odbc/libtdsodbc.so 7f6fff3d6000-7f6fff3d8000 r-xp 00000000 08:01 784470 /usr/lib/x86_64-linux-gnu/gconv/ISO8859-1.so 7f6fff3d8000-7f6fff5d7000 ---p 00002000 08:01 784470 /usr/lib/x86_64-linux-gnu/gconv/ISO8859-1.so 7f6fff5d7000-7f6fff5d8000 r--p 00001000 08:01 784470 /usr/lib/x86_64-linux-gnu/gconv/ISO8859-1.so 7f6fff5d8000-7f6fff5d9000 rw-p 00002000 08:01 784470 /usr/lib/x86_64-linux-gnu/gconv/ISO8859-1.so 7f6fff5d9000-7f70035d9000 rw-s 00000000 00:04 574933 /dev/zero (deleted) 7f70035d9000-7f70035e4000 r-xp 00000000 08:01 918697 /lib/x86_64-linux-gnu/libnss_files-2.13.so 7f70035e4000-7f70037e3000 ---p 0000b000 08:01 918697 /lib/x86_64-linux-gnu/libnss_files-2.13.so 7f70037e3000-7f70037e4000 r--p 0000a000 08:01 918697 /lib/x86_64-linux-gnu/libnss_files-2.13.so 7f70037e4000-7f70037e5000 rw-p 0000b000 08:01 918697 /lib/x86_64-linux-gnu/libnss_files-2.13.so 7f70037e5000-7f70037ef000 r-xp 00000000 08:01 918707 /lib/x86_64-linux-gnu/libnss_nis-2.13.so 7f70037ef000-7f70039ee000 ---p 0000a000 08:01 918707 /lib/x86_64-linux-gnu/libnss_nis-2.13.so 7f70039ee000-7f70039ef000 r--p 00009000 08:01 918707 /lib/x86_64-linux-gnu/libnss_nis-2.13.so 7f70039ef000-7f70039f0000 rw-p 0000a000 08:01 918707 /lib/x86_64-linux-gnu/libnss_nis-2.13.so 7f70039f0000-7f7003a05000 r-xp 00000000 08:01 918699 /lib/x86_64-linux-gnu/libnsl-2.13.so 7f7003a05000-7f7003c04000 ---p 00015000 08:01 918699 /lib/x86_64-linux-gnu/libnsl-2.13.so 7f7003c04000-7f7003c05000 r--p 00014000 08:01 918699 /lib/x86_64-linux-gnu/libnsl-2.13.so 7f7003c05000-7f7003c06000 rw-p 00015000 08:01 918699 /lib/x86_64-linux-gnu/libnsl-2.13.so 7f7003c06000-7f7003c08000 rw-p 00000000 00:00 0 7f7003c08000-7f7003c0f000 r-xp 00000000 08:01 918711 /lib/x86_64-linux-gnu/libnss_compat-2.13.so 7f7003c0f000-7f7003e0e000 ---p 00007000 08:01 918711 /lib/x86_64-linux-gnu/libnss_compat-2.13.so 7f7003e0e000-7f7003e0f000 r--p 00006000 08:01 918711 /lib/x86_64-linux-gnu/libnss_compat-2.13.so 7f7003e0f000-7f7003e10000 rw-p 00007000 08:01 918711 /lib/x86_64-linux-gnu/libnss_compat-2.13.so 7f7003e10000-7f7003e21000 r-xp 00000000 08:01 923664 /usr/local/lib64/kamailio/modules/jsonrpc-s.so 7f7003e21000-7f7004021000 ---p 00011000 08:01 923664 /usr/local/lib64/kamailio/modules/jsonrpc-s.so 7f7004021000-7f7004022000 rw-p 00011000 08:01 923664 /usr/local/lib64/kamailio/modules/jsonrpc-s.so 7f7004022000-7f700402f000 r-xp 00000000 08:01 923743 /usr/local/lib64/kamailio/modules/xhttp.so 7f700402f000-7f700422f000 ---p 0000d000 08:01 923743 /usr/local/lib64/kamailio/modules/xhttp.so 7f700422f000-7f7004230000 rw-p 0000d000 08:01 923743 /usr/local/lib64/kamailio/modules/xhttp.so 7f7004230000-7f700426f000 r-xp 00000000 08:01 923652 /usr/local/lib64/kamailio/modules/dispatcher.so 7f700426f000-7f700446f000 ---p 0003f000 08:01 923652 /usr/local/lib64/kamailio/modules/dispatcher.so 7f700446f000-7f7004471000 rw-p 0003f000 08:01 923652 /usr/local/lib64/kamailio/modules/dispatcher.so 7f7004471000-7f70044af000 r-xp 00000000 08:01 923689 /usr/local/lib64/kamailio/modules/permissions.so 7f70044af000-7f70046af000 ---p 0003e000 08:01 923689 /usr/local/lib64/kamailio/modules/permissions.so 7f70046af000-7f70046b1000 rw-p 0003e000 08:01 923689 /usr/local/lib64/kamailio/modules/permissions.so 7f70046b1000-7f70046b2000 rw-p 00000000 00:00 0 7f70046b2000-7f70046bf000 r-xp 00000000 08:01 923660 /usr/local/lib64/kamailio/modules/group.so 7f70046bf000-7f70048bf000 ---p 0000d000 08:01 923660 /usr/local/lib64/kamailio/modules/group.so 7f70048bf000-7f70048c0000 rw-p 0000d000 08:01 923660 /usr/local/lib64/kamailio/modules/group.so 7f70048c0000-7f70048d7000 r-xp 00000000 08:01 923655 /usr/local/lib64/kamailio/modules/domain.so 7f70048d7000-7f7004ad6000 ---p 00017000 08:01 923655 /usr/local/lib64/kamailio/modules/domain.so 7f7004ad6000-7f7004ad7000 rw-p 00016000 08:01 923655 /usr/local/lib64/kamailio/modules/domain.so 7f7004ad7000-7f7004aea000 r-xp 00000000 08:01 923540 /usr/local/lib64/kamailio/modules/auth_db.so 7f7004aea000-7f7004cea000 ---p 00013000 08:01 923540 /usr/local/lib64/kamailio/modules/auth_db.so 7f7004cea000-7f7004ceb000 rw-p 00013000 08:01 923540 /usr/local/lib64/kamailio/modules/auth_db.so 7f7004ceb000-7f7004d13000 r-xp 00000000 08:01 923539 /usr/local/lib64/kamailio/modules/auth.so 7f7004d13000-7f7004f13000 ---p 00028000 08:01 923539 /usr/local/lib64/kamailio/modules/auth.so 7f7004f13000-7f7004f14000 rw-p 00028000 08:01 923539 /usr/local/lib64/kamailio/modules/auth.so 7f7004f14000-7f7004f28000 r-xp 00000000 08:01 923690 /usr/local/lib64/kamailio/modules/pike.so 7f7004f28000-7f7005128000 ---p 00014000 08:01 923690 /usr/local/lib64/kamailio/modules/pike.so 7f7005128000-7f7005129000 rw-p 00014000 08:01 923690 /usr/local/lib64/kamailio/modules/pike.so 7f7005129000-7f7005163000 r-xp 00000000 08:01 923661 /usr/local/lib64/kamailio/modules/htable.so 7f7005163000-7f7005362000 ---p 0003a000 08:01 923661 /usr/local/lib64/kamailio/modules/htable.so 7f7005362000-7f7005364000 rw-p 00039000 08:01 923661 /usr/local/lib64/kamailio/modules/htable.so 7f7005364000-7f700537b000 r-xp 00000000 08:01 918717 /lib/x86_64-linux-gnu/libpthread-2.13.so 7f700537b000-7f700557a000 ---p 00017000 08:01 918717 /lib/x86_64-linux-gnu/libpthread-2.13.so 7f700557a000-7f700557b000 r--p 00016000 08:01 918717 /lib/x86_64-linux-gnu/libpthread-2.13.so 7f700557b000-7f700557c000 rw-p 00017000 08:01 918717 /lib/x86_64-linux-gnu/libpthread-2.13.so 7f700557c000-7f7005580000 rw-p 00000000 00:00 0 7f7005580000-7f7005589000 r-xp 00000000 08:01 801498 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7f7005589000-7f7005788000 ---p 00009000 08:01 801498 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7f7005788000-7f7005789000 rw-p 00008000 08:01 801498 /usr/lib/x86_64-linux-gnu/libltdl.so.7.3.0 7f7005789000-7f70057e9000 r-xp 00000000 08:01 801505 /usr/lib/x86_64-linux-gnu/libodbc.so.1.0.0 7f70057e9000-7f70059e8000 ---p 00060000 08:01 801505 /usr/lib/x86_64-linux-gnu/libodbc.so.1.0.0 7f70059e8000-7f70059f0000 rw-p 0005f000 08:01 801505 /usr/lib/x86_64-linux-gnu/libodbc.so.1.0.0 7f70059f0000-7f70059f1000 rw-p 00000000 00:00 0 7f70059f1000-7f7005a0d000 r-xp 00000000 08:01 923645 /usr/local/lib64/kamailio/modules/db_unixodbc.so 7f7005a0d000-7f7005c0c000 ---p 0001c000 08:01 923645 /usr/local/lib64/kamailio/modules/db_unixodbc.so 7f7005c0c000-7f7005c0d000 rw-p 0001b000 08:01 923645 /usr/local/lib64/kamailio/modules/db_unixodbc.so 7f7005c0d000-7f7005c17000 r-xp 00000000 08:01 923745 /usr/local/lib64/kamailio/modules/xlog.so 7f7005c17000-7f7005e16000 ---p 0000a000 08:01 923745 /usr/local/lib64/kamailio/modules/xlog.so 7f7005e16000-7f7005e17000 rw-p 00009000 08:01 923745 /usr/local/lib64/kamailio/modules/xlog.so 7f7005e17000-7f7005e37000 r-xp 00000000 08:01 923729 /usr/local/lib64/kamailio/modules/tmx.so 7f7005e37000-7f7006037000 ---p 00020000 08:01 923729 /usr/local/lib64/kamailio/modules/tmx.so 7f7006037000-7f7006038000 rw-p 00020000 08:01 923729 /usr/local/lib64/kamailio/modules/tmx.so 7f7006038000-7f700603a000 rw-p 00000000 00:00 0 7f700603a000-7f7006050000 r-xp 00000000 08:01 923723 /usr/local/lib64/kamailio/modules/textopsx.so 7f7006050000-7f7006250000 ---p 00016000 08:01 923723 /usr/local/lib64/kamailio/modules/textopsx.so 7f7006250000-7f7006251000 rw-p 00016000 08:01 923723 /usr/local/lib64/kamailio/modules/textopsx.so 7f7006251000-7f700628b000 r-xp 00000000 08:01 923722 /usr/local/lib64/kamailio/modules/textops.so 7f700628b000-7f700648b000 ---p 0003a000 08:01 923722 /usr/local/lib64/kamailio/modules/textops.so 7f700648b000-7f700648c000 rw-p 0003a000 08:01 923722 /usr/local/lib64/kamailio/modules/textops.so 7f700648c000-7f700648d000 rw-p 00000000 00:00 0 7f700648d000-7f70064a7000 r-xp 00000000 08:01 923718 /usr/local/lib64/kamailio/modules/sqlops.so 7f70064a7000-7f70066a6000 ---p 0001a000 08:01 923718 /usr/local/lib64/kamailio/modules/sqlops.so 7f70066a6000-7f70066a8000 rw-p 00019000 08:01 923718 /usr/local/lib64/kamailio/modules/sqlops.so 7f70066a8000-7f70066be000 r-xp 00000000 08:01 923715 /usr/local/lib64/kamailio/modules/sl.so 7f70066be000-7f70068be000 ---p 00016000 08:01 923715 /usr/local/lib64/kamailio/modules/sl.so 7f70068be000-7f70068bf000 rw-p 00016000 08:01 923715 /usr/local/lib64/kamailio/modules/sl.so 7f70068bf000-7f70068ef000 r-xp 00000000 08:01 923714 /usr/local/lib64/kamailio/modules/siputils.so 7f70068ef000-7f7006aef000 ---p 00030000 08:01 923714 /usr/local/lib64/kamailio/modules/siputils.so 7f7006aef000-7f7006af0000 rw-p 00030000 08:01 923714 /usr/local/lib64/kamailio/modules/siputils.so 7f7006af0000-7f7006b09000 r-xp 00000000 08:01 923707 /usr/local/lib64/kamailio/modules/sanity.so 7f7006b09000-7f7006d08000 ---p 00019000 08:01 923707 /usr/local/lib64/kamailio/modules/sanity.so 7f7006d08000-7f7006d09000 rw-p 00018000 08:01 923707 /usr/local/lib64/kamailio/modules/sanity.so 7f7006d09000-7f7006d32000 r-xp 00000000 08:01 923703 /usr/local/lib64/kamailio/modules/rr.so 7f7006d32000-7f7006f32000 ---p 00029000 08:01 923703 /usr/local/lib64/kamailio/modules/rr.so 7f7006f32000-7f7006f33000 rw-p 00029000 08:01 923703 /usr/local/lib64/kamailio/modules/rr.so 7f7006f33000-7f7006fbe000 r-xp 00000000 08:01 923699 /usr/local/lib64/kamailio/modules/pv.so 7f7006fbe000-7f70071be000 ---p 0008b000 08:01 923699 /usr/local/lib64/kamailio/modules/pv.so 7f70071be000-7f70071c2000 rw-p 0008b000 08:01 923699 /usr/local/lib64/kamailio/modules/pv.so 7f70071c2000-7f70071c9000 r-xp 00000000 08:01 923674 /usr/local/lib64/kamailio/libbinrpc.so.0.1 7f70071c9000-7f70073c9000 ---p 00007000 08:01 923674 /usr/local/lib64/kamailio/libbinrpc.so.0.1 7f70073c9000-7f70073ca000 rw-p 00007000 08:01 923674 /usr/local/lib64/kamailio/libbinrpc.so.0.1 7f70073ca000-7f70073d1000 r-xp 00000000 08:01 923677 /usr/local/lib64/kamailio/modules/mi_rpc.so 7f70073d1000-7f70075d1000 ---p 00007000 08:01 923677 /usr/local/lib64/kamailio/modules/mi_rpc.so 7f70075d1000-7f70075d2000 rw-p 00007000 08:01 923677 /usr/local/lib64/kamailio/modules/mi_rpc.so 7f70075d2000-7f70075ed000 r-xp 00000000 08:01 923673 /usr/local/lib64/kamailio/modules/mi_fifo.so 7f70075ed000-7f70077ed000 ---p 0001b000 08:01 923673 /usr/local/lib64/kamailio/modules/mi_fifo.so 7f70077ed000-7f70077ee000 rw-p 0001b000 08:01 923673 /usr/local/lib64/kamailio/modules/mi_fifo.so 7f70077ee000-7f70077f4000 r-xp 00000000 08:01 923670 /usr/local/lib64/kamailio/modules/maxfwd.so 7f70077f4000-7f70079f4000 ---p 00006000 08:01 923670 /usr/local/lib64/kamailio/modules/maxfwd.so 7f70079f4000-7f70079f5000 rw-p 00006000 08:01 923670 /usr/local/lib64/kamailio/modules/maxfwd.so 7f70079f5000-7f7007a76000 r-xp 00000000 08:01 918696 /lib/x86_64-linux-gnu/libm-2.13.so 7f7007a76000-7f7007c75000 ---p 00081000 08:01 918696 /lib/x86_64-linux-gnu/libm-2.13.so 7f7007c75000-7f7007c76000 r--p 00080000 08:01 918696 /lib/x86_64-linux-gnu/libm-2.13.so 7f7007c76000-7f7007c77000 rw-p 00081000 08:01 918696 /lib/x86_64-linux-gnu/libm-2.13.so 7f7007c77000-7f7007c87000 r-xp 00000000 08:01 923646 /usr/local/lib64/kamailio/libsrutils.so.1.0 7f7007c87000-7f7007e86000 ---p 00010000 08:01 923646 /usr/local/lib64/kamailio/libsrutils.so.1.0 7f7007e86000-7f7007e87000 rw-p 0000f000 08:01 923646 /usr/local/lib64/kamailio/libsrutils.so.1.0 7f7007e87000-7f7007e99000 r-xp 00000000 08:01 923665 /usr/local/lib64/kamailio/modules/kex.so 7f7007e99000-7f7008098000 ---p 00012000 08:01 923665 /usr/local/lib64/kamailio/modules/kex.so 7f7008098000-7f700809a000 rw-p 00011000 08:01 923665 /usr/local/lib64/kamailio/modules/kex.so 7f700809a000-7f70080e3000 r-xp 00000000 08:01 923565 /usr/local/lib64/kamailio/modules/ctl.so 7f70080e3000-7f70082e2000 ---p 00049000 08:01 923565 /usr/local/lib64/kamailio/modules/ctl.so 7f70082e2000-7f70082e3000 rw-p 00048000 08:01 923565 /usr/local/lib64/kamailio/modules/ctl.so 7f70082e3000-7f70082f4000 rw-p 00000000 00:00 0 7f70082f4000-7f7008303000 r-xp 00000000 08:01 923561 /usr/local/lib64/kamailio/modules/corex.so 7f7008303000-7f7008503000 ---p 0000f000 08:01 923561 /usr/local/lib64/kamailio/modules/corex.so 7f7008503000-7f7008504000 rw-p 0000f000 08:01 923561 /usr/local/lib64/kamailio/modules/corex.so 7f7008504000-7f700850c000 r-xp 00000000 08:01 923544 /usr/local/lib64/kamailio/libkmi.so.1.0 7f700850c000-7f700870b000 ---p 00008000 08:01 923544 /usr/local/lib64/kamailio/libkmi.so.1.0 7f700870b000-7f700870c000 rw-p 00007000 08:01 923544 /usr/local/lib64/kamailio/libkmi.so.1.0 7f700870c000-7f700871b000 r-xp 00000000 08:01 923559 /usr/local/lib64/kamailio/modules/cfgutils.so 7f700871b000-7f700891a000 ---p 0000f000 08:01 923559 /usr/local/lib64/kamailio/modules/cfgutils.so 7f700891a000-7f700891b000 rw-p 0000e000 08:01 923559 /usr/local/lib64/kamailio/modules/cfgutils.so 7f700891b000-7f7008928000 r-xp 00000000 08:01 923533 /usr/local/lib64/kamailio/libkcore.so.1.0 7f7008928000-7f7008b27000 ---p 0000d000 08:01 923533 /usr/local/lib64/kamailio/libkcore.so.1.0 7f7008b27000-7f7008b28000 rw-p 0000c000 08:01 923533 /usr/local/lib64/kamailio/libkcore.so.1.0 7f7008b28000-7f7008b54000 r-xp 00000000 08:01 923529 /usr/local/lib64/kamailio/libsrdb1.so.1.0 7f7008b54000-7f7008d53000 ---p 0002c000 08:01 923529 /usr/local/lib64/kamailio/libsrdb1.so.1.0 7f7008d53000-7f7008d54000 rw-p 0002b000 08:01 923529 /usr/local/lib64/kamailio/libsrdb1.so.1.0 7f7008d54000-7f7008d97000 r-xp 00000000 08:01 923543 /usr/local/lib64/kamailio/modules/avpops.so 7f7008d97000-7f7008f97000 ---p 00043000 08:01 923543 /usr/local/lib64/kamailio/modules/avpops.so 7f7008f97000-7f7008f98000 rw-p 00043000 08:01 923543 /usr/local/lib64/kamailio/modules/avpops.so 7f7008f98000-7f7008f99000 rw-p 00000000 00:00 0 7f7008f99000-7f7008fb6000 r-xp 00000000 08:01 923542 /usr/local/lib64/kamailio/modules/avp.so 7f7008fb6000-7f70091b5000 ---p 0001d000 08:01 923542 /usr/local/lib64/kamailio/modules/avp.so 7f70091b5000-7f70091b6000 rw-p 0001c000 08:01 923542 /usr/local/lib64/kamailio/modules/avp.so 7f70091b6000-7f70092c4000 r-xp 00000000 08:01 923727 /usr/local/lib64/kamailio/modules/tm.so 7f70092c4000-7f70094c3000 ---p 0010e000 08:01 923727 /usr/local/lib64/kamailio/modules/tm.so 7f70094c3000-7f70094c7000 rw-p 0010d000 08:01 923727 /usr/local/lib64/kamailio/modules/tm.so 7f70094c7000-7f7009cdb000 rw-p 00000000 00:00 0 7f7009cdb000-7f7009e5c000 r-xp 00000000 08:01 918702 /lib/x86_64-linux-gnu/libc-2.13.so 7f7009e5c000-7f700a05c000 ---p 00181000 08:01 918702 /lib/x86_64-linux-gnu/libc-2.13.so 7f700a05c000-7f700a060000 r--p 00181000 08:01 918702 /lib/x86_64-linux-gnu/libc-2.13.so 7f700a060000-7f700a061000 rw-p 00185000 08:01 918702 /lib/x86_64-linux-gnu/libc-2.13.so 7f700a061000-7f700a066000 rw-p 00000000 00:00 0 7f700a066000-7f700a079000 r-xp 00000000 08:01 918715 /lib/x86_64-linux-gnu/libresolv-2.13.so 7f700a079000-7f700a278000 ---p 00013000 08:01 918715 /lib/x86_64-linux-gnu/libresolv-2.13.so 7f700a278000-7f700a279000 r--p 00012000 08:01 918715 /lib/x86_64-linux-gnu/libresolv-2.13.so 7f700a279000-7f700a27a000 rw-p 00013000 08:01 918715 /lib/x86_64-linux-gnu/libresolv-2.13.so 7f700a27a000-7f700a27c000 rw-p 00000000 00:00 0 7f700a27c000-7f700a27e000 r-xp 00000000 08:01 918708 /lib/x86_64-linux-gnu/libdl-2.13.so 7f700a27e000-7f700a47e000 ---p 00002000 08:01 918708 /lib/x86_64-linux-gnu/libdl-2.13.so 7f700a47e000-7f700a47f000 r--p 00002000 08:01 918708 /lib/x86_64-linux-gnu/libdl-2.13.so 7f700a47f000-7f700a480000 rw-p 00003000 08:01 918708 /lib/x86_64-linux-gnu/libdl-2.13.so 7f700a480000-7f700a4a0000 r-xp 00000000 08:01 918714 /lib/x86_64-linux-gnu/ld-2.13.so 7f700a68b000-7f700a68c000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a68c000-7f700a68d000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a68d000-7f700a68e000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a68e000-7f700a695000 r--s 00000000 08:01 784486 /usr/lib/x86_64-linux-gnu/gconv/gconv-modules.cache 7f700a695000-7f700a698000 rw-p 00000000 00:00 0 7f700a698000-7f700a699000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a699000-7f700a69a000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a69a000-7f700a69b000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a69b000-7f700a69c000 rw-s 00000000 00:04 0 /SYSV790116c2 (deleted) 7f700a69c000-7f700a69f000 rw-p 00000000 00:00 0 7f700a69f000-7f700a6a0000 r--p 0001f000 08:01 918714 /lib/x86_64-linux-gnu/ld-2.13.so 7f700a6a0000-7f700a6a1000 rw-p 00020000 08:01 918714 /lib/x86_64-linux-gnu/ld-2.13.so 7f700a6a1000-7f700a6a2000 rw-p 00000000 00:00 0 7ffe7071f000-7ffe70740000 rw-p 00000000 00:00 0 [stack] 7ffe707bb000-7ffe707bc000 r-xp 00000000 00:00 0 [vdso] ffffffffff600000-ffffffffff601000 r-xp 00000000 00:00 0 [vsyscall] 13(9728) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 16 0(9715) ALERT: <core> [main.c:784]: handle_sigs(): child process 9724 exited by a signal 6 0(9715) ALERT: <core> [main.c:787]: handle_sigs(): core was not generated 0(9715) INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD 12(9727) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 8(9723) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 4(9719) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 7(9722) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 3(9718) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 10(9725) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 2(9717) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 5(9720) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 1(9716) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 13(9728) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 6(9721) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 11(9726) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 0(9715) ERROR: ctl [ctl.c:382]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1)

