- sr-users - kamailio.org

selective logging with debugger module
by Sharath Kumar 16 Feb '15

16 Feb '15

Hello, I followed the documentation from http://kamailio.org/docs/modules/4.2.x/modules/debugger.html#idp84752. I have the global debug flag at 9. modparam("debugger", "cfgtrace", 1) modparam("debugger", "mod_level_mode", 1) modparam("debugger", "mod_level", "core=3") My Kamailio complains with a parsing error of some sort and fails to start. - "dbg_mod_level_param(): cannot store parameter: core=3" My intention is to turn off/reduce the "core" module tracing while keeping the "script" module at debug. Any suggestions ? I am running version 4.1.4. thanks Sharath

1 0

Amazon VPS, long UDP packets are seen by sniffer, but don't reach application
by Andrey Utkin 16 Feb '15

16 Feb '15

We experience strange networking issue, not exactly specific to kamailio, but still related to it. Rtpengine's "ng" interface uses UDP. Protocol messages contains SDP, and for encrypted video call those messages exceed 1500 bytes. Everything works fine within localhost, but when rtpengine and Kamailio are on different hosts, and when hosts are Amazon-hosted, we have trouble. This is experienced with l3.large, t2.micro with Ubuntu 14. I believe we don't have any special settings over system defaults. We send a large datagram from remote host, e.g. with such trivial app in python: import socket UDP_IP = "123.123.123.123" # remote host IP UDP_PORT = 33333 MESSAGE = """ .....0010......0020......0030......0040......0050......0060......0070......0080......0090......0100 .....0110......0120......0130......0140......0150......0160......0170......0180......0190......0200 .....0210......0220......0230......0240......0250......0260......0270......0280......0290......0300 .....0310......0320......0330......0340......0350......0360......0370......0380......0390......0400 .....0410......0420......0430......0440......0450......0460......0470......0480......0490......0500 .....0510......0520......0530......0540......0550......0560......0570......0580......0590......0600 .....0610......0620......0630......0640......0650......0660......0670......0680......0690......0700 .....0710......0720......0730......0740......0750......0760......0770......0780......0790......0800 .....0810......0820......0830......0840......0850......0860......0870......0880......0890......0900 .....0910......0920......0930......0940......0950......0960......0970......0980......0990......1000 .....1010......1020......1030......1040......1050......1060......1070......1080......1090......1100 .....1110......1120......1130......1140......1150......1160......1170......1180......1190......1200 .....1210......1220......1230......1240......1250......1260......1270......1280......1290......1300 .....1310......1320......1330......1340......1350......1360......1370......1380......1390......1400 .....1410......1420......1430......1440......1450......1460......1470......1480......1490......1500 .....1510......1520......1530......1540......1550......1560......1570......1580......1590......1600 .....1610......1620......1630......1640......1650......1660......1670......1680......1690......1700 .....1710......1720......1730......1740......1750......1760......1770......1780......1790......1800 .....1810......1820......1830......1840......1850......1860......1870......1880......1890......1900 .....1910......1920......1930......1940......1950......1960......1970......1980......1990......2000""" print "UDP target IP:", UDP_IP print "UDP target port:", UDP_PORT print "message:", MESSAGE sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM) sock.sendto(MESSAGE, (UDP_IP, UDP_PORT)) Then we listen on that port with such trivial python app: import socket UDP_IP = "172.31.4.102" # local ip UDP_PORT = 33333 sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM) sock.bind((UDP_IP, UDP_PORT)) while True: data, addr = sock.recvfrom(0x10000) print "received message:", data Meanwhile, we monitor the traffic with e.g. ngrep: ngrep -t -e -d any -W byline -O large_udp.pcap port 33333 or '(ip[6:2]' '&' '0x1fff)' '!=' '0' (the part after "or" catches segments of segmented packets) About the host: # uname -a Linux hostname 3.16.0-30-generic #40~14.04.1-Ubuntu SMP Thu Jan 15 17:43:14 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux Also linux-image-3.13.0-36-generic and linux-image-3.13.0-45-generic behave in same way. What we see: - ngrep shows the packets with correct contents. All segments are delivered. - application doesn't get any data at all Rarely dmesg shows such messages: [ 102.161679] UDP: bad checksum. From 123.123.123.124:56439 to 172.31.4.102:33333 ulen 2008 but it is logged really rarely, so this is surely not what happens on every packet transmission. This test works fine on e.g. cheapest DigitalOcean VPS. I am concerned with this issue because rtpengine software has UDP interface. So on Amazon hosts this interface works only within localhost, and I cannot distribute software to different nodes. Any thoughts? What's wrong, how to fix? -- Andrey Utkin

2 4

Log Media IP
by Ryan Brindley 16 Feb '15

16 Feb '15

Hey community, What's the best way to pull out the media ip from the SIP INVITE body (for logging)? Ryan Brindley

2 1

Event reg - questions
by Stefan Ljung 16 Feb '15

16 Feb '15

I made a test configuration for trying the pua_regEvent module. There were two things I want to ask about. First - when the SUBSCRIBE to event 'reg' following a UEs registration to the Kamailio is responded with a NOTIFY - there was no body in the NOTIFY. I expected a XML body with registration status. Second - When the UE re-registers to Kamailio - I expected a NOTIFY to be send on the existing 'reg' subscription, but nothing happened. Or am I missing some configuration ? Here's some of the conf script: ####### Modules Section ######## loadmodule "db_mysql.so" 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 "xcap_client.so" loadmodule "pua.so" loadmodule "pua_reginfo.so" loadmodule "presence.so" loadmodule "presence_reginfo.so" loadmodule "presence_xml.so" # ----------------- setting module-specific parameters --------------- #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" # ----- xlog params ----- modparam("xlog", "prefix", "-xlog: ") # ----- 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) modparam("registrar", "max_expires", 86400) # ----- pua_reginfo params ----- modparam("pua_reginfo", "default_domain", "system.test") modparam("pua_reginfo", "server_address", "sip:reginfo@10.171.0.4") modparam("pua_reginfo", "publish_reginfo", 0) modparam("pua_reginfo", "default_domain", "system.test") # ----- presence_xml params ----- modparam("presence_xml", "integrated_xcap_server", 1) # ----- presence params ----- modparam("presence", "db_url", DBURL)

2 1

dialog module 4.2 REFER
by Uri Shacked 16 Feb '15

16 Feb '15

Hi, I am trying to use dlg_refer. I set the side to refer and the final destination. But, the contact header of the refer stays "controler(a)kamailio.org". So, the refer fails. Is it a bug or should i change the contact header by myself before doing the dlg_refer? same with the dlg_bridge..... Cheers, Uri

3 2

Using dialog_ng
by Shane Harrison 15 Feb '15

15 Feb '15

Hi all, I think the lack of response to my previous thread is probably because it was too complicated. I have now narrowed my problem down after further archive readings . I am trying to use the dialog_ng module along with the ims_charging_module. Upon reception of an INVITE, a call to the Ro_CCR() from the request_route script is made. That function does a check that the dialog record exists (binds to dialog module and does a "get_dlg(msg)" call) - this fails i.e. dialog is not found. Running kamailio 4.2.2 and relevant sections of kamailio.cfg are: modparam("dialog_ng", "dlg_flag", FLT_DIALOG) modparam("dialog_ng", "timeout_avp", "$avp(DLG_TIMEOUT_AVP)") modparam("dialog_ng", "detect_spirals", 1) modparam("dialog_ng", "profiles_no_value", "orig ; term") request_route { # NAT detection route(NATDETECT); # Set DLG flag to track dialogs using dialog2 if (!is_method("REGISTER|SUBSCRIBE")) setflag(FLT_DIALOG); # 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(); if (is_method("INVITE")) { set_dlg_profile("orig"); Ro_CCR("CHARGING_CCR_REPLY", "orig", "SCUR", "", "30","CCR_location"); } So my question is, "should the above logic create a dialog record on the initial INVITE?" and if yes then "what possible reasons might there be for the Ro_CCR() function not be able to detect it exists?" Cheers Shane Harrison

3 7

rtpengine - Error when sending message. Error: Invalid argument
by Amit Patkar 14 Feb '15

14 Feb '15

Hi I am getting error with rtpengine. Running Kamailio 4.2.3 I am trying to call from conventional SIP client to WebRTC client Google Chrome v38.0.2125.104 Firefox v33.0 Using sipML5 as WebRTC client root@rtcpbx:/home/avhan# /usr/sbin/rtpengine --interface=127.0.0.1\!192.168.2.161 --listen-ng=7722 --pidfile=/var/run/ngcp-rtpengine-daemon.pid --tos=184 --table=42 --foreground --log-level=7 --log-stderr INFO: Generating new DTLS certificate DEBUG: Dump of DTLS certificate: DEBUG: --- -----BEGIN CERTIFICATE----- DEBUG: --- MIIBpDCCAQ2gAwIBAAIJAIH/X+7BeV8gMA0GCSqGSIb3DQEBBQUAMBQxEjAQBgNV DEBUG: --- BAMTCXJ0cGVuZ2luZTAeFw0xNTAyMTMxMjM5MjJaFw0xNTAzMTYxMjM5MjJaMBQx DEBUG: --- EjAQBgNVBAMTCXJ0cGVuZ2luZTCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEA DEBUG: --- tfOduffYlJeuTeA6GRRKpeuUougYgu33QKjPZ9RuQNl6AIsmJQrqPJ9bTv93BlNd DEBUG: --- I/COq89SRxxjhAHEw6v3F283/lZAitySVa+TazbpZrwKsWbfPTXtLMC5oJvEPUr/ DEBUG: --- 8xcX+rnImLHZPrECMZj4pjSI6htclKok+YFf3incSmECAwEAATANBgkqhkiG9w0B DEBUG: --- AQUFAAOBgQBA2zHxUAv4J/dRLMboF9y3h6hQNnoEQNDwlmE+M4bBbOAxNYNxpmy3 DEBUG: --- mmZ54GBZFbvIWO4MVYrGiEMXBsQ3LSHo+MeExtY7Jq1RbXHU+DT/hHg5DGHHeFyr DEBUG: --- d46TUaHIFG/XhKSAaaCX1EbqtezMaUWrJw+mtOfzfZMEVzMP72CmiA== DEBUG: --- -----END CERTIFICATE----- DEBUG: Dump of DTLS private key: DEBUG: --- -----BEGIN PRIVATE KEY----- DEBUG: --- MIICdgIBADANBgkqhkiG9w0BAQEFAASCAmAwggJcAgEAAoGBALXznbn32JSXrk3g DEBUG: --- OhkUSqXrlKLoGILt90Coz2fUbkDZegCLJiUK6jyfW07/dwZTXSPwjqvPUkccY4QB DEBUG: --- xMOr9xdvN/5WQIrcklWvk2s26Wa8CrFm3z017SzAuaCbxD1K//MXF/q5yJix2T6x DEBUG: --- AjGY+KY0iOobXJSqJPmBX94p3EphAgMBAAECgYBJG9sLZz/Z7Nx3pBoUC6g25OKz DEBUG: --- VmjK6w0HKQ01XhDy869m2oJ4LduSgkFxC4rqusJY94aMGGCp/AkseH10VfuQ/Dn4 DEBUG: --- 6Zv9l8xGlprRmW0do6w0AR7F5JpcuoMT7ae8Jf7hAJpa//LbAg2s0lJBMofIUtd/ DEBUG: --- 0xs0y9uX/stV9hRAAQJBAOUocKdTbN1J9W6EB+r64/QBPxB7jhbs8J76RRfCjhE2 DEBUG: --- CS8JRBajMJJ3XgjOWQuh44xmydGCJeSSisDKsONVpgECQQDLQ6QCHYJxeXJ0R4w9 DEBUG: --- 5XqUEQhb03GlwoTIas3YQfJJyk/zKnQAkBKwjwatrNEldCGBOCOgeQhTERPwSxcx DEBUG: --- kGRhAkBxjmSkxjaAQhIZPdwemU00sRlrNY7n1Wo3nZV3fLWh2cEbSeAvx5ZePkCe DEBUG: --- 5IJDvWlhPd5sr63PYr7XYpndTagBAkEAj27Nvwhf0f8nOtJ6LftPAkj6owGUAqo+ DEBUG: --- 1MY5N09hz5xJyoe8j+EaryheSBPZChrmwa6BxW1i6Gr2rOvwE/OWQQJACxjVriL8 DEBUG: --- mKFqDLN1rQuDqyA6f0jSaxuPsA3TGCcEbgw4e8hQD9pFJ6NK7/X/UeqqWsW0sAZS DEBUG: --- mNrnLY9VvZPSpw== DEBUG: --- -----END PRIVATE KEY----- INFO: Startup complete, version 3.3.0.0+0~mr3.8.0.0 DEBUG: Adding a proxy for control ng stats:127.0.0.1 INFO: Got valid command from 127.0.0.1:60075: offer - { "sdp": "v=0 o=- 28314700 28314700 IN IP4 192.168.2.18 s=http://www.portsip.com c=IN IP4 192.168.2.18 t=0 0 m=audio 20670 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ", "ICE": "force", "flags": [ "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/SAVPF", "call-id": "FjDYirMD4b8ldLCQFrUMaw..", "received-from": [ "IP4", "192.168.2.18" ], "from-tag": "7a3b9012", "command": "offer" } NOTICE: [FjDYirMD4b8ldLCQFrUMaw..] Creating new call DEBUG: [FjDYirMD4b8ldLCQFrUMaw..] Opened ports 30000..30001 for media relay DEBUG: [FjDYirMD4b8ldLCQFrUMaw..] Opened ports 30010..30011 for media relay INFO: [FjDYirMD4b8ldLCQFrUMaw..] Returning to SIP proxy: d3:sdp669:v=0 o=- 28314700 28314700 IN IP4 192.168.2.161 s=http://www.portsip.com c=IN IP4 192.168.2.161 t=0 0 a=ice-lite m=audio 30000 RTP/SAVPF 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:30001 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:JBeKZmgOzhC3A5Ays2nm4Jx3GqSbsoVStGSjSFL/ a=setup:actpass a=fingerprint:sha-1 CE:47:48:60:04:D0:A5:D7:FD:00:89:6E:8F:D9:25:48:9D:2F:B9:89 a=ice-ufrag:bMtYwXGC a=ice-pwd:q053zJtRE8bxgCEu3eebdaqoWSmr a=candidate:EqkUFjVat8drCK70 1 UDP 2130706431 192.168.2.161 30000 typ host a=candidate:EqkUFjVat8drCK70 2 UDP 2130706430 192.168.2.161 30001 typ host 6:result2:oke INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] Successful STUN binding request from 192.168.2.18:7427 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] Successful STUN binding request from 192.168.2.18:7428 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] Successful STUN binding request from 192.168.2.18:7427 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] Successful STUN binding request from 192.168.2.18:7428 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] Successful STUN binding request from 192.168.2.18:7427 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] Successful STUN binding request from 192.168.2.18:7428 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] Successful STUN binding request from 192.168.2.18:7427 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] Successful STUN binding request from 192.168.2.18:7428 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] Successful STUN binding request from 192.168.2.18:7427 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] Successful STUN binding request from 192.168.2.18:7428 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] Successful STUN binding request from 192.168.2.18:7427 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30000] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] Successful STUN binding request from 192.168.2.18:7428 INFO: [FjDYirMD4b8ldLCQFrUMaw.. port 30001] STUN: using this candidate INFO: Got valid command from 127.0.0.1:42832: ping - { "command": "ping" } INFO: Returning to SIP proxy: d6:result4:ponge INFO: Got valid command from 127.0.0.1:35595: offer - { "sdp": "v=0 o=- 28373892 28373892 IN IP4 192.168.2.18 s=http://www.portsip.com c=IN IP4 192.168.2.18 t=0 0 m=audio 20674 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ", "ICE": "force", "flags": [ "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/SAVPF", "call-id": "fJTfptxXKrVoGdalt_2MMA..", "received-from": [ "IP4", "192.168.2.18" ], "from-tag": "bd78da3f", "command": "offer" } NOTICE: [fJTfptxXKrVoGdalt_2MMA..] Creating new call DEBUG: [fJTfptxXKrVoGdalt_2MMA..] Opened ports 30022..30023 for media relay DEBUG: [fJTfptxXKrVoGdalt_2MMA..] Opened ports 30044..30045 for media relay INFO: [fJTfptxXKrVoGdalt_2MMA..] Returning to SIP proxy: d3:sdp669:v=0 o=- 28373892 28373892 IN IP4 192.168.2.161 s=http://www.portsip.com c=IN IP4 192.168.2.161 t=0 0 a=ice-lite m=audio 30022 RTP/SAVPF 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:30023 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:YUVDWqqE+RNOPLu4d1J81/gee3OY8tnfGSjc2zjB a=setup:actpass a=fingerprint:sha-1 CE:47:48:60:04:D0:A5:D7:FD:00:89:6E:8F:D9:25:48:9D:2F:B9:89 a=ice-ufrag:1d16gzNH a=ice-pwd:qpNBX4eBwfNKq0TPpNHLeuMdHOjX a=candidate:EqkUFjVat8drCK70 1 UDP 2130706431 192.168.2.161 30022 typ host a=candidate:EqkUFjVat8drCK70 2 UDP 2130706430 192.168.2.161 30023 typ host 6:result2:oke INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: [FjDYirMD4b8ldLCQFrUMaw..] Closing call due to timeout INFO: [FjDYirMD4b8ldLCQFrUMaw..] Final packet stats: INFO: [FjDYirMD4b8ldLCQFrUMaw..] --- Tag '', created 1:06 ago, in dialogue with '(none)' INFO: [FjDYirMD4b8ldLCQFrUMaw..] ------ Media #1, port 30000 <> [::]:0 , 6 p, 600 b, 0 e, 1423917572 last_packet INFO: [FjDYirMD4b8ldLCQFrUMaw..] ------ Media #1, port 30001 <> [::]:0 (RTCP), 6 p, 600 b, 0 e, 1423917572 last_packet INFO: [FjDYirMD4b8ldLCQFrUMaw..] --- Tag '7a3b9012', created 1:06 ago, in dialogue with '' INFO: [FjDYirMD4b8ldLCQFrUMaw..] ------ Media #1, port 30010 <> 192.168.2.18:20670, 0 p, 0 b, 0 e, 1423917566 last_packet INFO: [FjDYirMD4b8ldLCQFrUMaw..] ------ Media #1, port 30011 <> 192.168.2.18:20671 (RTCP), 0 p, 0 b, 0 e, 1423917566 last_packet INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] Successful STUN binding request from 192.168.2.18:7449 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30022] STUN: using this candidate INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] Successful STUN binding request from 192.168.2.18:7450 INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30023] STUN: using this candidate INFO: Got valid command from 127.0.0.1:36029: answer - { "sdp": "v=0 o=Mozilla-SIPUA-33.0 15128 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:524a14d8 a=ice-pwd:f23ceb82b6f9a86a2c1f180a013a9ff3 a=fingerprint:sha-256 6A:5D:49:F6:7B:FE:5B:5E:23:43:EE:31:43:20:46:C0:E8:12:B7:AD:AB:A2:09:66:94:55:42:E2:6B:D3:D9:63 m=audio 7449 UDP/TLS/RTP/SAVPF 0 101 c=IN IP4 192.168.2.18 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=setup:active a=candidate:0 1 UDP 2128609535 192.168.2.18 7449 typ host a=candidate:0 2 UDP 2128609534 192.168.2.18 7450 typ host ", "ICE": "remove", "flags": [ "trust-address" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "fJTfptxXKrVoGdalt_2MMA..", "received-from": [ "IP4", "192.168.2.18" ], "from-tag": "bd78da3f", "to-tag": "mUTC7w3qs5ka6fJWKCiM", "command": "answer" } INFO: [fJTfptxXKrVoGdalt_2MMA..] Returning to SIP proxy: d3:sdp240:v=0 o=Mozilla-SIPUA-33.0 15128 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 m=audio 30044 RTP/AVP 0 101 c=IN IP4 192.168.2.161 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:30045 6:result2:oke *DEBUG: [fJTfptxXKrVoGdalt_2MMA.. port 30044] Error when sending message. Error: Invalid argument WARNING: [fJTfptxXKrVoGdalt_2MMA.. port 30044] Write error on RTP socket: Invalid argument* INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] Final packet stats: INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] --- Tag 'mUTC7w3qs5ka6fJWKCiM', created 0:13 ago, in dialogue with 'bd78da3f' INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] ------ Media #1, port 30022 <> 192.168.2.18:7449 , 7 p, 700 b, 0 e, 1423917638 last_packet INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] ------ Media #1, port 30023 <> 192.168.2.18:7450 (RTCP), 7 p, 700 b, 0 e, 1423917638 last_packet INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] --- Tag 'bd78da3f', created 0:13 ago, in dialogue with 'mUTC7w3qs5ka6fJWKCiM' INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] ------ Media #1, port 30044 <> 192.168.2.18:20674, 0 p, 0 b, 1 e, 1423917626 last_packet INFO: [fJTfptxXKrVoGdalt_2MMA.. port 30044] ------ Media #1, port 30045 <> 192.168.2.18:20675 (RTCP), 0 p, 0 b, 0 e, 1423917626 last_packet Regards Amit

2 2

Kamailio and rtpengine
by Marc Soda 14 Feb '15

14 Feb '15

How does Kamailio load balance traffic to rtpengine? Is it load based, round robin, etc? The module makes mention of this but I don't see how it works. Also, it talks about weighting the proxies. How is that accomplished? Thanks, Marc

3 2

WebRTC to PSTN call, proxied through Kamailio
by Rahul MathuR 14 Feb '15

14 Feb '15

Hello gents, I was trying my hands on getting a successful RTCweb call (JSsip, since Peter Dunkley mentioned that he's been using JSsip for most of the testing scenarios..) to PSTN, making my kamailio as proxy + protocol converter (sip over web-sockets to sip over udp). And yes, I've referred Carlos' config; the main problem is I get 'Bad Media Description' error in Google Chromium (Version 40.0.2214.111 m) & my SIP server even sends 200 OK, but my phone doesn't ring. To make it worse, I can see rtpengine throwing this error - "SRTCP output wanted, but no crypto suite was negotiated" BTW, I have - [root@localhost log]# openssl version OpenSSL 1.0.1j 15 Oct 2014 I even tried building kamailio & rtpengine using this openssl but in-vain. One thing that baffles me is that, apparently kamailio has started receiving RTP packets (perhaps early media) but the mobile phone hasn't ringed :-( I am attaching all possible logs & seek some guidance from the array of experts in this list. Files attached: a) tcpdump on ext. interface b) tcpdump on loopback c) syslogs d) Chromium JS logs UAC (14.98.55.38), Kamailio (125.99.186.126), SIP Server (157.238.178.153), Media Server (199.27.244.6) -- Warm Regds. MathuRahul

3 6

Issues after Ubuntu Server Update and WebRTC
by Manuel Camargo Lominchar 13 Feb '15

13 Feb '15

>* Is the SIP signaling still going fine via websocket? Do you get callee *>* ringing? *>>* Cheers, *>* Daniel* Hello Daniel Actually SIP signalling goes perfect, and i can see in RTP asterisk log that RTP packets are being sent, so I suppose there is something stuck somewhere in RTP maybe related with RTP Engine >Hi, > >If you are using rtpengine for the rtp you might have to upgrade to a >newer version. > >On my system crypto negotiation in rtpengine started failing after a >openssl update in January. Recompiling a newer version from git fixed that. > >regards > >M I updated to latest version through git but still stuck there. Now I can see that the config file (ngcp-rtpengine in etc default) has changed so I'm not sure how should this impact (not ADDRESS neither ADV ADDRESS now I see an Interfaces parameter) I also find for some reason very possible that somehow rtp traffic is not getting through. ICE candidates are being negotiated, so I feel that the route is clear, so maybe there might be some crypto issues as you point out, but update has not been enough :( Any other ideas? *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com <https://twitter.com/SirLouen> [image: Ver el perfil de Manuel Camargo Lominchar en LinkedIn] <http://es.linkedin.com/in/louen>

1 1

change_reply_status() 415 -> 200 change
by Juha Heinanen 13 Feb '15

13 Feb '15

i tried to use change_reply_status() to change 415 to 200, but got: ERROR: textopsx [textopsx.c:283]: change_reply_status_f(): ERROR: textops: the class of provisional or positive final replies cannot be changed textopsx/README does not mention anything about what can or cannot be changed. why is it that the above change cannot be made? -- juha

2 1

Issues after Ubuntu Server Update and WebRTC
by Manuel Camargo Lominchar 13 Feb '15

