- sr-users - kamailio.org

LM_ERR and LM_DBG
by aawaise 04 Aug '14

04 Aug '14

I want to ask some queries regarding logging in kamailio. 1. First of all if I want to add LM_DBG or LM_ERR in lookup.c file of registrar module, where to add the command as on extraction of kamailio-3.1.1.zip, I found registrar module files in modules_k folder as well as modules_s folder. 2. I was advised to us macros from dprint.h to see LM_DBG and LM_ERR statements in log file, how to do so ?? 3. Third, will these logs also print in /var/log/messages in debian linux along with other kamailio.cfg logs ?? 4. What changes should be made in kamailio.cfg to see these logs, i.e log level etc ?? Any help will be highly appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/LM-ERR-and-LM-DBG-tp129392.html Sent from the Users mailing list archive at Nabble.com.

2 1

See XDBG and XERR Messages in Logs
by aawaise 04 Aug '14

04 Aug '14

I have added some XLOG and XDBG commands in registrar module of kamailio. I want to see messages in result of these commands being printed in logs to make sure that changes are being made surely. How can it be done ? I am using fedora and I see kamailio related logs in /var/log/messages.log -- View this message in context: http://sip-router.1086192.n5.nabble.com/See-XDBG-and-XERR-Messages-in-Logs-… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration - Problem
by Reza Kiani Moghaddam 03 Aug '14

03 Aug '14

Hi, I Installed Kamailio & Asterisk on Ubuntu 14.04 LTS, and both of them started successfully. I can register with Jitsi, But I can't call other users.. I don't know what to do ...

2 1

Kamailio behind NAT- alias issue
by Yuriy Gorlichenko 03 Aug '14

03 Aug '14

Hello. I have Kamailio running behind NAT. It lesten eth0 with ip 192.168.0.3 and I have external IP that have domain name (for example sip.myserver.com). Register packets from clients comes from external IP. If I write at kamailio.cfg: alias=sip.myserver.com I see error at log - bad_uri sip.myserver.com I think it happens because kamailio does not know atything about external ip pecause kamamiliol working with server interfaces (eth0, eth1, etc.) So my question - how to listen external IP with domain name on kamailio that running behind NAT? P.S. I do not have any access to router, that present NAT for me.

2 2

SIPp load testing
by bob sacamano 02 Aug '14

02 Aug '14

I am trying to follow the guide shown here:http://www.kamailio.org/docs/openser-performance-tests/ to load test my kamailio system.I am a little unclear as to how many instances of sipp are running. For the first part i see the command ./sipp -sf uac_msg.xml -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 10000 -d 1 -l 70for generating the UAC part but is there another instance of sipp running on the kamailio computer acting as a UAS? I have seen come sites use for example: sipp 192.168.1.100:5060 -sn uas -p 5060 but is this not necessary? thanks for any help

2 3

Sample Kamailio Config for Siremis
by Noah Mehl 02 Aug '14

02 Aug '14

I have managed to install Kamailio and Siremis 4.x, but I have noticed that neither Kamaliio or Siremis come with a sample config that makes sense for Siremis (as far as I can tell). For instance, if I want to use LCR with dispatchers, the sample Kamilio configs don’t have the dispatcher module loaded or configured. Am I missing something? Can someone please share theirs? Thanks! ~Noah

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ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene>
by Rene Montilva 01 Aug '14

01 Aug '14

Hi list i've been working with t_suspend, t_continue and async sleep, now the problem it's with register users, the softphones are frozen when try to establish a session with kamailio, now my users can't register and the only error i got is this: ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene> my kamailio config: # ----- 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", 30) # max value for expires of registrations #modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "hash_size", 15) #!endif