2 9

Nonce lifetime in RADIUS auth module
by Måns Nilsson 31 Mar '15

31 Mar '15

Hi, I have a problem with stale nonces being reused by a UA. I'd like to do two things; * Set the nonce lifetime to a lower value * Handle stale nonces in my authentication route For the first, I'm interpreting the RADIUS module documentation so that it gets such parameters from the generic auth module and, therefore, I should be good with something like: ... modparam("auth", "nonce_expire", 60) # Set nonce_expire to 60s ... For the second, I'd appreciate pointers on how to set stale=true in the 401 reply, if special handling is required. I have incomplete packet traces to work from right now, so can't tell if it is being set already. Thanks, -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 I'm wearing PAMPERS!!

1 0

htable persistation question
by Attila Megyeri 31 Mar '15

31 Mar '15

Hi, We are having problem with persisting htable. There are 4 hashtables defined in kamailio config, like: modparam("htable", "db_url", DBURL) modparam("htable", "htable", "vtp=>size=10;autoexpire=120;dbtable=htable;dbmode=1;") modparam("htable", "htable", "a=>size=6;") modparam("htable", "htable", "rpn=>size=10;autoexpire=0;dbtable=htable;dbmode=1;"); modparam("htable", "htable", "rpnn=>size=10;autoexpire=0;dbtable=htable;dbmode=1;"); The strange behavior is that if I restart kamailio, the hash table is not persisted into the database. If I change e.g. rpn's dbtable to e.g. htable_1 (which should be created in the kamailio mysql database schema upfront, then the rpn hashtable will be presisted successfully at every stop and will be reload at every start. What is wrong in my config? Kamailio version is 4.2.3 Thanks, Attila

2 2

Share location table 3.x 4.x
by Jon Bonilla (Manwe) 31 Mar '15

31 Mar '15

Hi all I'm sharing a location table between two kamailio 3.3 and kamailio 4.2 servers. Table versions are 5 and 6 but I don't know how to skip table version checking to prevent one of them from running. Any tips? thanks, Jon

2 2

keep Kamailio running
by Mihaly Zachar 31 Mar '15

31 Mar '15

Hi All, As far as I can see if a child dies due to segmentation fault Kamailio stops all the childs and it exits. Is there a possibility to change this behaviour to a kind of "respawn the child" way ? What do you think is the best way to keep Kamailio running on Ubuntu ? On RHEL ? Thanks, Misi

4 4

please help
by mail＠sathees.co.uk 31 Mar '15

31 Mar '15

Hi, I’m trying to setup kamailio as proxy to asterisk. Asterisk server running on public IP. I want to hide the asterisk behind kamailio. UAC à Kamailio à Asterisk media services à PSTN PSTN à Asterisk media services à Kamailio à UAC I like to use dispatcher or load balancer modules. I have tried Kamailio 4.0.x and Asterisk 11.3.0 Real time Integration no luck. Can some mail me working scripts or some sort of idea? Appreciate. Sathees

2 1

is_present_hf error on select
by Kyle Kurz 30 Mar '15

30 Mar '15

I'm trying to use the is_present_hf function to detect the presence of a user-defined header and getting the following errors in my Kamailio logs: 2(12149) BUG: <core> [select.c:426]: run_select(): Select structure has not been resolved 2(12149) ERROR: <core> [lvalue.c:416]: lval_assign(): assignment failed at pos: (99,58-99,58) 2(12149) ERROR: <script>: A quick Google search yielded no results for why this could be happening. Originally, I had my code in route[REPLY], but noticed that was not one of the paths it was listed as available in the textops document, so I moved it to request_route, but got the same errors. Does anyone have any feedback on how to move forward? My code snippet is below: if (is_present_hf("X-Digium-AppServer-Response-URI")) { $var(a) = @hf_value.X_Digium_AppServer_Response_URI[1]; xlog ("\n\nHERE RESPONSE URI $var(a)\n\n"); } Thanks, Kyle -- *Kyle Kurz* Digium, Inc. · Senior Software Engineer 9115 Brown Deer Rd · San Diego, CA 92121 · US direct +1 858.537.1166 Check us out at: http://digium.com · http://asterisk.org <http://www.asterisk.org/>

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Re: [SR-Users] NAT Contact:
by Slava Bendersky 29 Mar '15

29 Mar '15

Hello Alex, This is debug might be helpful. Mar 29 12:24:05 canlvprx01 kamailio: 11(31941) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Mar 29 12:24:05 canlvprx01 kamailio: 11(31941) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Mar 29 12:24:05 canlvprx01 kamailio: 11(31941) DEBUG: sl [sl_funcs.c:412]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, March 29, 2015 11:56:58 AM Subject: Re: [SR-Users] NAT Contact: Hello Alex, I switched to aliases method, but in case like this, AU ----> Internet ---> Kamailio ----> Asterisk AU2 <--Internet <---------| <------------| Call between two extensions not working, because kamailio mark as external domain and LOCATION not saved, but even if I save location I see in kamctl ul show more then 5 records for each AOR. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, March 29, 2015 9:52:18 AM Subject: Re: [SR-Users] NAT Contact: Hello Alex, I switched to alias method, but there some CRITICAL message. Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=901 a=24 n=is_first_hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) CRITICAL: <core> [route.c:1237]: comp_num(): Invalid right operand (1) Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=902 a=5 n=route Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=976 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=971 a=24 n=add_contact_alias Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: nathelper [nathelper.c:970]: add_contact_alias_0_f(): no need to add alias param From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, March 28, 2015 7:31:54 PM Subject: Re: [SR-Users] NAT Contact: On 03/28/2015 07:26 PM, Slava Bendersky wrote: > Is for mix of tcp transport and alias is required add any additional > parameters in global configuration ? I wouldn't think so. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] NAT Contact:
by Slava Bendersky 29 Mar '15

29 Mar '15

Hello Alex, I switched to aliases method, but in case like this, AU ----> Internet ---> Kamailio ----> Asterisk AU2 <--Internet <---------| <------------| Call between two extensions not working, because kamailio mark as external domain and LOCATION not saved, but even if I save location I see in kamctl ul show more then 5 records for each AOR. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, March 29, 2015 9:52:18 AM Subject: Re: [SR-Users] NAT Contact: Hello Alex, I switched to alias method, but there some CRITICAL message. Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=901 a=24 n=is_first_hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) CRITICAL: <core> [route.c:1237]: comp_num(): Invalid right operand (1) Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=902 a=5 n=route Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=976 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=971 a=24 n=add_contact_alias Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: nathelper [nathelper.c:970]: add_contact_alias_0_f(): no need to add alias param From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, March 28, 2015 7:31:54 PM Subject: Re: [SR-Users] NAT Contact: On 03/28/2015 07:26 PM, Slava Bendersky wrote: > Is for mix of tcp transport and alias is required add any additional > parameters in global configuration ? I wouldn't think so. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] NAT Contact:
by Slava Bendersky 28 Mar '15

28 Mar '15

Hello Alex, Is for mix of tcp transport and alias is required add any additional parameters in global configuration ? Slava. From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, March 28, 2015 5:26:16 PM Subject: Re: [SR-Users] NAT Contact: Hello Slava, On 03/28/2015 02:26 PM, Slava Bendersky wrote: > How is possible add to Contact: received=ip ? Is this should be > handled by fix_nated_contact() This is not the function of fix_nated_contact(). Have a look at a add_rcv_param(): http://kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.a… Someone might correct me if I'm wrong, but I believe the problem with the ;received approach is that it's only useful for registrations, since only the registrar module has a provision that allows it to be used automatically. So, for larger applications outside of registration, you should take a look at the new add_contact_alias() / handle_ruri_alias() facility: http://kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.a… http://kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.h… -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

NAT Contact:
by Slava Bendersky 28 Mar '15

28 Mar '15

Hello Everyone, How is possible add to Contact: received=ip ? Is this should be handled by fix_nated_contact() ? I am using default NATDETECT route, but I see in debug fix_nated_contact() executed, but in SDP Contact: stay Contact: <sip:102@private_ip:5065;transport=tcp>;audio. >From public side client use TCP inbound to kamailio and on local side to asterisk is use UDP. Kamailio have 2 interfaces one public one private. Any help thank you. Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=895 a=55 n=force_rport Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=16 n=if Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=896 a=25 n=nat_uac_test Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=901 a=16 n=if Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=897 a=25 n=is_method Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=900 a=24 n=fix_nated_contact Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=902 a=39 n=setflag Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=2 n=return

2 1

DNS resolver.
by Slava Bendersky 28 Mar '15

28 Mar '15

Hello Everyone, How I can disable DNS resolution ? Currently I have in configuration file, but I still see in log DNS SRV resolution and forward. dns_cache_init=off use_dns_cache=off use_dns_failover=off dns=no rev_dns=no Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [resolve.c:1239]: srv_sip_resolvehost(): srv_sip_resolvehost: domain.org:0 proto=0 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [resolve.c:1329]: srv_sip_resolvehost(): srv_sip_resolvehost: SRV(_sip._udp.domain.org) = canlpbx02.domain.org:5060 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [resolve.c:1360]: srv_sip_resolvehost(): srv_sip_resolvehost: returning 0x7f2bf81f9680 (domain.org:5060 proto=1) Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [forward.c:211]: get_out_socket(): DEBUG: get_out_socket: socket determined: 0x7f2bf765e2d8 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [msg_translator.c:2893]: create_via_hf(): create_via_hf: id added: <;i=2>, rcv proto=2 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: tm [t_funcs.c:394]: t_relay_to(): SER: new transaction fwd'ed

1 0

Asking for help in configuration Kamailio
by Mikael Sarkisyan 28 Mar '15

28 Mar '15

Hello Dear Sip Router Team I’m asking you to help me please with kamailio configuration I’ve made some laboratory system following this document http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb <http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb> and it works perfect. But this scenario is not good for my tests. What i really want is: UAC going to Kamailio IP address and Kamailio forwarding request to Asterisk IP without auth on Kamailio and auth with asterisk .conf files (i mean without mysql at all) But i wanna keep data about what to do will presence and other stuff Can you make changes in attached files please?