13 Feb '15

This might be a weird question I've been operating for some months with my kam + asterisk + webrtc sipml5 based system inside an Ubuntu Server Today I had the idea to apt-get upgrade my system and... now the whole system webrtc communication service is dead Here is the VAST list of packages updated... But I cannot figure out who is the "destroyer" package. No prompts appeared during the update so I believe no config files have been updated I'm trying to debug but I cant find a solution The handshake is done, and in Wireshark the only thing I find is that no UDP (RTP) packages are being processed, but in the SIP log I can see ICE candidates and all process is done I tried to do a direct SIP connection without WebRTC, and it worked pretty fine, so this have something to do with maybe crypt libs, or something to do with kamailio and imcompatibilities with one of this packages. Kamailio Version: # kamailio -v version: kamailio 4.1.5 (x86_64/linux) 2b3d4b 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: 2b3d4b compiled on 14:39:42 Aug 28 2014 with gcc 4.8.2 Any ideas of where to start digging? Start-Date: 2015-02-12 14:40:19 Install: linux-image-extra-3.13.0-45-generic:amd64 (3.13.0-45.74, automatic), linux-image-3.13.0-45-generic:amd64 (3.13.0-45.74, automatic), linux-headers-3.13.0-45-generic:amd64 (3.13.0-45.74, automatic), linux-headers-3.13.0-45:amd64 (3.13.0-45.74, automatic) Upgrade: bind9-host:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), python3-problem-report:amd64 (2.14.1-0ubuntu3.5, 2.14.1-0ubuntu3.6), mime-support:amd64 (3.54ubuntu1, 3.54ubuntu1.1), liblwres90:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), apt:amd64 (1.0.1ubuntu2.5, 1.0.1ubuntu2.6), libevent-2.0-5:amd64 (2.0.21-stable-1ubuntu1, 2.0.21-stable-1ubuntu1.14.04.1), libsystemd-login0:amd64 (204-5ubuntu20.7, 204-5ubuntu20.9), linux-headers-generic:amd64 (3.13.0.37.44, 3.13.0.45.52), mysql-server-core-5.5:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), ppp:amd64 (2.4.5-5.1ubuntu2, 2.4.5-5.1ubuntu2.1), mysql-server-5.5:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), multiarch-support:amd64 (2.19-0ubuntu6.3, 2.19-0ubuntu6.5), libdns100:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), krb5-multidev:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), mysql-client-core-5.5:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), libisccfg90:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), systemd-services:amd64 (204-5ubuntu20.7, 204-5ubuntu20.9), libssl1.0.0:amd64 (1.0.1f-1ubuntu2.7, 1.0.1f-1ubuntu2.8), apt-transport-https:amd64 (1.0.1ubuntu2.5, 1.0.1ubuntu2.6), procps:amd64 (3.3.9-1ubuntu2, 3.3.9-1ubuntu2.2), libbind9-90:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), tcpdump:amd64 (4.5.1-2ubuntu1, 4.5.1-2ubuntu1.1), dbus:amd64 (1.6.18-0ubuntu4.2, 1.6.18-0ubuntu4.3), libgssapi-krb5-2:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libapache2-mod-php5:amd64 (5.5.9+dfsg-1ubuntu4.4, 5.5.9+dfsg-1ubuntu4.5), php5-mysql:amd64 (5.5.9+dfsg-1ubuntu4.4, 5.5.9+dfsg-1ubuntu4.5), python-urllib3:amd64 (1.7.1-1build1, 1.7.1-1ubuntu0.1), dkms:amd64 (2.2.0.3-1.1ubuntu5, 2.2.0.3-1.1ubuntu5.14.04), php5-common:amd64 (5.5.9+dfsg-1ubuntu4.4, 5.5.9+dfsg-1ubuntu4.5), coreutils:amd64 (8.21-1ubuntu5, 8.21-1ubuntu5.1), libkrb5-3:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), apt-utils:amd64 (1.0.1ubuntu2.5, 1.0.1ubuntu2.6), cpio:amd64 (2.11+dfsg-1ubuntu1, 2.11+dfsg-1ubuntu1.1), plymouth-theme-ubuntu-text:amd64 (0.8.8-0ubuntu17, 0.8.8-0ubuntu17.1), python3-software-properties:amd64 (0.92.37.1, 0.92.37.3), libkadm5clnt-mit9:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), mount:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), libsystemd-daemon0:amd64 (204-5ubuntu20.7, 204-5ubuntu20.9), python-apt:amd64 (0.9.3.5, 0.9.3.5ubuntu1), libc-dev-bin:amd64 (2.19-0ubuntu6.3, 2.19-0ubuntu6.5), libcgmanager0:amd64 (0.24-0ubuntu7, 0.24-0ubuntu7.2), file:amd64 (5.14-2ubuntu3.2, 5.14-2ubuntu3.3), libapparmor1:amd64 (2.8.95~2430-0ubuntu5, 2.8.95~2430-0ubuntu5.1), libkrb5-dev:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libmagic1:amd64 (5.14-2ubuntu3.2, 5.14-2ubuntu3.3), libpam-systemd:amd64 (204-5ubuntu20.7, 204-5ubuntu20.9), libc-bin:amd64 (2.19-0ubuntu6.3, 2.19-0ubuntu6.5), libc6:amd64 (2.19-0ubuntu6.3, 2.19-0ubuntu6.5), libglib2.0-0:amd64 (2.40.0-2, 2.40.2-0ubuntu1), php5-readline:amd64 (5.5.9+dfsg-1ubuntu4.4, 5.5.9+dfsg-1ubuntu4.5), libprocps3:amd64 (3.3.9-1ubuntu2, 3.3.9-1ubuntu2.2), libsepol1:amd64 (2.2-1, 2.2-1ubuntu0.1), libglib2.0-dev:amd64 (2.40.0-2, 2.40.2-0ubuntu1), libapt-inst1.5:amd64 (1.0.1ubuntu2.5, 1.0.1ubuntu2.6), libuuid1:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), libmount1:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), dnsutils:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), libdbus-1-3:amd64 (1.6.18-0ubuntu4.2, 1.6.18-0ubuntu4.3), udev:amd64 (204-5ubuntu20.7, 204-5ubuntu20.9), curl:amd64 (7.35.0-1ubuntu2.1, 7.35.0-1ubuntu2.3), update-notifier-common:amd64 (0.154.1, 0.154.1ubuntu1), plymouth:amd64 (0.8.8-0ubuntu17, 0.8.8-0ubuntu17.1), libglib2.0-data:amd64 (2.40.0-2, 2.40.2-0ubuntu1), mysql-client-5.5:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), krb5-locales:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libxml2:amd64 (2.9.1+dfsg1-3ubuntu4.3, 2.9.1+dfsg1-3ubuntu4.4), ntp:amd64 (4.2.6.p5+dfsg-3ubuntu2, 4.2.6.p5+dfsg-3ubuntu2.14.04.2), libudev1:amd64 (204-5ubuntu20.7, 204-5ubuntu20.9), binutils:amd64 (2.24-5ubuntu3, 2.24-5ubuntu3.1), bsdutils:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), libxml2-dev:amd64 (2.9.1+dfsg1-3ubuntu4.3, 2.9.1+dfsg1-3ubuntu4.4), libapparmor-perl:amd64 (2.8.95~2430-0ubuntu5, 2.8.95~2430-0ubuntu5.1), lshw:amd64 (02.16-2ubuntu1, 02.16-2ubuntu1.2), libssl-dev:amd64 (1.0.1f-1ubuntu2.7, 1.0.1f-1ubuntu2.8), wget:amd64 (1.15-1ubuntu1, 1.15-1ubuntu1.14.04.1), libapt-pkg4.12:amd64 (1.0.1ubuntu2.5, 1.0.1ubuntu2.6), uuid-runtime:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), python3-apt:amd64 (0.9.3.5, 0.9.3.5ubuntu1), mysql-common:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), php5-cli:amd64 (5.5.9+dfsg-1ubuntu4.4, 5.5.9+dfsg-1ubuntu4.5), libmysqlclient18:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), git-man:amd64 (1.9.1-1, 1.9.1-1ubuntu0.1), libplymouth2:amd64 (0.8.8-0ubuntu17, 0.8.8-0ubuntu17.1), libmysqlclient-dev:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), git:amd64 (1.9.1-1, 1.9.1-1ubuntu0.1), python3-distupgrade:amd64 (0.220.5, 0.220.7), libcurl3:amd64 (7.35.0-1ubuntu2.1, 7.35.0-1ubuntu2.3), apport:amd64 (2.14.1-0ubuntu3.5, 2.14.1-0ubuntu3.6), linux-firmware:amd64 (1.127.7, 1.127.11), libssl-doc:amd64 (1.0.1f-1ubuntu2.7, 1.0.1f-1ubuntu2.8), libcurl4-gnutls-dev:amd64 (7.35.0-1ubuntu2.1, 7.35.0-1ubuntu2.3), landscape-common:amd64 (14.01-0ubuntu3, 14.12-0ubuntu0.14.04), libblkid1:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), python3-apport:amd64 (2.14.1-0ubuntu3.5, 2.14.1-0ubuntu3.6), apparmor:amd64 (2.8.95~2430-0ubuntu5, 2.8.95~2430-0ubuntu5.1), tzdata:amd64 (2014e-0ubuntu0.14.04, 2015a-0ubuntu0.14.04), mysql-server:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), software-properties-common:amd64 (0.92.37.1, 0.92.37.3), openssl:amd64 (1.0.1f-1ubuntu2.7, 1.0.1f-1ubuntu2.8), libelf1:amd64 (0.158-0ubuntu5.1, 0.158-0ubuntu5.2), libglib2.0-bin:amd64 (2.40.0-2, 2.40.2-0ubuntu1), libkdb5-7:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), ubuntu-release-upgrader-core:amd64 (0.220.5, 0.220.7), linux-libc-dev:amd64 (3.13.0-37.64, 3.13.0-45.74), util-linux:amd64 (2.20.1-5.1ubuntu20.2, 2.20.1-5.1ubuntu20.3), python-apt-common:amd64 (0.9.3.5, 0.9.3.5ubuntu1), php5-gd:amd64 (5.5.9+dfsg-1ubuntu4.4, 5.5.9+dfsg-1ubuntu4.5), libgssrpc4:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libdrm2:amd64 (2.4.52-1, 2.4.56-1~ubuntu1), language-pack-en:amd64 (14.04+20140707, 14.04+20141110), libmysqld-dev:amd64 (5.5.40-0ubuntu0.14.04.1, 5.5.41-0ubuntu0.14.04.1), linux-image-generic:amd64 (3.13.0.37.44, 3.13.0.45.52), libkadm5srv-mit9:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libkrb5support0:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libk5crypto3:amd64 (1.12+dfsg-2ubuntu4.2, 1.12+dfsg-2ubuntu5.1), libisccc90:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), dh-apparmor:amd64 (2.8.95~2430-0ubuntu5, 2.8.95~2430-0ubuntu5.1), libc6-dev:amd64 (2.19-0ubuntu6.3, 2.19-0ubuntu6.5), libisc95:amd64 (9.9.5.dfsg-3, 9.9.5.dfsg-3ubuntu0.1), libcurl3-gnutls:amd64 (7.35.0-1ubuntu2.1, 7.35.0-1ubuntu2.3), rsyslog:amd64 (7.4.4-1ubuntu2.3, 7.4.4-1ubuntu2.5), ntpdate:amd64 (4.2.6.p5+dfsg-3ubuntu2, 4.2.6.p5+dfsg-3ubuntu2.14.04.2), linux-generic:amd64 (3.13.0.37.44, 3.13.0.45.52) End-Date: 2015-02-12 14:43:37 Kind regards, *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com <https://twitter.com/SirLouen> [image: Ver el perfil de Manuel Camargo Lominchar en LinkedIn] <http://es.linkedin.com/in/louen>

4 3

Unable to route outbound with multiple VRRP IP's
by mark＠brightvoip.co.uk 13 Feb '15

13 Feb '15

Hi, I am trying to install a Kamailio server in a HA configuration, with IP's for 3 attached networks. I am using Keepalived/VRRP to manage the VIP's. I have tested v 4.1.6 and latest 4.2.2 I have 3 x listen entries, one for each network and mhomed=1. Also, for info, we have net.ipv4.ip_nonlocal_bind = 1 in sysctl.conf listen=udp:my.public.net.46:5060 listen=udp:192.168.106.46:5060 listen=udp:192.168.116.46:5060 mhomed=1 I have dispatcher set up to load balance over two seperate groups of Asterisk servers, one on each private network. Dispatcher does probing to know which Asterisks are 'alive'. When I ran the system, Kamailio/Dispatcher was not able to send any probing OPTIONS to any Asterisk (normal PINGS work fine in all networks in/out of Asterisk/Kamailio) Looking at the log, I see these entries for all 3 networks: Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: <core> [forward.c:218]: get_out_socket(): no socket found Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: <core> [forward.c:220]: get_out_socket(): no corresponding socket found for(u dp:192.168.116.38:5060) Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: tm [ut.h:345]: uri2dst2(): no corresponding socket found for "192.168.116.38" af 2 (udp:192.168.116.38:5060) Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: tm [uac.c:307]: t_uac_prepare(): t_uac: no socket found Feb 12 17:28:19 app-srv-dev-1-01 /usr/local/sbin/kamailio[2992]: ERROR: dispatcher [dispatch.c:2564]: ds_check_timer(): unable to ping [sip:192.168.1 16.38:5060] I notice from other logging that Kamailio is receiving and processing incoming OPTIONS messages, and sending appropriate replies. Now, this might not be a Kamailio issue, as when I run the same config with 3 IP's, but NOT using VRRP/Keepalived/Aliased IP's, everything works normally. However, I do need to run this setup in HA, so would welcome any suggestions as to how I might resolve this issue. Kind regards, Mark Hall --- This email has been checked for viruses by Avast antivirus software. http://www.avast.com

2 2

Possible memory leak dealing with presence in kamailio
by Nuno Reis 12 Feb '15

12 Feb '15

Hello all. I'm consistently watching a memory increase in kamailio when dealing with PRESENCE events, namely SIP PUBLISH events. The system eventually hangs running out of memory. This behavior is seen at least in kamailio 4.1 and 4.2. I'm currently using the latest stable 4.2.2. If I disable the SIP PUBLISH handling in kamailio i don't observe the issue anymore but as a side effect I don't have presence (name BLFs) also. What do you think can be the right approach here? Should I open an issue in github for this? Should I run kamailio under valgrind for some time? Are there any other possible debug hints here? Please find attached a code snippet with the presence related parts I'm using right now. Looking forward to hear from you. Best Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

4 13

Roadmap to next major release 4.3.0
by Daniel-Constantin Mierla 12 Feb '15

12 Feb '15

Hello, being discussed yesterday during the IRC devel meeting, the proposed planning for next major release (v4.3.0) is: - freeze the development sometime in April - after mid of March decide the exact date when the development is going to be frozen - testing during last part of April and May - based on testing results, release in the time frame: end of May till mid of June >From git activity, there are already several new modules and plenty of new features -- don't forget to add them to: - http://www.kamailio.org/wiki/features/new-in-devel Also, any detail that is relevant for upgrading has to added to: - https://www.kamailio.org/wiki/install/upgrade/stable-to-devel 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

Using MariaDB with sqlops sql_query_async
by davy van de moere 12 Feb '15

12 Feb '15

Does anyone know if the async queries of kamailio work with mariadb? I'm on kamailio 4.2.3, and I can't get the async way of querying to work. I don't get any errors, but I don't see the queries to be executed. the relevant part of my config is: loadmodule "db_mysql.so" loadmodule "sqlops.so" modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost /kamailio") sql_query_async("ca", "insert into cdr_events (callid) values ('$ci') "); When just replace sql_query_async with sql_query, I do see inserts passing by. Do I just bluntly miss something? Or does someone have some good pointers? Thx! Davy

2 3

long l_uuid in uac_reg
by kovarik＠odorik.cz 11 Feb '15

11 Feb '15

Hello, rpc funcions: reg_info,reg_enable,reg_disable not work if l_uuid is number and size > than 10, for 1234567890 is O.K. and for 12345678901 try to find 18446744073170328629. kamcmd uac.reg_enable l_uuid 1234567890 -- found kamcmd uac.reg_enable l_uuid 12345678901 -- not found and if number start with zero then zero is cut. kamcmd uac.reg_enable l_uuid 0034567890 -- 34567890 I add debug to uac_reg.c 1615 LM_ERR("START %.*s,%.*s\n", attr.len,attr.s,val.len, val.s); Thaks

2 1

Trouble with DB APIs
by Davidson, Erik 11 Feb '15

11 Feb '15

Hey Everyone! I'm currently working on a database module and I've come across some issues/confusion that I was hoping someone could clear up for me. I've been looking at the existing modules and the docs online for help but I can't find a definitive answer for them. 1. I see there are 2 DB APIs (srdb1 and srdb2) and I gather that srdb2 is the newer version? Should I aim to use srdb2 exclusively or should I also use srdb1? 2. It looks like srdb2 provides a db_uri command that can be used to implement custom URI parsing but I can't figure out how to hook into it. I added it to my exported commands but it was never called. When I looked through the code I saw the only place that ever really uses it is db_ctx, which itself looks to be used in the child_init of modules. I'm developing a database module that I want to be a drop-in replacement for others, but it needs multiple hosts specified in the connection uri (similar to the URI example here http://docs.mongodb.org/manual/reference/connection-string/) and I can't figure out the best place to do it. Should I be doing it in db_func_t.init because it has the full string version of the uri and parse it there? Do I then have to reimplement everything that db_do_init would be doing? 3. Do I need to implement multiple database interfaces to get full module coverage in Kamailio? I was looking at db_berkeley as a reference and it looks like it has the KAMAILIO_MOD_INTERFACE as well as the SER_MOD_INTERFACE implemented. 4. Is there a preferred way to handle the API binding (such as db_ commands vs db_bind_api)? Thanks for any help or direction you can give me! - Erik Davidson ----------------------------------------- CONFIDENTIALITY NOTICE: The information contained in this e-mail message and any attachments is for the sole use of the intended recipients and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient and have received this communication in error, please contact the sender by reply e-mail and destroy all copies of the original message. Thank you.

2 1

Next IRC development meeting
by Daniel-Constantin Mierla 11 Feb '15

11 Feb '15

Hello, I thought to propose organizing the next IRC meeting to discuss and sync each other about Kamailio development and surrounding ecosystem. First date proposal is next week, Wednesday, 15:00GMT, Feb 11, 2015, on the usual #kamailio channel at freenode.net. The date can be changed, based on feedback from the people that want to attends. I made a wiki page for the event at: - http://www.kamailio.org/wiki/devel/irc-meetings/2015a Feel free to propose topics to the agenda and/or new date. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 4

[Kamailio 4.0.6] sometimes route specified by RTIMER module doesn't be executed just after finishing starting Kamailio server
by jack wang 11 Feb '15

11 Feb '15

Hello everyone, I got a little trouble on using the RTIMER module. Sometimes the route block I set using the "exec" parameter in RTIMER module doesn't be executed, this will happen from Kamailio server being started, and only work (not always) after restarting server. I have no idea why this happen, does anyone ever met this? Which part should I notice? Thx :)

1 0

Presence module for BLF issue
by SamyGo 11 Feb '15

11 Feb '15

Hi community, I'm dealing with a problem here related to presence module handling BLF. My BLF phones are Yealink T28p with latest firmware and work perfectly with Asterisk and FreeSwitch but not with Kamailio as expected. What I'm observing here is a malfunctioning statuses due to accumulation of multiple entries of just one status in "presentitiy" table. These entries neither get expired not cleared from DB hence I see atleast 21 XML tags combined in the NOTIFY sent to Phones. Here is what I've figured out a manual way to make the BLF work fine again. *Step -1* Make call between two endpoints. see entries in presentitiy table showing up. *Step -2* NOTIFYs sent to phones subscribing those users from Kamailio and phones show correct flashing red BLF lights *Step -3* clear the entries from presentity table manually i.e mysql> truncate presentity; *Step -4* Hangup the phones, presentity table shows few more entries with state='terminated' NOTIFY sent again to subscribers.. BLF lighst turn green happily. Here are my module params: modparam("presence", "db_url", DBURL) modparam("presence", "notifier_processes", 1) modparam("presence", "server_address", "sip:1.2.5.5" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 1) modparam("presence", "clean_period", 4) modparam("presence", "subs_db_mode", 2) modparam("presence", "max_expires", 36) modparam("presence", "expires_offset", 10) modparam("presence", "subs_htable_size", 12) modparam("presence", "pres_htable_size", 12) modparam("presence", "fetch_rows", 1000) (these params were set as default earlier but I changed them as above hoping they'll help) Kindly share your expert opinion. Best Regards, Sammy

2 10

Issues while using Kamailio as proxy with Asterisk
by Juned Khan 10 Feb '15

10 Feb '15

Hi All, I am using Kamailio as proxy server with Asterisk but having some issues with Inbound and outbound calls. Asterisk is a registrar, Kamailio is a just for proxy. Using softphone call are working fine but using hardphone, its creating issues. Here are the test cases. No Test case *98 Inbound 1 ATA registered with Proxy Domain No No 2 ATA registered with Proxy IP Yes Yes 3 ATA registered with Asterisk domain No No 4 ATA registered with Asterisk IP Yes Yes 5 Soft phone registered with Proxy Domain Yes Yes 6 Soft phone registered with Proxy IP Yes Yes 7 Soft phone registered with Asterisk domain Yes Yes 8 Soft phone registered with Asterisk IP Yes Yes Here I am dialling *98 with the mentioned scenario, if it goes to voicemail options then Yes, else No I am using ALLO ATA. Here is my Kamailio.cfg http://pastie.org/9887806 Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

2 1

Kamailio v4.2.3 Released
by Daniel-Constantin Mierla 10 Feb '15

10 Feb '15

Hello, Kamailio SIP Server v4.2.3 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.2. 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 v4.2.3. For more details about version 4.2.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/02/kamailio-v4-2-3-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

Re: [SR-Users] Wrong location entries when using usrloc with Mongo
by Mickael Marrache 10 Feb '15

10 Feb '15

I forgot to precise that I allow only one contact per AOR. modparam("registrar", "max_contacts", 1) save("location", "0x04") From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Friday, January 30, 2015 10:51 AM To: sr-users(a)lists.sip-router.org Subject: Wrong location entries when using usrloc with Mongo Hi, I start with no location nor in Mongo, nor in memory. My UA registers successfully and I can see the location entry in Mongo. Then, I close my UA which unregisters (by setting Expires header to 0). Then, I open the app again and a new registration is made. The entry after first registration. Looks okay. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "username" : "A", "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" : ISODate("2015-01-30T08:55:34Z"), "q" : -1, "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8809, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip:XXXXXXXXXXX:54217;transport=tcp", "path" : "<sip:XXXXXXXXXX;lr;received=sip:XXXXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp:XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:55:34Z"), "ruid" : "uloc-54cb38df-3378-2", "instance" : null, "reg_id" : 0 } The same entry after un register (Expires 0). Note that the username field is missing. In any case, I expected the entry to be deleted. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" : ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0, "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" } The entries after second registration. The new entry looks okay. But, the old entry is still here. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" : ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0, "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" } { "_id" : ObjectId("54cb3a7884e581337a25b895"), "username" : "A", "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" : ISODate("2015-01-30T09:02:00Z"), "q" : -1, "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8813, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T08:02:00Z"), "ruid" : "uloc-54cb38df-337a-1", "instance" : null, "reg_id" : 0 } The issue is the entry is not deleted after un registering. I precise that registrations are dispatched over multiple registrars with all accesses the same Mongo cluster. Thanks, Mickael