1 0

kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached kamailio.cfg does this work. There is an problem when handling SUBSCRIBE requests (as required for BLF and voicemail indications). My configuration is written so that these SUBSCRIBE requests are not handled by kamailio, but instead routed to asterisk. There is a failure to check From/To headers to see whether NOTIFY packets generated as part of a subscription can be restored using the information in Record-Route. The end result is that kamailio ends up sending packets with garbled tags that are (rightly) rejected by the SIP endpoint. The following is an example that demonstrates the issue (using Jitsi as endpoint): After registration, Jitsi sends a SUBSCRIBE request: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com> Max-Forwards: 70 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Proxy-Authorization: Digest username="avillacisIM",realm="pbx.villacis.com",nonce="U9lZJlPZV/r06Xep/ukc1UzAIO0V3TbS",uri="sip:avillacisIM@pbx.villacis.com",response="0e18f4913c2693f6154c91f158fb17fe" Content-Length: 0 This packet is mangled by the configuration, and is sent to asterisk like this: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> Max-Forwards: 69 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Content-Length: 0 The asterisk response for the SUBSCRIBE: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080>;expires=3600 Content-Length: 0 This is in turn transformed back by kamailio, and sent to Jitsi like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080;alias=127.0.0.1~5080~1>;expires=3600 Content-Length: 0 Now asterisk wants to send a NOTIFY to the endpoint for the subscription. The NOTIFY looks like this: NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport Max-Forwards: 70 Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes>,<sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Here is where the bug appears. The autoprocessing does not recognize that the From header (From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e) from the above request has nothing to do with the saved information (vsf parameter). Instead, it blindly mangles the From header, and does not even run a sanity check on the result before routing it. The end result is shown below. NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as5562e95e> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as5562e95e> Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bK8333.8bfe7bc2bd554a8631f0d00d463b28ee.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport=5080 Max-Forwards: 69 From: "asterisk" <sip:asterisk@12(.0.0.1:5080.....@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) From examination of the source code, the vsf and vst strings are base64 encodings of the result of XORing the byte strings of the old and new tags. For this to work, the headers of future packets should match. However, here kamailio does not realize that the header does not match (by the ftag), and also does not check that the resulting "restored" header is a valid header.

2 2

SCA module - sca_call_info_update() function for 180 responses
by João Vitor Arruda 01 Aug '14

01 Aug '14

Hello all, I've had proposed a fix to SCA module in the tracker ( https://sip-router.org/tracker/index.php?do=details&task_id=446) but did´t hear anything about it until now. I would appreciate some comments on that to see if its feasible to be committed or even hear for anyone else using SCA module with Polycom phones to see if they have the same issue described. Thanks, João Vitor Arruda

4 5

Kamailio V4.1.3 - Dialog timeout -ACC ERROR
by Yasin Caner 01 Aug '14

01 Aug '14

Hello; i am trying to limit call duration with dialog module. So i use *timeout_avp *for limiting it. there is no problem to close calls but when it happens i got some errors and kamailio can not write Bye result in DB. There is code sample and result in below. 2. question : is there a big difference between dialog_ng and dialog? Which one is more stable? 3. question : After restarting kamailio , Dialog hash vanish so timeout isn't working. Can DB mode solve this problem? PS: event_route[dialog:end] isn't working in Kamailio V 4.1.4. I got some error like t_w_relay. Thanks for helps. ------------------------------------------------------------------------ #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif ------------------------------------------------------------------------ route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # 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()) { 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; } } ------------------------------------------------------------------------ Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 host-91-93-189-136 /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

1 0

sip
by Nar 01 Aug '14

01 Aug '14

Hi, I am developing a social application where I want to enable voice & video calling function and still I don't know what to use in server side, do you have any product that I can use to develop the server side(for 10.000+ users)? Thank you in advance.

2 1

[Kamailio]list members
by Djamel Bahamid 01 Aug '14

01 Aug '14

Hi, I would like to subscribe to the list. Best Regards, Djamel BAHAMID.

2 1

SIGUSR1 for memory status not working as documented - only one process reports back
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a kamailio process will dump a report of the allocation map when shutting down, or when receiving a SIGUSR1. I have configured my kamailio.cfg with memlog=1 and no other change, and I see the memory report on shutdown for all processes. However, when I send a SIGUSR1 to a kamailio process, the process does absolutely nothing, with the exception of the first kamailio process (the one reported as Type=attendant by "kamctl ps"). All of the other processes just ignore SIGUSR1. What is going on? It is inconvenient to force a shutdown of all the kamailio processes just to get the memory report.