3 3

Reload htable
by Grant Bagdasarian 28 Mar '15

28 Mar '15

Hello, I need to reload a htable from the database when a certain condition is met inside my kamailio script. I found the htable.reload htable RPC command, but is it possible to run this from within the script? Is there perhaps another way to force reload the htable inside the script? Thanks, Grant

4 6

evapi_relay some problems
by dolphin 28 Mar '15

28 Mar '15

Hello, I test the evapi modules function , but not full operation ,only receive not send to client I capture the important setup parameters are as follows: debug=3 loadmodule "evapi.so" #----- evapi params ----- # modparam("evapi", "bind_addr", "0.0.0.0:8448") modparam("evapi", "netstring_format", 0) modparam("evapi", "workers", 4) event_route[evapi:connection-new] { evapi_relay("{ \"event\": \"test\",\n \"data\": { \"fU\": \"fU\" }\n}"); xlog("L_WARN", "new connection from $evapi(srcaddr):$evapi(srcport)\n"); } scene : One client connected kamailio server x.x.x.x:8448 use nc tools , I get garbled , kamailioserver not sent to me client , Because this is the right format to display josn ? sh-4.2# nc -n x.x.x.x 8448 ; øX±š^C any suggestions or missing something ? my syslog : Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: DEBUG: evapi [evapi_dispatch.c:385]: evapi_accept_client(): new connection - pos[0] from: [x.x.x.x:56215] Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: DEBUG: evapi [evapi_dispatch.c:531]: evapi_relay(): relaying event data [{ "event": "test",#012 "data": { "fU": "fU" }#012}] Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: DEBUG: evapi [evapi_dispatch.c:561]: evapi_relay(): sent [0x7f939ab158f8] [{ "event": "test",#012 "data": { "fU": "fU" }#012}] (44) Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: WARNING: <script>: new connection from x.x.x.x:56215 kamailio server : root@SIP:/etc/init.d# dpkg -l | grep libev-dev ii libev-dev 1:4.11-1 amd64 static library, header files, and docs for libev root@SIP:/etc/init.d# uname -a Linux SIP 3.2.0-4-amd64 #1 SMP Debian 3.2.46-1+deb7u1 x86_64 GNU/Linux root@SIP:/home/pkg/kamailio-4.2.3/sbin# ./kamailio -V version: kamailio 4.2.3 (x86_64/linux) 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: unknown compiled on 15:13:09 Mar 27 2015 with gcc 4.7.2 download : http://www.kamailio.org/pub/kamailio/latest/src/kamailio-4.2.3_src.tar.gz

1 0

Routing PSTN calls from Asterisk through Kamailio to UAC & loose_route
by Anthony Messina 27 Mar '15

27 Mar '15

I've been working on integration of Asterisk and Kamailio, currently on the same host with different ports, and have come across a problem with calls that originate from the Asterisk side (PSTN/DAHDI) and route through Kamailio to a SIP UAC. In short, when the SIP UAC (10.1.1.9) sends the BYE, loose_route() is returning -1 and the BYE is routed back to Kamailio (10.1.1.1:5060) instead of Asterisk (10.1.1.1:5080). I am using the stock WITHINDLG route configuration. RR module settings are as follows: modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) The BYE from the SIP UAC contains the following Route header which only contains the contents of Kamailio's Record-Route header. I have attached the full sip trace for review as well. Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>. What would be the best method to resolve this issue in either Asterisk or Kamailio? Should I manually add a Record-Route header for the Asterisk host:port to Kamailio config? Is there something to be done in Asterisk? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 2

Kamailio - Used By Directory
by Daniel-Constantin Mierla 27 Mar '15

27 Mar '15

Hello, in order to facilitate easier interaction and updates, references (http://www.kamailio.org/w/references/) and some of the business use cases already listed on kamailio.org are in the process of being moved to a new page: - http://www.kamailio.org/w/used-by/ This offers now the possibility to submit new entries without the interaction with Kamailio admins that could lead to delays. The form is available at: - http://www.kamailio.org/w/used-by-form/ The goals is to show where Kamailio is used as well as increase the visibility of the products, solutions and services offered by the Kamailio community. Anyone is welcome to add own references. Besides a spambot protection of the form, we rely for now on the good manners of human nature, if you notice anything inadequate there, please report to be corrected. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

tsilo module
by Alex Balashov 27 Mar '15

27 Mar '15

Hello, I don't understand the tsilo module. What is the practical use case for it? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 3

Kamailio BLF Issue TCP vs UDP
by Fred Posner 27 Mar '15

27 Mar '15

Greetings all: Having a weird issue with BLF relay from Kamailio <-> FreeSWITCH on the EC2 network. Set-up: Natted Client <-> Natted Kamailio <-> Natted FreeSWITCH Kamailio has a public advertised IP (Amazon Cloud) and sends to FreeSWITCH's public IP (Amazon Cloud). Clients register to Kamailio and the registration sent to FreeSWITCH (via uac). What works: When a client registers to kamailio via UDP, all works beautifully. What doesn't work: When a client registers to kamailio via TCP, a NOTIFY is sent with almost no information. Example of UDP traffic: U 2015/03/26 22:12:30.483767 FREESWITCH_IPADDR:5060 -> KAMAILIO_NAT_IPADDR:5060 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKv7X88FmKg04UF. Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Record-Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Max-Forwards: 70. From: <sip:3000@CLIENT_DOMAIN;user=phone>;tag=YF4HMSQNlOOt. To: <sip:3000@CLIENT_DOMAIN>;tag=xew5nmpr9q. Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj. CSeq: 366877453 NOTIFY. Contact: <sip:3000@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=2686. Content-Type: application/dialog-info+xml. Content-Length: 546. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="14" state="full" entity="sip:3000@CLIENT_DOMAIN"> <dialog id="312b3bee-d405-11e4-a8b6-41b65d0bd045" direction="initiator"> <state>confirmed</state> <local> <identity display="3000">sip:3000@CLIENT_DOMAIN</identity> <target uri="sip:3000@CLIENT_DOMAIN"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="*98">sip:*98@CLIENT_DOMAIN</identity> <target uri="sip:**3000@CLIENT_DOMAIN"/> </remote> </dialog> </dialog-info> U 2015/03/26 22:12:30.483877 KAMAILIO_NAT_IPADDR:5060 -> CLIENT_PUBLIC_IPADDR:42477 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0. Via: SIP/2.0/UDP KAMAILIO_PUBLIC_IPADDR:5060;branch=z9hG4bKfabf.4908255bf1623d250ddf7d51a3b49fb8.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKv7X88FmKg04UF. Record-Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Max-Forwards: 69. From: <sip:3000@CLIENT_DOMAIN;user=phone>;tag=YF4HMSQNlOOt. To: <sip:3000@CLIENT_DOMAIN>;tag=xew5nmpr9q. Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj. CSeq: 366877453 NOTIFY. Contact: <sip:3000@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=2686. Content-Type: application/dialog-info+xml. Content-Length: 546. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="14" state="full" entity="sip:3000@CLIENT_DOMAIN"> <dialog id="312b3bee-d405-11e4-a8b6-41b65d0bd045" direction="initiator"> <state>confirmed</state> <local> <identity display="3000">sip:3000@CLIENT_DOMAIN</identity> <target uri="sip:3000@CLIENT_DOMAIN"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="*98">sip:*98@CLIENT_DOMAIN</identity> <target uri="sip:**3000@CLIENT_DOMAIN"/> </remote> </dialog> </dialog-info> Example of TCP failure: U 2015/03/26 22:19:45.963196 FREESWITCH_IPADDR:5060 -> KAMAILIO_NAT_IPADDR:5160 NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKjH5Nt8QKUU9Sp. Route: <sip:KAMAILIO_PUBLIC_IPADDR:5160;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes>. Route: <sip:KAMAILIO_PUBLIC_IPADDR:5160;transport=tcp;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes>. Max-Forwards: 70. From: "3000" <sip:3000@CLIENT_DOMAIN>;tag=9jSFr02tgvHym. To: <sip:3001@CLIENT_DOMAIN>;tag=1dba0f3ceb32436bi0. Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0. CSeq: 73368127 NOTIFY. Contact: <sip:mod_sofia@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: refer;id=101. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Content-Length: 16. . SIP/2.0 200 OK. T 2015/03/26 22:19:45.963407 KAMAILIO_NAT_IPADDR:5160 -> CLIENT_PUBLIC_IPADDR:5065 [AP] NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0. Via: SIP/2.0/TCP KAMAILIO_PUBLIC_IPADDR:5160;branch=z9hG4bKe961.8d1f19f1acc0295e5e397afa76910115.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKjH5Nt8QKUU9Sp. Max-Forwards: 69. From: "3000" <sip:3000@CLIENT_DOMAIN>;tag=9jSFr02tgvHym. To: <sip:3001@CLIENT_DOMAIN>;tag=1dba0f3ceb32436bi0. Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0. CSeq: 73368127 NOTIFY. Contact: <sip:mod_sofia@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: refer;id=101. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Content-Length: 16. . SIP/2.0 200 OK. My thoughts is that this is on FreeSWITCH, just not sending a NOTIFY with anything of value. That being said, I'd love to know if (a) anyone else has had this issue or (b) if someone has a different theory. Thanks! -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

4 11

rtpengine DTLS
by Marc Soda 25 Mar '15

25 Mar '15

I'm running a new kamailio/rtpengine instance and, using webrtc, I can't get audio to flow between the browser and rtpengine. Kamailio seems to be handing it off properly. Based on the rtpengine logs and a packet capture it doesn't look like DTLS is being negotiated properly: https://gist.github.com/marcantonio/ea0077a5884e4e4b6b45 I see "SRTCP output wanted, but no crypto suite was negotiated" and I've never seen that one before (the SRTCP part). I have this exact config working elsewhere, but this new setup is not working? Any advice? Thanks, Marc

1 1

Siremis error Call to a member function getQuoted()
by canuck15 25 Mar '15

25 Mar '15

When you add/update/view Administrator usernames To reproduce Administration > User Management > Click on any existing username. The browser goes blank. Error in apache logs is: PHP Fatal error: Call to a member function getQuoted() on a non-object in /var/www/html/openbiz/bin/data/BizDataSql.php on line 237 This is on the latest Siremis v4.2.0 downloaded today I am using CentOS v6.6, php v5.3.3

3 11

active_watchers - dbt_table_check_row(): null value not allowed - field 17
by Alexandru Raileanu 25 Mar '15

25 Mar '15

Hi, I have kamailio 4.2 with presence, auth, textdb activated. The scenario is that Kamailio receives a SUBSCRIBE, sends 200ok then a NOTIFY. (like it should do) The problem is what when I kill kamailio and run it again, it gives me this error: 0(30580) ERROR: db_text [dbt_tb.c:474]: dbt_table_check_row(): null value not allowed - field 17 0(30580) DEBUG: db_text [dbt_file.c:480]: dbt_load_file(): error at row=1 col=26 c=1 0(30580) ERROR: db_text [dbt_lib.c:307]: dbt_db_get_table(): could not load database from file [active_watchers] 0(30580) ERROR: db_text [dbt_base.c:460]: dbt_delete(): failed to load table <active_watchers>! The field with null values is record_route, so if I put some value I manage to run kamailio. my conf, maybe I missed some field? #!define DBURL "text:///etc/kamailio/dbtext" #!ifdef WITH_PRESENCE modparam("presence", "db_url", DBURL) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:10.0.37.1:5060" ) modparam("pua_reginfo", "default_domain", "10.0.37.1") modparam("pua_reginfo", "server_address", "sip:10.0.37.1") modparam("pua_reginfo", "outbound_proxy", "sip:10.0.37.1") modparam("pua_reginfo", "publish_reginfo", 1) modparam("pua", "db_mode", 2) modparam("pua", "hash_size", 11) modparam("pua", "db_url", DBURL) modparam("pua", "db_table", "pua") modparam("pua", "min_expires", 0) modparam("pua", "default_expires", 3600) modparam("pua", "update_period", 1000) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "reginfo_increase_version", 1) modparam("pua", "check_remote_contact", 0) modparam("pua", "fetch_rows", 1000) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif Or as a workaround, can I reset the active_watchers table every time I start kamailio? because *kamdbctl create/drop/init* wipes all the other tables... thanks a lot! Alex

2 2

File transferring issue
by Wingsravi R 25 Mar '15

25 Mar '15

Dear Kamailio'ns, I am working on kamailio Server (V 4.0) installed on Ubuntu 12.04 system. Currently i am trying file transfer feature using IMSDroid SIP clients. file transferring is fine between two IMSDroid clients but at the receiving end (file recipient) i cont see the sender identity/ sender name. it is just displaying as 'null' (Screen shot is attached). But the full file is transferring to destination address without any problem. I have configured MSRP module related script in my kamailio configuration file (find the attachment). but i guess it is not invoking anyway when file transferring and the file transferring is negotiating client-to-client basis. And also find Sip capture of file transfer (ngrep based) for better understanding. What could be the problem? Why at the file recipient side it is not displaying sender's identity (instead it is 'null')? how can i resolve this issue? Anyone please help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

5 6

[4.2.x] "socket" feature missing in dispatcher module
by Sven Neuhaus 25 Mar '15

25 Mar '15

Hello, in 4.2.2, the "socket" module was added to the documentation of the dispatcher module (see http://kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatcher.e… ), however the code is missing, the feature isn't implemented in 4.2.x. This was in commit 9bb59f922. Please include the patch that adds this feature from the trunk in the 4.2.x branch, as documented. A brief search turned up the commits aae775859f54f84734ba2cfc2945c02c52b95e1f and 2a605383d5c524d9875256c7f95f64216b970331 with this feature, perhaps there are more. Thanks, -Sven Neuhaus

2 1

Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 24 Mar '15

24 Mar '15

Hello Daniel, Look like transactions are not match when it send reply for NOTIFY. I think it should reuse connection. Here SDP output where cseq are match and asterisk does retransmissions until time out. Also I wonder where Path route is reused I see %* instead ; as delimiter between transport and port. Retransmitting #8 (NAT) to 10.18.130.46:5060: NOTIFY sip:191@192.168.88.246:5062;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport Max-Forwards: 70 Route: <sip:inbound@10.18.130.46;lr;received=sip:client_public_ip:5062%3Btransport%3Dtcp> From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp> Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 24/0 (0/0) --- Retransmitting #9 (NAT) to 10.18.130.46:5060: NOTIFY sip:191@192.168.88.246:5062;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport Max-Forwards: 70 Route: <sip:inbound@10.18.130.46;lr;received=sip:99.251.238.38:5062%3Btransport%3Dtcp> right here From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp> Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 24/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport=5060;received=10.18.130.51 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp>;tag=dc299d9ac16f3377ac0b8dfa22eaf0a6-2fe2 Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Mar 23 08:45:47 canlvprx01 /usr/sbin/kamailio[20143]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect client_pub_ip:5063 failed (timeout) Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 23, 2015 4:21:47 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, as you can see the 408 is for another request, because the cseq is 103. The first one has cseq 102, but has no Record-Route header, which is mandated by new RFCs. In kamailio, if you added in request_route when handling the NOTIFY, you can try to add record route via reply_route if the client is not keeping it in reply, kind of following fix: reply_route { if(is_method("NOTIFY")) { if(!is_present_hf("Record-Route")) { insert_hf("Record-Route: < sip:yourserverip:port;transport=...\r\n "); } } Cheers, Daniel On 22/03/15 18:11, Slava Bendersky wrote: Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: <sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes> --->Kamailio route From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2> ;tag=3b27a2a409 Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2> ;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as00936a6d To: <sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2> ;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

5 8

msilo module
by Rajat Yadav 24 Mar '15

24 Mar '15

Hello i m very new to kamailio.I m trying to develop an application using webrtc and kamailio. In this i tried using msilo module so as to store offline messages but wasnt able to achieve it.After that i searched a lot on internet about working msilo codes but i didnt find it. Can someone please send me a working code of msilo which i can embed in the kamailio configuration file. Thank you.

2 1

ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1]
by Satish Patel 24 Mar '15

24 Mar '15

Hello, Recently i have installed Homer capture server with kamailio. It works! but in logs i am seeing many error like following, I am using Opensips 2.1 siptrace server to send all trace to Homer. ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1] ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1] ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1]

1 1

Compiling with db_unixodbc
by Grant Bagdasarian 24 Mar '15

24 Mar '15

Hello, I'm trying to compile Kamailio 4.2.3 on Debian 7.8 with db_unixodbc but the sql.h file can't be found. I've used the following command to compile and install unixODBC. tar -zxvf unixODBC-2.3.2.tar.gz && cd unixODBC-2.3.2 && ./configure --sysconfdir=/etc --prefix=/usr/local/unixODBC --disable-gui && make && make install #####################: locate odbc_config /usr/local/unixODBC/bin/odbc_config /usr/src/unixODBC-2.3.2/exe/odbc_config #####################: locate sql.h /usr/local/unixODBC/include/sql.h /usr/src/kamailio-4.2.3/modules/db_mysql/km_db_mysql.h /usr/src/kamailio-4.2.3/modules/db_postgres/pg_sql.h /usr/src/unixODBC-2.3.2/Drivers/Postgre7.1/isql.h /usr/src/unixODBC-2.3.2/Drivers/nn/isql.h /usr/src/unixODBC-2.3.2/Drivers/nn/nnsql.h /usr/src/unixODBC-2.3.2/exe/isql.h /usr/src/unixODBC-2.3.2/include/sql.h Kamailio Compile Error CC (gcc) [M db_unixodbc.so] connection.o In file included from connection.c:34:0: connection.h:42:17: fatal error: sql.h: No such file or directory compilation terminated. make[1]: *** [connection.o] Error 1 make: *** [modules] Error 1 Modules.lst include_modules= carrierroute db_unixodbc dialog_ng lcr tls uuid Removed these includes from the exclude_modules Any ideas?