2 7

Kamailio appends firs inserted branch twice
by Yuriy Gorlichenko 10 Feb '15

10 Feb '15

Hello I use this version of kamailio kamailio -v version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7 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: 8cdbe7 compiled on 17:54:50 Jan 20 2015 with gcc 4.8.2 I hav an issue with append branches to branch route when I need fork call to one endpoint woth different destionations. I use my own algorithm for call to this devices because with lookup("location") I can not use RTPENGINE for different types of endpoints (web endoints and standart UDP endpoints) My alg is here: { sql_query("ca", "select contact from location where username='$tU'", "ra"); xlog("L_INFO","rows: $dbr(ra=>rows) cols: $dbr(ra=>cols)\n"); if($dbr(ra=>rows)>0){ $var(i)=0; while($var(i)<$dbr(ra=>rows)){ xlog("L_INFO","SQL query return contact {$dbr(ra=>[$var(i),0])} for {$tU} at step {$var(i)}\n"); if ($dbr(ra=>[$var(i),0])=~"transport=ws"){ xlog("L_INFO", "This is a Websocket call to endpoint"); sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); $du=$var(recieved); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}. Destination is {$du}\n"); append_branch("sip:$tU@$(du{s.select,1,:})"); } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $var(recieved)=''; sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO", "SQL query return RECIEVED {$var(recieved)}"); if ($var(recieved)==0){ xlog("L_INFO", "Recieved string is EMPTY"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); } else { xlog("L_INFO", "Recieved string is {$var(recieved)}"); $du=$var(recieved); } $var(UDP_contact)="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); append_branch("sip:$tU@$(du{s.select,1,:})"); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}}. Destination is {$du}\n"); } $var(i) = $var(i) + 1; } } else{ exit; } t_on_branch("1"); return; } } } branch_route[1]{ if($du=~"transport=ws"){ xlog("L_INFO","Websocket Branch is {$du} for {$tU}\n"); rtpengine_manage("internal extenal force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else{ xlog("L_INFO","UDP Branch is {$du)} for {$tU}\n"); rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } So as you see I choose array of devices and then set its to branches. At the kamailio console I see output This is output of alg that above. I have 3 devices: 2 websocket devices and 1 udp But when you look at dump where rtpengine make changes with packets you can see that it runs for 4 devices and 1 and 4 devices is the same. You can see start of every branch from words of log "Websocket Branch is" or "UDP Branch is" So this bug gives a mistake because some of hardphones can not handlie double INVITE and kamilio response for this devices cancel with 200 cause because hardphone set 482 reply to kamialio. (you can see it reply at the log) Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: rows: 3 cols: 1 Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: SQL query return contact {sip:5ue13vsh@2t4iielj109h.invalid;transport=ws} for {123455678} at step {0} Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: This is a Websocket call to endpoint Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: SQL query return recieved {sip:85.6.14.8:4328;transport=WS} for {123455678}. Destination is {sip:85.6.14.8:4328;transport=WS} Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: SQL query return contact {sip:a7uo9vsq@7c0oo5kvc71e.invalid;transport=ws} for {123455678} at step {1} Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: This is a Websocket call to endpoint Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: SQL query return recieved {sip:85.2.10.3:58509;transport=WS} for {123455678}. Destination is {sip:85.2.10.3:58509;transport=WS} Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: SQL query return contact {sip:upsjchv2@5c88tisd29d4.invalid;transport=ws} for {123455678} at step {2} Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: This is a Websocket call to endpoint Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: SQL query return recieved {sip:18.9.18.10:56917;transport=WS} for {123455678}. Destination is {sip:18.9.18.10:56917;transport=WS} Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: Websocket Branch is {sip:18.9.18.10:56917;transport=WS} for {123455678} Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Got valid command from 127.0.0.1:52689: offer - { "sdp": "v=0#015#012o=root 1141861088 1141861088 IN IP4 12.34.56.78#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 12.34.56.78#015#012t=0 0#015#012m=audio 12230 RTP/AVP 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012", "ICE": "force", "direction": [ "internal" ], "flags": [ "extenal", "force", "trust-address" ], "replace": [ "origin", ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: ... "session-connection" ], "transport-protocol": "RTP/SAVPF", "call-id": "220725c425526bc941dff8e972bd9458@12.34.56.78:50600", "received-from": [ "IP4", "12.34.56.78" ], "from-tag": "as4f0c032f", "command": "offer" } Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'extenal' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'force' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] Creating new call Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] Returning to SIP proxy: d3:sdp747:v=0#015#012o=root 1141861088 1141861088 IN IP4 23.101.139.216#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 23.101.139.216#015#012t=0 0#015#012a=ice-lite#015#012m=audio 38518 RTP/SAVPF 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012a=rtcp:38519#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:f31jYQjjKBdYoqWYijx0V/ZtTfP17pTVW8NJGVQ4#015#012a=setup:actpass#015#012a=fingerprint:sh ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] ... a-1 00:E3:0F:B6:C0:9F:D4:16:52:D2:41:A8:01:24:51:CD:3F:FB:B2:55#015#012a=ice-ufrag:VfqUSGBP#015#012a=ice-pwd:t5bLzDaNcqWUwzh7iEtyAZfweGo6#015#012a=candidate:04GQy5683m4TP1Da 1 UDP 2130706431 23.101.139.216 38518 typ host#015#012a=candidate:04GQy5683m4TP1Da 2 UDP 2130706430 23.101.139.216 38519 typ host#015#0126:result2:oke Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: Websocket Branch is {sip:85.6.14.8:4328;transport=WS} for {123455678} Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Got valid command from 127.0.0.1:52689: offer - { "sdp": "v=0#015#012o=root 1141861088 1141861088 IN IP4 12.34.56.78#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 12.34.56.78#015#012t=0 0#015#012m=audio 12230 RTP/AVP 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012", "ICE": "force", "direction": [ "internal" ], "flags": [ "extenal", "force", "trust-address" ], "replace": [ "origin", ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: ... "session-connection" ], "transport-protocol": "RTP/SAVPF", "call-id": "220725c425526bc941dff8e972bd9458@12.34.56.78:50600", "received-from": [ "IP4", "12.34.56.78" ], "from-tag": "as4f0c032f", "command": "offer" } Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'extenal' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'force' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] Returning to SIP proxy: d3:sdp747:v=0#015#012o=root 1141861088 1141861088 IN IP4 23.101.139.216#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 23.101.139.216#015#012t=0 0#015#012a=ice-lite#015#012m=audio 38518 RTP/SAVPF 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012a=rtcp:38519#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:f31jYQjjKBdYoqWYijx0V/ZtTfP17pTVW8NJGVQ4#015#012a=setup:actpass#015#012a=fingerprint:sh ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] ... a-1 00:E3:0F:B6:C0:9F:D4:16:52:D2:41:A8:01:24:51:CD:3F:FB:B2:55#015#012a=ice-ufrag:VfqUSGBP#015#012a=ice-pwd:t5bLzDaNcqWUwzh7iEtyAZfweGo6#015#012a=candidate:04GQy5683m4TP1Da 1 UDP 2130706431 23.101.139.216 38518 typ host#015#012a=candidate:04GQy5683m4TP1Da 2 UDP 2130706430 23.101.139.216 38519 typ host#015#0126:result2:oke Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: Websocket Branch is {sip:85.2.10.3:58509;transport=WS} for {123455678} Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Got valid command from 127.0.0.1:52689: offer - { "sdp": "v=0#015#012o=root 1141861088 1141861088 IN IP4 12.34.56.78#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 12.34.56.78#015#012t=0 0#015#012m=audio 12230 RTP/AVP 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012", "ICE": "force", "direction": [ "internal" ], "flags": [ "extenal", "force", "trust-address" ], "replace": [ "origin", ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: ... "session-connection" ], "transport-protocol": "RTP/SAVPF", "call-id": "220725c425526bc941dff8e972bd9458@12.34.56.78:50600", "received-from": [ "IP4", "12.34.56.78" ], "from-tag": "as4f0c032f", "command": "offer" } Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'extenal' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'force' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] Returning to SIP proxy: d3:sdp747:v=0#015#012o=root 1141861088 1141861088 IN IP4 23.101.139.216#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 23.101.139.216#015#012t=0 0#015#012a=ice-lite#015#012m=audio 38518 RTP/SAVPF 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012a=rtcp:38519#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:f31jYQjjKBdYoqWYijx0V/ZtTfP17pTVW8NJGVQ4#015#012a=setup:actpass#015#012a=fingerprint:sh ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] ... a-1 00:E3:0F:B6:C0:9F:D4:16:52:D2:41:A8:01:24:51:CD:3F:FB:B2:55#015#012a=ice-ufrag:VfqUSGBP#015#012a=ice-pwd:t5bLzDaNcqWUwzh7iEtyAZfweGo6#015#012a=candidate:04GQy5683m4TP1Da 1 UDP 2130706431 23.101.139.216 38518 typ host#015#012a=candidate:04GQy5683m4TP1Da 2 UDP 2130706430 23.101.139.216 38519 typ host#015#0126:result2:oke Feb 8 23:04:43 Kamailio2 kamailio[59406]: INFO: <script>: Websocket Branch is {sip:18.9.18.10:56917;transport=WS} for {123455678} Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Got valid command from 127.0.0.1:52689: offer - { "sdp": "v=0#015#012o=root 1141861088 1141861088 IN IP4 12.34.56.78#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 12.34.56.78#015#012t=0 0#015#012m=audio 12230 RTP/AVP 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012", "ICE": "force", "direction": [ "internal" ], "flags": [ "extenal", "force", "trust-address" ], "replace": [ "origin", ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: ... "session-connection" ], "transport-protocol": "RTP/SAVPF", "call-id": "220725c425526bc941dff8e972bd9458@12.34.56.78:50600", "received-from": [ "IP4", "12.34.56.78" ], "from-tag": "as4f0c032f", "command": "offer" } Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'extenal' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: Unknown flag encountered: 'force' Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] Returning to SIP proxy: d3:sdp747:v=0#015#012o=root 1141861088 1141861088 IN IP4 23.101.139.216#015#012s=Asterisk PBX 13.1.0#015#012c=IN IP4 23.101.139.216#015#012t=0 0#015#012a=ice-lite#015#012m=audio 38518 RTP/SAVPF 9 8 107 0 101#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:107 opus/48000/2#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=maxptime:60#015#012a=sendrecv#015#012a=rtcp:38519#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:f31jYQjjKBdYoqWYijx0V/ZtTfP17pTVW8NJGVQ4#015#012a=setup:actpass#015#012a=fingerprint:sh ... Feb 8 23:04:43 Kamailio2 rtpengine[55410]: [220725c425526bc941dff8e972bd9458@12.34.56.78:50600] ... a-1 00:E3:0F:B6:C0:9F:D4:16:52:D2:41:A8:01:24:51:CD:3F:FB:B2:55#015#012a=ice-ufrag:VfqUSGBP#015#012a=ice-pwd:t5bLzDaNcqWUwzh7iEtyAZfweGo6#015#012a=candidate:04GQy5683m4TP1Da 1 UDP 2130706431 23.101.139.216 38518 typ host#015#012a=candidate:04GQy5683m4TP1Da 2 UDP 2130706430 23.101.139.216 38519 typ host#015#0126:result2:oke Feb 8 23:04:43 Kamailio2 kamailio[59438]: INFO: <script>: Reply from webrtc client {18.9.18.10:56917} for method {INVITE}: 100 Feb 8 23:04:43 Kamailio2 kamailio[59438]: INFO: <script>: Manage_Reply from webrtc client {18.9.18.10:56917} for method {INVITE}: 100 Feb 8 23:04:43 Kamailio2 kamailio[59438]: INFO: <script>: Reply from webrtc client {18.9.18.10:56917} for method {INVITE}: 180 Feb 8 23:04:43 Kamailio2 kamailio[59438]: INFO: <script>: Manage_Reply from webrtc client {18.9.18.10:56917} for method {INVITE}: 180 Feb 8 23:04:43 Kamailio2 kamailio[59438]: INFO: <script>: Reply from webrtc client {18.9.18.10:56917} for method {INVITE}: 482 Feb 8 23:04:43 Kamailio2 kamailio[59442]: INFO: <script>: Reply from webrtc client {85.2.10.3:58509} for method {INVITE}: 100 Feb 8 23:04:43 Kamailio2 kamailio[59442]: INFO: <script>: Manage_Reply from webrtc client {85.2.10.3:58509} for method {INVITE}: 100

2 3

Setting dialog variable when BYE is received before accounting
by Mickael Marrache 10 Feb '15

10 Feb '15

Hi, Is there a way to set a dialog variable when a BYE is received so that it can be written in a CDR? I tried as follows but it didn't work: if (loose_route()) { .. if(is_method("BYE")) { if(is_direction("downstream")) { #Call disconnected by caller $dlg_var(dr) = 700; } else { #Call disconnected by callee $dlg_var(dr) = 701; } xlog("Setting disconnect reason to $dlg_var(dr)\n"); } .. exit; } The xlog statement prints: ERROR: <script>: Setting disconnect reason to <null> The doc states: This pseudo-variable will be available only for subsequential requests after doing loose_route(). Thanks, Mickael

2 1

Planing to release Kamailio v4.2.3
by Daniel-Constantin Mierla 10 Feb '15

10 Feb '15

Hello, I am planning to release a new version from latest stable branch - 4.2. Most likely it will be Tuesday next week (Feb 10). If there is something important to take in consideration for this version, reply to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

Facing kamailio issue
by taha zafar 10 Feb '15

10 Feb '15

Dear Kamailo team, I am facing problem in connecting kamailio on TLS, even kamailio is listening perfectly. TLS certificate are also tested. Kindly find attached logs and kamailio configuration file and give me your valuable feedback. *Eocean Pvt. Ltd.* *Taha Zafar* VoIP Engineer Mobile: 03318244432 Disclaimer: This email and any attachments to it may be confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of Eocean (Pvt) Ltd. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error.

2 1

Parallel call forward
by Raúl Alexis Betancor Santana 09 Feb '15

09 Feb '15

Hi all, I'm getting crazy trying to setup a kamailio to do a parallel call forward to multiple destinations based on a query to the DB. I do the db query and get the RURI's of the destinations I could forward the call to in parallel, then when I get the resulset I do: [...] $var(cf_idx) = 0; while(is_avp_set("$(avp(s:cf_destinations)[0])")) { $ru = $(avp(s:cf_destinations)[0]); # Set the new RURI to call to $(avp(s:cf_destinations)[0]) = $null; # Setting it to $null pull the value from the stack $(avp(s:callee_fr_inv_timer)[*]) = $(avp(s:cf_timeouts)[0]) * 1000; t_set_fr("$avp(s:callee_fr_inv_timer)"); $(avp(s:cf_timeouts)[0]) = $null; route(ROUTE_ACC_CF); # Do the ACC $(avp(s:acc_state)[*]) = "cfna"; $(avp(s:hih_cause)[*]) = "404"; $var(cf_idx) = $var(cf_idx) + 1; append_branch(); } route(ROUTE_EXECUTE_CF_LOOP); # Some final task and then t_relay [...] But only the last RURI get called How do I do to accomplish this? ... I need to change the $ru of each new branch, so when request get loop-back to the proxy, it do again all the lookup logic. Best regards

2 6

dmq_t_replicate fails with ERROR: t_forward_nonack: no branches for forwarding
by Leonardo Arena 09 Feb '15

09 Feb '15

Hi all, I have three Registrar-only proxies DMQ interconnected [1]. I'm attempting to use a client to register to one of them and replicate REGISTRAR message to the other two. * What it works: - Clients registration to proxy1 - Proxy1 replication of REGISTRAR message to proxy2 * What does not work: - Proxy replication of REGISTRAR message to proxy3 * Error message of proxy1: ERROR: tm [t_fwd.c:1770]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding ERROR: dmq [dmq_funcs.c:410]: cfg_dmq_t_replicate(): error calling t_replicate append_branch() did't help me and a search on the ML either. What am I doing wrong? Thanks! - leonardo [1] #-------------------------------------------------------- # Global Definitions #-------------------------------------------------------- debug = 2 fork = yes log_stderror = no # listen = 0.0.0.0 port = 5060 children = 4 dns = no rev_dns = no use_dns_failover = on dns_srv_lb = yes #dns_try_naptr = no mpath = "/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #-------------------------------------------------------- # Modules #-------------------------------------------------------- loadmodule "tm.so" loadmodule "sl.so" loadmodule "tmx.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "pv.so" loadmodule "xlog.so" loadmodule "ctl.so" #loadmodule "db_text.so" loadmodule "db_sqlite.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "memcached.so" loadmodule "dmq.so" loadmodule "corex.so" #-------------------------------------------------------- # Module Configuration #-------------------------------------------------------- modparam ( "usrloc", "db_mode", 0 ) modparam ( "usrloc", "use_domain", 1 ) modparam ( "usrloc", "hash_size", 20 ) #modparam ( "ctl", "binrpc_max_body_size", 10000 ) #modparam ( "ctl", "binrpc_struct_max_body_size", 200 ) #default #modparam ( "db_text", "db_mode", 0 ) modparam ( "auth", "use_domain", 1 ) #modparam ( "auth_db", "db_url", "text:///etc/kamailio/dbtext/" ) modparam ( "auth_db", "db_url", "sqlite:///etc/kamailio/kamailiodb" ) modparam ( "auth_db", "use_domain", 1 ) #use plaintext passwords for testing modparam ( "auth_db", "calculate_ha1", 1 ) modparam ( "auth_db", "version_table", 0 ) modparam ( "auth_db", "password_column", "password" ) modparam ( "auth_db", "load_credentials", "$avp(s:username)=username; $avp(s:domain)=domain;$avp(s:password)=password" ) modparam( "memcached", "servers", "localhost:11211" ) modparam( "dmq", "server_address", "sip:10.0.0.1:5060" ) modparam( "dmq", "notification_address", "sip:dmq.my.domain.net" ) #modparam( "dmq", "num_threads", 4 ) # modparam( "dmq", "ping_interval", 60 ) #-------------------------------------------------------- # Main Route Block #-------------------------------------------------------- include_file "kamailio-errors.cfg" route { # Uncomment next line if you want to see # each incoming message xlog( "L_INFO", "$rm: $fu -> $ru" ); if (!mf_process_maxfwd_header("20")) { route(ERROR_TOO_MANY_HOPS); } if (msg:len > 4096) { route(ERROR_MESSAGE_OVERFLOW); } if ($rm == "REGISTER" ) { if ( dmq_is_from_node() ) { xlog( "L_INFO", "REGISTER from node" ); route(REGISTER); exit; } else { # if we don't have credentials in cache authenticate against DB if ( $mct($tU@$td) == $null ) { if (!www_authenticate("$td", "subscriber")) { www_challenge("$td", "1"); exit; } else { route(REGISTER); xlog( "L_INFO", "Authenticated user $tU for domain $td against DB" ); # store credentials into cache $mct($avp(s:username)@$avp(s:domain)) = $avp(s:password); exit; } } # credentials are in cache, authenticate against it else { if (!pv_www_authenticate("$td", "$mct($tU@$td)", "0")) { # turn last param to "1" if using ha1 passwords # keep a counter of authentication attempts # if above 1 user is denied auth from cache so we discard the cache and force auth from DB at next attempt if ( $mct(count_$tU@$td) ) == $null { $mct(count_$tU@$td) = 1; } else { $mct(count_$tU@$td) = $null; $mct($tU@$td) = $null; xlog( "L_INFO", "Authentication count limit for user $tU for domain $td against cache reached. Cache discarded" ); } www_challenge("$td", "1"); exit; } else { route(REGISTER); $mct( count_$tU@$td ) = $null; xlog( "L_INFO", "Authenticated user $tU for domain $td against cache" ); exit; } } } } if ( $rm == "KDMQ" ) { dmq_handle_message(); } route(ERROR_INVALID_METHOD); } route[REGISTER] { if (!save("location")) { sl_reply_error(); } else { append_branch(); dmq_t_replicate(); } }

3 11

Get the SDP IP in Kamailio
by Sebastian Damm 09 Feb '15

09 Feb '15

Hi, since we integrated IPv6 into our existing routing logic, in some cases "fix_nated_sdp" gets called for IPv6 calls, too. (Actually, this just happens because of the mistakenly detected NAT.) But that ends up with a an error in the log. ERROR: nathelper [nathelper.c:1685]: replace_sdp_ip(): not an IPv4 address in 'c=' SDP So I'm looking for a way to exempt IPv6 calls from that. But I can't find a pseudovariable which carries the SDP contact IP. I guess I could take some function from the sdpops module, but is there a more convenient way to get just the contact IP? Best Regards, Sebastian

2 1

Registrar module and ipv6 addresses
by Sergey Okhapkin 09 Feb '15

09 Feb '15

Registrar module save() function stores "received" field in wrong format like sip:2601:3:8805:107:41E6:C7A1:7724:CF01:5062 When lookup() config function is executed for ipv6 client, kamailio log shows error Feb 3 12:07:34 east /usr/local/sbin/kamailio[16069]: ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: sip:2601:3:8 805:107:41E6:C7A1:7724:CF01:5062 Feb 3 12:07:34 east /usr/local/sbin/kamailio[16069]: ERROR: tm [t_fwd.c:1709]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Is it kamailio bug or a problem with my config?

4 5

Understanding t_relay
by Ryan Brindley 09 Feb '15

09 Feb '15

Hey community, I'm trying to understand t_relay () when a forward times out. This is an abbreviated version of what i have: Request_route { ... Route(do1) } Route [do1] { ... T_on_reply (1reply) T_on_failure (1fail) T_relay () } Reply_route[1reply] { ... If (t_check_status (302)) { Route (do2) } } Failure_route [1fail] { Xlog (dafail1) } Route [do2]{ ... T_on_reply (2reply) T_on_failure (2fail) T_relay () } Reply_route [2reply]{ Xlog (twerked) } Failure_route [2fail]{ Xlog (failured) } The case im currently interested in is when the first relay (do1) returns a 302 and then the second (do2) times out. What happens on the second when it times out, it hits the 1fail failure_route. This messes with my logic as i would've expected (and want to find out how to make it) hit the 2nd failure_route. I also noticed that if i loop and try the do2 again after the first failure it will then hit the 2fail route. Any clarification on this subject would be greatly appreciated, Ryan

2 3

Inconsistency between dlg_list and 'get_statistics dialog:'
by mayamatakeshi 09 Feb '15

09 Feb '15

Hello, I am counting the number o dialogs using this: # kamctl fifo dlg_list |grep dialog:: |wc -l 126 However when I use 'get_statistics dialog:' I get this: # kamctl fifo get_statistics dialog: dialog:active_dialogs = 67 dialog:early_dialogs = 3 dialog:expired_dialogs = 253 dialog:failed_dialogs = 29529 dialog:processed_dialogs = 338783 The number of dialogs don't match (i tested this several times). What could be causing this? (I am using kamailio 4.1). Regards, Takeshi

2 3

One Way RTP from destination SIP server using rtpengine
by Don Fanning 07 Feb '15

07 Feb '15

Hello All, I currently am running Kamailio in a WSS configuration with sipML5. I use rtpengine to convert a RTP/SAVPF packet to a RTP/AVP packet as the destination server only supports AVP or SAVP. RTPEngine has no issues rewriting the packet going out, the SIP session comes up and handshakes correctly to start the session. Then the remote server sends the RTP stream back and I'm having issues getting Kamailio or RTPEngine or something to take the RTP stream, process it back to SAVPF and send it back out the WSS port. I see no RTP data come from Kam/RTPengine going towards the remote server. On the server side, I do see STUN being called and the sipML5 bind successfully but that's it. Both the destination and server sit without firewalls in the way so the problem has to be what I'm doing in Kamailio. My config is located here: http://pastebin.com/dWLdUz5j Packet dumps available upon request. Thanks for any assistance!

4 7

RTPEngine Intermittent One Way Audio Issue SRTP => RTP
by Tim Chubb 07 Feb '15

07 Feb '15

Hi, I am currently experiencing an intermittent one way audio issue, when using RTPEngine and proxying between srtp and rtp. There is no apparent pattern, I have managed to repeat the on the 7th, 23rd, and 63rd calls to a test number, where upon connection only audio in the RTP => SRTP direction was working. All other test calls worked as normal. Digging through the RTPEngine log file the only abnormalities I can see are: Discarded invalid SRTP packet: authentication failed entries on the calls that had the one way audio issue. What could be causing this? Thanks Tim.

2 3

Multiple OR into a if
by Igor Potjevlesch 06 Feb '15

06 Feb '15

Hello, I'm very disappointed because of the following behaviour: if ($tu=~"^sip:0[1-9]{9}") { [.] In that case, Kamailio returns TRUE because the instructions in the block are executed. } if ($rU=~"^33" || $rU=~"^0033" || $rU=~"^0[1-9]{9}" || $rU=~"^\+33" || $tu=~"^sip:33" || $tu=~"^sip:0033" || $tu=~"^sip:0[1-9]{9}" || $tu=~"^sip:\+33" ) { [.] In that case, Kamailio should return FALSE because the instructions in the block are not executed. } Am I missed something regarding 'OR' ? Regards, Igor.