2 1

Unable to SUBSCRIBE for presence using sip.js through WSS
by Alex Villacís Lasso 31 Jul '14

31 Jul '14

My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the NOTIFY for the presence event. However, MESSAGE packets are routed correctly between peers, even through websockets. The location table after registration looks like this: +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | id | ruid | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods | instance | reg_id | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | 192 | uloc-53d96398-1afc-1 | admin | pbx.villacis.com | sip:nfinhooe@192.0.2.90;transport=ws | sip:192.168.3.2:41478;transport=WS | NULL | 2014-07-30 17:18:57 | -1.00 | 6a2dot9ihoobhng9ackqmq | 90 | 2014-07-30 17:08:57 | 0 | 64 | SIP.js/0.5.0 | tls:192.168.2.18:5061 | 782 | <urn:uuid:b8888e95-7833-46a4-a21e-8d2e809a66b8> | 1 | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ When running the following javascript code: var subs = ua.subscribe('admin(a)pbx.villacis.com', 'presence'); subs.on('notify', function (n) { alert(n.request.body); }); ... I see the following traffic in the browser console area: SUBSCRIBE sip:admin@pbx.villacis.com SIP/2.0 Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279 Max-Forwards: 70 To: <sip:admin@pbx.villacis.com> From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="admin", realm="pbx.villacis.com", nonce="U9lswVPZa5WSu5jz1P/jIOJTvthKXw6v", uri="sip:admin@pbx.villacis.com", response="f7246e64573b5888d69896a70d4204aa" Event: presence Expires: 3600 Contact: <sip:nfinhooe@192.0.2.90;transport=ws> Allow: ACK,CANCEL,BYE,OPTIONS,MESSAGE Supported: outbound User-Agent: SIP.js/0.5.0 Content-Length: 0 SIP/2.0 202 OK Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279;rport=41478;received=192.168.3.2 To: <sip:admin@pbx.villacis.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-84c3 From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.2.18:5061;transport=ws> Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 However, I also get the following messages in the kamailio log: Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription I have found a discussion at http://marc.info/?l=sr-users&m=136624316816288&w=2 which recommends fix_nated_contact(), but I am not really sure about how to add it, or what conditions to check on whether it should be used. I added the call to fix_nated_contact() like this: # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); # # 1 - Contact header field is searched for occurrence of RFC1918 or rfc6598 addresses. # 2 - the "received" test is used: address in Via is compared against source IP address of signaling # 16 - test if the source port is different from the port in Via # 64 - test if the source connection of signaling is a WebSocket # ----- # 83 #!ifdef WITH_WEBSOCKET # 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. if (nat_uac_test("83")) { #!else if (nat_uac_test("19")) { #!endif if (is_method("REGISTER")) { fix_nated_register(); #!ifdef WITH_WEBSOCKET if (nat_uac_test("64")) fix_nated_contact(); #!endif } else { if (is_first_hop()) if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); } #!endif return; } After registration, "kamctl ul show" reports this: [root@elx3 js]# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: admin(a)pbx.villacis.com Contact:: sip:85017jar@192.168.3.2:41777;transport=ws Q= Expires:: 584 Callid:: argl25kg0r4qe8lbtbhn9f Cseq:: 82 User-agent:: SIP.js/0.5.0 Received:: sip:192.168.3.2:41777;transport=WS State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: tls:192.168.2.18:5061 Methods:: 782 Ruid:: uloc-53d971c7-297f-1 Instance:: <urn:uuid:75464840-96b8-4e22-8eaa-5012ba327086> Reg-Id:: 1 Last-Keepalive:: 1406759390 Last-Modified:: 1406759390 However, it still does not work: Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription

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Obscuring SIP traffic and using with NoSIP
by Muhammad Shahzad 31 Jul '14