3 3

Inter Kamailio communication: UDP vs TCP vs SCTP
by Mickael Marrache 24 Mar '15

24 Mar '15

Hi, We are deploying a VoIP system based on Kamailio. The system is composed of load balancers, proxies, registrars, etc What is the recommended transport protocol to use for internal communication (i.e. load balancer <-> proxy/registrar)? We started with UDP but we also see that Kamailio supports TCP and SCTP. For information, at first stage, the UAs will communicate using UDP but we may need to use TCP in case messages are too large (we had such issues when using ICE). Thanks, Mickael

2 1

Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Everyone, In my case I am using Path and I asterisk generate NOTIFY with route based on path, then I see on kamailio side tries reach on client on public IP side and open new connection not follow ESTABLISHED/RELATED link then it times out. Log: Mar 23 11:00:17 canlvprx01 /usr/sbin/kamailio[20193]: INFO: <script>: [NOTIFY], Inbound connection from <10.18.130.51> traffic from Asterisk Mar 23 11:00:17 canlvprx01 /usr/sbin/kamailio[20193]: INFO: <script>: [NOTIFY], Present Route: Client_public_IP. Mar 23 11:00:28 canlvprx01 /usr/sbin/kamailio[20204]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect Client_public_IP:5060 failed (timeout) Here trace on kamailio side http://fpaste.org/201813/15287114/ Here trace from soft phone client side. http://fpaste.org/201814/14271530/ Slava. From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 23, 2015 6:42:59 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Anthony, The Contact presented by the subscriber in the initial subscription is: m: "Test User" <sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme>;+sip.ice And the subsequent NOTIFY is correctly targeted to this RURI: NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme SIP/2.0 The subscriber returns this Contact in the 200 OK response for the NOTIFY: m: "Test User" <sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O>;+sip.ice And subsequent NOTIFYs appear to use this RURI instead of the original Contact of the subscriber: NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O SIP/2.0 NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O SIP/2.0 etc. A SUBSCRIBE is a dialog-creating event, so in-dialog messages (e.g. NOTIFYs) should be targeted at a Request URI corresponding to the Contact URI of the SUBSCRIBE-er. It seems to me that this is the problem here, rather than truncation. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamalio call issue
by Yogendra Gupta 23 Mar '15

23 Mar '15

Hello, We have setup kamalio at server and it is working at some DNS with call/chat. At some IP chat is working but when we call then it is not working. At some DNS(Net IP) it is working . What can be problem at server? Following is login for SIP test user: tester1(a)23.253.110.48 123456 Host IP : 23.253.110.48 I have attached mu kamalio config file. Let me know what we need to change at config file or server Thanks

2 9

What is the best SIP trunk authentication strategy
by canuck15 23 Mar '15

23 Mar '15

Been struggling with this for awhile now. So far I am finding it rather difficult to come up with way to authenticate SIP trunks taking into account all possible scenarios. My setup is Kamailio combined with Asterisk realtime. Everything is in a MySQL database. All authentication is done by Kamailio. Kamailio handles SIP extension (user) authentication quite well but seems poorly equipped for SIP trunk (peer) authentication. The biggest problem I have run into so far is that SIP trunks typically use DNS names and Kamailio is not at all designed to used DNS names. Also SIP trunks do not use named "realms" whereas that is a key part of Kamailio authentication. For IP authentication about the only solution I have found is to DNS lookup and save all returned IP addresses in the ip_addr mysql table. Then I do: #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address() && $au == "") { # source IP allowed return; } I can run a cron job every hour to DNS lookup and update the ip_addr table as needed so I think this is a satisfactory solution for IP authentication. SIP trunk user/pass authentication is the one I am now struggling with. This standard Kamailio authentication section does not work for SIP trunks. if (is_method("REGISTER|INVITE") || from_uri==myself) { # authenticate requests #if (!auth_check("$fd", "subscriber", "1")) { if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); This is a multidomain setup and therefore: modparam("auth_db", "use_domain", 1) So the authentication section will try authenticate the realm of the trunk which will always be an IP address. However the SIP trunk realm will usually be saved in the "subscriber" database as the FQDN. There is no mechanism for auth_check() to directly or indirectly check against FQDN. The other problem is that even if I used the ip_addr table somehow it will only try use the user/pass of the first instance of a matching IP address. For SIP trunks it is possible I could have multiple subaccounts and therefore multiple instances of the same IP with different user/pass. So auth_check() is not designed for being used this way as far as I can tell. Seems to me like there should almost be a special kamailio module just for SIP trunks. I had a look a carrierroute module it's not designed for this either.

4 6

Websocket kamailio.cfg doubt
by Austin Einter 23 Mar '15

23 Mar '15

The README of websocket suggests to modify kamailio.cfg as below *# Handle requests within SIP dialogsroute[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } route(RELAY); } else { if ( is_method("ACK") ) {...* But the original script was *# Handle requests within SIP dialogsroute[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit;}* Now I have to include the websocket README suggested script in original script. I am not finding logical way where shall I include websocket part. Can somebody please help me. Thanks Austin

2 1

Kamailio sending NOTIFY messages to private IP
by Olli Attila 23 Mar '15

23 Mar '15

Hello, Im running kamailio 4.2.3 and I'm having troubles with reporting user presence data from Kamailio server to a registered UA behind nat. The NOTIFY message is being sent from Kamailio to the UA after receiving PUBLISH, but because the UA is behind nat the NOTIFY message is being sent to the UAs private (=unroutable) ip and therefor never reaching the real UA. My server is using nat helping modules so that REGISTER messages and INVITE messages are handled ok even if they are originated behind nat. Any suggestions how to solve this and make the NOTIFY messages to be sent via routabe ip addresses? my kamailio.cfg PRESENCE routing: #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } -- Olli

2 1

Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: <sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes> --->Kamailio route From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2>;tag=3b27a2a409 Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2>;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as00936a6d To: <sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2>;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

Re: [SR-Users] remove_hf_value
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Daniel, I end up insert routes like this. And right now on client side I see only public ip route and private side only private ip routes. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("NOTIFY")) { if(is_ip_rfc1918($si)) { xlog("L_INFO", "[$rm], Present Route: $(route_uri{uri.host})\n"); remove_hf("Route"); insert_hf("Record-Route: <"sip" + $sel(cfg_get.kamailio.bindpublicip) + ":" $sel(cfg_get.kamailio.bindpublicport);transport=tcp;lr=on;ftag=$ft;nat=yes>\r\n"); $fs = "tcp" + ":" + $sel(cfg_get.kamailio.bindpublicip) + ":" + $sel(cfg_get.kamailio.bindpublicport); } else { xlog("L_INFO", "[$rm], Present Route: $(route_uri{uri.host})\n"); remove_hf("Route"); insert_hf("Record-Route: <"sip" + ":" $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes>\r\n"); $fs = "udp" + ":" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport); } } } } And for initial SUBSCRIBE INVITE if (is_method("INVITE|SUBSCRIBE")) { remove_hf("Route"); insert_hf("Record-Route: <"sip" + ":" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport);lr=on;ftag=$ft;nat=yes>\r\n"); #record_route(); } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 18, 2015 4:15:28 AM Subject: Re: [SR-Users] remove_hf_value Hello, but the INVITE request goes out with 2 Record Route headers, if you get them in the 200ok. Not sure how it is going to work overall, you will have to manually set local sockets for requests within dialog, because you are doing also trasport layer gatewaying (tcp-udp). Anyhow, as a solution, maybe you should just not be using record_route() function and just add the header with insert_hf() for the invite, remove the Record-Route header for 200ok with remove_hf() and insert a new one again with insert_hf() If you know the local address (IP/port/transport) you want to put in the Record-Route, you can build the header as: insert_hf("Record-Route: <sip:_LOCAL_IP_:_LOCAL_PORT_;transport=_LOCAL_TRANSPORT_;lr=on;ftag=$ft;nat=yes> \r\n"); Requests within the dialog can be handled with loose_route(), but again, you have to do more in config for selecting the local socket to send to next hop. Cheers, Daniel On 16/03/15 13:35, Slava Bendersky wrote: Hello Daniel, I want do clean up on routes from first SUBSCRIBE, that when traffic goes to public side of kamailio will be only public route and when traffic is internal the Record-Route will stay internal only. In SDP I see like this. I tired double routes set 0 in rr, but didn't clean enough. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TCP 192.168.88.252:5062;received=client_pub_ipbranch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> On public side remote this Record-Route Record-Route: <sip:kamailio_pub_ip:5084;transport=tcp;r2=on;lr=on;ftag=a185d974ec;nat=yes> On private side remote this Record-Route From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec To: <sips:10101@networklab.ca> ;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060> ;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 16, 2015 7:01:54 AM Subject: Re: [SR-Users] remove_hf_value Hello, can you give an example of incoming header and how you want to be after modification? It is not clear for me what end result you want/expect. Cheers, Daniel On 14/03/15 06:41, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Here updated version. Trying remove Record-Route after the coma. My log shows Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Inbound connection from <Client Public IP> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: <script>: <null> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Route for removal: '<null>' route snippet. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("SUBSCRIBE")) { if(is_ip_rfc1918($si)) { xlog( "$sel((a)hf_value.Record_Route[-1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[-1])'\n"); remove_hf_value("Record-Route[-1]"); } else { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } } return; } Slava, From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:00:44 PM Subject: [SR-Users] remove_hf_value Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

Presence data to users behind NAT
by Olli Attila 22 Mar '15

22 Mar '15

Hello, Im running kamailio 4.2.3 and I'm trying to build a system where I could have the following features: SIP proxy, RTP proxy, Presence & XCAP service for IM messaging. I have a problem with the Presence service with users that are located behind nat. When Kamailio sends presence info to a watcher with sip NOTIFY message, the destaination ip for the pacage seems to be an un-routable (=private) ip, and so the watcher never receives the NOTIFY message making the watchers sip client unaware of the other users presence. Any suggestions how to make Kamailio handle the watchers behind nat so that the NOTIFY messages will get delivered succesfully? I tried the scenario with a client with public ip and the presence works fine. my kamailio.cfg: http://tny.cz/7d85773c -- Olli

2 1

Kamailio with websocket as transport
by Austin Einter 22 Mar '15

22 Mar '15

Dear All I have a sip user agent. Today it works fine with Kamailio using tcp as tranasport. Now I am planning to support websocket in my sip ua. My question is does Kamailio support websocket transport. If so how can I configure it or while building kamailio do I need to make some flags on. I am using latest Kamailio 4.2.3 (version: kamailio 4.2.3 (x86_64/linux)). Thanks Austin

2 2

Hardware config to run 1000 Kamailio TLS users
by elektau＠yandex.ru 20 Mar '15

20 Mar '15

Hi All, 1. What are resource requirements for running Kamailio server with TLS module active for 1000 registered users (concurrent calls) when running on VPS? In other words, what amount of RAM and processor power required? 2. What is the maximum number of Kamailio TLS users/concurrent calls can be made for the below default values? # Amount of shared and private memory to allocate # for the running Kamailio server (in Mb) SHM_MEMORY=64 PKG_MEMORY=8 Thanks.

3 3

Kamailio hardware requirements
by Mickael Marrache 20 Mar '15

20 Mar '15

Hi, We are currently deploying an entire architecture composed of load balancers, proxies and media relays. All the components except the media relays are Kamailio instances. The media relays are RTP proxy instances. We are trying to determine the hardware requirements for the different servers. We will start with an architecture composed of: . 2 load balancers (one kept as backup to ensure availability) . 2 proxies (load is balanced over the 2 using the DISPATCHER module) . 2 media relays We are trying to determine the recommended hardware for the different components taking into account we will need to serve around 400000 users. How many registrations per second can Kamailio support? How many concurrent calls can Kamailio support? What is the recommended number of children processes? What is the recommended size for PKG memory? What is the recommended size for SHM memory? Thanks for your help, Mickael

2 1

auth_diameter module
by Grant Bagdasarian 19 Mar '15

19 Mar '15

Hello, Does Kamailio still support authentication using the auth_diameter module? There is a note on the modules page stating the module is now obsolete. "NOTE: diameter support was developed for DISC (DIameter Server Client project at http://developer.berlios.de/projects/disc/). This project seems to be no longer maintained and DIAMETER specifications were updated in the meantime. Thus, the module is obsolete and needs rework to be usable with opendiameter or other DIAMETER servers." Is the page referring to this module or the DISC? Are there perhaps any other ways to do authentication with Kamailio using diameter?

2 1

RTPProxy issue?
by Igor Potjevlesch 19 Mar '15

19 Mar '15

Hello, I discovered an issue related to the handling of "timestamp" and/or "Marker bit" with rtpproxy (I use the latest Extension 20081224). The call-flow is the following: one UA places a call to A and put this call on hold. Then, the same UA call another number B. Individual streams are ok. When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with a huge timestamp and the Marker bit set to "True". Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. B to C is still working. Anyone have an idea? Regards, Igor.

3 9

Sending PUSH notification asynchronously
by Mickael Marrache 19 Mar '15

19 Mar '15

Hi, I'm trying to add PUSH support to my system using Kamailio. When a call is received and the callee has no active registration, a PUSH is sent to the callee endpoint. In order to send this PUSH, my proxy sends an HTTP requests to another server that will send the PUSH. So, my route block is as follows: route[PUSH] { #Send PUSH notification using a custom module I wrote #Suspend transaction execution. t_on_failure("MANAGE_FAILURE"); if(!t_suspend()) { send_reply("503", "Service Unavailable"); exit; } $sht(push=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); exit; } I would like to execute this route block asynchronously (i.e. in a separate process). I tried using the async_task_route command, and I can see the route block is execute in a separate process, however some stuffs are broken (e.g. transaction timeout, failure route). So, I thought it may be related to the fact asynchronous execution is implemented using t_suspend/t_continue and in this case, t_suspend/t_continue would be called twice. The important point is to make the interaction with the PUSH server asynchronous. Any idea? Thanks, Mickael

2 6

Timeout after t_suspend and failure route
by Mickael Marrache 19 Mar '15

19 Mar '15

Hi, When receiving an INVITE, I arm a failure route and set the fr_timer to 30 seconds. After the fr_timer expires, I need to do some operations however I see the failure route is not executed. I guess it is related to the fact the INVITE has not been forwarded? If yes, how can I execute code when a transaction times out after calling t_suspend? Thanks, Mickael

2 7

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 19 Mar '15

19 Mar '15

This is the SIP log when using location routing for BYE from peer server.

1 0

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 19 Mar '15

19 Mar '15

This is the SIP log with the 500 error for BYE.

1 0

SQLOPS bug
by Mihaly Zachar 18 Mar '15

18 Mar '15

Hi Gents, Please consider applying this patch: --- ./modules/sqlops/sql_api.c.orig 2015-03-18 16:33:56.994320505 +0100 +++ ./modules/sqlops/sql_api.c 2015-03-18 16:34:11.254419341 +0100 @@ -743,7 +743,7 @@ LM_ERR("row index out of bounds [%d/%d]\n", i, res->nrows); goto error; } - if(i>=res->ncols) + if(j>=res->ncols) { LM_ERR("column index out of bounds [%d/%d]\n", j, res->ncols); goto error; Regards, Misi

2 3

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 18 Mar '15

18 Mar '15

Hi, all I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio severs share the the same database with db_mode=3, and no registration replication. Use pjsua2 as SIP client for testing. The test setup is as follows: kamailio server 1(k1): 10.0.1.30:5061 kamailio server 2(k2): 10.0.1.32:5061 sip client c1: sip:16317@10.0.1.30, client ip: 10.0.1.254 sip client c2: sip:72316@10.0.1.30, client ip: 10.0.1.254 c1 is registered with k1. c2 is registered with k2. When c1 calls c2, the call flow is as follows: INVITE: c1------>k1------>k2------>c2 200 OK: c1<------k1<------k2<------c2 So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error. The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself instead of using the existing TLS connection to c1 to forward the BYE. BYE: c1 X k1<------k2<------c2 | |------->Connection refused/500 Here is the kamailio log on k1: Mar 16 10:34:21 kamailio.sip.com /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 10.0.1.30:55999 failed (RST) Connection refused Mar 16 10:34:21 kamailio.sip.com /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 10.0.1.30:55999: connect & send for 0x7fc96a68a1a0 failed: Connection refused (111) This is the routing logic for BYE in kamailio.cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... If I add location based routing for BYE from peer kamailio, then the BYE seems to be forwarded correctly: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails # If BYE coming from Kamailio peer, route BYE by location $var(peerlist)=$sel(cfg_get.trusted.peers); $var(i) = 0; while($var(i)<$(var(peerlist){param.count})) { xlog("L_DBG", "=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n"); if(src_ip==$(var(peerlist){param.valueat,$var(i)})) { lookup("location"); xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n"); break; } $var(i) = $var(i) + 1; } } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces are enclosed. Thanks

1 0

Kamailio and Busy Lamp Fields (BLF)
by Daniel-Constantin Mierla 18 Mar '15

18 Mar '15

Hello, a bit of time ago I played with Kamailio and BLF, therefore I published kind of getting started configuration file for this feature. It is available at: - http://kb.asipto.com/kamailio:presence:k43-blf The plan was to add more to the tutorial, some guidelines for configuring the phones, but the time didn't allow it so far. Keep an eye on it for some new content. Contributions are also welcome, send them to me or take the content and publish it with new additions on another place. Hopefully is going to be useful for some people out there. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

siremis login issue
by Boris Ivanov 18 Mar '15

18 Mar '15

Hi Hope this is right place to post my question. I have installed siremis under FreeBSD 10 It went through well. DB created with /usr/local/sbin/kamdbctl create. Problem happen when I trying to login as admin. [{"target":"ERROR","content":" \n[2015-03-11 17:31:37 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::getConfiguration() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error:\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstract.php:268<\/font> \n function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstrac...\", 268, Array(0) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/sysheader.inc 117\n function:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstrac...\", 268, Array(0) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstract.php 268\n function:<\/b> setQuoteIdentifiers ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj_Abstract.php 177\n function:<\/b> __construct ( Array(1) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/data\/BizDataObj.php 37\n function:<\/b> __construct ( Array(1) ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/ObjectFactory.php 166\n function:<\/b> constructObject ( \"system.do.UserDO\" ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/ObjectFactory.php 56\n function:<\/b> getObject ( \"system.do.UserDO\", 0 ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizSystem.php 252\n function:<\/b> getObject ( \"system.do.UserDO\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/modules\/service\/authService.php 92\n function:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/modules\/service\/authService.php 66\n function:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/modules\/user\/form\/LoginForm.php 92\n function:<\/b> Login ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php 310\n function:<\/b> invoke ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php 110\n function:<\/b> dispatchRequest ( ) @ \/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php 32\n function:<\/b> include_once ( \"\/usr\/local\/www\/apache24\/data\/openbiz\/bin\/BizController.php\" ) @ \/usr\/local\/www\/apache24\/data\/siremis\/bin\/controller.php 6<\/div>\n _____ Please ask system administrator for help...<\/div>\n"}] Anyone knows how to fix? Boris ____________________________________________________________ This message and any attachments contain confidential information and must not be disclosed to a third party without the prior written consent of setcom. If you have received it erroneously, please discard it and contact us by return. ____________________________________________________________

2 5

Redirect server for load balancing and HA
by Markus 18 Mar '15

18 Mar '15

Hi list, I'm new here, also to Kamailio. My goal is to achieve the "standard solution": Kamailio for load balancing and high availability of several Asterisk servers. For starters, I've set it up with Kamailio+dispatcher+permissions acting as a proxy and it was working fine, but I'd like to try to keep it much more simple. For these reasons: 1) I'd like to maintain different peer settings in Asterisk (DTMF, codecs). 2) I don't want/need a database for Kamailio. I think it is mandatory for permissions module + IP auth. So, let's do Asterisk the authentication, it's already there. 3) Accounting. I'm using a software where it's really useful for accounting purposes if every peer/user has its own entry in Asterisk's sip.conf / in realtime table. Long story short, my config now looks like this and it seems to be working fine for me: request_route { route(DISPATCHER); } route[DISPATCHER] { if ( method=="REGISTER") { ds_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } if ( method=="INVITE" ) { ds_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } if ( method=="SUBSCRIBE" ) { ds_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } } My questions: 1) There are other approaches to the DTMF+codecs issue. Like this thread from 2013: http://lists.kamailio.org/pipermail/sr-users/2013-January/076379.html - but I have dozens of changing PSTN peers, so authentication/accounting based on port would be cumbersome. Are there new/alternative ways? 2) I searched the archive and the Web but did not find many questions about a "I-don't-want-Kamailio-to-show-as-the-only-peer-to-Asterisk" solution. Therefore I'm wondering: Is everyone a) running such homogeneous setups that they can survive with the same codec + DTMF settings in Asterisk for all their peers + users? Or b) doing some rewriting magic within Kamailio to deal with it? Or c) I'm overlooking a simple solution. Which one is it? :) 3) And the biggest question: Everywhere I read basically "Redirect server for LB + HA is not recommended, better use proxy". Is this still true, and if so, what are the reasons? Because I should be worried that not all clients/gateways out there respect a SIP 300 Redirect? Other reasons? Thanks so much! :) Markus

4 8

msilo clarification
by mayamatakeshi 18 Mar '15

18 Mar '15

Hello, maybe this phrase in the msilo doc is misleading: *Every time when a user registers with Kamailio, the module is looking in database for offline messages intended for that user. All of them will be sent to contact address provided in REGISTER request.* http://kamailio.org/docs/modules/4.2.x/modules/msilo.html The above seems to imply that msilo would send the call directly to the contact address. But this doesn't seem to be the case based on what is mentioned in several threads in the mailing list and in what I saw in the msilo.c code. If so, is there a reason to be a such? Would it be a problem to allow msilo m_dump to get the values of contact, received and path and use them when calling t_request? Regards, Takeshi.