2 8

Kamailio load balansing 2 servers issues
by Yuriy Gorlichenko 06 Feb '15

06 Feb '15

Hello. We use 2 kamailio server 4.3 master brancher for load balansing cluster. We have some problems with this deplooiment fist of all we have REgstering issue: Witth one server all works fine (we have some endpoints with same creditians) all endpoints reinging well. But at 2 servers we hawe problem with this- call ringing only at one endpoint. We use Database for USRLOC module modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) As i think it mean al connections to servers ives at database. We have one database for 2 servers and this is means it only one table LOCATION for its. The second problen id that when loadbalanser sends bye to kamailio from the outside- bye is not recieves to endpoint. It say that no transaction with this CallID. I think it is because transaction goues through another kamailio and kamiaio that recieves BYE does not know about it transaction. So my first qestion- Does anybody have some information about loafd balansiong between kamailios? And second question - how to resolve issue with LOCATION table? Thanks

3 3

Siremis domain querry reports SQL error
by Marek Moravcik 06 Feb '15

06 Feb '15

Hello, after upgrading Siremis from 3.1 to 4.1 he starts report SQL errors on more places. But most annoying is error in domain list. I've tried to reinstall Siremis, drop database, but nothing helps. On other server after same upgrade, everything goes well. Only difference is, that error server has 64 bit architecture. Both servers have Kamailio 4.2.1. Where shall I look for problem? Here is domain list error: [2015-02-05 18:45:32 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.domain, T0.did, T0.last_modified FROM domain T0 LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.did' in 'field list' Script name and line number of error: /home/palo/sip/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /home/palo/sip/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /home/palo/sip/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /home/palo/sip/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/home/palo/sip/siremis/modules/ser/template..." ) @ /home/palo/sip/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /home/palo/sip/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /home/palo/sip/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /home/palo/sip/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/home/palo/sip/siremis/modules/ser/template..." ) @ /home/palo/sip/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /home/palo/sip/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /home/palo/sip/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /home/palo/sip/openbiz/bin/BizController.php 221 function: renderView ( "ser.view.DomainListView", "", "", Null, "" ) @ /home/palo/sip/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /home/palo/sip/openbiz/bin/BizController.php 32 function: include_once ( "/home/palo/sip/openbiz/bin/BizController.ph..." ) @ /home/palo/sip/siremis/bin/controller.php 6 function: include ( "/home/palo/sip/siremis/bin/controller.php" ) @ /home/palo/sip/siremis/bin/_forward.php 102 function: include ( "/home/palo/sip/siremis/bin/_forward.php" ) @ /home/palo/sip/siremis/index.php 3 Thanks! Marek Moravcik

2 1

How to install kamailio-4.2.2 in DEBIAN -7.1.0
by Yanko Marín Muro 06 Feb '15

06 Feb '15

Hello: How to install kamailio-4.2.2 in DEBIAN -7.1.0-amd64? This URL is the last version: http://www.kamailio.org/pub/kamailio/4.2.2/src/kamailio-4.2.2_src.tar.gz Best regards, Yanko Antonio Marín 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

Issue with dlg_setflag vs. setflag
by Anthony Messina 06 Feb '15

06 Feb '15

I'm currently using Kamailio built from master@e59db79 and have been following the recent threads about the dialog module as I'm moving toward adding dialog support for presence and CDRs in my setup. However, I seem to have come across an issue where dlg_setflag() does nothing, while setflag() works properly with the following (trimmed) script. So for now, I issue setflag(FLT_DIALOG) for INVITE rather than dlg_setflag(FLT_DIALOG). Is there something I'm missing about dlg_setflag()? #!define FLT_DIALOG 4 request_route { route(REQINIT); route(NATDETECT); if(is_method("CANCEL")) { if(t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); route(AUTH); remove_hf("Route"); if(is_method("INVITE|SUBSCRIBE")) record_route(); if(is_method("INVITE")) { setflag(FLT_ACC); # Enable dialog support (dlg_setflag not working) #dlg_setflag(FLT_DIALOG); setflag(FLT_DIALOG); } route(SIPOUT); route(PRESENCE); route(REGISTRAR); if($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); route(LOCATION); route(RELAY); } route[WITHINDLG] { if(!has_totag()) return; if(loose_route()) { route(DLGURI); if(is_method("BYE")) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); # Testing dialog flag xlog("L_INFO", "Completed $dlg(from_uri) to $dlg(to_uri) - $DLG_lifetime duration\n"); } else if(is_method("ACK")) { route(NATMANAGE); } else if(is_method("NOTIFY")) { record_route(); } route(RELAY); exit; } if(is_method("SUBSCRIBE") && uri==myself) { route(PRESENCE); exit; } if(is_method("ACK")) { if(t_check_trans()) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); exit; } -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

2 2

Siremis v4.2.0 Released
by Elena-Ramona Modroiu 05 Feb '15

05 Feb '15

Hi, Siremis v4.2.0 has been released, web management interface aiming at compatibility with Kamailio v4.2.x series - notes about release are available at: - http://siremis.asipto.com/2015/02/02/siremis-v4-2-0-released/ Ramona

3 3

Best module to use for interrupting a call ???
by Uri Shacked 05 Feb '15

05 Feb '15

Hi, I need to create some kind of check with an application outside of kamailio every X seconds. The check is per call, to get its status or something and according to the result do something with the call (transfer, send info or hangup...). I read about call_control and cnxcc modules. It seems these last ones are to much to do in order to get what i need. Is there a simpler module to use? BR, Uri

2 2

Kamailio v4.1.8 Released
by Daniel-Constantin Mierla 05 Feb '15

05 Feb '15

Hello, Kamailio SIP Server v4.1.8 stable release is out. This is a maintenance release of the previous stable branch, 4.1, that includes fixes since release of v4.1.7. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.1.x versions are strongly recommended to be upgraded to v4.1.8 or latest 4.2.x. For more details about version 4.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/02/kamailio-v4-1-8-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.2. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ 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

Lint and syntax check
by Manuel Rubio 05 Feb '15

05 Feb '15

Hi, I want to know if someone is using something similar to lint-like behaviour[1] and syntax highlighting in editors like Sublime, TextMate, emacs, ... for Kamailio configuration files. [1] http://en.wikipedia.org/wiki/Lint_%28software%29 Thanks. Manuel Rubio.

3 2

kamailio passthrough user auth
by Kai Arif 05 Feb '15

05 Feb '15

Hi, I have to implement a rather uncommon SIP setup and wanted to ask, if it is even possible with Kamailio. We are among other things a VoIP-provider for a couple of hundred customers. Right now we use a rather old piece of software as a SIP-server, that was written in-house and is deeply integrated in our user interfaces and the billing system and can't be replaced easily. But now we need to make some modifications because our PSTN-gateway-provider is changing things. The idea is to use Kamailio as proxy in front of our old SIP-server. But to do this we need to pass through the REGISTER requests of our users to the old SIP-server. So the question is if Kamailio supports this kind of passthrough and I would also be rely grateful for a piece of example config, if somebody already has a similar setup running. So far neither the documentation nor google have been very helpful. Best regards Kai -- Kai Arif - System Administrator Inter.net Germany GmbH Knesebeckstraße 59-61 10719 Berlin Germany Fon +49 30 25430 0 Fax +49 30 25430 499 arif(a)team.de.inter.net www.de.inter.net Legal-Information: Inter.net Germany GmbH, HRB #79136, Amtsgericht Berlin Charlottenburg, UST-IdNr:: DE 813 165 159 FA für Körperschaften I Berlin, Geschäftsführer: Jörn Lubkoll, Christian Röhl Zuständige Regulierungsbehörde: Bundesnetzagentur Chemnitz, Liselotte-Herrmann-Str. 20a, 09127 Chemnitz, Registriernummer: 06/164

2 1

Does the event_route[tm:local-request] support stateless forwarding configured in the script?
by Klaus Feichtinger 05 Feb '15

05 Feb '15

Hello, I´m searching for a way to keep track about success or failure of locally generated SUBSCRIBE transactions, which are typically triggered by the RLS module. In case that a subscription is rejected by the subscribee with any negative response code or the transaction is timing out, I have to trigger a specific action (e.g. sending a PUBLISH message to another user agent). This seems to be not so easy.... As Kamailio does not support referencing a failure_route to locally generated requests, I had the idea to "forward()" locally generated requests in the event_route "tm:local-request" to itself and handle the forwarded requests in the standard request_route, which would offer all usual options. But that seems not to be supported / allowed by Kamailio, as I have in practical tests (1) an outgoing request that is sent to the original Destination-URI (with a malformed "Content-Length" SIP hdr) and (2) the second request that is sent to the manual target (loop). I´ve found in the mailing list that elder Kamailio versions did not support any common way for this feature. However, can anybody give m a hint if / how this could eventually be solved with the event_route or in any alternative route? It seems that the Kamailio internally generated requests are still below the radar of TM or any other route blocks except tm:local-request..... regards, Klaus

2 1

Out of Memory Error in Kamailio
by Neeraj Swarnkar 05 Feb '15

05 Feb '15

Hi Guys, I am using Kamailio-4.2.1. Earlier I was using shared memory of 16GB, which started popping out of memory error, within few seconds of generating load from sipp. I used sipp following command: sipp -s 999999999999 MY_KAM_IP -l 2500 -r 50 *Error coming is :* *Feb 4 14:12:23 kamailio_us_ha1 /usr/local/sbin/kamailio[27389]: ERROR: tm [t_reply.c:1946]: relay_reply(): ERROR: relay_reply: cannot alloc reply shmem* After changing the shared memory to 1 GB, this error came quite late when generate load at 200 CPS. sipp -s 999999999999MY_KAM_IP -l 2500 -r 200 Why this is happening? I am using following configuration file( kamailio.cfg). Earliest possible help will be appreciated a lot!! #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database # define WITH_LCR to enable LCR # define WITH_CR to enable carrier route #!define WITH_MYSQL #!define DBURL #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_ACCDB #!define WITH_AUTH #!define WITH_DEBUG #!define WITH_NAT #----#!define WITH_ANTIFLOOD #!define WITH_CR #!ifdef WITH_FREESWITCH #freeswitch.bindip = "10.60.23.81" desc "FreeSWITCH IP Address" freeswitch.bindip = "54.254.97.209" desc "FreeSWITCH IP Address" freeswitch.bindport = "5060" desc "FreeSWITCH Port" #!endif #!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" import_file "ip.cfg" ####### 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. #!ifndef DBURL #!define DBURL "mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio " #!endif #!endif #!define MULTIDOMAIN 0 # - 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=-1 log_stderror=no #!else debug=-1 log_stderror=no #!endif memdbg=2 memlog=2 log_facility=LOG_LOCAL0 fork=yes children=8 /* 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 #listen=udp:172.31.42.191:5060 advertise 54.254.97.209:5060 listen=LOCAL_BIND_PROTO_IP_UDP_PORT advertise ADVERTISED_BIND_IP_PORT listen=LOCAL_BIND_PROTO_IP_TCP_PORT advertise ADVERTISED_BIND_IP_PORT /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 mhomed=0 #!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 ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.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" loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "dispatcher.so" loadmodule "pdt.so" loadmodule "mi_datagram.so" loadmodule "domain.so" loadmodule "htable.so" loadmodule "pike.so" #!ifdef WITH_LCR #Neeraj: Load LCR Module loadmodule "lcr.so" #!endif #!ifdef WITH_CR #Neeraj: Load CR Module loadmodule "carrierroute.so" loadmodule "avpops.so" #!endif #loadmodule "nat_traversal.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # for mi_datagram modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # for sqlops modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") # for domain modparam("domain", "db_url", "mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") #modparam("domain", "db_mode", 1) # Use caching modparam("domain", "domain_table", "domain") modparam("domain", "register_myself", 1) # ----- 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: 3sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 5sec modparam("tm", "fr_inv_timer", 30000) # ----- dispatcher params ----- modparam("dispatcher", "db_url","mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") # ---- pdt params------ modparam("pdt", "db_url","mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") #modparam("pdt", "table_name", "pdt") modparam("pdt", "fetch_rows", 4000) modparam("pdt", "check_domain", 0) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 1 contacts per AOR */ modparam("registrar", "max_contacts", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 1) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", "mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") # Neeraj : Disabled Saving Location in DB #modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_mode", 0) # Neeraj : Enabling Multiple Domain modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url","mysql:// kamailio:kamailiorw@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- #modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:10.0.0.121:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 0) #!endif #!ifdef WITH_LCR modparam("lcr", "db_url", "mysql:// kamailioro:kamailioro@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") #!endif #!ifdef WITH_CR #Neeraj: ----carrierroute params------- modparam("carrierroute", "config_source", "db") modparam("carrierroute", "db_url", "mysql:// kamailioro:kamailioro@kamailio-us.cejtf390jur8.us-east-1.rds.amazonaws.com/kamailio ") modparam("carrierroute", "carrierroute_table", "carrierroute") modparam("carrierroute", "default_tree", "default") modparam("carrierroute", "fetch_rows", 2000) #!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 { # 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(); # 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 } #!ifdef WITH_LCR #Neeraj: oute request to LCR route(LCR); #!endif # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } #!ifdef WITH_CR # route(CARRIER_ROUTE); #!endif # user location service route(LOCATION); } #!ifdef WITH_CR route[CARRIER_ROUTE] { if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not Carrier Found"); exit; } else { # In case of failure, re-route the request #t_on_failure(1); t_on_failure("MANAGE_FAILURE"); # Relay the request to the gateway # t_relay(); } } failure_route[1] { xlog("route[FAILOVER_ROUTE: called"); # In case of failure, send it to an alternative route: if (t_check_status("403|5[0-9][0-9]")) { #choose route domain 1 of the default carrier if(!cr_route("failover", "failover", "$rU", "$rU", "call_id")){ t_reply("403", "Not allowed"); } else { t_on_failure("2"); # t_relay(); } } } #!endif #!ifdef WITH_CR_OLD route[CARRIER_ROUTE] { xlog("route[CARRIER_ROUTE: called"); # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){ if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){ xlog("CARRIER_ROUTE:ROUTING FAILED: no route found for $rU"); sl_send_reply("604", "Unable to route this call"); exit; } else { xlog("CARRIER_ROUTE:LOOKUP FOUND: $rd $avp(s:route_desc)"); avp_pushto("$ru/username", "$avp(s:route_desc)"); sl_send_reply("302", "$rd"); exit; } } else { xlog("route[CARRIER_ROUTE: method is not either INVITE or X-ROUTE LOOKUP"); } } #!endif #!ifdef WITH_LCR route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR route\n"); } if(method=="INVITE") { xlog("We got an invite for: '$rU'"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'\n"); xlog("Domain of destination: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gateway\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); } }; } #!endif route[FSDISPATCH] { # dial number selection route(FSRELAY); exit; } route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); if($var(newbranch)==1) { append_branch(); $var(newbranch) = 0; } route(RELAY); exit; } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } #!endif # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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()) { 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("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",5)) sl_reply_error(); exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not Carrier Found"); exit; } break; 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; } # 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(); fix_nated_contact(); } 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; } route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Regards Neeraj Swarnkar

2 1

Planing to release Kamailio v4.1.8
by Daniel-Constantin Mierla 05 Feb '15

05 Feb '15

Hello, I am planning to release a new version from previous stable branch - 4.1. Most likely it will be Thursday or Friday this week. If there is something important to take in consideration for that version, reply to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

Linking Kamailio to Asterisk
by Brian Echesa 05 Feb '15

05 Feb '15

Hi, Kindly assist. I have successfully installed Kamailio and Asterisk. Asterisk is acting as an PSTN in my case. The trunk between Kamailio and Asterisk is up. Issue is that i can not route calls to the PSTN. I am using the default configuration version: kamailio 4.2.2 (i386/linux) OS: Debian wheezy -- Kind regards, Brian Linux registered user: . #565878 www.brianechesa.com

2 1

RTPEngine - No support for kernel packet forwarding available
by Muhammad Shahzad 05 Feb '15

05 Feb '15

Hi, I have latest stable release of RTPEngine deployed in a virtual machine (KVM) along with Kamailio v4.2. All is working fine except i see this message in RTPEngine logs, -- rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] Kernelizing media stream rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] No support for kernel packet forwarding available -- using lsmod command i see that module xt_RTPENGINE is load. -- Module Size Used by iptable_mangle 12488 0 xt_RTPENGINE 22068 3 iptable_nat 12800 0 nf_nat 17924 1 iptable_nat ... -- Also the iptables rule for RTPEngine exists, i.e. -- Chain INPUT (policy ACCEPT 224K packets, 37M bytes) pkts bytes target prot opt in out source destination 224K 37M rtpengine all -- any any anywhere anywhere Chain FORWARD (policy ACCEPT 0 packets, 0 bytes) pkts bytes target prot opt in out source destination Chain OUTPUT (policy ACCEPT 224K packets, 37M bytes) pkts bytes target prot opt in out source destination Chain rtpengine (1 references) pkts bytes target prot opt in out source destination 2022 414K RTPENGINE udp -- any any anywhere anywhere RTPENGINE id:0 -- Yet still RTPEngine is not able to kernelize the RTP stream and does user-space packet forwarding? Can you guys suggest what could be wrong here? Thank you.

3 4

Issues while using Kamailio as proxy with Asterisk
by Juned Khan 05 Feb '15

05 Feb '15

Hi All, I am using Kamailio as proxy server with Asterisk but having some issues with Inbound and outbound calls. Asterisk is a registrar, Kamailio is a just for proxy. Using softphone call are working fine but using hardphone, its creating issues. Here are the test cases. No Test case *98 Inbound 1 ATA registered with Proxy Domain No No 2 ATA registered with Proxy IP Yes Yes 3 ATA registered with Asterisk domain No No 4 ATA registered with Asterisk IP Yes Yes 5 Soft phone registered with Proxy Domain Yes Yes 6 Soft phone registered with Proxy IP Yes Yes 7 Soft phone registered with Asterisk domain Yes Yes 8 Soft phone registered with Asterisk IP Yes Yes Here I am dialling *98 with the mentioned scenario, if it goes to voicemail options then Yes, else No I am using ALLO ATA. Here is my http://pastie.org/9887806 Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

1 0

Send sip INFO req from kamailio
by Uri Shacked 04 Feb '15

04 Feb '15

Hi, Any idea how to send a SIP INFO req ? I am familiar with uac_req_send. but how do I send it with in a specific dialog and with data in the INFO req ? BR, Uri

2 1

kamailio 4.2.1 core in timer.c:877
by 张顺通 03 Feb '15

03 Feb '15

I user rtimer module : loadmodule "rtimer.so" modparam("rtimer", "timer", "name=p_route;interval=1;mode=1;") modparam("rtimer", "exec", "timer=p_route;route=P_ROUTE") application will coredump in sometimes, and core file in below, what's the reason? Program terminated with signal 11, Segmentation fault. #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 877 timer.c: No such file or directory. in timer.c (gdb) bt full #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 tl = 0x7f9ed47f7e98 ret = 0 #1 0x00000000005fc686 in timer_handler () at timer.c:953 saved_ticks = 1043586690 run_slow_timer = 0 i = 268 __FUNCTION__ = "timer_handler" #2 0x00000000005fdd4d in timer_main () at timer.c:992 No locals. #3 0x00000000004aa571 in main_loop () at main.c:1700 i = 12 pid = 0 si = 0x0 si_desc = "udp receiver child=11 sock=10.1.*.*:5160\000\000\b\000\000\000\000\000\000\000\000\060S\324\236\177\000\000\060=#z\001\000\000\000(\205V\324\236\177\000\000\060=#z\377\177\000\000\214\216N\000\000\000\000\000\037t\372D\000\000\000\000(\205V\324\236\177\000\000\220@ #z\377\177\000\000\000\000\000\000\001\000\000" nrprocs = 12 __FUNCTION__ = "main_loop" #4 0x00000000004af166 in main (argc=11, argv=0x7fff7a234098) at main.c:2561 cfg_stream = 0x1da75e0 c = -1 r = 0 tmp = 0x7fff7a234681 "" tmp_len = 32767 port = 2049130512 proto = 32673 options = 0x706ec0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 220012340 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xb p = 0x0 __FUNCTION__ = "main"

2 5

[OT] ssl certificates updated for kamailio.org
by Daniel-Constantin Mierla 03 Feb '15

03 Feb '15

Hello, short note to say that I updated the ssl certificates for kamailio.org web site. They are issued by cacert.org, which is probably not trusted by most of the browsers, but they are well known in the open source environment. If you get a warning about certificate change and the issuer is cacert.org, then you should be safe to login over https (e.g., to edit the wiki pages). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Issues while using Kamailio as proxy with Asterisk
by Juned Khan 03 Feb '15

03 Feb '15

Hi All, I am using Kamailio as proxy server with Asterisk but having some issues with Inbound and outbound calls. Here are the test cases. No Test case *98 Inbound 1 ATA registered with Proxy Domain No No 2 ATA registered with Proxy IP Yes Yes 3 ATA registered with Asterisk domain No No 4 ATA registered with Asterisk IP Yes Yes 5 Soft phone registered with Proxy Domain Yes Yes 6 Soft phone registered with Proxy IP Yes Yes 7 Soft phone registered with Asterisk domain Yes Yes 8 Soft phone registered with Asterisk IP Yes Yes Here I am dialling *98 with the mentioned scenario, if it goes to voicemail options then Yes, else No I am using ALLO ATA. I have attached my kamailio.cfg with this email. Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

1 0

Ro_CCR function
by Shane Harrison 03 Feb '15

03 Feb '15

Hi all, In a previous post I was alerted to the (perhaps obvious) fact that that ims_charging module is design to run within an IMS environment: http://lists.sip-router.org/pipermail/sr-users/2015-January/086675.html I have subsequently attempted to modify the module to work with the standard usrloc Kamailio module. I decided to branch from the 4.2.2 release rather than the master since I was unsure of the state of the ims_charging module in the master. It now compiles fine and installs. I also have the cdp module talking watchdog_requests to a freeDiameter server so at that level things seem OK. However inserting the Ro_CCR() call into the kamailio cfg file shows up some problems, primarily: Feb 2 01:05:52 hh-rcs-sipproxy3 /usr/sbin/kamailio[6473]: WARNING: usrloc [dlist.c:624]: register_udomain(): Registering a new domain called 'voip.we-rcs.flowcloud.systems' with usrloc Feb 2 01:05:52 hh-rcs-sipproxy3 /usr/sbin/kamailio[6473]: ERROR: <core> [db.c:443]: db_check_table_version(): invalid version 0 for table voip.we-rcs.flowcloud.systems found, expected 6 (check table structure and table "version") Exploring this error leads me to a few questions: 1) The 4.2.2 Ro_CCR() has the diameter domain as a parameter. The addition of this function to my kamailio.cfg file causes ro_fixup() -> domain_fixup() -> ul.register_domain which as per the log above, attempts to look for a table with the same name as the domain. Is it expected that such a table exists, for standard Kamailio I thought all usrloc records simply go into the Location table , regardless of the domain. 2) What is the purpose of the ro_fixup() 3) I notice that the 4.3 module documentation for ims_charging module has 6 parameters for the function call while the example code given in the same section as only 5 (missing domain). The code from the master branch for the function is different again - not sure if 4.3 is still in quite a bit of flux or not for the ims_charging module? 4) Should I be using 4.3 ims_charging module as a starting point or is 4.2.2 good enough i.e are there major changes? At this stage I am going to create a table for the domain but would like some clarity on how the usrloc and domains relate to table names. 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 2

Grouped dispatching
by Grant Bagdasarian 02 Feb '15

02 Feb '15

Hello, Would the below scenario be possible using the dispatcher module or any other routing module? Destination 1: 192.168.1.1 Destination 2: 192.168.1.2 Destination 3: 172.168.1.1 These three destinations belong to the same distribution group(set), but Kamailio has to do the following: Perform round-robin using destination 1 and 2, but if both are down fail-over to destination 3. I hope I'm being clear enough. Regards, Grant

2 1

Video Key-Frame Request using RTCP FIR or SIP INFO message
by Muhammad Shahzad 02 Feb '15

02 Feb '15

Hi, This may be a bit out of focus topic for this forum but i am posting it here anyway with hope that some guru would shed some light on it and point me to right direction. The problem is that i want to establish video call between a webrtc and a sip client using kamailio (for signalling) and RTPEngine (for media relay). Both signalling and the audio stream seems to work perfectly fine The remote video on webrtc client side (i.e. video stream from sip client) takes about 20-30 seconds to establish but once it starts it works fine. However, the remote video on sip client side (i.e. video stream from webrtc client) starts almost immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then it goes blank after about 30 seconds. After a long discussion with sip client developer, we now understand the fact that sip client sends a request for so called key-frame, which is ignored by webrtc client. This request is sent through both RTCP stream and SIP INFO message. The SIP INFO message seems to be pointless as media is internally managed by chrome/firefox and these browsers don't give us such sophisticated access and control over media streams. Please let me know if this assumption is wrong. For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP packet type" error message in RTPEngine logs (not sure if it drops this packet or relay it anyway). Does anyone has any idea on how can we either, 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request? OR 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream on webrtc client's behalf? If there is any other solution to this, please feel free to share. Thank you.