31 Jul '14

Hi, As the mobile voip is getting more and more popular these days, there has been a strong opposition from GSM operators against mobile voip apps. They often use tactics like blocking voip ports, or detect and block voip traffic and in some cases restricting udp traffic altogether to very low upload and download speeds. See below link for some details, http://www.linphone.org/eng/blog/linphone-over-3g.html While not all the problems can be solved right now (especially the limiting udp traffic, since RTP always uses udp transport) I was wondering if we can at least handle the sip related problems. The most important of them is SIP traffic detection. While some forks would suggest using TCP/TLS to encrypt SIP traffic, it has a few problems, e.g. 1. It requires somewhat high resources on mobile devices, so many low-end android phones simply can't use it. 2. There is possibility that encryption signature may identify it as SIP traffic. There exists firewalls (often deployed in middle eastern countries) which have huge database of encryption signatures and patterns which although may not decrypt the sip packet but at least identify it as sip packet and block it. Also with rough agencies of evil empires spying over millions of users worldwide makes the current encryption standards pretty much pointless, at least in terms of user privacy and network security. So there is a strong need to experiment with new ideas and concepts to regain internet freedom. Some of such ideas are, 1. Convert sip traffic which is plain text to binary format just before transmitting it and revert it to plain text upon reception. 2. XOR the sip traffic (pretty much same as binary sip). 3. Use some very lightweight but effective / non-standard encryption algorithm, e.g. https://github.com/mshary/itv All these ideas require that SIP server such as Kamailio is able to adopt to these, preferably with minimal or no change in native code. The NoSIP module seems an interesting module in this regard. It provides all traffic it thinks is not the SIP traffic to configuration script, where we can do our own parsing and do whatever we want with it. I have two questions about this, 1. If parsed message is SIP, we can we send it back to kamailio core to get it processed as if it is a normal SIP message received by kamailio? 2. Can this module or any other module available in kamailio, that can provide us full sip packet that is about to be transmitted over sip socket, so we can "encode" it just before it is sent to next hop? I know this would be like writing a SIP transport in kamailio script which would be very tough if not impossible to implement in native core. But it will really help in winning the modern mobile voip challenges. Thank you.

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[Serusers] how to enable logs in SER
by pawan 31 Jul '14

31 Jul '14

Sir pl help for registration of soft phone to my local sip server my email vijay8538(a)gmail.com --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

1 0

multi-homed rtpengine
by Paul Belanger 30 Jul '14

30 Jul '14

Greetings, We have had good success in the past using rtpproxy with internal and external interfaces however, we are attempting to migrate to rtpengine now but are running into an issue (apparently this feature is not supported). So, my question is as follows, does this mean we need to multiple rtpengine and bind them to each interface? Additionally, in this setup can audio be bridged between the 2 rtpengine processes? If not, how do people over come this difference? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [SR-Users] LCR
by Keith 30 Jul '14

30 Jul '14

Hi, I want to do LCR based on from uri. I get I need to do regular expression matching but have no idea where to look to start. Any pointers would be good. Thanks Keith

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Warning message on SIP websocket
by Alex Villacís Lasso 29 Jul '14

29 Jul '14

Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Via: SIP/2.0/TLS 192.168.3.2:58657#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: Er/CgVyBNzVGREBmt+w3FTff8s4=#015#012Server: kamailio (4.1.4 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012>

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Plz help: MSILO on TLS + cert based client auth enabled server
by Marc M. 29 Jul '14

29 Jul '14

Hi, I have a SIP server with TLS + client certificate based authentication. The server is listening for SIP(UDP) on port 5060, and for SIPS(TLS) on port 5061) The client certificate based authentication is enforced on port 5061. I would like to fire up MSILO and face now a strange problem: At each REGISTER, the stored messages get dumped. Dumped messages get sent to the clients from senders logged in over port 5060(UDP) as SIP:xxxxxxxxx@xx.xx.xx.xx however Dumped messages are NOT sent to clients from senders logged in on port 5061(TLS) as SIPS:xxxxxxxx@xx.xx.xx.xx It looks like the messages get dumped so M-dump() is working, but they will never enter the route. I would assume TLS is stopping the dumped messages to entering the route logic. I would either 1. somehow force m_dump() to pipe the messages over SIP(and not SIPS), or 2. disable client auth for messages recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc

2 1

Only a redirect server
by Jack Smith 29 Jul '14

29 Jul '14

Dear friends, Just new to this mailing list, new to Kamailio, and new to running a sip server. I would like to try only a redirect server. I had a look at http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg which was friendly, but then the default file kamailio.cfg looked frightening ! Could someone help me set up a minimal Kamailio installation only to handle state-less redirect? Best, Jack

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append_branch and t_replicate issue
by Asgaroth 28 Jul '14

28 Jul '14

Hi All, I have an issue that I cant seem to get to the bottom of, I would appreciate if someone could point me in the right direction. I have 3 registrar's and would like to replicate the registration regest from 1 to the other 2, I am trying this with t_replicate, the docs say that to send to multiple destinations an append_branch should be performed prior to the t_replicate, which is what I am trying to do. What appears to happen is that the replication only sends the REGISTER to the server defined in the t_replicate command, it doesnt appear to be sending it to the destination defined in append_branch. Kamailio Version: version: kamailio 4.1.4 (x86_64/linux) 84c1ff 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, DBG_QM_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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 84c1ff compiled on 14:36:04 Jul 20 2014 with gcc 4.4.7 Here is what I have tried so far (on registrar 1): #!define BACKUP_REGISTRAR_1 '1.1.1.1' #!define BACKUP_REGISTRAR_2 '1.1.1.2' #!define BACKUP_REGISTRAR_3 '1.1.1.3' save("location"); if ( src_ip != BACKUP_REGISTRAR_2 && src_ip != BACKUP_REGISTRAR_3 ) { append_branch("sip:" + BACKUP_REGISTRAR_2 + ":5060"); t_replicate("sip:" + BACKUP_REGISTRAR_3 + ":5060"); }; With the above in place, I only see the system attempting to replicate to BACKUP_REGISTRAR_3, I dont see any request to BACKUP_REGISTRAR_2. Am I doing something wrong here? Thanks Bruce

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possible regression on 4.1.3 vs 4.1.4
by Kelvin Chua 26 Jul '14

26 Jul '14

using the default cfg file, whenever a WITHINDLG BYE is received, 4.1.4 needs a retransmission before passing it through, the first BYE would generate this in syslog tm [tm.c:1518]: _w_t_relay_to(): ERROR: w_t_relay_to: unsupported route type: 64 and the 2nd retransmitted BYE would go on fine. i believe route type 64 is branch failure, but using the same cfg and reverting back to 4.1.3 solves it. so whatever changes was done to 4.1.4 broke it. Kelvin Chua

2 4

Rtpengine vs. TURN?
by Peter Villeneuve 25 Jul '14

25 Jul '14

Hi, On my server, I have the option of using either Rtpengine for NAT traversal or pure TURN without rtpengine. Rtpengine has the obvious plus that it only needs 1 public IP, while TURN (with STUN) will need 2 public IPs, although that's not a problem in my case. Having said that, I'd like to take advantage of the huge experience that users of this list have in real world deployments. in your experience, which option is more reliable in a real world deployment? Cheers, Peter

2 3

MSILO not checking Expires header if Allow: is absent and Contact: is present but without expires
by Anthony Messina 25 Jul '14

25 Jul '14

In setting up Kamailio with MSILO, I've noticed that some sip clients (I'm using CSipSimple), upon un-REGISTER-ing do not send an Allow header, but do send an "Expires: 0" header as well as a Contact header. However as the Contact header doesn't contain "expires=0", it appears as though the MSILO skips over checking the Expires header. As such, I've had to add the following workaround to my route[REGISTRAR] block or else these clients trigger the m_dump. # Ensure we aren't unregistering # Right now, CSipSimple doesn't send an Allow header with an un-REGISTER, # so Kamailio skips to checking the Contact header for expires, and appears # to ignore the fact that an Expires: 0 header is present. if($hdr(Expires) != 0) m_dump(); Looking through the msilo check_message_support function, it appears that it never checks for the Expires header if there is no Allow header and the Contact header does not specify an expire setting. I do believe that we're missing the check for the Expires header, regardless of whether or not there is an Allow header, or a Contact header with expires=0 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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