2 3

Re: [SR-Users] remove_hf_value
by Slava Bendersky 18 Mar '15

18 Mar '15

Hello Daniel, I want do clean up on routes from first SUBSCRIBE, that when traffic goes to public side of kamailio will be only public route and when traffic is internal the Record-Route will stay internal only. In SDP I see like this. I tired double routes set 0 in rr, but didn't clean enough. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TCP 192.168.88.252:5062;received=client_pub_ipbranch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> On public side remote this Record-Route Record-Route: <sip:kamailio_pub_ip:5084;transport=tcp;r2=on;lr=on;ftag=a185d974ec;nat=yes> On private side remote this Record-Route From: "Slava Bendersky" <sips:10101@networklab.ca>;tag=a185d974ec To: <sips:10101@networklab.ca>;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060>;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 16, 2015 7:01:54 AM Subject: Re: [SR-Users] remove_hf_value Hello, can you give an example of incoming header and how you want to be after modification? It is not clear for me what end result you want/expect. Cheers, Daniel On 14/03/15 06:41, Slava Bendersky wrote: Hello Everyone, Here updated version. Trying remove Record-Route after the coma. My log shows Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Inbound connection from <Client Public IP> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: <script>: <null> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Route for removal: '<null>' route snippet. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("SUBSCRIBE")) { if(is_ip_rfc1918($si)) { xlog( "$sel((a)hf_value.Record_Route[-1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[-1])'\n"); remove_hf_value("Record-Route[-1]"); } else { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } } return; } Slava, From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:00:44 PM Subject: [SR-Users] remove_hf_value Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog( "$sel((a)hf_value.Record_Route[1])" ); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Dialogs not deleted on BYE
by Shane Harrison 18 Mar '15

18 Mar '15

Hi all, Have found a few threads on this before which have clarified my understanding a little but not yet presented me with a solution. I am using dialog_ng. dlg_manage() is deprecated and as per the documentation, I am simply setting a dialog flag early in the request_route. Kamctl dialog show does indeed show the dialogs are tracked, but they are not destroyed on the receipt of the BYE. Dialog ref count remains at 2 and in state=4 Log shows the following Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=16 , global msg id=15 , T on entrance=(nil) Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=5943, isACK=0 Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG: trans=0x7fc8d79af798, callback type 1, id 0 entered Mar 16 22:43:42 hh-rcs-sipproxy3 /usr/sbin/kamailio[15484]: DEBUG: dialog_ng [dlg_handlers.c:855]: dlg_new_dialog(): starting dlg_new_dialog and method is [BYE] Looks to me as if fails to find a dialog match and actually creates a new dialog..... Dialog_ng only has one matching mode - DID, others have been deprecated. I have captured the outgoing INVITE and there is no dialog_id parameter added to the record_route. Is that causing my matching problem? Record-Route: <sip:64.208.160.211;transport=tcp;lr=on;ftag=5419eecb;nat=yes> Record-Route: <sip:203.171.39.49:5061;transport=tls;lr=on> Cheers Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

2 10

PRACK not sent
by Dharm S 17 Mar '15

17 Mar '15

Hi All, I am using kamailio 3.1.5. I am trying to send the PRACK message but it sends back 404 Not Here. I am new to using this. I am adding details in the configuration file below: I guess route[WITHINDLG] ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } }

2 1

xlog from 1.5 to 4.2
by Juan Perez 17 Mar '15

17 Mar '15

I installed 4.2 and I am using the same config file form my current version 1.5.2, but none of my custom logs (xlog) are being printed in the log file, even after rising the debug level, the only messages I see are DEBUG these are my settings: loadmodule "xlog.so" # ----- xlog parms ------ modparam("xlog", "buf_size", 4096) Sample uses:xlog("L_INFO","mylog: Main logic started.\n"); xlog("L_INFO","mylog: Switchover Activated!\n"); xlog("L_INFO","mylog: Redundancy test, last 11 digits [$var(number)].\n"); xlog("L_INFO","mylog: Too Many Hops.\n"); am I missing something ?

2 1

PKG and Shared Memory statistics
by Dirk Teurlings Signet B.V. 17 Mar '15

17 Mar '15

Hi, We'd like to be able to get statistics of the used amount of memory in Kamailio. # kamctl stats shmem The above command only tells us what the shared memory is doing. Is there any way to get the committed size of memory for each of the PKG_MEMORY pools? If not, could this be added at some point? Kind regards, -- Dirk Teurlings

3 3

Deleted Domain
by Eric Fetty 17 Mar '15

17 Mar '15

Hi, I accidently deleted my domain from iptel.org. I went to re-create my account and it will not let me. Domain: kingfetty.com

2 1

478/sl and 478/sm on BYE?
by Don Fanning 17 Mar '15

17 Mar '15

I am getting these both when one end of the client sends a BYE - after the other client has already sent a BYE and got a 200 back. Config: http://pastebin.com/5gcFpJ3m /usr/local/sbin/kamailio[3259]: INFO: <script>: Request in RELAY - call is {<null>} for {1070} /usr/local/sbin/kamailio[3259]: INFO: <script>: sip:1070@df7jal23ls0d.invalid /usr/local/sbin/kamailio[3259]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "df7jal23ls0d.invalid" /usr/local/sbin/kamailio[3259]: ERROR: <core> [forward.c:531]: forward_request(): bad host name df7jal23ls0d.invalid, dropping packet /usr/local/sbin/kamailio[3259]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL)

2 3

Re: [SR-Users] [RTPproxy] Re: [OpenSIPS-Users] Announcing rtpproxy v2.0.0
by Maxim Sobolev 17 Mar '15

17 Mar '15

Do you have any particular RFC in mind? On Mar 12, 2015 10:28 AM, "John Mathew" <john.mathew(a)divoxmedia.com> wrote: > Hi, > > Maxim, > Is there any plans for rtp header compression in future. I can't see > anything in the change log for 2.0.0 > > On Tuesday, 10 March 2015, Maxim Sobolev <sobomax(a)sippysoft.com> wrote: > >> Hi All, >> >> I'm happy to announce that we have released rtpproxy v2.0.0. >> >> You can review the release notes here: >> https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 >> >> -sobomax >> >> > > -- > Sent from iPhone 6 > > -- > You received this message because you are subscribed to the Google Groups > "rtpproxy" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to rtpproxy+unsubscribe(a)googlegroups.com. > To post to this group, send email to rtpproxy(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B… > <https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B…> > . > For more options, visit https://groups.google.com/d/optout. >

2 1

Kamailio IMS: can not register
by dfretes＠ing.una.py 16 Mar '15

16 Mar '15

Hi all. I'm trying to setup Kamailio for IMS server, but I've reached a point where I can not move. Trying to register from the Boghe client, I get these errors from the pcscf: tail -f /var/log/syslog|grep './pcscf' Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=402 a=25 n=xlog Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: <script>: REGISTER (sip:alice@ open-ims.test (192.168.1.105:62044) to sip:alice@open-ims.test, 20236156-7eed-01 e4-daac-5072e67f99e9) Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=414 a=5 n=route Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=594 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=576 a=24 n=has_totag Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=584 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=590 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=584 a=24 n=pike_check_req Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=606 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=611 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=606 a=25 n=mf_process_maxfwd_heade r Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=617 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=611 a=26 n=sanity_check Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=623 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=617 a=24 n=has_totag Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=629 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=623 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=633 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=629 a=24 n=t_lookup_request Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=636 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=633 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE QINIT] c=[/etc/kamailio/pcscf/kamailio.cfg] l=634 a=26 n=send_reply Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=417 a=5 n=route Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=673 a=55 n=force_rport Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=694 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=675 a=25 n=nat_uac_test Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=679 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=676 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=679 a=39 n=setflag Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=687 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=680 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=681 a=24 n=set_contact_alias Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=683 a=26 n=xlog Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=684 a=24 n=nat_keepalive Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[NA T] c=[/etc/kamailio/pcscf/kamailio.cfg] l=694 a=2 n=return Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=424 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=420 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=458 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=424 a=24 n=has_totag Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=465 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=458 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=465 a=24 n=t_check_trans Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=472 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=467 a=25 n=is_method Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[DE FAULT_ROUTE] c=[/etc/kamailio/pcscf/kamailio.cfg] l=468 a=5 n=route Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=812 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=810 a=25 n=is_present_hf Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=811 a=25 n=pcscf_save_pending Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=812 a=25 n=t_on_reply Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=813 a=25 n=t_on_failure Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=861 a=16 n=if Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=856 a=25 n=is_present_hf Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=860 a=25 n=append_hf Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=872 a=24 n=t_relay Jan 29 22:36:29 kamailioast ./pcscf[3841]: ERROR: *** cfgtrace:request_route=[RE GISTER] c=[/etc/kamailio/pcscf/kamailio.cfg] l=873 a=2 n=exit Jan 29 22:36:39 kamailioast ./pcscf[3845]: ERROR: *** cfgtrace:failure_route=[RE GISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=910 a=16 n=if Jan 29 22:36:39 kamailioast ./pcscf[3845]: ERROR: *** cfgtrace:failure_route=[RE GISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=908 a=25 n=t_check_status Jan 29 22:36:39 kamailioast ./pcscf[3845]: ERROR: *** cfgtrace:failure_route=[RE GISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=909 a=26 n=send_reply All the time, this messages from the scscf: tail -f /var/log/syslog|grep './scscf' Jan 29 23:50:55 kamailioast ./scscf[3874]: INFO: cdp [peermanager.c:278]: peer_timer(): peer_timer(): Peer hss.open-ims.test #011State 6 Jan 29 23:50:55 kamailioast ./scscf[3874]: WARNING: cdp [peermanager.c:313]: peer_timer(): Inactivity on peer [hss.open-ims.test], sending DWR... - if we don't get a reply, the peer will be closed and nothing from the icscf. Thank you very much for any advice, I've run out of ideas.

2 4

seturi analog for Kamailio 4.3
by Yuriy Gorlichenko 16 Mar '15

16 Mar '15

Hello. I try to call multi[ple endpoints from my server using append_branch. It works fine but when I have only one endpoint - kamailio generate 2 INVITE requests to it. As I understand it is original request and the next one is branch. I used seturi() before for sending original reqest to destination, but I can not see this function at kamailio 4.3. Kamailio 4.3 uses send() dunnction? but it works at stateless mode. I need analog of it or seturi that works at statefull mode.

3 8

Interoperability issues with older SIP clients
by Magnus Nordström 16 Mar '15

16 Mar '15

Hi, We got interoperability issues with older SIP-clients. When e.g. some parameters are included in the SIP header Record-Route the call does not get connected. Now we are looking for a solution for that. We are using kamailio as SIP-registrar/proxy and would like to know if we can keep the call state and rewrite headers that causes interoperability issues. We are aware that kamailio is not a B2BUA. The basic idea would be e.g.: 1. kamailio proxy identifies a call from a modern SIP-client to an old SIP-client 2. kamailio send the call to a interoperability module (?) 3. the interoperabilty module removes sip-headers known to cause issues and saves them. 4. the interoperabilty module makes sure it is in the SIP signaling path 5. the interoperabilty module send the call forward to the old SIP-client 6. the interoperabilty module stays in path rewriting the SIP-headers. The modern SIP-client receives the SIP-signaling as if it was not rewritten. Do you think this is possible with kamailio? Thankful for any hints or ideas. Magnus Nordström

2 1

does ipops module dns_int_match_ip(hostname, ipaddr) search all returned records?
by canuck15 16 Mar '15

16 Mar '15

I am reading the documentation for the ipops module dns_int_match_ip() function and it states the following: "Returns TRUE if ipaddr is associated by DNS to hostname. FALSE otherwise. It uses internal DNS resolver. At this moment, the function might not check all the IP addresses as returned by dns_sys_match_ip(), because the internal resolver targets to discover the first address to be used for relaying SIP traffic. Thus is better to use dns_sys_match_ip() if the host you want to check has many IP addresses, in different address famililies (IPv4/6)." I am not sure what that statement means. I will be using this for IP authentication where the records it is checking against are in a mysql database and may or may not be be using the DNS name instead of the IP. Many of those DNS names could return several IP addresses and the incoming call could be on any one of those IP addresses. So when I do a check of a hostname against an IP address I want all returned A or srv IP's to be checked against the IP. Not just the first IP returned when checked. I do not believe dns_sys_match() will be fast enough on a busy server so I would prefer to using dns_int_match_ip(). If anyone can clarify what that statement means regarding "internal resolver targets to discover the first address" and whether that would affect my use it would be greatly appreciated. On a related note, is there any way to check the contents of the dns resolver cache? I looked through kamctl and kamcmd but didn't find anything. I tried "kamctl address show" but that didn't return anything.

2 1

Re: [SR-Users] remove_hf_value
by Slava Bendersky 16 Mar '15

16 Mar '15

Hello Everyone, Here updated version. Trying remove Record-Route after the coma. My log shows Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Inbound connection from <Client Public IP> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: ERROR: <script>: <null> Mar 14 01:25:21 canlvprx01 /usr/sbin/kamailio[751]: INFO: <script>: [SUBSCRIBE], Route for removal: '<null>' route snippet. route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Route")) { xlog("L_INFO", "[$rm], Inbound connection from <$si>\n"); if(is_method("SUBSCRIBE")) { if(is_ip_rfc1918($si)) { xlog("$sel((a)hf_value.Record_Route[-1])"); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[-1])'\n"); remove_hf_value("Record-Route[-1]"); } else { xlog("$sel((a)hf_value.Record_Route[1])"); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } } return; } Slava, From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:00:44 PM Subject: [SR-Users] remove_hf_value Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog("$sel((a)hf_value.Record_Route[1])"); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

What does it mean when auth_check and auth_challenge flags are "0"?
by canuck15 16 Mar '15

16 Mar '15

I am trying to decipher a kamailio.cfg file for a particular open source project. There are problems with authentication and I am trying to understand why. They use the following lines for authentication check if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "0"); exit; The documentation only talks about using "1" and not "0". So I am trying to understand what the above statement with "0" is actually doing. Any help in this regard would be greatly appreciated.

2 1

Kamailio Freeswitch TCP call origination
by Niraj Roy 16 Mar '15

16 Mar '15

Hi, Here is the setup I have configured. --Two Kamailio K1, K2 Works as a registrar and call load balancer -- Three freeswitch works as the media server. -- They follow common MySQL database. Regular call flow User A---> Kamailio (K1)---> Freeswitch (F1)----Kamailio(K1)--->User B. Everything works fine as long as all users are registered and making call with single Kamailio. Now the problem starts when we distribute users on more than one kamailio. Let us take the case. UserA -- Registered over TCP with Kamailio (K1) UserB -- Registered over TCP with Kamailio (K2) Now User A initiates a call.. It will follow the same path User A---> Kamailio (K1)---> Freeswitch (F1)----Kamailio(K1)--->User B. Since User B is registered with K2. K1 being a public server won't be able to initiate any call to User B because there is no TCP connection established with it. Can anybody shade some light on this? Thanks,

3 2

how can I use WITH_IPAUTH when the IP may be stored in the DB as a FQDN?
by canuck15 14 Mar '15

14 Mar '15

Here is is the relevant section of kamailio.cfg $var(tempfU) = $fU; #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address() && $au == "") { # Loading $fU from database using IP sql_pvquery("elxpbx", "SELECT name FROM sip WHERE host = '$si' AND sippasswd IS NULL", "$var(tempfU)"); # source IP allowed return; } The problem is that when host= somefqdn.com the above will fail since $si will always be an IP address as far as I can tell. More often than not host= is a fqdn and requiring it to always be an IP address is not an option. Converting it to IP before storing it in the DB is also not an option because it needs to be able to work of the IP address changes. So how can the above be done to accomodate the possibility that host= somefqdn.com or an IP address. Preferably in such a way that it can scale to hundreds/thousands of rows in the database without slowing things down or crashing.