4 5

What could cause an intermittent fault (28) with http_query()?
by Tim Chubb 01 Feb '15

01 Feb '15

Hi All I seem to be experiencing an intermittent fault with the utils http_query() method. We are implementing a routing and white list component that is accessed via a REST api, however i have observed several occasions where this is logged: Jan 30 16:04:17 vs-kam-prod02 /usr/local/sbin/kamailio[13184]: ERROR: utils [functions.c:149]: http_query(): failed to perform curl (28) The indicated error number (28) seems to suggest a timeout is occurring with curl, however examining a capture of network traffic when this happens shows that a http request is not sent from the server to the destination at all, usually attempting a second call results in everything working correctly. As stated its intermittent in its nature and so I cannot reliably trigger this issue, other than through sheer repetition, so any ideas as to what could be causing this issue would be gratefully received,

2 2

Async module taking down our server
by Will Ferrer 31 Jan '15

31 Jan '15

Hi All We are trying to use the async module to to delay sending a bye on from one end of the call to the other. We are using async_route(routename, seconds) to delay the WITHINDLG route. The idea is that in the future we want to be able to have our billing min duration enforced (though currently we are having issues with the dialog module that we are discussing in another thread). After running this on our deploy servers, the delays before sending on the byes get longer and longer, and then kamailio goes down. Then the receive udp buffer fills up. We tried it with both 4 and 400 async workers, and it made no difference. I am including a screen capture of the servers stats when this happens taken from voip monitor. Here are the relevant parts of the config: ... loadmodule "async.so" ... modparam("async", "workers", ASYNC_THREADS) ... request_route { ... route(DELAYED_BYE_STATIC); ... route[DELAYED_BYE_STATIC] { xlog("L_DEBUG","route DELAYED_BYE_STATIC"); #!ifdef WITH_DELAYED_BYE_STATIC if (is_method("BYE")) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, from self \n"); #if (from_uri == myself) { if ((allow_trusted() || allow_source_address()) && from_uri == myself) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, Bye detected, from self \n"); send_reply("200", "OK"); xlog("L_DEBUG","route DELAYED_BYE_STATIC, sent 200 about to sleep \n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (has_totag()) { xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG_DELAYED \n"); async_route("WITHINDLG_DELAYED", MIN_DURATION); } else { xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG \n"); async_route("WITHINDLG", MIN_DURATION); } xlog("L_DEBUG","route DELAYED_BYE_STATIC, slept\n"); exit; } } #!endif return; } ... route[WITHINDLG_DELAYED] { xlog("L_DEBUG", "route WITHINDLG_DELAYED: triggered \n"); $avp(was_delayed) = 1; route(WITHINDLG); } ... route[WITHINDLG] { xlog("L_DEBUG", "route WITHINDLG: will -- DLG triggered, request method: $rm \n"); #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL")) { xlog("L_DEBUG","route WITHINDLG: cancel or bye detected, request method: $rm \n"); #!ifdef WITH_DISPATCHER_LOAD_AWARE xlog("L_DEBUG","route WITHINDLG: running ds_load_update, request method: $rm \n"); ds_load_update(); #dlg_get ("$ci","$ft","$tt"); #dlg_bye ("all"); #!endif } #!endif if (has_totag() || $avp(was_delayed) == 1) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG has totag or was_delayed: $avp(was_delayed) \n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG has loose route \n"); route(DLGURI); if (is_method("BYE")) { xlog("L_DEBUG","route WITHINDLG: BYE detected"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails xlog("L_DEBUG","route WITHINDLG: ACC flag set"); } 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(); } xlog("L_DEBUG", "route WITHINDLG: will -- DLG RELAY 1\n"); route(RELAY); } else { xlog("L_DEBUG", "route WITHINDLG: will -- DLG else \n"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG is ack \n"); if ( t_check_trans() ) { xlog("L_DEBUG", "route WITHINDLG: will -- DLG t_check_trans \n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_DEBUG", "route WITHINDLG: will -- DLG RELAY 2\n"); route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } ... Does any one know if this is a bug or a leak with in the async module, or perhaps something I am doing in my config? Thanks in advance for an assistance you can offer me. All the best. Will Ferrer Switchsoft

3 5

Dispatcher weight dont work
by Yuriy Gorlichenko 30 Jan '15

30 Jan '15

Hello I use dipatcher algorithm 8 that works with weight. I added 2 Asterisks and try to call its with my kam.We use 4.3 version. Tthis config select needed dst from database with my scenario. if(!ds_select_dst("$var(setid)", "8")) $var(setid)- is variable for setting setid that i get from database with my own scenario. IT does not matter. When running asterisk with weight 90 - all calls goes through it. When I starting asterisk with weight 10 -calls going through asterisk 90. When I shut down asterisk with weight 90 -calls goes through asterisk 10? but when i start asterisk weight 90 all calls goes through sterisk 10 until I shut down it. root@Kamailio:~# kamailio -v version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 8cdbe7 compiled on 01:17:56 Jan 21 2015 with gcc 4.8.2 id setid destination flags priority attrs 1 2 sip:34.25.123.45:506000 0 weight=10 2 2 sip:10.0.1.6:506000 0weight=90 modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.1.1") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0) Thanks

2 3

NDB_REDIS password to REDIS remote DB
by Yuriy Gorlichenko 30 Jan '15

30 Jan '15

Hello. I try to use NDB_REDIS with remote REDIS DB and can not to connect because remote DB use password, but Kamailio module have no any variable or attr of modparam that implenets password for DB. How I can connect to my REDIS?

3 5

why do not support tcp:eth1:5062，only support eht1:5062
by 张顺通 30 Jan '15

30 Jan '15

why do not support tcp:eth1:5062，only support eht1:5062? I need to bind on eth1 only tcp.

2 6

tls_port_no maybe do not work
by 张顺通 30 Jan '15

30 Jan '15

I set cfg like this: tls_port_no=9004 listen=bond0:9002 but result is: udp: 10.232.98.23 [10.232.98.23]:9002 tcp: 10.232.98.23 [10.232.98.23]:9002 tls: 10.232.98.23 [10.232.98.23]:9003

1 0

Wrong location entries when using usrloc with Mongo
by Mickael Marrache 30 Jan '15

30 Jan '15

Hi, I start with no location nor in Mongo, nor in memory. My UA registers successfully and I can see the location entry in Mongo. Then, I close my UA which unregisters (by setting Expires header to 0). Then, I open the app again and a new registration is made. The entry after first registration. Looks okay. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "username" : "A", "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" : ISODate("2015-01-30T08:55:34Z"), "q" : -1, "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8809, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip:XXXXXXXXXXX:54217;transport=tcp", "path" : "<sip:XXXXXXXXXX;lr;received=sip:XXXXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp:XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:55:34Z"), "ruid" : "uloc-54cb38df-3378-2", "instance" : null, "reg_id" : 0 } The same entry after un register (Expires 0). Note that the username field is missing. In any case, I expected the entry to be deleted. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" : ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0, "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" } The entries after second registration. The new entry looks okay. But, the old entry is still here. { "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" : ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0, "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" } { "_id" : ObjectId("54cb3a7884e581337a25b895"), "username" : "A", "contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" : ISODate("2015-01-30T09:02:00Z"), "q" : -1, "callid" : "297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8813, "flags" : 0, "cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip: XXXXXXXXX:54217;transport=tcp", "path" : "<sip: XXXXXXXXX;lr;received=sip: XXXXXXXXX:54217%3Btransport%3Dtcp>", "socket" : "udp: XXXXXXXXX:5060", "methods" : 4751, "last_modified" : ISODate("2015-01-30T08:02:00Z"), "ruid" : "uloc-54cb38df-337a-1", "instance" : null, "reg_id" : 0 } The issue is the entry is not deleted after un registering. I precise that registrations are dispatched over multiple registrars with all accesses the same Mongo cluster. Thanks, Mickael

1 0

Can kamailio generate a PRACK
by Rahul MathuR 30 Jan '15

30 Jan '15

Hello, I was wondering whether Kamailio (as proxy) can generate a PRACK on its own ( since one of the custom written dialer is not sending PRACK) ? Is there any way I can achieve this ? -- Warm Regds. MathuRahul

2 1

Core dump while saving user location in mongo
by Mickael Marrache 30 Jan '15

30 Jan '15

Hi, When my proxy receives a REGISTER request, it crashes while saving the user location in MongoDB. This is the backtrace from the core dump: #0 0x0000003f08e32625 in raise () from /lib64/libc.so.6 #1 0x0000003f08e33e05 in abort () from /lib64/libc.so.6 #2 0x00000000006196f9 in fm_free (qm=0x7f4368bf7010, p=0x210add0, file=0x7f4367fd7608 "<core>: db_res.c", func=0x7f4367fd8100 "db_free_columns", line=86) at mem/f_malloc.c:588 #3 0x00007f4367fc37ab in db_free_columns (_r=0x7f4368ca0430) at db_res.c:86 #4 0x00007f4367fc479e in db_free_result (_r=0x7f4368ca0430) at db_res.c:134 #5 0x00007f4367da37cb in db_mongodb_free_result (_h=0x7f4368c94718, _r=0x7f4368ca0430) at mongodb_dbase.c:646 #6 0x00007f4367da5d65 in db_mongodb_query (_h=0x7f4368c94718, _k=0x7fffa03d5550, _op=0x0, _v=0x7fffa03d5510, _c=0x7fffa03d5560, _n=1, _nc=16, _o=0x7f4365ed2c80, _r=0x7fffa03d5508) at mongodb_dbase.c:783 #7 0x00007f4365cac7d2 in db_load_urecord (_c=0x7f4368c94718, _d=0x7f436154a0e8, _aor=0x7fffa03d59d0) at udomain.c:601 #8 0x00007f4365cb1e8e in get_urecord (_d=0x7f436154a0e8, _aor=0x7fffa03d59d0, _r=0x7fffa03d5760) at udomain.c:1106 #9 0x00007f4365a75dea in add_contacts (_m=0x7f4368cbdc40, _d=0x7f436154a0e8, _a=0x7fffa03d59d0, _mode=1, _use_regid=1) at save.c:832 #10 0x00007f4365a77d32 in save (_m=0x7f4368cbdc40, _d=0x7f436154a0e8, _cflags=4, _uri=0x0) at save.c:986 #11 0x00007f4365a606db in w_save2 (_m=0x7f4368cbdc40, _d=0x7f436154a0e8 "H\240TaC\177", _cflags=0x4 <Address 0x4 out of bounds>) at reg_mod.c:447 #12 0x000000000041d489 in do_action (h=0x7fffa03d6180, a=0x7f4368c71e98, msg=0x7f4368cbdc40) at action.c:1100 #13 0x0000000000429a7a in run_actions (h=0x7fffa03d6180, a=0x7f4368c71e98, msg=0x7f4368cbdc40) at action.c:1583 #14 0x000000000042a0df in run_actions_safe (h=0x7fffa03d7470, a=0x7f4368c71e98, msg=0x7f4368cbdc40) at action.c:1648 #15 0x0000000000540b7c in rval_get_int (h=0x7fffa03d7470, msg=0x7f4368cbdc40, i=0x7fffa03d6658, rv=0x7f4368c72830, cache=0x0) at rvalue.c:924 #16 0x0000000000544db4 in rval_expr_eval_int (h=0x7fffa03d7470, msg=0x7f4368cbdc40, res=0x7fffa03d6658, rve=0x7f4368c72828) at rvalue.c:1918 #17 0x00000000005451aa in rval_expr_eval_int (h=0x7fffa03d7470, msg=0x7f4368cbdc40, res=0x7fffa03d6ae0, rve=0x7f4368c72070) at rvalue.c:1926 #18 0x000000000041ce4e in do_action (h=0x7fffa03d7470, a=0x7f4368c73148, msg=0x7f4368cbdc40) at action.c:1064 #19 0x0000000000429a7a in run_actions (h=0x7fffa03d7470, a=0x7f4368c71cc0, msg=0x7f4368cbdc40) at action.c:1583 #20 0x0000000000419dea in do_action (h=0x7fffa03d7470, a=0x7f4368c3bdb8, msg=0x7f4368cbdc40) at action.c:712 #21 0x0000000000429a7a in run_actions (h=0x7fffa03d7470, a=0x7f4368c33898, msg=0x7f4368cbdc40) at action.c:1583 #22 0x000000000042a1a7 in run_top_route (a=0x7f4368c33898, msg=0x7f4368cbdc40, c=0x0) at action.c:1669 #23 0x000000000050747e in receive_msg ( buf=0xa6f780 "REGISTER sip:XXXXXXXXXXXX:5060;transport=tcp SIP/2.0\r\nVia: SIP/2.0/UDP XXXXXXXXX;branch=z9hG4bK242c.26de52a75c8675f6757e0958454a0ad5.0;i=1\r\nv: SIP/2.0/TCP XXXXXXXXXX:62051;received=XXXXXXXX"..., len=1004, rcv_info=0x7fffa03d7760) at receive.c:216 #24 0x0000000000606e76 in udp_rcv_loop () at udp_server.c:521 #25 0x00000000004a5c97 in main_loop () at main.c:1629 #26 0x00000000004aaf2a in main (argc=13, argv=0x7fffa03d7b78) at main.c:2561 I tested multiple times, it occurs everytime. Let me know if you need more information. Thanks, Mickael

2 6

Does forking impact max_inv_lifetime?
by Alex Balashov 30 Jan '15

30 Jan '15

Hi, In this serial forking-based failover scenario UAC --> Proxy --> UAS 1 UAS 2 ... UAS 3 I've got a requirement to terminate the call and return an error to the UAC if there is no final response within 60 seconds. The usual fr_inv_timer / restart_fr_on_each_reply combination won't do, because even when restart_fr_on_each_reply == 0, the timer is still restarted on provisional replies of increasing response code value (e.g. 183 after 180). As the documentation states, fr_inv_timer is a branch-level construct anyway, so this approach doesn't make sense here. That's why I looked into max_inv_lifetime. However, when I set it to 60000, I still get the same result, but only when there are new branches being formed on the proxy -> UAS side. It works fine if there is no forking and a single gateway does not provide a final reply within 60 seconds. So, my sense is that the max_inv_lifetime timeout is reset when a new branch is created to UAS 2...UAS N. Is this accurate? Thanks, -- 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 6

gruu within dialog
by samuel 29 Jan '15

29 Jan '15

Hi all, I'm having some issues treating requests within dialogs with gruu enabled with kamailio 4.1.2. I've got the "standard" configuration of WITHIN route with the adition of the next lines: if(is_gruu()){ route(LOCATION); }; before the the RELAY route call in the loose_route section. The "problem" is that the ACK with a pub-gruu on the Req-URI is not properly lookup. In the logs I can see the following statements: 2(4232) DEBUG: registrar [lookup.c:123]: lookup(): looking up pub gruu [urn:uuid:d63b1c4f-d7dc-4f4e-87f1-948123266dc0] 2(4232) DEBUG: registrar [lookup.c:158]: lookup(): 'sam(a)A.B.C.D' Not found in usrloc Where A.B.C.D is the local IP of the UA. Looking at the code, this last line looks like is looking for the "standard" URI (username@domain) instead of using the pub gruu. Am I right with this assumption or am I missing something from the code? As far as I could look, it looks like there's an exit -1 statement in the line 158 of lookup.c which disables the following gruu treatment. Since the username with IP is not registered, this ACK is lost and the sesion is not stablished (lost ACK). Can anyone provide some hints why is this failing? Thanks a lot in advance! Samuel.

3 8

Dialog module is crashing after setting track_cseq_updates to 1
by Fernando Aquilino 29 Jan '15

29 Jan '15

Hi, after setting track_cseq_updates to 1 in module dialog (and calling dlg_manage() from request_route) I'm getting reproducible crashes in both stable (4.2.2) and dev (4.3.0-dev3, 2015-01-28) releases. I made the change to cope with CSeq not being incremented when sending another INVITE with authentication information with by calling uac_auth()/append_branch()/t_relay(). The SIP trunk on the asterisk server I'm forwarding the request to requires so and sends me an "401 Unauthorized" message after the first INVITE. Anyone using it would help me with the configuration? Should I post to the dev list? Thanks for any help, Fernando Aquilino -------------------- kamailio.cfg -------------------- modparam("uac","credential","username:domain:password") modparam("uac","auth_realm_avp","$avp(i:101)") modparam("uac","auth_username_avp","$avp(i:102)") modparam("uac","auth_password_avp","$avp(i:103)") modparam("uac","restore_mode","none") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "dlg_flag", 4) ... failure_route["serial"] { if (t_check_status("401|407")) { if (isflagset(auth_sent)) { xlog("Authentication failed on SIP Trunk: $ru"); } else { #realm $avp(i:101) = "asterisk"; #username $avp(i:102) = "user123"; #password $avp(i:103) = "pass123"; if (uac_auth()) { setflag(auth_sent); t_on_failure("serial"); append_branch(); t_relay(); } } exit; } if (!t_next_contacts()) { exit; } t_on_failure("serial"); t_relay(); } ------------------------------------------------------------ -------------------- debug log -------------------- Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_cseq.c:215]: dlg_cseq_msg_sent(): traking cseq updates Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_hash.c:633]: dlg_lookup(): ref dlg 0x7f9eeb4aec20 with 1 -> 2 Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_hash.c:635]: dlg_lookup(): dialog id=9870 found on entry 2115 Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_cseq.c:249]: dlg_cseq_msg_sent(): uac auth request - cseq inc needed Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_var.c:248]: print_lists(): Internal var-list ((nil)): Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_var.c:258]: print_lists(): Dialog var-list (0x7f9eeb4b2f38): Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9838]: DEBUG: dialog [dlg_var.c:263]: print_lists(): cseq_diff=1 (flags 1) Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9856]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 20 Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9856]: DEBUG: <core> [tcp_main.c:3445]: handle_ser_child(): dead child 10, pid 9838 (shutting down?) Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9856]: DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0x96cfe0, 20, -1, 0x0) fd_no=37 called Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9828]: ALERT: <core> [main.c:727]: handle_sigs(): child process 9838 exited by a signal 11 Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9828]: ALERT: <core> [main.c:730]: handle_sigs(): core was not generated Jan 28 17:40:12 kamailio /usr/sbin/kamailio[9828]: INFO: <core> [main.c:742]: handle_sigs(): terminating due to SIGCHLD ------------------------------------------------------------ -------------------- core dump* -------------------- root@kamailio:/etc/kamailio# gdb kamailio core GNU gdb (GDB) 7.4.1-debian Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...Reading symbols from /usr/lib/debug/.build-id/62/86ce2a5e0241120986e5ec3ed803be5b945076.debug...done. done. [New LWP 3077] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by ` '. Program terminated with signal 11, Segmentation fault. #0 0x00007fbe4ba33dae in memcpy (__len=18446744073709551522, __src=0x7fbe47c9f121, __dest=0x7ffff12ada6d) at /usr/include/x86_64-linux-gnu/bits/string3.h:52 52 return __builtin___memcpy_chk (__dest, __src, __len, __bos0 (__dest)); (gdb) bt #0 0x00007fbe4ba33dae in memcpy (__len=18446744073709551522, __src=0x7fbe47c9f121, __dest=0x7ffff12ada6d) at /usr/include/x86_64-linux-gnu/bits/string3.h:52 #1 dlg_cseq_msg_sent (data=0x7ffff12bda40) at dlg_cseq.c:298 #2 0x0000000000000000 in ?? () (gdb) bt full #0 0x00007fbe4ba33dae in memcpy (__len=18446744073709551522, __src=0x7fbe47c9f121, __dest=0x7ffff12ada6d) at /usr/include/x86_64-linux-gnu/bits/string3.h:52 No locals. #1 dlg_cseq_msg_sent (data=0x7ffff12bda40) at dlg_cseq.c:298 msg = {id = 0, pid = 0, tval = {tv_sec = 0, tv_usec = 0}, fwd_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, rpl_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, first_line = {type = 1, flags = 1, len = 44, u = {request = {method = { s = 0x7fbe47c9f038 "INVITE sip:030636111@192.168.2.151 SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0"..., len = 6}, uri = { s = 0x7fbe47c9f03f "sip:030636111@192.168.2.151 SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0337c5a7"..., len = 27}, version = { s = 0x7fbe47c9f05b "SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0337c5a75bfeebaef0700dd30e433800.1\r\n"..., len = 7}, method_value = 1}, reply = {version = { s = 0x7fbe47c9f038 "INVITE sip:030636111@192.168.2.151 SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0"..., len = 6}, status = { s = 0x7fbe47c9f03f "sip:030636111@192.168.2.151 SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0337c5a7"..., len = 27}, reason = { s = 0x7fbe47c9f05b "SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0337c5a75bfeebaef0700dd30e433800.1\r\n"..., len = 7}, statuscode = 1}}}, via1 = 0x7fbe504594b8, via2 = 0x7fbe50456778, headers = 0x7fbe50459230, last_header = 0x7fbe504598c8, parsed_flag = 134284414, h_via1 = 0x7fbe50458810, h_via2 = 0x7fbe50459320, callid = 0x7fbe5044fee0, to = 0x7fbe504598c8, cseq = 0x7fbe50459020, from = 0x7fbe5045a3b0, contact = 0x0, maxforwards = 0x0, route = 0x0, record_route = 0x7fbe50459230, content_type = 0x0, content_length = 0x0, authorization = 0x0, expires = 0x0, proxy_auth = 0x0, supported = 0x7fbe50459108, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, accept = 0x0, accept_language = 0x0, organization = 0x0, priority = 0x0, subject = 0x0, user_agent = 0x7fbe50459a60, server = 0x0, content_disposition = 0x0, diversion = 0x0, rpid = 0x0, refer_to = 0x0, session_expires = 0x0, min_se = 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x0, identity = 0x0, identity_info = 0x0, pai = 0x0, ppi = 0x0, path = 0x0, privacy = 0x0, body = 0x0, eoh = 0x0, unparsed = 0x7fbe47c9f2d8 "Contact: \"Fernando Aquilino\" < sip:200@192.168.1.72:5060>\r\nAllow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK\r\nContent-Length: 252\r\nContent-Type: application/sdp\r\nMax-F"..., rcv = {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, addr = '\000' <repeats 15 times>}}, src_port = 0, dst_port = 0, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = { sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x0, proto = 0 '\000'}, buf = 0x7fbe47c9f038 "INVITE sip:030636111@192.168.2.151 SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0"..., len = 1357, new_uri = {s = 0x0, len = 0}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = {user = { s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}}, parsed_orig_ruri_ok = 0, parsed_orig_ruri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = { s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = { s = 0x0, len = 0}}, add_rm = 0x0, body_lumps = 0x0, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, hash_index = 0, msg_flags = 0, flags = 0, set_global_address = {s = 0x0, len = 0}, set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec = {s = 0x0, len = 0}, instance = {s = 0x0, len = 0}, reg_id = 0, ruid = {s = 0x0, len = 0}, location_ua = {s = 0x0, len = 0}, ldv = {flow = {decoded = 0, rcv = {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, addr = '\000' <repeats 15 times>}}, src_port = 0, dst_port = 0, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = { sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x0, proto = 0 '\000'}}}} ---Type <return> to continue, or q <return> to quit--- obuf = 0x7ffff12bda40 direction = 1 t = <optimized out> ninc = <optimized out> vinc = <optimized out> dlg = 0x7fbe47c9a8e0 nval = {s = 0xa0ce84 "2", len = 1} pval = <optimized out> tbuf = "INVITE sip:030636111@192.168.2.151 SIP/2.0\r\nRecord-Route: <sip:192.168.2.160;lr;did=34e.1782>\r\nRecord-Route: <sip:192.168.2.163;lr>\r\nCSeq: 1 INVITE\r\nVia: SIP/2.0/UDP 192.168.2.160;branch=z9hG4bKa7ee.0"... tbuf_len = 237 via = <optimized out> __FUNCTION__ = "dlg_cseq_msg_sent" #2 0x0000000000000000 in ?? () No symbol table info available. (gdb) ------------------------------------------------------------ *I generated a core dump by running kamailio from the shell as "kamailio -D -d -E -l 192.168.2.160:5060"

2 1

Issue with non static values for sleep in the Async module
by Will Ferrer 28 Jan '15

28 Jan '15

Hi I am having an issue with the Async module. The async_route function doesn't seem able to take a dynamic value as is it's duration. The module docs states: The sleep parameter represent the number of seconds to suspend the processing of a SIP request. Maximum value is 100. The parameter can be a static integer or a variable holding an integer. Docs here: http://kamailio.org/docs/modules/4.2.x/modules/async.html#async.f.async_rou… However when I try: async_route("WITHINDLG_DELAYED", $avp(buffer_duration) ); I get the following error when trying to restart kamailio: parse error in config file /etc/kamailio/switchsoftk amailio/includes/deployments/develop/callprocessing/../../../default.cfg, line 866, column 41-61: function async_route: parameter 2 is not constant ... failed! I also tried with $var instead of an AVP, and even tried having perl call the function (since we have gotten around this sort of limitation in the past using perl). Perl gives: Jan 24 03:27:07 develop-callprocessing /usr/sbin/kamailio[1424]: ERROR: app_perl [kamailioxs.xs:289]: moduleFunc(): Module function 'async_route' is unsafe. Call is refused. Jan 24 03:27:07 develop-callprocessing /usr/sbin/kamailio[1424]: ERROR: app_perl [kamailioxs.xs:986]: XS_Kamailio__Message_moduleFunction(): calling module function 'async_route' failed. Missing loadmodule? I have gotten around this issue with a large if else structure like the following: if ($avp(buffer_duration) == 1) { async_route("WITHINDLG_DELAYED", 1); } else if ($avp(buffer_duration) == 2) { async_route("WITHINDLG_DELAYED", 2); ......... This is messy but works for us. The bug is not blocking our progress due to this work around but I wanted to post it any way incase it helps improve the module. All the best. Will Ferrer Switchsoft

2 2

CDRs for failed calls
by Mickael Marrache 28 Jan '15

28 Jan '15

Hi, Is there a way to write CDRs for failed calls? I tried calling the acc_db_request command from my script, but it inserts a transaction log not a CDR. Thanks, Mickael

2 3

How the README file in each module's directory is generated?
by Ding Ma 28 Jan '15

28 Jan '15

Is there a doc tool? Or just use an editor? Thanks

1 0

usrloc and MongoDB
by Mickael Marrache 28 Jan '15

28 Jan '15

Hi, Is it possible to save user location data in a MongoDB database? If not, are you guys using p_usrloc in your projects? Can you please give me some feedback? Thanks, Mickael

2 3

How to run Kamailio as a stateful proxy
by Thomas Bäumer 28 Jan '15

28 Jan '15

Hello all, we are are a small network-provider and selling internet via FTTC and FTTH. To complete the portfolio we also sell telephony via a rented ClassV Service with provisioning API, but its not our system and not in our network. To get more informations about Registration-Status and calls I have now installed a kamailio as a statefull proxy between the subscriber and the softwitch: |----------| |subscriber|\ --------- |----------| \ /| Mobile| \ -------------- / --------- |----------| \ ------------ | Class V | / -------- |subscriber|---------| Kamailio |--------| Softswitch |-------| PSTN | |----------| / ------------ | | \ -------- / -------------- \ --------- |----------| / \| World | |subscriber|/ --------- |----------| I am really new in kamailio, so I just use the configuration named "Edge Proxy Configuration" shown in the documentation to the outbound module, see http://kamailio.org/docs/modules/4.2.x/modules/outbound.html example 1.1. We dont need NAT and STUN because all our subscribers get official IPv4 addresses. Outgoing calls (subscriber -> PSTN) are working fine, all SIP messages pass the Kamailio Proxy. But incoming Calls (World -> subscriber) miss the proxy and will be routed directly from the Softwitch to the subscriber. I think because of the Contact-Header in the subscriber REGISTER: Contact: <sip:49xxxxxxyyyyyy@134.xx.yy.zz;uniq=E41120B46C92D3818E72B> This is the IP of the subscriber FritzBox. So I just added some lines to replace the Contact-Header in the REGISTER to get all incoming Calls from the Softswitch to the Kamailio: if (is_method("REGISTER")) { if (src_ip==CUSTOMER_NET) { remove_hf("Route"); add_path(); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; if ($ct != $null) { subst('/^Contact:(.*)@[a-zA-Z0-9.]+(.*)$/Contact:\1@KAMAILIO_IP\2/ig'); } } } and now the Contact-Header looks better: Contact: <sip:49xxxxxxyyyyyy@KAMAILIO_IP;uniq=E41120B46C92D3818E72B> Now the subscriber is registered on the Softswitch with the IP from the Kamailio. And now the Kamailio get the incoming Call, but directly route it back to the Softwitch and the call failed. Now my questions: - Is the "Edge Proxy Configuration" right for this setup? - How the Kamailio keep all registered IPs and Lines to route an incoming Call to the right subscriber? - Why the Kamailio route an incoming call directly back to the Softswitch? Thanks a lot. Regards Thomas

1 0

Can't start Kamailio with both db_postgres and tls
by Øyvind Kolbu 28 Jan '15

28 Jan '15

Hi, I'm trying to add tls to our services, but seem hindered by "something". When enabling the tls module it seems to turn off the SSL setting in postgresql, and since we require SSL on our postgres servers Kamailio can't connect and it fails to start. If I disable WITH_TLS it will start. If I comment out the sqlops modparam it will also start. Kamailio on RHEL6, from ~4.1.6 git rev 2f690887b45dbc49a8038b1fa041d47cd9ae39ea. # kamailio -V version: kamailio 4.1.6 (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, 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: unknown compiled on 16:48:10 Sep 26 2014 with gcc 4.4.7 Very simpel config file: debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=2 #!define TLSFILE "/kamailio/tls-fooserver1.cfg" port=5060 #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif include_file "/kamailio/databases.cfg" loadmodule "tm.so" # Transaction (stateful) module loadmodule "tmx.so" # Extensions from Kamailio TM module loadmodule "sl.so" # Stateless replier module loadmodule "rr.so" # Record-Route and Route module loadmodule "pv.so" # Module holding Pseudo-Variables loadmodule "sqlops.so" # SQL operations loadmodule "db_postgres.so" # POSTGRES-backend for database API module #!ifdef WITH_TLS loadmodule "tls.so" #!endif modparam("sqlops","sqlcon",SQLOPS_DATA) #!ifdef WITH_TLS modparam("tls", "config", TLSFILE) #!endif route{ exit; } SQLOPS_DATA is just a normal "data=>postgres://user:pass@server/db". TLSFILE contains: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /ssl/key certificate = /ssl/cert ca_list = /ssl/terena_chain2.pem [client:default] verify_certificate = no require_certificate = no From messages on startup: Dec 15 13:50:19 fooserver1 kamailio[12115]: INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... Dec 15 13:50:19 fooserver1 kamailio[12115]: INFO: <core> [tcp_main.c:4836]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected) Dec 15 13:50:19 fooserver1 kamailio[12118]: WARNING: <core> [daemonize.c:352]: daemonize(): pid file contains old pid, replacing pid Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_mod.c:346]: mod_init(): With ECDH-Support! Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_mod.c:349]: mod_init(): With Diffie Hellman Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_init.c:549]: init_tls_h(): tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, compression: on Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_init.c:557]: init_tls_h(): tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1e-fips 11 Feb 2013" (0x1000105f), kerberos support: on, zlib compression: on#012 compiler: gcc -fPIC -DO PENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -DTERMIO -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack -DPURIFY -DOPEN SSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: WARNING: tls [tls_init.c:611]: init_tls_h(): tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [cfg/cfg_ctx.c:613]: cfg_set_now(): INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [udp_server.c:176]: probe_max_receive_buffer(): INFO: udp_init: SO_RCVBUF is initially 124928 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [udp_server.c:227]: probe_max_receive_buffer(): INFO: udp_init: SO_RCVBUF is finally 249856 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [udp_server.c:176]: probe_max_receive_buffer(): INFO: udp_init: SO_RCVBUF is initially 124928 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [udp_server.c:227]: probe_max_receive_buffer(): INFO: udp_init: SO_RCVBUF is finally 249856 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSs<default>: tls_method=9 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSs<default>: certificate='/path/ssl/fooserver1.uio.no.crt' Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSs<default>: ca_list='/voip/packages/mgmt/ssl/terena_chain2.pem' Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSs<default>: crl='(null)' Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSs<default>: require_certificate=0 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSs<default>: cipher_list='(null)' Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSs<default>: private_key='/path/ssl/fooserver1.uio.no.key' Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSs<default>: verify_certificate=0 Dec 15 13:50:19 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSs<default>: verify_depth=9 Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:670]: set_verification(): TLSs<default>: No client certificate required and no checks performed Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:275]: fill_missing(): TLSc<default>: tls_method=9 Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:287]: fill_missing(): TLSc<default>: certificate='(null)' Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:294]: fill_missing(): TLSc<default>: ca_list='(null)' Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:301]: fill_missing(): TLSc<default>: crl='(null)' Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:305]: fill_missing(): TLSc<default>: require_certificate=0 Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:312]: fill_missing(): TLSc<default>: cipher_list='(null)' Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:319]: fill_missing(): TLSc<default>: private_key='(null)' Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:323]: fill_missing(): TLSc<default>: verify_certificate=0 Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:326]: fill_missing(): TLSc<default>: verify_depth=9 Dec 15 13:50:23 fooserver1 /usr/sbin/kamailio[12118]: INFO: tls [tls_domain.c:673]: set_verification(): TLSc<default>: Server MAY present invalid certificate Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: db_postgres [km_pg_con.c:82]: db_postgres_new_connection(): SSL SYSCALL error: Resource temporarily unavailable#012FATAL: no pg_hba.conf entry for host "129.240.1.1", user "foo_test_user", database " foo_test", SSL off#012 Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: db_postgres [km_pg_con.c:95]: db_postgres_new_connection(): cleaning up 0x7fce98be0c78=pkg_free() Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: sqlops [sql_api.c:166]: sql_connect(): failed to connect to the database [data] Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: <core> [sr_module.c:927]: init_mod_child(): init_mod_child(): Error while initializing module sqlops (/usr/lib64/kamailio/modules/sqlops.so) Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: <core> [pt.c:490]: fork_tcp_process(): ERROR: fork_tcp_process(): init_child failed for process 7, pid 12125, "tcp receiver (generic) child=0" Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12125]: ERROR: <core> [tcp_main.c:4962]: tcp_init_children(): ERROR: tcp_main: fork failed: Success Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12118]: ALERT: <core> [main.c:774]: handle_sigs(): child process 12125 exited normally, status=255 Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12118]: INFO: <core> [main.c:792]: handle_sigs(): INFO: terminating due to SIGCHLD Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12126]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12124]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12123]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12122]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12121]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12120]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Dec 15 13:50:26 fooserver1 /usr/sbin/kamailio[12127]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received Any ideas? Best regards, Øyvind

4 22

Registration to Kamailio World 2015 is open
by Daniel-Constantin Mierla 28 Jan '15

28 Jan '15

Hello, I would like to announce that registration to Kamailio World 2015 is open: - http://conference.kamailio.com/k03/registration/ The schedule is still work in progress, a preview list with several speakers is available at: - http://conference.kamailio.com/k03/speakers/ The structure of the event contains: - pre-conference technical tutorials in the afternoon of May 27, 2015 - conference presentations during May 28-29, 2015 More details at: - http://conference.kamailio.com/k03/schedule/ Looking forward to meeting many of you and have a great time again in Berlin! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Trouble getting start time / call duration from dialog module
by Will Ferrer 28 Jan '15

28 Jan '15

Hi All I am in need of being able to see what the duration of the call was at the time of hang out. I tried turning on the dialog module, but the result is that the values I need are either null or always show as zero. I tried to follow the suggestions in the thread about this here: http://lists.sip-router.org/pipermail/sr-users/2010-October/065889.html In the end my config looks like this: loadmodule "dialog.so" ... modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_match_mode", 1) ... request_route { if (is_method("INVITE") && (! has_totag() ) ) { dlg_manage(); } if (is_method("BYE")) { $var(elapsed) = ( $TV(s) - $dlg(start_ts) ); xlog ("L_INFO", "request_route DIALOG TEST: Completed $dlg(from_uri) to $dlg(to_uri), elapsed: $var(elapsed), now seconds: $TV(s), dlg start time: $dlg(start_ts), DLG_lifetime: $DLG_lifetime"); } .... Note that I put at the top of the request route just for testing purposes The result I get in the log is this: INFO: <script>: request_route DIALOG TEST: Completed sip:willf1976Test@develop-sbc.switchsoft.com to sip:+18054515526@develop-sbc.switchsoft.com, elapsed: 1421305210, now seconds: 1421305210, dlg start time: 0, DLG_lifetime: <null> I was wondering if any one could point out to me what I have set up wrong. Thanks in advance All the best. Will Ferrer Switchsoft

3 9

GeoIP Module and IPv6 seems not to match
by Jöran Vinzens 28 Jan '15

28 Jan '15

Hi All, i'm trying to get GEO IP Location for an IPv6 Setup. We downloaded the GeoLiteCityv6.dat from maxmind and referred to it in modparams of geoip. If i now place a call from an V6 User and do something like: if(geoip_match("$si", "src")) { xlog("L_NOTICE", "Call comes from IP '$si' ($gip(src=>cc)) CID=$ci F=$fU URI=$ru\n"); append_hf("X-GeoIP: $gip(src=>cc)\r\n"); } it will not match for any reason if i xlog the src IP it look like: "Src IP = 2A01:abc:321:123:0:0:0:2" so it should match to the GEO IP Database. In CSV version of geo IP database there is an entry for my IP Address. Since the Module is not that verbose, i cannot tell whether the entry exists or not. Just the result in my "if" is False. Has anybody experience with GEOIP and IPv6? thanks Best regards, Jöran

3 4

Rpc question
by Pars3c 27 Jan '15

27 Jan '15

Hi, I need to request to kamailio the list of all the active dialog (with some dialog variable). Now , i'm doing it with a program that call the proxy with the xmlrpc module. The problem is that when it arrive to 350-400 calls, the reply fail because it has no memory free. Now ,i have setted the pkg memory to 12MB but at peak it reply with error. Now i shoul increase it. Is there another method to optimize the request of these data, to avoid every time to increase the memory parameter? Thanks to all

1 0

Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
by rahul.ultimate 27 Jan '15

27 Jan '15

Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs. This architectue uses a TURN server and the browser is chrome with latest updates. The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do).. Also, does it necessitates to have Apache ruuning https on 443 ? Thanks in advance Sent from Samsung Mobile -------- Original message -------- From: Gonzalo Gasca Meza <gascagonzalo(a)gmail.com> Date:27/01/2015 4:07 AM (GMT+05:30) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server? Are you using latest Chrome version or FF ? A working sample config using the following architecture: https://github.com/spicyramen/llamato/tree/LlamatoReg signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip udp--> * media: sipml5 ------------------------------------------------------------------------> * On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR <rahul.ultimate(a)gmail.com> wrote: Hi Richard, Thanks for spending some cycles on it. It is OpenSSL 1.0.1e-fips 11 Feb 2013 On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs <rfuchs(a)sipwise.com> wrote: On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - "STUN: using this candidate" "Successful STUN binding request from .." "SRTP output wanted, but no crypto suite was negotiated" This is fairly strange: Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Are you running a very old OpenSSL version by any chance? cheers _______________________________________________ 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 -- Warm Regds. MathuRahul _______________________________________________ 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

3 2

tcp_max_connection to 4096
by Varghese Paul 27 Jan '15

27 Jan '15

Hi Team, We are seeing some errors in our kamilio for TCP max conn (ERROR) : 2048 (the default). We are thinking to double the TCP connection for our kamailio registrar server. tcp_max_connections=4096 Is there any performance issue if we double the tcp_max_connections ?. Currently we are setting these parameters for TCP. tcp_connection_lifetime=3605 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 Do we need to tune any other variable if we are setting max tcp connections to 4096 for better performance ? Thanks for looking into this. Regards Varghese Paul

2 1

remote mysql server
by Slava Bendersky 27 Jan '15

27 Jan '15

Hello Everyone, I am trying start kamailio 4.2 with remote mysql server and on debug I see DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/kamailio but in cfg defined actual mysql server ip and not localhost 0(28885) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: found <db_bind_api> in module db_mysql [/usr/lib64/kamailio/modules/db_mysql.so] 0(28885) DEBUG: <core> [db.c:213]: db_bind_mod(): using db bind api for db_mysql 0(28885) DEBUG: <core> [db.c:318]: db_do_init2(): connection 0x7faca59eced8 not found in pool 0(28885) DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/kamailio 0(28885) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2 "No such file or directory") 0(28885) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(28885) ERROR: dialog [dlg_db_handler.c:147]: init_dlg_db(): unable to connect to the database Slava.

4 4

Re: [SR-Users] Wrong dialog selected in on_reply/failure route in case of spirals.
by Julia Boudniatsky 27 Jan '15

27 Jan '15

Hello Daniel, Today I installed kamailio from GIT last devel 4.3.0-dev3. *The problem is solved!* Thank you so much for your help ! Julia On Mon, Jan 26, 2015 at 8:38 PM, Julia Boudniatsky <juliabo(a)gmail.com> wrote: > Yes it /usr/local/sbin/kamailio, > I haven't internet in the test server. > I get git from another server and copy received directory kamailio to test > server, then make cfg/all/install. > > BR, > Julia > > On Mon, Jan 26, 2015 at 8:10 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> I did the same right now and I get: >> >> version: kamailio 4.3.0-dev3 (x86_64/darwin) 1b334f >> >> Can you check if you have another instance installed on a different path >> that takes precendence? >> >> Do: >> >> which kamailio >> >> When installed from sources, it should be: >> >> /usr/local/sbin/kamailio >> >> Cheers, >> Daniel >> >> >> >> On 26/01/15 18:55, Julia Boudniatsky wrote: >> >> I used from your link * http://www.kamailio.org/wiki/#installation >> >> >> - Install Kamailio Devel Version From GIT >> <http://www.kamailio.org/wiki/install/devel/git> >> >> #mkdir -p /usr/local/src/kamailio-devel >> #cd /usr/local/src/kamailio-devel >> #git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio >> kamailio >> #cd kamailio >> >> BR, >> >> Julia >> >> On Mon, Jan 26, 2015 at 7:04 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> that is not the last devel version, it should be with -dev3. How did you >>> get the sources? >>> >>> I get: >>> >>> version: kamailio 4.3.0-dev3 (x86_64/darwin) 1b334f >>> >>> And yes, if you still get the error, send the logs with the description. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 26/01/15 17:16, Julia Boudniatsky wrote: >>> >>> Hello Daniel, >>> In devel installed, I received the same problem. >>> >>> kamailio -V >>> version: kamailio 4.3.0-dev2 (x86_64/linux) ecd5c5 >>> 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: ecd5c5 >>> compiled on 18:02:05 Jan 26 2015 with gcc 4.4.6 >>> >>> Do you want a log files? >>> >>> Thank you, >>> >>> Julia >>> >>> On Mon, Jan 26, 2015 at 12:47 PM, Daniel-Constantin Mierla < >>> miconda(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> none of existing releases are good because the patches are only in git >>>> branches, added after the last relesea (I added it iver this weekend). You >>>> must install from git, as pointed by one of the tutorials at: >>>> >>>> * http://www.kamailio.org/wiki/#installation >>>> >>>> You must use devel (master), 4.2 (v4.2.x) or 4.1 (v4.1.x) branches. >>>> >>>> We will have new releases in the near future, like 1-2 weeks from now, >>>> but by then I hope to get this bug sorted out. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 26/01/15 11:42, Julia Boudniatsky wrote: >>>> >>>> Hello Daniel, >>>> I installed last 4.2 >>>> >>>> version: kamailio 4.2.1 (x86_64/linux) d80dfc >>>> 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: d80dfc >>>> compiled on 10:06:01 Jan 26 2015 with gcc 4.4.6 >>>> >>>> >>>> *Problem in call with Call-ID : 5-5028(a)10.25.153.150 >>>> <5-5028(a)10.25.153.150> * >>>> >>>> *Short log* >>>> >>>> *dialog [3748:5628]* >>>> >>>> *100 trying* >>>> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: >>>> DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:637]: parse_msg(): status: <100> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Trying> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: >>>> tag=9313591363960470767 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header >>>> reached, state=29 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: >>>> <To> [71]; uri=[sip:039951004@10.25.153.149:5060;user=phone] >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [ >>>> <sip:039951004@10.25.153.149:5060;user=phone>] >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq >>>> <CSeq>: <1> <INVITE> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 234, >>>> <received> = <10.25.153.149>; state=6 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, >>>> <branch> = <z9hG4bKbde6.9c95fbe06c33a54c13ef2ebc4b519dcc.0>; state=9 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:2715]: parse_via(): parse_via: next_via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, >>>> <branch> = <z9hG4bKbde6.785d27b1.0>; state=9 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:2715]: parse_via(): parse_via: next_via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, >>>> <branch> = <z9hG4bKbde6.5be423cd6c755cf13d68e74be1f301a1.0>; state=9 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:2715]: parse_via(): parse_via: next_via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, >>>> <branch> = <z9hG4bK-5028-5-0>; state=16 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, >>>> state=5 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via >>>> found, flags=2 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is >>>> the first via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [receive.c:154]: receive_msg(): After parse_msg... >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=132 global >>>> id=131 T start=0xffffffffffffffff >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash >>>> 28379 label 0 branch 0 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply >>>> matched (T=0x7f3ec89a3e68)! >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=132 global >>>> id=132 T end=0x7f3ec89a3e68 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> tm [t_reply.c:2200]: reply_received(): DEBUG: reply_received: org. status >>>> uas=100, uac[0]=0 local=0 is_invite=1) >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: >>>> tag=5028SIPpTAG005 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header >>>> reached, state=29 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : >>>> content_length=0 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header >>>> *Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: >>>> ERROR: KAMnet: processing INVITE >>>> (5-5028(a)10.25.153.150/5028SIPpTAG005/9313591363960470767/SIP/2.0/UDP >>>> <http://5-5028@10.25.153.150/5028SIPpTAG005/9313591363960470767/SIP/2.0/UDP> >>>> 10.25.153.149:5060;received=10.25.153.149;branch=z9hG4bKbde6.9c95fbe06c33a54c13ef2ebc4b519dcc.0,SIP/2.0/UDP >>>> 10.25.154.104;branch=z9hG4bKbde6.785d27b1.0,SIP/2.0/UDP >>>> 10.25.153.149:5060;branch=z9hG4bKbde6.5be423cd6c755cf13d68e74be1f301a1.0,SIP/2.0/UDP >>>> 10.25.153.150:5062;branch=z9hG4bK-5028-5-0)[sip:39988004@10.25.153.150 >>>> <sip%3A39988004(a)10.25.153.150>;user=phone=>sip:039951004@10.25.153.149:5060;user=phone]* >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> dialog [dlg_hash.c:711]: internal_get_dlg(): ref dlg 0x7f3ec89b6d28 with 1 >>>> -> 2 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> dialog [dlg_hash.c:714]: internal_get_dlg(): *dialog >>>> callid='5-5028(a)10.25.153.150 <5-5028(a)10.25.153.150>' found* >>>> * on entry 3748, dir=1* >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20649]: DEBUG: >>>> dialog [dlg_var.c:243]: print_lists(): Internal var-list ((nil)): >>>> >>>> >>>> *301 Moved Permanently* >>>> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <301> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Moved >>>> Permanently> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: >>>> add_param: tag=9313591363960470767 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of >>>> header reached, state=29 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: >>>> get_hdr_field: <To> [71]; uri=[ >>>> sip:039951004@10.25.153.149:5060;user=phone] >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [ >>>> <sip:039951004@10.25.153.149:5060;user=phone>] >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: >>>> cseq <CSeq>: <1> <INVITE> >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param >>>> type 234, <received> = <10.25.153.149>; state=6 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param >>>> type 232, <branch> = <z9hG4bKbde6.9c95fbe06c33a54c13ef2ebc4b519dcc.0>; >>>> state=9 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:2715]: parse_via(): parse_via: next_via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param >>>> type 232, <branch> = <z9hG4bKbde6.785d27b1.0>; state=9 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:2715]: parse_via(): parse_via: next_via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param >>>> type 232, <branch> = <z9hG4bKbde6.5be423cd6c755cf13d68e74be1f301a1.0>; >>>> state=9 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:2715]: parse_via(): parse_via: next_via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param >>>> type 232, <branch> = <z9hG4bK-5028-5-0>; state=16 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header >>>> reached, state=5 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: >>>> Via found, flags=2 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: >>>> this is the first via >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=136 >>>> global id=135 T start=0xffffffffffffffff >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: >>>> hash 28379 label 0 branch 0 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: >>>> reply matched (T=0x7f3ec89a3e68)! >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=136 >>>> global id=136 T end=0x7f3ec89a3e68 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: tm [t_reply.c:2200]: reply_received(): DEBUG: reply_received: org. >>>> status uas=100, uac[0]=100 local=0 is_invite=1) >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: >>>> add_param: tag=5028SIPpTAG005 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of >>>> header reached, state=29 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: >>>> get_hdr_body : content_length=0 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of >>>> header >>>> *Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[20653]: >>>> ERROR: KAMnet: processing INVITE >>>> (5-5028(a)10.25.153.150/5028SIPpTAG005/9313591363960470767/SIP/2.0/UDP >>>> <http://5-5028@10.25.153.150/5028SIPpTAG005/9313591363960470767/SIP/2.0/UDP> >>>> 10.25.153.149:5060;received=10.25.153.149;branch=z9hG4bKbde6.9c95fbe06c33a54c13ef2ebc4b519dcc.0,SIP/2.0/UDP >>>> 10.25.154.104;branch=z9hG4bKbde6.785d27b1.0,SIP/2.0/UDP >>>> 10.25.153.149:5060;branch=z9hG4bKbde6.5be423cd6c755cf13d68e74be1f301a1.0,SIP/2.0/UDP >>>> 10.25.153.150:5062;branch=z9hG4bK-5028-5-0)[sip:39988004@10.25.153.150 >>>> <sip%3A39988004(a)10.25.153.150>;user=phone=>sip:039951004@10.25.153.149:5060;user=phone]* >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: dialog [dlg_hash.c:654]: dlg_lookup(): ref dlg 0x7f3ec895f650 with 1 >>>> -> 2 >>>> Jan 26 10:20:49 vm-KAMnet-dev01 /usr/local/sbin/kamailio[*20653*]: >>>> DEBUG: dialog [dlg_hash.c:656]: dlg_lookup(): *dialog id=2299 found on >>>> entry 422* >>>> >>>> >>>> >>>> Thank you, >>>> Julia >>>> >>>> >>>> >>>> On Sun, Jan 25, 2015 at 7:47 PM, Daniel-Constantin Mierla < >>>> miconda(a)gmail.com> wrote: >>>> >>>>> I looked at the logs, but I don't see any log messages like: >>>>> >>>>> ... resetting the local dialog shortcuts ... >>>>> >>>>> I did some tests locally and it is printed even after processing 100 >>>>> Trying. The log message appears after 301 or 200ok as well. Are you sure >>>>> you run the latest version from the git branch (4.1, 4.2, or master) or all >>>>> log messages were sent over? >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> On 25/01/15 12:27, Julia Boudniatsky wrote: >>>>> >>>>> >>>>> *dialog [3692:448]* >>>>> >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: >>>>> ERROR: KAMnet: processing INVITE ( >>>>> 2-17858(a)10.25.153.150/17858SIPpTAG002/9313591363960453592/SIP/2.0/UDP >>>>> 10.25.153.149:5060;received=10.25.153.149;branch=z9hG4bK3ae5.3e00b5a33eaac375191362e27e930fba.0,SIP/2.0/UDP >>>>> 10.25.154.104;branch=z9hG4bK3ae5.89c39796.0,SIP/2.0/UDP 10.25.153.149:5060;branch=z9hG4bK3ae5.42e72ca36c3875261747250c959a8d70.0,SIP/2.0/UDP >>>>> 10.25.153.150:5062;branch=z9hG4bK-17858-2-0)[ >>>>> sip:39988001@10.25.153.150;user=phone=> >>>>> sip:039951001@10.25.153.149:5060;user=phone] >>>>> >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: DEBUG: >>>>> dialog [dlg_hash.c:872]: dlg_unref(): unref dlg 0x7faf1c35fa48 with 1 -> 1 >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: >>>>> NOTICE: KAMnet: REPLY_TO_SOURCE: 301 Q.850[410] [1 INVITE] from C5tst to >>>>> IVRcustomer >>>>> ru=<null> >>>>> fu=sip:39988001@10.25.153.150;user=phone >>>>> tu=sip:039951001@10.25.153.149:5060;user=phone >>>>> ai=<null> >>>>> ci=2-17858(a)10.25.153.150 >>>>> ct=sip:D39739951001@10.25.153.150 >>>>> cl=0 >>>>> branch[1] >>>>> >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: >>>>> DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=8 >>>>> global id=8 T start=0x7faf1c372048 >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: DEBUG: >>>>> tm [t_lookup.c:1144]: t_check_msg(): DEBUG: t_check_msg: T already found! >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: DEBUG: >>>>> tm [tm.c:1035]: t_check_status(): DEBUG:t_check_status: checked status is >>>>> <301> >>>>> Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: DEBUG: >>>>> dialog [dlg_hash.c:654]: dlg_lookup(): ref dlg 0x7faf1c35fa48 with 1 -> 2 >>>>> *Jan 25 11:41:31 vm-KAMnet-dev01 /usr/local/sbin/kamailio[2588]: >>>>> DEBUG: dialog [dlg_hash.c:656]: dlg_lookup(): dialog id=9780 found on entry >>>>> 1905* >>>>> >>>>> *Thank you,* >>>>> >>>>> *Julia* >>>>> >>>>> On Sun, Jan 25, 2015 at 12:48 PM, Daniel-Constantin Mierla < >>>>> miconda(a)gmail.com> wrote: >>>>> >>>>>> Can you point shortly again the lines where the wrong matching >>>>>> happens? Like last time -- it helped to spot in the file faster and analyze >>>>>> from there. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> >>>>>> On 25/01/15 11:44, Julia Boudniatsky wrote: >>>>>> >>>>>> >>>>>> >>>>>> On Sun, Jan 25, 2015 at 12:03 AM, Daniel-Constantin Mierla < >>>>>> miconda(a)gmail.com> wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> got the time to analyze your last logs and I discovered that the >>>>>>> previous patch was not resetting properly the dialog shortcuts. I pushed a >>>>>>> fix, can you try again with the latest version from one of the branches: >>>>>>> master, 4.2 or 4.1?Let me know the results. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> On 15/01/15 23:42, Daniel-Constantin Mierla wrote: >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I looked at the logs, but I cannot see the log message I added with >>>>>>> the patch for reseting local dialog shortcuts after onreply and failure >>>>>>> route. It shuld be something like: >>>>>>> >>>>>>> "... resetting the local dialog shortcuts" >>>>>>> >>>>>>> Are you sure you are running the latest branch 4.1? You can send the >>>>>>> output of kamailio -v to double check here. >>>>>>> >>>>>>> I separated the logs per process and looked when a 301 was received, >>>>>>> apparently the ones for Call-ID: 4-21712 are matching the same dialog. I >>>>>>> found something not printing the same kind of logs like for other 301 for >>>>>>> Call-ID: 2-21712. If you can point the line in the logs where you think it >>>>>>> was the wrong dialog match, I can investigate from there, but now I >>>>>>> couldn't find any lead. >>>>>>> >>>>>>> Anyhow, if it is the latest branch 4.1 and the problem still >>>>>>> persists, can you add the following xlog at the beginning of main >>>>>>> request_route (or the route {...} ), plus in each reply_route, >>>>>>> onreply_route and failure_route: >>>>>>> >>>>>>> xlog("processing $rm ( $ci / %ft / $tt / $hdr(Via) ) [$fu => >>>>>>> $tu]\n"); >>>>>>> >>>>>>> Then redo the tests to catch the issue and send me the new logs. >>>>>>> Because you use sipp with tags like '4', it is hard to search on a single >>>>>>> tocken and I want to have the all message attribute when a relevant route >>>>>>> block is executed. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> On 14/01/15 15:07, Daniel-Constantin Mierla wrote: >>>>>>> >>>>>>> Hello Julia, >>>>>>> >>>>>>> ok, I will look over the logs very soon. >>>>>>> >>>>>>> As a short note for the future, make the archives zip or tgz, being >>>>>>> more common formats that can be unarchived everywhere without installing >>>>>>> additional app, being handy to open even on small devices. Now it is ok >>>>>>> with rar, being at the desk. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> On 14/01/15 14:43, Julia Boudniatsky wrote: >>>>>>> >>>>>>> Hello Daniel, >>>>>>> >>>>>>> Please find in attached file the debug 3 log. >>>>>>> Failed call with Call-ID: 4-21712(a)10.25.153.150. >>>>>>> The problem occurred with all modes of "dlg_match_mode " >>>>>>> The same load test in 3.3 executed without failures. >>>>>>> >>>>>>> Thank you, >>>>>>> >>>>>>> Julia. >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> >>>>>>> >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>> >>>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>> >>>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