3 4

Help with 407 Proxy Auth. Required
by Agiftel 14 Mar '15

14 Mar '15

Hi all, i cannot understand where is the problem with this transaction: Kamailio ask for Proxy authorization and in the second INVITE credentials are present. Can you help me understand? Regards U 2015/03/09 15:03:42.191831 10.160.21.51:5060 -> 10.160.20.18:5060 INVITE sip:1000@sip.longwave.labo;user=phone SIP/2.0. Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, NOTIFY, UPDATE. Supported: 100rel,from-change,timer,histinfo. User-Agent: OXO_GW_820/092.001. Session-Expires: 43200. P-Asserted-Identity: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>. History-Info: <sip:1000@sip.longwave.labo;user=phone>;index=1. To: <sip:1000@sip.longwave.labo;user=phone>. From: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>;tag=11056089a66d6a43ec04cd21b78039c5. Contact: "PIPPO" <sip:2000@10.160.21.51;transport=UDP;user=phone>. Content-Type: application/sdp. Call-ID: 346eded1901cb5b60b9d97a14f196e29(a)10.160.21.51. CSeq: 178945190 INVITE. Via: SIP/2.0/UDP 10.160.21.51;rport;branch=z9hG4bK1fd5f032d2296189ba85c9bf8553ff28. Max-Forwards: 70. Content-Length: 208. . v=0. o=default 1425909984 1425909984 IN IP4 10.160.21.51. s=-. c=IN IP4 10.160.21.51. t=0 0. m=audio 32000 RTP/AVP 18 106 4 8 0. a=sendrecv. a=rtpmap:106 telephone-event/8000. a=fmtp:106 0-15. a=maxptime:90. U 2015/03/09 15:03:42.193299 10.160.20.18:5060 -> 10.160.21.51:5060 SIP/2.0 407 Proxy Authentication Required. To: <sip:1000@sip.longwave.labo;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.aa4d. From: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>;tag=11056089a66d6a43ec04cd21b78039c5. Call-ID: 346eded1901cb5b60b9d97a14f196e29(a)10.160.21.51. CSeq: 178945190 INVITE. Via: SIP/2.0/UDP 10.160.21.51;rport=5060;branch=z9hG4bK1fd5f032d2296189ba85c9bf8553ff28;received=10.160.21.51. Proxy-Authenticate: Digest realm="oxo.longwave.labo", nonce="VP2palT9qD7LOD/nl465ujUSUW0NiDo/". Server: lw-sipproxysrv. Content-Length: 0. . U 2015/03/09 15:03:42.195800 10.160.21.51:5060 -> 10.160.20.18:5060 ACK sip:1000@sip.longwave.labo;user=phone SIP/2.0. Call-ID: 346eded1901cb5b60b9d97a14f196e29(a)10.160.21.51. From: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>;tag=11056089a66d6a43ec04cd21b78039c5. To: <sip:1000@sip.longwave.labo;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.aa4d. Via: SIP/2.0/UDP 10.160.21.51;rport;branch=z9hG4bK1fd5f032d2296189ba85c9bf8553ff28. CSeq: 178945190 ACK. Content-Length: 0. . U 2015/03/09 15:03:42.199176 10.160.21.51:5060 -> 10.160.20.18:5060 INVITE sip:1000@sip.longwave.labo;user=phone SIP/2.0. Proxy-Authorization: Digest username="oxo",realm="oxo.longwave.labo",nonce="VP2palT9qD7LOD/nl465ujUSUW0NiDo/",uri="sip:1000@sip.longwave.labo;user=phone",response="263ed33da80fb520af0fb0b2f246d310". Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, NOTIFY, UPDATE. Supported: 100rel,from-change,timer,histinfo. User-Agent: OXO_GW_820/092.001. Session-Expires: 43200. P-Asserted-Identity: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>. History-Info: <sip:1000@sip.longwave.labo;user=phone>;index=1. To: <sip:1000@sip.longwave.labo;user=phone>. From: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>;tag=11056089a66d6a43ec04cd21b78039c5. Contact: "PIPPO" <sip:2000@10.160.21.51;transport=UDP;user=phone>. Content-Type: application/sdp. Call-ID: 346eded1901cb5b60b9d97a14f196e29(a)10.160.21.51. CSeq: 178945191 INVITE. Max-Forwards: 70. Via: SIP/2.0/UDP 10.160.21.51;rport;branch=z9hG4bK503a3569cf5b057bd47d1154c5ba0842. Content-Length: 208. . v=0. o=default 1425909984 1425909984 IN IP4 10.160.21.51. s=-. c=IN IP4 10.160.21.51. t=0 0. m=audio 32000 RTP/AVP 18 106 4 8 0. a=sendrecv. a=rtpmap:106 telephone-event/8000. a=fmtp:106 0-15. a=maxptime:90. U 2015/03/09 15:03:42.200387 10.160.20.18:5060 -> 10.160.21.51:5060 SIP/2.0 407 Proxy Authentication Required. To: <sip:1000@sip.longwave.labo;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.d413. From: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>;tag=11056089a66d6a43ec04cd21b78039c5. Call-ID: 346eded1901cb5b60b9d97a14f196e29(a)10.160.21.51. CSeq: 178945191 INVITE. Via: SIP/2.0/UDP 10.160.21.51;rport=5060;branch=z9hG4bK503a3569cf5b057bd47d1154c5ba0842;received=10.160.21.51. Proxy-Authenticate: Digest realm="oxo.longwave.labo", nonce="VP2palT9qD7LOD/nl465ujUSUW0NiDo/". Server: lw-sipproxysrv. Content-Length: 0. . U 2015/03/09 15:03:42.203016 10.160.21.51:5060 -> 10.160.20.18:5060 ACK sip:1000@sip.longwave.labo;user=phone SIP/2.0. Call-ID: 346eded1901cb5b60b9d97a14f196e29(a)10.160.21.51. From: "PIPPO" <sip:2000@oxo.longwave.labo;user=phone>;tag=11056089a66d6a43ec04cd21b78039c5. To: <sip:1000@sip.longwave.labo;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.d413. Via: SIP/2.0/UDP 10.160.21.51;rport;branch=z9hG4bK503a3569cf5b057bd47d1154c5ba0842. CSeq: 178945191 ACK. Content-Length: 0. . -- View this message in context: http://sip-router.1086192.n5.nabble.com/Help-with-407-Proxy-Auth-Required-t… Sent from the Users mailing list archive at Nabble.com.

4 4

Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 13 Mar '15

13 Mar '15

Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

remove_hf_value
by Slava Bendersky 13 Mar '15

13 Mar '15

Hello Everyone, I am trying adjust Record-Route, but can't match headers value And I am call route(ROUTE_HDR_ORG); right after route(WITHINDLG); route[ROUTE_HDR_ORG] { # Getting record routes if(is_present_hf("Record-Route")) { if(is_method("SUBSCRIBE")) { xlog("$sel((a)hf_value.Record_Route[1])"); xlog("L_INFO", "[$rm], Route for removal: '$sel((a)hf_value.Record_Route[1])'\n"); remove_hf_value("Record-Route[1]"); } } return; } Any help Thank you, Slava.

1 0

using xHTTP module, bind port?
by SamyGo 13 Mar '15

13 Mar '15

Hi All, I am just trying xhttp_rpc module http://www.kamailio.org/docs/modules/4.2.x/modules/xhttp_rpc.html Where can I set/change the port for this module ? currently its only 5060 !? Thanks, Sammy.

2 3

Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 13 Mar '15

13 Mar '15

Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 2

Re: [SR-Users] Regd. Kamailio GPL
by Alex Balashov 13 Mar '15

13 Mar '15

I am also not at all an expert in GPL licencing or IP, but would like to add that any customisation you make specifically for the purpose of interacting with a billing server or other external application is likely to take the form of a Kamailio module, rather than modifications to the existing Kamailio modules, the Kamailio core, or any other part of the stock code tree. As far as I know, under the GPL--broadly speaking, for I certainly don't know or remember the specifics--modules get different treatment than modifications per se. As far sa I know, if you can stick your code into [a] completely self-contained module(s) (and given what you've said about what specifically you want to accomplish, it seems to me that you absolutely can) that interact(s) with Kamailio using its module APIs and requires no modifications to Kamailio itself, it's not subject to the same licence constraints as modifications per se. Please do not take my advice as stated, and consult a competent software/IP attorney. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 303 Perimeter Center North Suite 300 Atlanta, GA 30346 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/

3 5

How to install kamailio-4.2.3 in debian Wheezy
by Yanko Marín Muro 13 Mar '15

13 Mar '15

Hello: How to install kamailio-4.2.3 in debian Wheezy? I need the procedure!!!! we can install P-CSCF, I-CSCF, S-CSCF and HSS on separate servers? Best regards, YANKO MARIN MURO --- This message was processed by Kaspersky Mail Gateway 5.6.28/RELEASE running at host imx3.etecsa.cu Visit our web-site: <http://www.kaspersky.com>, <http://www.viruslist.com>

3 2

Loadmodule mi_xmlrpc.so gives error
by SamyGo 13 Mar '15

13 Mar '15

Hi All, Im trying to load up module 'mi_xmlrpc.so' into my v4.2.1 Kamailio. It compiles nicely but when I try to load the module in config and restart kamailio it gives following error(s) 0(30917) ERROR: <core> [sr_module.c:597]: load_module(): could not open module </usr/local/lib64/kamailio/modules/mi_xmlrpc.so>: /usr/local/lib64/kamailio/modules/mi_xmlrpc.so: undefined symbol: xmlrpc_strsol 0(30917) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 329, column 12-25: failed to load module 0(30917) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module matching <mi_xmlrpc> found Now, I've done some searching around and it seems like pre-req library version mismatch thing.. Do you guys have any solution to this. My end objective is to be able to send MI commands to my Kamailio server(s) via web-Interface. Any recommendations, examples are welcome. Best Regards, Sammy.

2 1

Kamailio and rtpengine Load Testing
by Marc Soda 12 Mar '15

12 Mar '15

I'm wondering if anyone has any experience load testing with Kamailio and rtpengine using WebRTC. I've had success load testing just Kamailio with sipp, but now I'd like to add the media piece and, if possible, do it via WebRTC. Anyone have any thoughts? Thanks, Marc

3 2

Continent Code GeoIP module, kamailio 4.2
by Rene Montilva 12 Mar '15

12 Mar '15

Hi list i'm trying to configure rtp engine routing by source ip, i'd like divide it by continent code, is there any pseudo-variables or possibilites to handle continent code. Team, any idea about it?

3 11

Re: [SR-Users] Regd. Kamailio GPL
by Daniel Tryba 12 Mar '15

12 Mar '15

On Tuesday 10 March 2015 14:52:22 Shankar wrote: > We are exploring kamailio source for use in our VOIP solution. This is the > first time we are looking at open source. We have few doubts in using > kamailio for providing our VOIP service. Under GPL any customization > (customization mainly with respect to interacting with our proprietary > applications e.g. Billing server) we do to kamailio also has to be provided > to the end user. Our doubt is whether we can procure commercial rights to > our customised code? You should really contact a lawyer specialised in IP to get good advice. But gnu.org points out that above assumption isn't always true: http://www.gnu.org/licenses/old-licenses/gpl-2.0-faq.html#GPLRequireSourceP… "The GPL does not require you to release your modified version. You are free to make modifications and use them privately, without ever releasing them. This applies to organizations (including companies), too; an organization can make a modified version and use it internally without ever releasing it outside the organization. But if you release the modified version to the public in some way, the GPL requires you to make the modified source code available to the program's users, under the GPL." So if you don't ship a binary to endusers, you don't have to release source (including modifications). It all depends on what you are supplying to your endusers (a service or a (complete) software product).

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Re: [SR-Users] Regd. Kamailio GPL
by Shankar 12 Mar '15

12 Mar '15

Hi, Thank you very much for the clarification. Our intention is to provide VOIP service and hence we need not expose our customized code to end users. Regards, Shankar --------------------------------------------------------------------- Date: Wed, 11 Mar 2015 10:56:02 +0100 From: "Olle E. Johansson" <oej(a)edvina.net> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Regd. Kamailio GPL Message-ID: <DD23FDE1-C5DA-4D98-B20E-DE7173C4902F(a)edvina.net> Content-Type: text/plain; charset=us-ascii On 11 Mar 2015, at 10:13, Daniel Tryba <d.tryba(a)pocos.nl> wrote: > On Tuesday 10 March 2015 14:52:22 Shankar wrote: >> We are exploring kamailio source for use in our VOIP solution. This is the >> first time we are looking at open source. We have few doubts in using >> kamailio for providing our VOIP service. Under GPL any customization >> (customization mainly with respect to interacting with our proprietary >> applications e.g. Billing server) we do to kamailio also has to be provided >> to the end user. Our doubt is whether we can procure commercial rights to >> our customised code? > > You should really contact a lawyer specialised in IP to get good advice. But > gnu.org points out that above assumption isn't always true: > http://www.gnu.org/licenses/old-licenses/gpl-2.0-faq.html#GPLRequireSourcePo stedPublic > > "The GPL does not require you to release your modified version. You are free > to make modifications and use them privately, without ever releasing them. > This applies to organizations (including companies), too; an organization can > make a modified version and use it internally without ever releasing it > outside the organization. > > But if you release the modified version to the public in some way, the GPL > requires you to make the modified source code available to the program's > users, under the GPL." > > So if you don't ship a binary to endusers, you don't have to release source > (including modifications). It all depends on what you are supplying to your > endusers (a service or a (complete) software product). > There is a big difference between GPLv2 and GPLv3 here. For GPL2, providing services is not seen as distribution of the source, and the commercial code can stay in the company. For GPLv3 it's different as far as I understand. Kamailio is GPLv2 and Siremis - the web interface by Asipto - is GPLv3. /O

1 0

Parser error with sql_pvquery?
by Anthony Messina 12 Mar '15

12 Mar '15

I'm trying to do an sql_pvquery() and have PostgreSQL return the default value using COALESCE (if the other parts of the query return NULL) rather than placing the $null check on the return in the Kamailio script. I seem to be coming up on a parser issue, perhaps too many nested parentheses: sql_pvquery("kamailio", "SELECT COALESCE((SELECT value FROM usr_preferences WHERE attribute='callee_rtp_transport' AND (uuid='$sht(rtpengine=>$ci::$T_branch_idx::callee_ip)' OR username='$rU') LIMIT 1), 'RTP/AVP')", "$sht(rtpengine=>$ci::$T_branch_idx::callee_rtp_transport)"); <core> [cfg.y:3428]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1444, column 14: syntax error Is this query "too complex" for the parser, or is there something I've missed above? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 11 Mar '15

11 Mar '15

Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 11 Mar '15

11 Mar '15

Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

SIREMIS error
by Agiftel 11 Mar '15

11 Mar '15

Hi, i tried to install SIREMIS v 4.2 and i get error, already posted on internet/forum, regarding versione of PHP 5.6 What i didn't find is a solution. So my question is, I still unsupported by SIREMIS php v 5.4 and upper? Is there a solution?' Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/SIREMIS-error-tp135884.html Sent from the Users mailing list archive at Nabble.com.

2 2

OT: results investigating why kamailio.org was not responding recently
by Daniel-Constantin Mierla 11 Mar '15

11 Mar '15

Hello, as some noticed and informed privately or on mailing list, several days ago kamailio.org was not accessible. It was caused by apache getting stuck on connecting to a remote host due to the pingback feature of wordpress. All the other services were fine. Listing running processes showed a large number of apache instances connecting to an address like 23.x.y.204 and being stuck in that state. Fetching the page from that address with wget resulted in some html with embedded obfuscated javascript. I couldn't find a reason why apache was not closing the connection, although there are some timeouts in configs. Also, as I expect apache/php/wordpress don't run javascript, the issue should be somewhere at the http layer. At some point apache reached the max number of threads and didn't serve other requests. The solution was to disable the pingbacks using a plugin for wordpress named: Disable XML-RPC Pingback . All seems ok for the last days. I posted here just in case it will be useful for someone else, but also to get hints on other tweaks that we should to do wordpress to prevent similar situation, if anyone can share from his/her experience. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Calling ims charging function in route_failure
by Shane Harrison 11 Mar '15

11 Mar '15

Hi all, What I want to do ~~~~~~~~~~~~~~ If a call to a SIP AoR fails then try the PSTN number associated with the AoR by routing the request to a PSTN gateway. Calls to a PSTN gateway need to be charged so need to call the Ro_CCR function in ims_charging module. How I tried to do it ~~~~~~~~~~~~~~ In the failure_route[MANAGE_FAILURE] block I call the Ro_CCR() Result was ~~~~~~~~ Kamailio fails to parse config file. yyparse(): misused command Ro_CCR yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 1162, column 91: Command cannot be used in the block So two questions: 1) What constraints are there for calling functions in the failure_route blocks i.e. how does parser decide what can be called and what can't 2) Any thoughts of how I can do effectively a late charging decision Cheers and thanks in advance Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

1 1

ims_charging module
by Shane Harrison 10 Mar '15

10 Mar '15

Hi all ims'ers I have modified the ims_charging module to cope with the CCA return code of "CREDIT CONTROL NOT APPLICABLE". This means the call is not a chargeable call and the charging module now returns a unique return code for this. Currently there are two scenarios handled: 1) CCA grants the credit required for the call --> route the INVITE and then automatically new CCR requests are made for further credit as the call progresses 2) CCA doesn't grant the credit ---> send an error response back to INVITE originator - I assume this will clear all dialogs and no further credit requests are made. The question I have, is how do I cope with this unique return code so that I can continue to route the call but not make any more CCR requests to the credit control server? As I am not sure how the ims_charging module continues to generate CCR requests, it is difficult for me to understand how to essentially clear the whole CCR session without clearing the call. Kind regards Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

1 0

kamailio asterisk NOTIFY
by Slava Bendersky 10 Mar '15

10 Mar '15

Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava.