1 0

Re: [SR-Users] Fosdem 2015
by DanB 27 Jan '15

27 Jan '15

I will be there this year also, if someone prepares an event. DanB.

3 4

usage of the select function
by Олег Першин 27 Jan '15

27 Jan '15

Hi! Please help me to figure out how to use "@select" or "$sel" in routes. For example: there is a sip message at the entrance of kamailo. This message contains an authorization header. Authorization: Digest username="oleg.pershin", realm="178.62.221.112", nonce="VMT1cVTE9EVT/6/MLuDZR8dA4g1Mfir4", uri="sip:178.62.221.112", response="6eb5427433ef0b51832829f9fd4340ea", algorithm=MD5 Authentication Scheme: Digest username="oleg.pershin" realm="178.62.220.111" nonce="VMT1cVTE9EVT/6/MLuDZR7dA4g1Mfir4" uri="sip:178.62.220.111" response="6eb5327433ef0b51832829f9fd4340ea" algorithm=MD5 So I want to get a value from the "nonce" with "@select" or "$sel" functions . How will it look like? Best regards, Oleg Pershin.

2 1

When does transaction logging trigger
by Ryan Brindley 27 Jan '15

27 Jan '15

Hey gents, In relation to my attempts to get transaction logging successful, I'm curious to know when transaction logging triggers. My current theory is that something I'm doing in my SIPp uac test messages (what's being sent to Kamailio) is preventing the transaction from fully completing and thus preventing logging. I'm currently setting up an Asterisk box to test this theory, but I'd still like to know more about this and see if I can understand more of what's going on under the hood. Thanks in advance, Ryan Brindley

2 6

lookup_branches and rtpengine_manage
by Yuriy Gorlichenko 27 Jan '15

27 Jan '15

Hello. I need parallel forking calls with the same username. (Call to all contacts with name for example User123), my endpoints may be WebSocket based and standart UDP endpoints. And I use rtpengine_manage for nmanaging calls wor webphones and standart softh/hard phones. I get all contacts manually and than at the branch route set rtpengine_manage settings for every call. It works fine but it works for one kamailio server. When I use 2 kamailio servers as load balansers Server that handle call get all endpoints from location but call to only one, that registred only at ths server for example I call user123 I have 3 contacts user123(a)1.2.3.4 - was registered at kamailio 1 user123(a)3.2.1.4 - was registered at kamailio 2 user123(a)4.3.2.1 - was registered at kamailio 1 So if call goes through kamailio 1 it call only to user123(a)1.2.3.4 and user123(a)4.3.2.1 I use this settings for usrloc at 2 kamailios to share all table between 2 servers modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "user_column", "username") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "cflags_column", "cflags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "path_column", "path") modparam("usrloc", "ruid_column", "ruid") modparam("usrloc", "instance_column", "instance") modparam("usrloc", "use_domain", 1) and this code for calling them [GET_CONTACTS] { sql_query("ca", "select contact from location where username='$tU'", "ra"); xlog("rows: $dbr(ra=>rows) cols: $dbr(ra=>cols)\n"); if($dbr(ra=>rows)>0){ $var(i)=0; while($var(i)<$dbr(ra=>rows)){ xlog("L_INFO","SQL query return contact {$dbr(ra=>[$var(i),0])} for {$tU} at step {$var(i)}\n"); if ($dbr(ra=>[$var(i),0])=~"transport=ws"){ xlog("L_INFO", "This is a Websocket call to endpoint"); sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); $du=$var(recieved); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}. Destination is {$du}\n"); append_branch("sip:$tU@$(du{s.select,1,:})"); } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $var(recieved)=''; sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO", "SQL query return RECIEVED {$var(recieved)}"); if ($var(recieved)==0){ xlog("L_INFO", "Recieved string is EMPTY"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); } else { xlog("L_INFO", "Recieved string is {$var(recieved)}"); $du=$var(recieved); } $var(UDP_contact)="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); append_branch("sip:$tU@$(du{s.select,1,:})"); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}}. Destination is {$du}\n"); } $var(i) = $var(i) + 1; } } t_on_branch("1"); return; } } } branch_route[1]{ if($du=~"transport=ws"){ xlog("L_INFO","Websocket Branch is {$du} for {$tU}\n"); rtpengine_manage("internal extenal force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else{ xlog("L_INFO","UDP Branch is {$du)} for {$tU}\n"); rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } When it try to branch endpoint without registration at server that handle call I get errors that tm module can not build Via header **via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [msg_translator.c:1725]: build_req_buf_from_sip_req(): could not create Via header* *ERROR: <core> [forward.c:607]: forward_request(): ERROR: forward_request: building failed* * UDP calls get errors something like above (sorry than can not share error code, This situation not often). So I think I have this trouble because I use manually handling call and tried to substitute to lookup_branches function. but I have no Idea how to set rtpengine_manage paraments for each endpoint depending this is websocket or standart call. IF there is write problem for callings thhrough 2 kamailios as loadbalansers pleas let me know about how to set rtpengine_manage parametrs wor endpoints for every fork. If not- can you tell me how I can call to all endpoints endepending of registration server (kamailio 1 or 2). Thanks.

1 0

Logging All Transactions
by Ryan Brindley 26 Jan '15

26 Jan '15

I've been working to try to get kamailio (4.1) to log every transaction to file, but can't seem to do it -- or even get it to reliably log. What the heck am i missing? It seems Kamailio logs the first time i try it on a day, but then won't do it again no matter what variation of configs I try. Is it something wrongly configured with flatstore? Permission issues? Any help/leads/hunches are greatly appreciated. The following is my truncated test config #!KAMAILIO #!define DBURL "flatstore:/var/log/kamailio-acc" #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 (the config and load mod stuff) modparam("acc", "log_level", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_facility", "LOG_LOCAL2") #confirmed in Ubuntu rsyslog LOG_LOCAL2 is configured modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "db_url", DBURL) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) request_route { (non-invite request stuff) setflag(FLT_ACC); setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); send_reply("420", "Test"); exit; } Ryan Brindley

2 5

Need help on WebRTC with Kamailio as proxy
by Rahul MathuR 26 Jan '15

26 Jan '15

Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - "STUN: using this candidate" "Successful STUN binding request from .." "SRTP output wanted, but no crypto suite was negotiated" a) What should I do to resolve this issue ? b) Why is that I never get 200 OK for the INVITE ? I am attaching the logfile & configs herewith. Can somebody please help me out here... -- Warm Regds. MathuRahul

4 6

Dialog variable lost in ACC
by Pars3c 26 Jan '15

26 Jan '15

Hi , I've updated kamailio from 4.1.4 to 4.1.7 and some times dialog variables are not writed on the db by the ACC module. The variable that is not writed ,is setted in the on reply route ,when it receive the 200ok and method is invite. If i come back to the 4.1.4 all is ok. If i patch kamailio 4.1.4 with the code "dialog: Set the dialog context on incoming replies" (commit:b12a01e553699786953ec601197669314bf414c7) sometime the variable are not writed. Any ideas? Thanks

2 3

Dialog module generates error since upgrade to 4.2.2
by Jan Hazenberg 26 Jan '15

26 Jan '15

Hi All, I'm running into a issue with the dialog module since the upgrade from 4.2.1 to 4.2.2. I use the dialog module to add extra vars to the cdr's generated by the ACC module. I use the following config: # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "rr_param", "did") modparam("dialog", "wait_ack", 1) # Create dialog if (method=="INVITE") { # Create Dialog dlg_manage(); # Add test dialog var $dlg_var(src_ua) = $hdr(User-Agent); } This seems to work fine on kamailio 4.2.1 but after the upgrade i see the following errors in the logs: Jan 22 13:16:57 sip /usr/sbin/kamailio[2773]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587(a)192.168.149.126' and tags '4017642921' '' Jan 22 13:17:01 sip /usr/sbin/kamailio[2770]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 7 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587(a)192.168.149.126' and tags '4017642921' '' If i'm correct "bogus event 6 in state 1" indicates that a ACK whas received while the dialog still was in unconfirmed state. Should i catch the 200 OK on the INVITE to update the dialog? Thanks, Jan Hazenberg

3 9

ims_charging module
by Shane Harrison 26 Jan '15

26 Jan '15

Hi all, I am exploring the use of the ims_charging module having set up a freediameter server (all very new to me). I have loaded and configured the dependencies i.e. cdp, dialog_ng, tm, cdp_avp but the startup gets stuck on: Jan 26 03:57:05 host-172-16-0-82 /usr/sbin/kamailio[2931]: ERROR: ims_charging [mod.c:269]: mod_init(): Could not import ul_register_ulcb Jan 26 03:57:05 host-172-16-0-82 /usr/sbin/kamailio[2931]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ims_charging (/usr/lib64/kamailio/modules/ims_charging.so) The corresponding code in mod.c is: /*Register for callback of URECORD being deleted - so we can send a SAR*/ if (ul.register_ulcb == NULL) { LM_ERR("Could not import ul_register_ulcb\n"); return -1; } But unfortunately this doesn't mean much to me - what is a URECORD, what is a SAR? Any words of wisdom around what a "ul.register_ulcb" is about would be appreciated. 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

2 1

How to forward sip message "statefully" so that the dialogue is preserved
by aft 26 Jan '15

26 Jan '15

We want to implement the following topology : INVITE--->kamailio1-----forward--->kamailio2--->route[NATMANAGE]--->Next Sip endpoint Client <---- route[NATMANAGE]<--------kamailio1<------forward-----kamailio2<---- reply with SDP So the caller to callee will be processed by one kamailio and callee to caller will be processed by another kamailio. Both kamailio will be up with a public IP. How this can be achieved?

1 0

lookup_branches and rtpengine_manage
by Yuriy Gorlichenko 26 Jan '15

26 Jan '15

Hello. I need parallel forking calls with the same username. (Call to all contacts with name for example User123), my endpoints may be WebSocket based and standart UDP endpoints. And I use rtpengine_manage for nmanaging calls wor webphones and standart softh/hard phones. I get all contacts manually and than at the branch route set rtpengine_manage settings for every call. It works fine but it works for one kamailio server. When I use 2 kamailio servers as load balansers Server that handle call get all endpoints from location but call to only one, that registred only at ths server for example I call user123 I have 3 contacts user123(a)1.2.3.4 - was registered at kamailio 1 user123(a)3.2.1.4 - was registered at kamailio 2 user123(a)4.3.2.1 - was registered at kamailio 1 So if call goes through kamailio 1 it call only to user123(a)1.2.3.4 and user123(a)4.3.2.1 I use this settings for usrloc at 2 kamailios to share all table between 2 servers modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "user_column", "username") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "cflags_column", "cflags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "path_column", "path") modparam("usrloc", "ruid_column", "ruid") modparam("usrloc", "instance_column", "instance") modparam("usrloc", "use_domain", 1) and this code for calling them [GET_CONTACTS] { sql_query("ca", "select contact from location where username='$tU'", "ra"); xlog("rows: $dbr(ra=>rows) cols: $dbr(ra=>cols)\n"); if($dbr(ra=>rows)>0){ $var(i)=0; while($var(i)<$dbr(ra=>rows)){ xlog("L_INFO","SQL query return contact {$dbr(ra=>[$var(i),0])} for {$tU} at step {$var(i)}\n"); if ($dbr(ra=>[$var(i),0])=~"transport=ws"){ xlog("L_INFO", "This is a Websocket call to endpoint"); sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); $du=$var(recieved); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}. Destination is {$du}\n"); append_branch("sip:$tU@$(du{s.select,1,:})"); } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $var(recieved)=''; sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO", "SQL query return RECIEVED {$var(recieved)}"); if ($var(recieved)==0){ xlog("L_INFO", "Recieved string is EMPTY"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); } else { xlog("L_INFO", "Recieved string is {$var(recieved)}"); $du=$var(recieved); } $var(UDP_contact)="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); append_branch("sip:$tU@$(du{s.select,1,:})"); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}}. Destination is {$du}\n"); } $var(i) = $var(i) + 1; } } t_on_branch("1"); return; } } } branch_route[1]{ if($du=~"transport=ws"){ xlog("L_INFO","Websocket Branch is {$du} for {$tU}\n"); rtpengine_manage("internal extenal force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else{ xlog("L_INFO","UDP Branch is {$du)} for {$tU}\n"); rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } When it try to branch endpoint without registration at server that handle call I get errors that tm module can not build Via header **via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [msg_translator.c:1725]: build_req_buf_from_sip_req(): could not create Via header* *ERROR: <core> [forward.c:607]: forward_request(): ERROR: forward_request: building failed* * UDP calls get errors something like above (sorry than can not share error code, This situation not often). So I think I have this trouble because I use manually handling call and tried to substitute to lookup_branches function. but I have no Idea how to set rtpengine_manage paraments for each endpoint depending this is websocket or standart call. IF there is write problem for callings thhrough 2 kamailios as loadbalansers pleas let me know about how to set rtpengine_manage parametrs wor endpoints for every fork. If not- can you tell me how I can call to all endpoints endepending of registration server (kamailio 1 or 2). Thanks.

1 0

Re: [SR-Users] remote mysql server
by Slava Bendersky 25 Jan '15

25 Jan '15

Hello Everyone, Thank you for help, fixed by adding to dialog module parameters. modparam("dialog", "db_url", DBURL) Slava. From: "Carsten Bock" <carsten(a)ng-voice.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, January 25, 2015 8:36:52 AM Subject: Re: [SR-Users] remote mysql server Hi Slava, is most modules a default database connection is defined. Probably this is the case here. As this error is coming from the dialog module: Did you configure some sort of "db_mode"? e.g.: modparam("dialog", "db_mode", 1) Thanks, Carsten 2015-01-25 9:27 GMT+02:00 Slava Bendersky <volga629(a)skillsearch.ca>: > Hello Everyone, > > I am trying start kamailio 4.2 with remote mysql server and on debug I see > DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening > connection: mysql://xxxx:xxxx@localhost/kamailio > but in cfg defined actual mysql server ip and not localhost > > 0(28885) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): > find_export_record: found <db_bind_api> in module db_mysql > [/usr/lib64/kamailio/modules/db_mysql.so] > 0(28885) DEBUG: <core> [db.c:213]: db_bind_mod(): using db bind api for > db_mysql > 0(28885) DEBUG: <core> [db.c:318]: db_do_init2(): connection 0x7faca59eced8 > not found in pool > 0(28885) DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): > opening connection: mysql://xxxx:xxxx@localhost/kamailio > 0(28885) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): > driver error: Can't connect to local MySQL server through socket > '/var/lib/mysql/mysql.sock' (2 "No such file or directory") > 0(28885) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection > to the pool 0(28885) ERROR: dialog [dlg_db_handler.c:147]: init_dlg_db(): > unable to connect to the database > > > 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 > -- 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/ _______________________________________________ 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 oon why I see in log connection attempt to localhost ? From: "Carsten Bock" <carsten(a)ng-voice.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, January 25, 2015 8:36:52 AM Subject: Re: [SR-Users] remote mysql server Hi Slava, is most modules a default database connection is defined. Probably this is the case here. As this error is coming from the dialog module: Did you configure some sort of "db_mode"? e.g.: modparam("dialog", "db_mode", 1) Thanks, Carsten 2015-01-25 9:27 GMT+02:00 Slava Bendersky <volga629(a)skillsearch.ca>: > Hello Everyone, > > I am trying start kamailio 4.2 with remote mysql server and on debug I see > DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening > connection: mysql://xxxx:xxxx@localhost/kamailio > but in cfg defined actual mysql server ip and not localhost > > 0(28885) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): > find_export_record: found <db_bind_api> in module db_mysql > [/usr/lib64/kamailio/modules/db_mysql.so] > 0(28885) DEBUG: <core> [db.c:213]: db_bind_mod(): using db bind api for > db_mysql > 0(28885) DEBUG: <core> [db.c:318]: db_do_init2(): connection 0x7faca59eced8 > not found in pool > 0(28885) DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): > opening connection: mysql://xxxx:xxxx@localhost/kamailio > 0(28885) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): > driver error: Can't connect to local MySQL server through socket > '/var/lib/mysql/mysql.sock' (2 "No such file or directory") > 0(28885) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection > to the pool 0(28885) ERROR: dialog [dlg_db_handler.c:147]: init_dlg_db(): > unable to connect to the database > > > 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 > -- 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/ _______________________________________________ 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

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remote mysql
by Slava Bendersky 25 Jan '15

25 Jan '15

Hello Everyone, I am trying start kamailio 4.2 with remote mysql server and on debug I see DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/kamailio but in cfg defined actual mysql server ip and not localhost Any help thank you, Slava.

3 2

CDRs and MongoDB
by Mickael Marrache 25 Jan '15

25 Jan '15

Hi, Is it possible to write CDRs directly to MongoDB using the new db_mongodb module? modparam("acc", "db_url", DB_MONGO_URL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_extra", " . ") modparam("acc", "cdr_log_enable", 0) modparam("acc", "cdrs_table", "cdrs") In the acc module documentation, I found the following statement: Note that CDR generation does not involve any kind of database storage (yet). In order to persist the CDRs into a database you will have to set up an exterior process (i.e., a script living outside of Kamailio) and implement the storage task yourself. Thanks, Mickael

2 2

Wrong dialog selected in on_reply/failure route in case of spirals.
by Julia Boudniatsky 25 Jan '15

25 Jan '15

After kamailio upgrade from v3.3 to v4.1.7 in production ~5% of calls with spirals dialog are failed with wrong dialog variables in on-reply/failure routes. The problem has been reproduced in the LAB (load test generated by SIPP). Please see the attached file with debug3 log. Scenario: Proxy Location Kamailio P1 Nbr translation Redirect | INVITE | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |------------------------------->| | | 100 Trying | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |<-------------------------------| | | | | | INVITE | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |<-------------------------------| | | 100 Trying | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |------------------------------->| | | | | | INVITE | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | |-------------------------------------------------------------->| | 100 Trying | | dlg 0x7f8f8504cda0 !!!!!!!! | | Call-ID:(5-8286@10.25.153.150)| |<--------------------------------------------------------------| | 301 Moved Permanently | | dlg 0x7f8f8504cda0 !!!!!!!! | | Call-ID:(5-8286@10.25.153.150)| |<--------------------------------------------------------------| | | Thank you, Julia.