2 1

Sipcapture module insert error
by Spencer Thomason 10 Mar '15

10 Mar '15

Hello, I’m not sure if this is the correct forum for this question but here goes… I’m facing an issue setting up a sipcapture server. I’m seeing several db errors in the logs: Mar 6 10:45:13 sipcapture /usr/local/sbin/kamailio[37872]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Data too long for column 'correlation_id' at row 1 I did a packet capture of the sql query and it appears that the value for the correlation_id is omitted. Interestingly its not all packets but maybe 60%. Do I have something misconfigured? Thanks, Spencer A sanitized version of the query is below: insert into sip_capture (date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,correlation_id,msg ) values ('2015-03-06 10:50:22',1425667824109829,'180','Ringing','','','+12052222222','gK007b315d','+12051111111','3Sr0m4vyjy2UK','','+12051111111','','2080378443_116641441(a)1.1.1.1','','SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00B5617ed64f1ebf7d7','z9hG4bK00B5617ed64f1ebf7d7','1888238783 INVITE','','','','','3.3.3.3',5060,'1.1.1.1',5060,'2.2.2.2',6038,'',0,1,2,'',2,'node1:2001','SIP/2.0 180 Ringing\r\nVia: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00B5617ed64f1ebf7d7\r\nRecord-Route: <sip:3.3.3.3;lr;ftag=gK007b315d>\r\nFrom: <sip:+12052222222@1.1.1.1>;tag=gK007b315d\r\nTo: <sip:+12051111111@3.3.3.3>;tag=3Sr0m4vyjy2UK\r\nCall-ID: 2080378443_116641441(a)1.1.1.1\r\nCSeq: 1888238783 INVITE\r\nContact: <sip:+12051111111@2.2.2.2:6038;transport=udp>\r\nAccept: application/sdp\r\nAllow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE\r\nSupported: path\r\nContent-Length: 0\r\n\r\n','SIP/2.0 180 Ringing\r\nVia: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00B5617ed64f1ebf7d7\r\nRecord-Route: <sip:3.3.3.3;lr;ftag=gK007b315d>\r\nFrom: <sip:+12052222222@1.1.1.1>;tag=gK007b315d\r\nTo: <sip:+12051111111@3.3.3.3>;tag=3Sr0m4vyjy2UK\r\nCall-ID: 2080378443_116641441(a)1.1.1.1\r\nCSeq: 1888238783 INVITE\r\nContact: <sip:+12051111111@2.2.2.2:6038;transport=udp>\r\nAccept: application/sdp\r\nAllow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE\r\nSupported: path\r\nContent-Length: 0\r\n\r\n')

2 2

building kamailio with MongoDB
by Tom Braarup Cuykens 10 Mar '15

10 Mar '15

I am trying to setup a Kamailio with MongoDB but the complilation fails: make include_modules="db_mongodb rtpproxy" cfg make all [...] In file included from mongodb_connection.c:24:0: mongodb_connection.h:26:20: fatal error: mongoc.h: No such file or directory compilation terminated. make[1]: *** [mongodb_connection.o] Error 1 make: *** [modules] Error 1 I have build the mongodb C driver from https://github.com/mongodb/mongo-c-driver/ Not sure if the issue is related to the Kamailio MongoDB module or the Mongo C driver. -- Med venlig hilsen Tom Braarup Cuykens plusTEL.dk

2 1

kamailio 4.2.3 Segmentation fault when using db_cluster module
by Jan Hazenberg 10 Mar '15

10 Mar '15

Hello, I have a issue on kamailio 4.2.3 when using the db_cluster module. I have the following config: #!define DBURL "cluster://cls1" # ----- db_cluster params ----- modparam("db_cluster", "connection", "con1=>mysql://kamailio:kamailiorw@10.121.0.120/kamailio") modparam("db_cluster", "connection", "con2=>mysql://kamailio:kamailiorw@10.121.0.121/kamailio") modparam("db_cluster", "connection", "con3=>mysql://kamailio:kamailiorw@10.121.0.122/kamailio") modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r") modparam("db_cluster", "inactive_interval", 180) modparam("db_cluster", "max_query_length", 5) # ----- sqlops params ----- modparam("sqlops","sqlcon","ca=>cluster://cls1") When i start kamailio it fails with: Mar 9 17:11:49 localhost kamailio[31063]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Mar 9 17:11:49 localhost kamailio[31063]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Mar 9 17:11:49 localhost kamailio[31063]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Mar 9 17:11:49 localhost kamailio[31063]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio/permissions.allow Mar 9 17:11:49 localhost kamailio[31063]: INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio/permissions.allow) not found => empty rule set Mar 9 17:11:49 localhost kamailio[31063]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio/permissions.deny Mar 9 17:11:49 localhost kamailio[31063]: INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio/permissions.deny) not found => empty rule set Mar 9 17:11:49 localhost kernel: kamailio[31063]: segfault at 80 ip 00007f7f4f9ffddc sp 00007fffde2f88a0 error 4 in db_cluster.so[7f7f4f9ec000+28000] Mar 9 17:11:50 localhost kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe Could this be a bug or is this a configuration issue? I tested the dbnodes and they seem to respond fine to any query's i send. Thanks, Jan Hazenberg

3 8

SIP 500 on WSS connection lookup failure
by Armen Babikyan 10 Mar '15

10 Mar '15

Hello, I am setting up this scenario using Kamailio 4.2.2: [UA] <-> [kamailio1] <-> [kamailio2] <-> [other stuff] UA connects to kamailio1 using SIP/WSS, kamailio1 proxies request along to kamailio2. Note that my intent here is to have kamailio2 provide the registrar functionality. The kamailio2 (or something behind it) can also initiate transactions to UA, like OPTIONS, for example. I'm having an issue though: when the websocket connection between kamailio1 and UA dies, transaction requests from kamailio2 show up to kamailio1, and I get errors in kamailio1's logs because the WSS connection is gone. I get messages as follows (debug=2): Mar 6 00:17:33 kamailio1 /usr/local/sbin/kamailio[5653]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Mar 6 00:17:33 kamailio1 /usr/local/sbin/kamailio[5653]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Mar 6 00:17:33 kamailio1 /usr/local/sbin/kamailio[5653]: ERROR: tm [t_fwd.c:527]: prepare_new_uac(): could not build request Mar 6 00:17:33 kamailio1 /usr/local/sbin/kamailio[5653]: ERROR: tm [t_fwd.c:1777]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches At debug=4, I get the output here: http://pastebin.com/d3RumekG In a running tshark, I see kamailio1 responding to kamailio2 with a "SIP/2.0 500 No error (2/TM)". It appears to me that this is happening before t_relay() even returns. Is there a way to prevent the SIP 500 from being sent back to the client? I am calling t_on_branch_failure("kam2bf") and t_on_failure("kam2f"), with each of those handlers (event_route[tm:branch-failure:kam2bf] and failure_route[kam2f]) written to print a log message, but it appears that neither of those handlers are being called. I tried installing these handlers independently to no avail. I also tried registering a t_on_reply() handler, thinking it was an internally-generated SIP response that might make it through my handler on its way out to kam2, but that is also not getting invoked. I want to return a 404 or something similar instead of having a 500 auto-returned for me. Related to this, is there a way to query kamailio's in-memory client connectivity data structure for the connectivity status of a client? Or is this something best done by keeping track of every client myself using (e.g.) localcache? Any pointers are appreciated. Thanks! Armen

2 3

help with a new install
by David Dunlap 10 Mar '15

10 Mar '15

Hello, I am seeking help with a new test server. I have SIP messages coming from a working FreeSwitch and getting posted into the sip_capture table. Homer web service is responding allowing me to manage log on accounts and other database settings however I do not see any packets in the web gui. I used the install script from the web page and did not notice any errors as it worked. I sure would appreciate some help. The IRC channel was empty. Thanks David Dunlap [cid:D854C04A-3D89-4B99-AD05-D242790BFC94] 3105 N. Main St. East Peoria, IL 61611 Direct: 309-427-7226 Main: 309-427-7000 Fax: 309-427-7326 Email: ddunlap(a)heart.net www.heart.net

2 1

Transfer call between devices
by Phil Daws 10 Mar '15

10 Mar '15

Hello: wish to ask how one may configure Kamailio so that if a user has a number of devices registered eg. deskphone, soft client on tablet, and soft client on cellphone when a call is originated or received how it can be passed between devices. Is it as simple as configuring a blind transfer ? Thanks for any help. Phil

2 1

4.2.3 textops: flag "f" doesn't work in subst_hf(hf, subexp, flags)
by Julia Boudniatsky 10 Mar '15

10 Mar '15

Hello, I received INVITE with 3 Diversion headers. I try to make substitutions in the body of a first Diversion header field by using flag "f", command *subst_hf("Diversion", /sip:(.*)@/sip:$var(dU)@/", "f") ;* substitutes *all headers *Diversion instead of only first header, command *subst_hf("Diversion", /sip:(.*)@/sip:$var(dU)@/", "*l*") ;* substitutes only *last* Diversion header. Thanks for help, Julia.

2 3

core parsing of SIP message failed errors
by Jeff Cohn 10 Mar '15

10 Mar '15

Hello, I finally got everything working but see to consistently receive the following 3 errors (see below) over and over when a call is being collected? Versions: Homer 3.6 kamailio 4.3.0-dev4 Captagent 4.2.0 Any assistance would be appreciated. Thanks, Jeff ------------------------- Errors: Mar 8 22:04:47 ce-homer2 /usr/local/kamailio/sbin/kamailio[9921]: ERROR: [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Mar 8 22:04:47 ce-homer2 /usr/local/kamailio/sbin/kamailio[9921]: ERROR: [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message=> Mar 8 22:04:47 ce-homer2 /usr/local/kamailio/sbin/kamailio[9921]: ERROR: [receive.c:129]: receive_msg(): core parsing of SIP message failed (127.0.0.1:33264/1) Mar 8 22:04:47 ce-homer2 /usr/local/kamailio/sbin/kamailio[9922]: ERROR: [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Mar 8 22:04:47 ce-homer2 /usr/local/kamailio/sbin/kamailio[9922]: ERROR: [parser/msg_parser.c:688]: parse_msg(): ERROR: parse_msg: message= Mar 8 22:04:47 ce-homer2 /usr/local/kamailio/sbin/kamailio[9922]: ERROR: [receive.c:129]: receive_msg(): core parsing of SIP message failed (127.0.0.1:33264/1)

2 4

We’ve got free Tickets for CeBIT 2015!
by Carsten Bock 10 Mar '15

10 Mar '15

Hi, Due to our presentation at CeBIT 2015 at the Open-Source Business Alliance’s Booth, we’ve received a limited number of free tickets for CeBIT 2015! If you want a free ticket, please answer the following question: - What is the key benefit of deploying VoLTE in a mobile network? Send your answer to info(a)ng-voice.com (hint: we’ve got some benefits posted on our website: www.ng-voice.com/volte), I will send the ticket by email to the 5 best answers! Don’t miss my presentation on Kamailio & VoLTE at the booth of the Open-Source Business Alliance (www.osb-alliance.de), Tuesday, March 17th, ~2 PM, Hall 6, Stand H16, (410)! Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

2 1

WARNING when querying data using db_mongodb
by Mickael Marrache 10 Mar '15

10 Mar '15

Hi, I'm continuously seeing the following WARNING in my logs: WARNING: db_mongodb [mongodb_dbase.c:454]: db_mongodb_get_columns(): unhandled data type column (instance) type id (10), use DB1_STRING as default It looks like the WARNING appears when data is fetched from the location collection. The instance field is always null in my case. Thanks, Mickael

2 1

Regd. Kamailio GPL
by Shankar 10 Mar '15

10 Mar '15

Hello, We are exploring kamailio source for use in our VOIP solution. This is the first time we are looking at open source. We have few doubts in using kamailio for providing our VOIP service. Under GPL any customization (customization mainly with respect to interacting with our proprietary applications e.g. Billing server) we do to kamailio also has to be provided to the end user. Our doubt is whether we can procure commercial rights to our customised code? Kindly clarify on this. Best Regards, Shankar

1 0

Dialog order of operations
by Alex Balashov 10 Mar '15

10 Mar '15

Hi, The dialog module documentation remains unclear about the order of operations with regard to when to call dlg_manage() or set the transaction flag. My impression is that dlg_manage() only registered TM callbacks, so it doesn't matter when you call it, as long as it's before t_relay(). However, the documentation neither confirms nor denies this. So, this raises the questions: 1) Is this okay? set_dlg_profile("caller", "$fU"); dlg_manage(); ... t_relay(); Or do I need to do this? dlg_manage(); set_dlg_profile("caller", "$fU"); ... t_relay(); 2) What about setting dialog-persistent variables? Is this okay? $dlg_var(account_id) = 49555; dlg_manage(); If so, where does the variable go if I never call dlg_manage() because the call is aborted beforehand, e.g. $dlg_var(account_id) = 49555; sl_send_reply("403", "Forbidden"); exit; dlg_manage(); ... t_relay(); 3) Any other gotchas or caveats in relation to the order of operations? I suppose my preference would be to set the dialog profiles in various places throughout call processing and call dlg_manage() at the very end, right before t_relay(). Is this acceptable? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 9

Announcing rtpproxy v2.0.0
by Maxim Sobolev 09 Mar '15

09 Mar '15

Hi All, I'm happy to announce that we have released rtpproxy v2.0.0. You can review the release notes here: https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 -sobomax

1 0

Enable SNMP causes issue when forking exec_avp request
by Daniel Goepp 09 Mar '15

09 Mar '15

I have an issue that from what I can tell is related to forking. My configuration works just fine, and makes a request out to execute curl to hit an external API: $var(my_query) = "curl -m 2 -s http://x.x.x.x/service/method?fromAddr=" + $var(fu) + "\&toAddr=" + $var(tu); Normally this gives us back a nice JSON response, no problems. All I do is enable snmpstats module, and it fails. Mar 5 16:51:30 server-name /usr/sbin/kamailio[1051]: ERROR: exec [exec.c:291]: exec_avp(): cmd curl -m 2 -s http://x.x.x.x/service/method?fromAddr=me@mydomain.com\&toAddr=someone@mydo… failed. exit_status=-1, errno=10: No child processes I have edited this for security reasons of course. If I disable SNMP, the problem goes away. Thoughts?

2 1

rtpproxy failed to load
by Tom Braarup Cuykens 09 Mar '15

09 Mar '15

kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I found this guide for an older version of Kamailio, but there seems to be some issues when trying to use it on the latest stable version. When I try to start Kamailio i get this : [....] 0(17153) INFO: tls [tls_init.c:399]: init_tls_compression(): tls: init_tls: disabling compression... 0(17153) ERROR: <core> [cfg.y:3286]: yyparse(): cfg. parser: failed to find command force_rtp_proxy (params 0) 0(17153) : <core> [cfg.y:3426]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 729, column 19: unknown command, missing loadmodule? 0(17153) ERROR: <core> [cfg.y:3286]: yyparse(): cfg. parser: failed to find command force_rtp_proxy (params 0) 0(17153) : <core> [cfg.y:3426]: yyerror_at(): parse error in config fil[FAILc/kamailio/kamailio.cfg, line 806, column 19: unknown command, missing loadmodule? ERROR: bad config file (2 errors) ... It seems it cannot load the rtpproxy module or that force_rtp_proxy command does not exist. For Debian the modules are located in /usr/lib/x86_64-linux-gnu/kamailio/modules/ and it was found there, seems to be loaded in the config file (line 5 and 235). According to the rtpproxy doc the module has this command. Does anyone have an idea what could be the issue ? I am planning to update this guide for the current version once I fixed all the issues. The most easy issues are the db/usernames/password and the path to the modules. -- Med venlig hilsen Tom Braarup Cuykens plusTEL.dk

2 1

lua_run seems to be working in asynchronous way
by José Seabra 09 Mar '15

09 Mar '15

Hi there, I'm using kamailio version 4.2.2 to a short time, and I'm using lua scripts to build some avps with information needed to make calls(lua functions uses redis as source data to build avp data.), what i have noticed today making some tests is that when i make a call my lua function: if(!lua_run("get_prefs","$fU","$fd")) { xdbg("SCRIPT: failed to execute lua function!\n"); } kamailio gets the avp's after send the invite to the destine,(the invite will fail because kamailio needs avp information). it seems that kamailio calls the lua_run in asynchronous way, i never noticed that before in old version. is that a normal behavior or it is an issue in this version? Best Regards -- Cumprimentos José Seabra

2 5

Re: [SR-Users] kamailio asterisk
by Slava Bendersky 08 Mar '15

08 Mar '15

Hello Everyone, I set up tcp and udp for right now to test all config and dialog is not established correctly. Call between extensions not working. Call from extesnsion to extension going right a way to voicemail. Any help thank you. Here one call debug. http://fpaste.org/195572/14258553/ Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 3, 2015 9:08:19 AM Subject: Re: [SR-Users] kamailio asterisk Hello Daniel, Is notify should follow subscribe routes ? Here SUBSCRIBE completely look normal to me. <--- SIP read from UDP:10.18.130.46:5060 ---> SUBSCRIBE sip:10102@networklab.ca SIP/2.0 Record-Route: <sip:kamailio_pub_ip:5081;transport=tls;lr=on;ftag=08c99307d3;nat=yes> Accept: application/simple-message-summary Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKdee2.b83c2028dc57c641d2b8cad968347d6a.0;i=2 Via: SIP/2.0/TLS 192.168.0.16:5063;rport=5063;received=client_pub_ip;branch=z9hG4bK0c54032bd8905d262 Max-Forwards: 69 From: "10102" <sips:10102@networklab.ca>;tag=08c99307d3 To: <sips:10102@networklab.ca> Call-ID: cea703eb8c21f709 CSeq: 1096440084 SUBSCRIBE Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Allow-Events: message-summary Authorization: Digest username="10102",realm="networklab.ca",nonce="29ac01f1",uri="sips:10102@networklab.ca",response="0298c2a93bdd4969a3cd0b757d671dbb",algorithm=MD5 Contact: <sips:10102@client_pub_ip:5063>;audio Event: message-summary Expires: 3600 Supported: eventlist, replaces, timer User-Agent: Media5-fone/4.1.6.3283 Android/4.1.2 Content-Length: 0 Path: <sip:outbound@10.18.130.46;lr;received=sip:client_pub_ip:5063%3Btransport%3Dtls> P-hint: outbound Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 3, 2015 1:50:03 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I pushed a fix for it -- the issue was in a LM_DBG() trying to print te address for local socket, but it was not set yet, it was affecting only when running in debug mode with debug=3. I pushed the patch to branch 4.2, so if you want the fix, then you have to install from git branch 4.2. For REGISTER, you need to add the Path header (see path module), otherwise the OPTIONS and INVITE sent out to the phone are not going via Kamailio. You have to check the version of your Asterisk to be one that supports Path. Cheers, Daniel On 27/02/15 14:40, Slava Bendersky wrote: Hello Daniel, Here paste for gdb http://fpaste.org/191338/25043949/ I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it. This is SUBSCRIBE route. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes> From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec To: <sips:10101@networklab.ca> ;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060> ;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, February 27, 2015 6:42:33 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for: p *tcpconn Daniel On 27/02/15 04:10, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here bt full from back trace. http://fpaste.org/191207/50064491/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, February 26, 2015 9:56:48 PM Subject: Re: [SR-Users] kamailio asterisk Hello Daniel, I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY. if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } Slava. Sent from mobile device typos are expected. From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Feb 25, 2015 1:04 PM To: Slava Bendersky Cc: sr-users Subject: Re: [SR-Users] kamailio asterisk Hello, On 25/02/15 17:19, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, substr you suggested didn't worked. See my previous email. your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri(). Or you can try the alternative with: if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today. Cheers, Daniel BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Group call advice: remove calling branch after alias_db_lookup/lookup_branches
by Anthony Messina 08 Mar '15

08 Mar '15

In terms of implementing "group" calling via the append_branch feature of both alias_db_lookup followed by lookup_branches, I'm looking for a reliable way to ensure that if the caller happens to be a member of the group (list of branches), the branch that's created to the original caller is dropped. Originally, I was thinking of comparing $rU and $fU in branch_route, but this would limit the ability for one contact of an AOR to call another contact of the same AOR. Can anyone offer an example of an efficient method to accomplish this? -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 3

ndb_redis set <null> reply to EXISTS query
by Yuriy Gorlichenko 06 Mar '15

06 Mar '15

Hello I try to get some replies from redis. Time after time redis request give me null result. But redis bs not disconnected. This happens only with websocket endpoints. My queries is: redis_cmd("srv1", "EXISTS $si", "s"); So at xLOG i see that $si correctly sended, but result is null. At db I keep key IP with value TIMESTAMP Intresting that route, that givesme this result idependend of WS or UDP endpoint. IT does not know about it anything.

2 11

Route Between WebSockets without rtpengine getting into the mix
by Don Fanning 05 Mar '15

05 Mar '15

Hello All, Prior to going down the rtpengine path for another need, calling was working fine between WSS client. Now that I've added in some pieces from caruizdiaz's config [ https://github.com/caruizdiaz/kamailio-ws/blob/master/kamailio-ws.cfg] relating to SETUP_BY_TRANSPORT, it has broken that functionality. Is there a way to skip rtpengine if the originator and destination are both on wss? route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { route(SETUP_BY_TRANSPORT); if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); if(sdp_with_transport("RTP/SAVPF")) { rtpengine_manage("rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=force DTLS=passive"); t_on_reply("REPLY_WS_TO_WS"); return; } #rtpengine_manage("rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=force DTLS=passive RTP/SAVP"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); rtpengine_manage("rtcp-mux-demux trust-address replace-origin replace-session-connection ICE=remove DTLS=passive RTP/SAVP"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); rtpengine_manage("replace-origin replace-session-connection"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } Thanks for any help! -Don

2 1

segfault problem with unixodbc module
by Lucas Wals Ochoa 05 Mar '15

05 Mar '15

Hello, this is the first time I send a question here, hope my english is good enough to explain my problem. I have a Kamailio 4.1.8+RTPProxy setup working, but recently I saw RTP Engine proxy, so I decided to give a try. I installed the latest kamailio stable version (4.2.3) and adapt the configuration I have(that uses a MariaDB database), to try the rtpengine module, but I couldn't even start kamailio process, cause I received a segfault problem at unixodbc module. The version of unixodbc that I installed is 2.3.1-10 That was the first time I encounter this error. I thought that maybe the database I have created with the version 4.1.8 wasn't compatible, so I decided to create a new database, but the problem persisted. So I'm here to know if someone know something about this, or have had a similar problem. I attached the system log for Kamailio, and a backtrace I made using "gdb" with the corefile that Kamailio creates when it crash. Log: Mar 4 12:46:35 localhost systemd: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 4 12:46:35 localhost kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 10.0.0.70 Mar 4 12:46:35 localhost /usr/sbin/kamailio[2369]: WARNING: tm [tm.c:845]: mod_init(): WARNING: mod_init: reparse_on_dns_failover is enabled on a multihomed host -- check the readme of tm module! Mar 4 12:46:35 localhost /usr/sbin/kamailio[2369]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Mar 4 12:46:35 localhost /usr/sbin/kamailio[2369]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Mar 4 12:46:35 localhost /usr/sbin/kamailio[2369]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Mar 4 12:46:35 localhost /usr/sbin/kamailio[2369]: INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Mar 4 12:46:35 localhost kernel: kamailio[2369]: segfault at 10070 ip 00007fc3d554df7e sp 00007fff89556450 error 4 in libmyodbc5w.so[7fc3d5527000+47000] Mar 4 12:46:35 localhost kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe Mar 4 12:46:35 localhost kamailio: Starting kamailio: Mar 4 12:46:35 localhost systemd: kamailio.service: control process exited, code=exited status=1 Mar 4 12:46:35 localhost systemd: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 4 12:46:35 localhost systemd: Unit kamailio.service entered failed state. Backtrace of gdb: Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 0x00007fc3d554df7e in get_ctype () from /usr/lib64/libmyodbc5.so Missing separate debuginfos, use: debuginfo-install kamailio-4.2.3-3.1.x86_64 (gdb) backtrace #0 0x00007fc3d554df7e in get_ctype () from /usr/lib64/libmyodbc5.so #1 0x00007fc3d554dfac in init_parser () from /usr/lib64/libmyodbc5.so #2 0x00007fc3d554ec41 in parse () from /usr/lib64/libmyodbc5.so #3 0x00007fc3d5553484 in prepare () from /usr/lib64/libmyodbc5.so #4 0x00007fc3d5559e9c in SQLPrepareWImpl () from /usr/lib64/libmyodbc5.so #5 0x00007fc3d5559eb9 in SQLExecDirectW () from /usr/lib64/libmyodbc5.so #6 0x00007fc3de43b05f in SQLExecDirect () from /lib64/libodbc.so.2 #7 0x00007fc3de695302 in db_unixodbc_submit_query () from /usr/lib64/kamailio/modules/db_unixodbc.so #8 0x00007fc3de21522e in db_do_submit_query () from /usr/lib64/kamailio/libsrdb1.so.1 #9 0x00007fc3de215cf7 in db_do_query_internal () from /usr/lib64/kamailio/libsrdb1.so.1 #10 0x00007fc3de21661a in db_do_query () from /usr/lib64/kamailio/libsrdb1.so.1 #11 0x00007fc3de696d90 in db_unixodbc_query () from /usr/lib64/kamailio/modules/db_unixodbc.so #12 0x00007fc3de20ce90 in db_table_version () from /usr/lib64/kamailio/libsrdb1.so.1 #13 0x00007fc3da209693 in init_ds_db () from /usr/lib64/kamailio/modules/dispatcher.so #14 0x00007fc3da1f4f1a in mod_init () from /usr/lib64/kamailio/modules/dispatcher.so #15 0x0000000000520e8d in init_mod () #16 0x0000000000520be8 in init_mod () #17 0x000000000052119a in init_modules () #18 0x0000000000517951 in main () Hope someone can help me. Thanks in advance.

3 5

WARNING: <core> [local_timer.c:83]: _local_timer_dist_tl(): 0 expire timer added"?
by Anthony Messina 05 Mar '15

05 Mar '15

Good day. Is anyone able to let me in on what this means "WARNING: <core> [local_timer.c:83]: _local_timer_dist_tl(): 0 expire timer added"? I just started seeing this after pulling down 5c1a9df from git and it seems like something that I probably don't want happening. -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

3 13

wss calls between devices works but strange error
by Don Fanning 05 Mar '15

05 Mar '15

Hi All, I'm placing calls between two devices over WSS and sipML5. Connection seems to work and stays up (despite some grumbling sometimes with 603 failed to get local SDP) but my question is the following log entries - could someone translate what is happening here? Thanks! Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: WARNING: <core> [tcp_main.c:1221]: tcp_do_connect(): 127.0.0.2:5061: could not find corresponding listening socket for 127.0.0.1, using default... Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 127.0.0.2:5061 failed (RST) Connection refused Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 127.0.0.2:5061: connect & send for 0x7f0eb3944ac0 failed: Connection refused (111) Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [forward.h:247]: msg_send(): tcp_send failed Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: INFO: <script>: Request in RELAY - call is {<null>} for {106E} Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: WARNING: <core> [tcp_main.c:1221]: tcp_do_connect(): 127.0.0.2:5061: could not find corresponding listening socket for 127.0.0.1, using default... Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 127.0.0.2:5061 failed (RST) Connection refused Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 127.0.0.2:5061: connect & send for 0x7f0eb3944ac0 failed: Connection refused (111) Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [forward.h:247]: msg_send(): tcp_send failed Mar 5 03:35:19 sandbox /usr/local/sbin/kamailio[4917]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Mar 5 03:35:21 sandbox /usr/local/sbin/kamailio[4917]: INFO: <script>: Request in RELAY - call is {<null>} for {106E} Mar 5 03:35:21 sandbox /usr/local/sbin/kamailio[4917]: WARNING: <core> [tcp_main.c:1221]: tcp_do_connect(): 127.0.0.2:5061: could not find corresponding listening socket for 127.0.0.1, using default... Mar 5 03:35:21 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 127.0.0.2:5061 failed (RST) Connection refused Mar 5 03:35:21 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 127.0.0.2:5061: connect & send for 0x7f0eb3944ac0 failed: Connection refused (111) Mar 5 03:35:21 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [forward.h:247]: msg_send(): tcp_send failed Mar 5 03:35:21 sandbox /usr/local/sbin/kamailio[4917]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Mar 5 03:35:25 sandbox /usr/local/sbin/kamailio[4917]: INFO: <script>: Request in RELAY - call is {<null>} for {106E} Mar 5 03:35:25 sandbox /usr/local/sbin/kamailio[4917]: WARNING: <core> [tcp_main.c:1221]: tcp_do_connect(): 127.0.0.2:5061: could not find corresponding listening socket for 127.0.0.1, using default... Mar 5 03:35:25 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 127.0.0.2:5061 failed (RST) Connection refused Mar 5 03:35:25 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 127.0.0.2:5061: connect & send for 0x7f0eb3944ac0 failed: Connection refused (111) Mar 5 03:35:25 sandbox /usr/local/sbin/kamailio[4917]: ERROR: <core> [forward.h:247]: msg_send(): tcp_send failed Mar 5 03:35:25 sandbox /usr/local/sbin/kamailio[4917]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Mar 5 03:35:33 sandbox /usr/local/sbin/kamailio[4918]: INFO: <script>: Request in RELAY - call is {<null>} for {106E} Mar 5 03:35:33 sandbox /usr/local/sbin/kamailio[4918]: WARNING: <core> [tcp_main.c:1221]: tcp_do_connect(): 127.0.0.2:5061: could not find corresponding listening socket for 127.0.0.1, using default... Mar 5 03:35:33 sandbox /usr/local/sbin/kamailio[4918]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 127.0.0.2:5061 failed (RST) Connection refused Mar 5 03:35:33 sandbox /usr/local/sbin/kamailio[4918]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 127.0.0.2:5061: connect & send for 0x7f0eb3944ac0 failed: Connection refused (111) Mar 5 03:35:33 sandbox /usr/local/sbin/kamailio[4918]: ERROR: <core> [forward.h:247]: msg_send(): tcp_send failed Mar 5 03:35:33 sandbox /usr/local/sbin/kamailio[4918]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Mar 5 03:35:49 sandbox /usr/local/sbin/kamailio[4918]: INFO: <script>: Request in RELAY - call is {<null>} for {106E} Mar 5 03:35:49 sandbox /usr/local/sbin/kamailio[4918]: WARNING: <core> [tcp_main.c:1221]: tcp_do_connect(): 127.0.0.2:5061: could not find corresponding listening socket for 127.0.0.1, using default... Mar 5 03:35:49 sandbox /usr/local/sbin/kamailio[4918]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 127.0.0.2:5061 failed (RST) Connection refused Mar 5 03:35:49 sandbox /usr/local/sbin/kamailio[4918]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 127.0.0.2:5061: connect & send for 0x7f0eb3944ac0 failed: Connection refused (111) Mar 5 03:35:49 sandbox /usr/local/sbin/kamailio[4918]: ERROR: <core> [forward.h:247]: msg_send(): tcp_send failed Mar 5 03:35:49 sandbox /usr/local/sbin/kamailio[4918]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL)

2 1

How to Use Multiple DMQ Notification Addresses
by Robert Boisvert 04 Mar '15

04 Mar '15

Hello all, I would like to setup a DMQ bus where servers can come on and offline and have DMQ continue working with the online servers. I'm having trouble understanding how to do that with the notification address. Can someone point me in the right direction? More specifically, suppose I have five servers named A, B, C, D, and E. I would like all of them to be on the DMQ bus when active but they may come up at different times. For example, when A, B and C start I want each of them to be on the same DMQ bus. However, the way I understand it, if I make E the notification address none will be on the bus because E is not active. I could create a circular notification chain where A uses B as a notification server, B uses C, and so on. Of course, there are problems when one or more servers are missing from the chain. Any help on this is greatly appreciated. Thanks, Bob

2 3

SCA fails to update appearance
by Andreas Granig 04 Mar '15

04 Mar '15

Hi guys, Playing with SCA for the first time, I'm hittin the following issue: User phone1(a)domain.org is registered on two devices A and B, and device B sends SUBSCRIBE to phone1(a)domain.org with call-info Event. Subscription is handled properly, and my DB content is this: id: 4 subscriber: sip:phone1@10.15.20.174:5100;alias=1.2.3.4~5100~1 aor: sip:phone1@domain.org event: 1 expires: 1425296906 state: 0 app_idx: 0 call_id: dacf5194-3d6a8e16d03338102a940080f0e9071c(a)10.15.20.174 from_tag: 4247246264 to_tag: f3067022b00c564156251ba2f28f331f-7c92 record_route: <sip:127.0.0.1;r2=on;lr=on;ftag=4247246264;nat=yes;ngcplb=yes;socket=udp:1.2.3.4:5060>,<sip:1.2.3.4;r2=on;lr=on;ftag=4247246264;nat=yes;ngcplb=yes;socket=udp:1.2.3.4:5060> notify_cseq: 3 subscribe_cseq: 2 Kamcmd reports this: kamcmd> sca.all_subscriptions phone1 10.15.20.174:5100 call-info 2697 active If phone1(a)domain.org on device A does a call, I get this in my logs when calling sca_call_info_update() right as first thing in my routing config handling the INVITE: Mar 2 12:02:33 sp2 proxy[34517]: WARNING: sca [sca_appearance.c:976]: sca_appearance_update_index(): Cannot update sip:phone1@domain.org index 0 to unknown: index 0 not in use Mar 2 12:02:33 sp2 proxy[34517]: ERROR: sca [sca_call_info.c:1046]: sca_call_info_invite_request_handler(): Failed to update sip:phone1@domain.org appearance-index 0 to unknown Is there something I'm obviously doing wrong here? Andreas

2 4

switch called number
by jaflong jaflong 04 Mar '15

04 Mar '15

Hi Guys I want to archive this setup Kamailio to switch called number and the switch to be hidden from the client so there should be no mention of the switched number in any of the messages back to the client. for example phone client calls "alex" kamailio to switch "alex" to "54321" and send to asterisk on response from asterisk, kamailio to revert all references to "54321" back to "alex" before send to client Any suggestions please

2 1

TLS conversion
by Kelvin Chua 04 Mar '15

04 Mar '15

I have 2 kamailio servers and 1 asterisk server. 1. asterisk calls kamailio1 2. kamailio1 relays INVITE to kamailio2 3. kamailio2 relays INVITE to client registered using TLS 4. client answers with 200 OK, sends to kamailio2 5. kamailio2 relays 200 OK to kamailio1 6. kamailio1 relays 200 OK to asterisk 7. asterisk sends ACK to kamailio1 8. kamailio1 complains "forward_request(): forward_req: ERROR: cannot forward to af 2, proto 3 no corresponding listening socket" which is because 200 OK has a contact header with tls transport from client 9. call drops in 30 secs which is expected because client never received the ACK any ideas on a fix or workaround? Kelvin Chua

2 7

Any suggestion if I just want to update one or some modules?
by jack wang 04 Mar '15

04 Mar '15

For example, If I modify only the registrar module, I thought the steps were: 1. make modules=modules\registrar modules 2. copy the registrar.so to the installed path And when starting kamailio I got the error message below: ERROR: <core> [sr_module.c:417]: version_control(): ERROR: module compile flags mismatch for [my installed path]/registrar.so I've searched for the answer and it seems to need to re-build and re-install kamailio and all the module again... Is there more efficient way for just updating the specified module ?? Thanks a lot.:)

2 2

Can following scenario be implemented using Kamilio
by Jayanth Acharya 03 Mar '15

03 Mar '15

Would like to know if Kamilio (in it's sip-proxy role) could be configured to somehow take note of the 'Contact:' header of a 302 response from B-party, to forward the INVITE to C-party (specified by the updated Contact URI) ? Alternatively, is there a way for the B-party SIP UAS to get the proxy (Kamilio) to forward the INVITE (from A-party) to C-party ? Does Kamilio have to be configured to act in a specific / non-default role ? In doing so, B-party receives no further SIP messages for this particular dialog. thanks, J

3 14

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