2 11

mistakenly detected NAT in IPv6 setup
by Sebastian Damm 23 Jan '15

23 Jan '15

Hi, it looks to me that Kamailio detects NAT without NAT being there, when used in IPv6 scenarios. I have a register, which comes in like this: U 2015/01/23 11:42:31.013284 2a01:1234:123:1234::2:16732 -> 2001:9876:98:9876::aa01:5060 REGISTER sip:domain SIP/2.0 Via: SIP/2.0/UDP [2a01:1234:123:1234::2]:16732;branch=z9hG4bK45528eba After it has run through Kamailio, the packet is sent out like this: U 2015/01/23 11:42:31.015126 1.2.3.4:5060 -> 9.8.7.6:5060 REGISTER sip:domain SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK8d21.7fa5fb060c2acc9583f05a93b93f9873.0 Via: SIP/2.0/UDP [2a01:1234:123:1234::2]:16732;received=2A01:1234:123:1234:0:0:0:2;branch=z9hG4bK45528eba As far as I know, the received part should only get appended, if the source IP of the packet differs from the IP advertised in the topmost Via header. But to me it looks like it always gets appended. In just IPv4 scenarios, the received parameter only gets appended if a difference is detected. And if it gets appended, shouldn't the received parameter IP be in square brackets, too? Later on, we have something like this in our code: if(method=="REGISTER") { if (nat_uac_test("23")) { append_hf("X-nathint: nat\r\n"); force_rport(); if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; } else { if (!add_path()) { sl_send_reply("503", "Internal Path Error"); }; }; } In our code, Kamailio always jumps into the add_path_received() part, so I guess, NAT detection triggers. But why? And then the Path header appended looks like this: Path: <sip:1.2.3.4;lr;received=sip:2A01:1234:123:1234:0:0:0:2:16732> The path module probably doesn't know anything about IPv6, too, so the IP in the received parameter is not enclosed in square brackets. When using the Path header later to send packets to the client, Kamailio complains about a parse error for the destination URI again. I tried looking into the path module, but I guess my C skills aren't sufficient to submit a patch. Best regards, Sebastian

2 2

Siremis Installation Web Page NOT Displayed.
by Mahmoud Ramadan Ali 23 Jan '15

23 Jan '15

Hi Dears, I'm trying to install Siremis 4.1 on Debian and whenever i open the URL in IE i get location.href='$url'"; exit; } ?> and in Firefox i see <?php if(is_file(dirname(__FILE__).'/install.lock')){ include 'bin/_forward.php'; }else{ $script_name = $_SERVER['SCRIPT_NAME']; $url = str_replace("index.php","install/",$script_name); echo "<script>location.href='$url'</script>"; exit; } ?> I've verified the permissions of - siremis/log - siremis/session - siremis/files - siremis/themes/default/template/cpl files without success also no logs gets written to the logs file...please help to fix this issue... Thanks in advance

1 0

RFC: best way to create dialogs
by Klaus Darilion 22 Jan '15

22 Jan '15

Hi! I use the dialog module to count and limit concurrent calls per user. It worked fine with 4.1.7 but fails with 4.2.2. My config basically looks like: route{ ... dlg_manage() ... authentication (stateless replies + exit) ... t_on_reply() t_on_branch() ... t_relay() exit; } initial_cbs_inscript uses the default value 1. Using 4.1.7, after stateless reply+exit, Kamailio executes dialog callbacks and deletes the dialog: [dlg_var.c:55]: dlg_cfg_cb(): new dialog with no trasaction after config execution [dlg_hash.c:872]: dlg_unref(): unref dlg 0xb233c3a0 with 1 -> 1 [dlg_hash.c:872]: dlg_unref(): unref dlg 0xb233c3a0 with 1 -> 0 [dlg_hash.c:872]: dlg_unref(): ref <=0 for dialog 0xb233c3a0 [dlg_hash.c:355]: destroy_dlg(): destroying dialog 0xb233c3a0 (ref 0) With 4.2.2, after stateless reply+exit, Kamailio just exits without calling any dialog callback (to destroy the dialog) I think I could work around the problem by creating the transaction later. Here are some ideas and I would be happy about your comments and best practices. - set initial_cbs_inscript to 0 - call dlg_manage() just before t_relay - do not use dlg_manage() but use the dialog flag and set the flag somewhere before t_relay Finally, do t_on_reply() and t_on_branch() already create the transaction or is it created with t_relay()? Thanks Klaus

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reINVITE with uac_req_send()
by Le Truong Giang 22 Jan '15

22 Jan '15

Hi ! I have the following simple scenario A <---> Kamailio <---> Asterisk <---> B now i'm trying to handle Timeout reply when B is unavailable. First i suspend transaction and wait for B available. i use uac_reg_send() to reINVITE B .The problem is when B available, it receive INVITE but A doesn't receive ACK therefore the call is incomplete I guess the reason is source IP in reINVITE is Kamailio's not A's IP, so ACK can't reach A. How can i set this source IP to A's IP ? Any idea is appreciated Thank you !

1 0

RedHat 7 packages issue
by Alberto Panizzo 22 Jan '15

22 Jan '15

Hi all, Thanks for your hard working on Kamailio! I was just trying to install kamailio 4.2.X on a RedHat 7 machine but I got checksum mismatch: I took the yum configuration from: http://rpm.kamailio.org/ [kamailio] name=RPMs for Kamailio on RHEL 7 type=rpm-md baseurl=http://rpm.kamailio.org/stable/RHEL_7/ gpgcheck=1 gpgkey=http://rpm.kamailio.org/stable/RHEL_7/repodata/repomd.xml.key enabled=1 Than started yum update $ sudo yum update Loaded plugins: amazon-id, rhui-lb kamailio | 1.2 kB 00:00:00 kamailio/primary FAILED ] 0.0 B/s | 0 B --:--:-- ETA http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz: [Errno -1] Metadata file does not match checksum] 0.0 B/s | 0 B --:--:-- ETA Trying other mirror. kamailio/primary | 7.4 kB 00:00:00 http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz: [Errno -1] Metadata file does not match checksum Trying other mirror. Indeed http://rpm.kamailio.org/stable/RHEL_7/repodata/repomd.xml says: <data type="primary"> <checksum type="sha">4cc322a95d4a978e383903feaf8fbbe8ccc2ff9e</checksum> <open-checksum type="sha">9fd04e59185a8e844133b834a662cae3cc0837b7</open-checksum> <location href="repodata/primary.xml.gz"/> <timestamp>1413465311</timestamp> <size>7344</size> <open-size>70829</open-size> </data> while http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz I can download is: $ ls -l primary.xml.gz -rw-rw-r-- 1 alberto alberto 7562 gen 22 10:58 primary.xml.gz $ shasum primary.xml.gz e8e052970984feb0cdb16b98f05aefc516295aba primary.xml.gz Do I have to switch to other repository? Thank you, Best Regards -- Alberto Panizzo CTO - Co-Founder Amarula Solutions BV Cruquiuskade 47 Amsterdam 1018 AM NL T. +31(0)851119171 F. +31(0)204106211 www.amarulasolutions.com

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How to make presence module allocating an etag only for an initial Publish??
by Nuno Reis 21 Jan '15

21 Jan '15

Hello all. I've been testing presence module for a while and I do notice that presentity table grows endlessly over time. Each time a call is made a new record is added in presentity with a different etag. In documentation, namely developers guide we can read this: int etag_not_new; /* * 0 - the standard mechanism (allocating new etag for each Publish) * 1 - allocating an etag only for an initial Publish */ How can I tell the presence module in kamailio config to work using the second form of the above? Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

2 2

Kamailio doesn't response to some INVITE packages
by Volkan Oransoy 21 Jan '15

21 Jan '15

Hi, My Kamailio box works very well for a year, in front of a Freeswitch box. Nowadays, some of my users live problems about receiving calls. But this problem occurs on a couple of specific users. %1 of users complain about it. When I capture the traffic, I see that, Kamailio receives an invite package from freeswitch but takes no action. Normally it should response with a 100 trying message. Here is the INVITE package. 1.2.3.6 is the Freeswitch box and 1.2.3.2 is the Kamailio box. I captured this package on Kamailio box. U 1.2.3.6:5060 -> 1.2.3.2:5060 INVITE sip:2000@test.example.net SIP/2.0. Via: SIP/2.0/UDP 1.2.3.6;rport;branch=z9hG4bKZU1Z76FvDZ26r. Route: <sip:1.2.3.2>;lr. Max-Forwards: 65. From: "908508850000" <sip:908508850000@test.example.net>;tag=KjcgDe0NyKN0B. To: <sip:2000@88.235.112.33:1028>. Call-ID: 72b344f4-1c09-1233-ad8d-0050568e9066. CSeq: 70585075 INVITE. Contact: <sip:mod_sofia@1.2.3.6:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-AUTH-IP: 159.253.40.138. X-BF-SRC: 2. X-FS-Support: update_display,send_info. Remote-Party-ID: "908508850000" <sip:908508850000@test.example.net >;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1421813895 1421813896 IN IP4 1.2.3.6. s=FreeSWITCH. c=IN IP4 1.2.3.6. t=0 0. m=audio 28384 RTP/AVP 3 8 0 101 13. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. I use standart location lookup and relay procedure in Kamailio default script. Is there anything wrong with this INVITE package and how can I trace this issue on Kamailio box? Thank you. /Volkan

3 3

FW: KAMAILIO IMS Questions
by Zaka Ul Isam 21 Jan '15

21 Jan '15

Hello Folks: Hope you can see this and share your ideas as to how to fix it? Thanks and regards, Zaka ________________________________________ From: Zaka Ul Isam Sent: Wednesday, January 21, 2015 12:20 PM To: Carsten Bock Cc: miconda(a)gmail.com Subject: RE: KAMAILIO IMS Questions hello All: First of all many thanks for your kind response. In fact my last few emails did not make it to the list (and nor bounced). It seems there is an error on the receiving mail server! However, I am regularly receiving list emails and can respond to threads but due to unknown reason, can not initiate a thread! Yes, it is an install from stable version source. Inorder to bouble check, we even compiled this module from source and copied to the kamailio modules folder replacing the actual module,but without luck! Please suggest the fix. Thanks and regards, Zaka ________________________________________ From: Carsten Bock [carsten(a)ng-voice.com] Sent: Wednesday, January 21, 2015 11:56 AM To: Zaka Ul Isam Cc: miconda(a)gmail.com Subject: Re: KAMAILIO IMS Questions Hi Zaka, please forward the request to the sr-users list, as neither me or Daniel provide direct support. How did you install Kamailio? Are you sure, that you have the SIP-Trace in (/usr/local/lib64/kamailio/modules/)? >From your log, i can see, that you did not install it from the Debian-Packages.... Thanks, Carsten 2015-01-21 12:17 GMT+02:00 Zaka Ul Isam <Zaka.Ul(a)albtelecom.al>: > Hi Gentlemen: > > Yet another stumbling block! > Now it is not setting module parameters or finding module <SIPTRACE> > The module is there, I have confirmed it. Below is the error log & relevant > code snippet: > > Error LOG > > kamailio -cf /usr/local/etc/kamailio/pcscf/kamailio.cfg > loading modules under config path: /usr/local/lib64/kamailio/modules/ > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 366, column 42: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 368, column 35: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 369, column 44: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 370, column 37: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 371, column 38: Can't set > module parameter > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 372, column 1: syntax error > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 372, column 1: > ERROR: bad config file (7 errors) > 0(22035) WARNING: <core> [ppcfg.c:219]: pp_ifdef_level_check(): different > number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 1 > [root@IMS-dta1 pcscf]# locate modparam.c > bash: locate: command not found > > Source File > > } > > regfree(&preg); > pkg_free(reg); > if (!mod_found) { > LM_ERR("No module matching <%s> found\n", regex); > return -4; > } > return 0; > } > > > Please advise! > > Kindest regards, > > Zaka > > > > ________________________________________ > From: Zaka Ul Isam > Sent: Friday, January 09, 2015 4:07 PM > To: Carsten Bock > Subject: RE: KAMAILIO IMS Questions > > > Hi Carsten: > > Thanks a lot for your quick response and elaboration. > > As regards HSS, for testbed, we plan to use FOKUS HSS, Hope it will serve > the purpose. Please note that in addition to DIALOG_NG old DIALOG Module was > also required! > > > Thanks again & regards, > > Zaka > > ________________________________________ > From: Carsten Bock [carsten(a)ng-voice.com] > Sent: Friday, January 09, 2015 3:05 PM > To: Zaka Ul Isam > Subject: Re: KAMAILIO IMS Questions > > Hi Zaka, > > Thanks :-) > > You questions: > 1) you should be able to start the three components with "kamailio -f > /path/to/kamailio.cfg". > > You may need to modify the following line: > # ------------------ module loading ---------------------------------- > mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" > > This defines, where Kamailio will find it's modules. After the > source-code install, this is probably not correct :-) > > 2) The PCRF (and the pcscf.xml-Configuration) is optional (this > functionality has to be enabled by enabling "WITH_RX" in the > configuration, otherwise the pcscf.xml is ignored) > > 3) I assume, you also installed FHoSS from source instead from the > Package? In any case, you need to edit the hibernate.properties (e.g. > /opt/FHoSS/deploy/hibernate.properties). > > Regarding the errors of the Webinterface: It doesn't matter, if it's > 32 or 64 bit. FHoSS and it's Webinterface are for showcase and > scientific work only, so i assume, that's fine (as long as it works). > In production environments, we use e.g. a Nokia Siemens HSS (if > already present, e.g. for the LTE access) or our own drop in > replacement (with a totally different interface). > > Kind regards, > Carsten > > 2015-01-08 15:27 GMT+01:00 Zaka Ul Isam <Zaka.Ul(a)albtelecom.al>: >> Hi, Carsten: >> >> Thanks for the informative posts and responses on IMS related topics. I >> was >> trying to configure based on your last post at: >> >> http://www.kamailio.org/wiki/tutorials/ims/installation-howto >> >> With respect to above, I have few questions: >> >> 1: If the Kamailio based IMS, compiled from source (not package manager >> install) and all components in one machine. How do we start different >> components? >> >> kamailio -f /PATH/TO/RESPECTIVE COMPONENTS EXECUTABLE (kamailio.cfg with >> an >> include file P/I/S-CSCF.CFG) >> >> 2: In case of P-CSCF there are three config files as well (.xml for >> Diameter >> interface). The latter has reference to PCRF PEER >> Is it must? as there is no corresponding DNS entry in >> Kamailio-IMS.DNSZONE file. >> 3: As regards FOKUS HSS, in order to adapt the config to our scenario, do >> we >> also need to edit FHOSS/hibernate.properties file? Besides, we have errors >> with HSS WEB interface on 64 bit machines. Is 32 bit architecture the >> recommended one for HSS? >> >> Thanking you in anticipation, >> >> Zaka >> >> >> >> >> ________________________________ >> >> Albtelecom & Eagle Mobile invite you to protect together the Environment. >> Please do not print the e-mail unless really necessary. >> >> Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi >> të >> mos e printoni këtë komunikim elektronik nëse nuk është vërtet i >> nevojshëm. >> >> ________________________________ >> >> This e-mail and any files transmitted with it are confidential and >> intended >> solely for the use of the addressee/s. If you are not the intended >> recipient >> you are hereby notified that any dissemination, forwarding, copying or use >> of any of the information is strictly prohibited. If you receive this >> e-mail >> in error, please notify the sender immediately and delete it! Albtelecom >> makes no warranty as to the accuracy or completeness of any information >> contained in this message and hereby excludes any liability of any kind >> for >> the information contained therein or for the information transmission, >> reception, storage or use of such in any way whatsoever. The opinions >> expressed in this message may belong to sender alone and may not >> necessarily >> reflect the opinions of Albtelecom. Albtelecom shall bear no liability for >> any loss or damage caused by software or e-mail viruses. >> >> Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi >> është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. >> Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, >> transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e >> merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e >> tij >> dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo >> plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban >> asnjë >> përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, >> ruajtjen >> apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë >> mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë >> domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë >> përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset. > > > > -- > 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/ > > ________________________________ > > This e-mail and any files transmitted with it are confidential and intended > solely for the use of the addressee/s. If you are not the intended recipient > you are hereby notified that any dissemination, forwarding, copying or use > of any of the information is strictly prohibited. If you receive this e-mail > in error, please notify the sender immediately and delete it! Albtelecom > makes no warranty as to the accuracy or completeness of any information > contained in this message and hereby excludes any liability of any kind for > the information contained therein or for the information transmission, > reception, storage or use of such in any way whatsoever. The opinions > expressed in this message may belong to sender alone and may not necessarily > reflect the opinions of Albtelecom. Albtelecom shall bear no liability for > any loss or damage caused by software or e-mail viruses. > > Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi > është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. > Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, > transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e > merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij > dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo > plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë > përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen > apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë > mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë > domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë > përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset. -- 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/ ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

1 0

Kamailio doesn't response to some INVITE packages
by Volkan Oransoy 21 Jan '15

21 Jan '15

My Kamailio box works very well for a year, in front of a Freeswitch box. Nowadays, some of my users live problems about receiving calls. But this problem occurs on a couple of specific users. %1 of users complain about it. When I capture the traffic, I see that, Kamailio receives an invite package from freeswitch but takes no action. Normally it should response with a 100 trying message. Here is the INVITE package. 1.2.3.6 is the Freeswitch box and 1.2.3.2 is the Kamailio box. I captured this package on Kamailio box. U 1.2.3.6:5060 -> 1.2.3.2:5060 INVITE sip:2000@test.example.net SIP/2.0. Via: SIP/2.0/UDP 1.2.3.6;rport;branch=z9hG4bKZU1Z76FvDZ26r. Route: <sip:1.2.3.2>;lr. Max-Forwards: 65. From: "908508850000" <sip:908508850000@test.example.net>;tag=KjcgDe0NyKN0B. To: <sip:2000@88.235.112.33:1028>. Call-ID: 72b344f4-1c09-1233-ad8d-0050568e9066. CSeq: 70585075 INVITE. Contact: <sip:mod_sofia@1.2.3.6:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-AUTH-IP: 159.253.40.138. X-BF-SRC: 2. X-FS-Support: update_display,send_info. Remote-Party-ID: "908508850000" <sip:908508850000@test.example.net >;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1421813895 1421813896 IN IP4 1.2.3.6. s=FreeSWITCH. c=IN IP4 1.2.3.6. t=0 0. m=audio 28384 RTP/AVP 3 8 0 101 13. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. I use standart location lookup and relay procedure in Kamailio default script. Is there anything wrong with this INVITE package and how can I trace this issue on Kamailio box? Thank you.

1 0

Removing unwanted characters from From-header
by Sebastian Thörn 21 Jan '15

21 Jan '15

Hi, Hello! I'm having trouble with matching massages for some Patton equipment, from the looks of it, Patton don't like </> in headers. The initial INVITE from Patton is sent without </> in To and From-headers. In the logs from Patton I can see that it matches the "100 trying" that Kamailio sends back to it, and this message has To and From-headers without </>, but the "180 Ringing" and "200 OK" that Kamailio sends is not matched, and these two has </>. I would like to remove < and > in my onreply_route, how can I do this? Or does anyone have experience on how Patton matches messages? Br Sebastian Thörn

1 0

need some help please
by mail＠sathees.co.uk 20 Jan '15

20 Jan '15

Hi need some help please, kamailio v-4.2.2 asterisk 13.1 ;#### sip.conf [general] context=incoming calls allowguest=yes bindport=5080 bindaddr=192.168.1.77 rtcachefriends=yes [kamailio] type=friend host=192.168.1.77 port=5060 context=kamailio-trunk canreinvite=no insecure=port,invite dtmfmode=rfc2833 nat=force_rport,comedia qualify=yes i have followed this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb My DDI provider sending call to ip address via DDNS. DDI à DDNS à kamailio & asterisk in one server [192.168.1.77]. kamailio passing DDI to asterisk, but its sending to wrong context. If any of the sip users is registered, the calls were sending to registered user context. If no sip user is online then calls has been send to kamailio-trunk. How can I get all DDI to kamailio-trunk? How can I solve this problem? Many thanks Sathees

3 2

OMA/RCS Presence compliance
by Osbun, Alex 20 Jan '15

20 Jan '15

Hello, Are the Kamailio presence modules (presence, xcap, pua, rls, etc.) RCS 5.X and OMA Presence Simple spec compliant. I found the following bullet from this announcement - http://www.kamailio.org/w/2012/09/kamailio-at-11-years/ * SIMPLE Presence & IM extensions – embedded XCAP server, MSRP relay, RLS full OMA/RCS extensions implementation, IMS registrar records replication, a.s.o. Does that mean that only the RLS functionality is fully OMA/RCS compliant? Thanks and I appreciate any guidance. Regards, Alex

2 1

dialog module shared memory
by Grant Bagdasarian 20 Jan '15

20 Jan '15

Hello, I'm using the dialog module to control the maximum number of calls allowed to be passed through. Is it possible to share the dialogs stored in memory with multiple instances of kamailio? So if both kamailio instances process calls, and the maximum is set to a global of 10, both combined should never let more than 10 calls through. Regards, Grant

4 8

PCSCF Installation got issues
by Sravana R 20 Jan '15

20 Jan '15

Hi, Downloaded Kamailio-4.2.1.tar.gz file from Kamailio.org site and compiled it on RHEL 6.2 server. Configured as per setup instruction given in attached document. While start pcscf, getting the following errors: Jan 20 01:25:17 localhost pcscf: INFO: <core> [ppcfg.c:80]: pp_subst_add(): ### added subst expression: /PORT/4060/ Jan 20 01:25:17 localhost pcscf: INFO: <core> [ppcfg.c:80]: pp_subst_add(): ### added subst expression: /NETWORKNAME/open-ims.test/ Jan 20 01:25:17 localhost pcscf: INFO: <core> [ppcfg.c:80]: pp_subst_add(): ### added subst expression: /HOSTNAME/pcscf.open-ims.test/ Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: path [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: path [path_mod.c:147]: mod_init(): outbound module not available Jan 20 01:25:17 localhost ./pcscf[4201]: ERROR: registrar [reg_mod.c:352]: mod_init(): can't bind usrloc Jan 20 01:25:17 localhost ./pcscf[4201]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module registrar (/usr/local/lib64/kamailio/modules/registrar.so) Please help us to resolve above issue. Thanks, Sravan. ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally http://tim.techmahindra.com/tim/disclaimer.html internally within TechMahindra. ============================================================================================================================

1 0

REGISTER request and save() from onreply_route
by Dmitry Sytchev 19 Jan '15

19 Jan '15

Hi all! Seems to be an old topic but still unclean is it possible or not, please correct me if I'm wrong. There are some old threads like http://lists.sip-router.org/pipermail/users/2007-May/011278.html I use Kamailio 4.2.1 trying to create intercepting registrar proxy, which handles NAT (rewrites contact), forwards REGISTER to backend and then stores REGISTER in its internal location on 200 OK from backend, so I call save() from reply route in noreply mode. The problem is, it stores IP source of reply as 'received' and modified contact which i sent out to backend server as 'contact', so it loses original information from initial REGISTER. Is there a way to make save() use original request fields like T_req($ct), or some way to call save() with full set of custom params? Of course, I can save() all REGISTERs in request_route and then delete on backend fail, but it seems to be waste of resources, especially when REGISTER traffic is high :( I'd be glad to hear any advice.

2 3

REGISTER request and save() from onreply_route
by Dmitry Sytchev 19 Jan '15

19 Jan '15

Hi all! Seems to be an old topic but still unclean is it possible or not, please correct me if I'm wrong. There are some old threads like http://lists.sip-router.org/pipermail/users/2007-May/011278.html I use Kamailio 4.2.1 trying to create intercepting registrar proxy, which handles NAT (rewrites contact), forwards REGISTER to backend and then stores REGISTER in its internal location on 200 OK from backend, so I call save() from reply route in noreply mode. The problem is, it stores IP source of reply as 'received' and modified contact which i sent out to backend server as 'contact', so it loses original information from initial REGISTER. Is there a way to make save() use original request fields like T_req($ct), or some way to call save() with full set of custom params? Of course, I can save() all REGISTERs in request_route and then delete on backend fail, but it seems to be waste of resources, especially when REGISTER traffic is high :( I'd be glad to hear any advice. -- Best regards, Dmitry Sytchev, IT Engineer

1 0

Diameter problem on Kamailio
by Kowalski Piotr 3 - Hurt 19 Jan '15

19 Jan '15

Dear sr-users! We are facing serious problem. We built openIMS based on Kamailio. Moreover, we have RCS users registered, but some users are automatically unregistered on the openIMS core: 1) We don't know what a piece of log on S-CSCF means: /usr/sbin/kamailio[17392]: ERROR: --- EXPLOIT --- We need to do an UNREG server SAR assignemnt 2) In log files there is "404 Not Found: Unknown destination" or "404 user offline". Probably there is some diameter problem. Has anyone of you already faced such problem? Many thanks for your answers. Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 www.hurt-orange.pl<http://www.hurt-orange.pl/> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

2 1

Search Un-Register Expires header
by Emilio Soler Muñoz 19 Jan '15

19 Jan '15

Hi, I find a problem when I try to search the header "Expires" with value 0. At my kamailio.cfg the configuration is: if ( search_hf("Expires", ":0@", "a") ) { xlog("L_INFO", "Encontrado expires 0 ($rm)"); } This way find nothing. I test with 0 (no wildcards regular expression), but this make matches in any Register with header Expires content 0. Example: ... if ( search_hf("Expires", "0", "a") ) .... I saw some people control expires=0 at contact header, but not my case. Example: if (search("expires=0")) { xlog("L_INFO", "Encontrado expires 0 ($rm)"); } Regards, -- Emilio Soler emiliosoler(a)gmail.com

3 6

RTPengine
by Kalala Alexander 19 Jan '15

19 Jan '15

4 11

TCP connectivity issue
by Mickael Marrache 19 Jan '15

19 Jan '15

Hi, I use Kamailio as a stateless proxy with UDP + TCP. When TCP is used, I get the following TCP errors: WARNING: <core> [receive.c:265]: receive_msg(): error while trying onreply script ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect xxxxxxxxx failed (timeout) WARNING: <core> [receive.c:265]: receive_msg(): error while trying onreply script ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect xxxxxxxxxx failed (timeout) The TCP config in my script: tcp_connection_lifetime=3605 I get this issue when I receive replies to INVITE from the UAS. I can see the replies using ngrep but Kamailio doesn't seem to be able to process them correctly. I precise that the UAS is behind NAT and NAT configuration is correct (the INVITE does reach the UAS). I also precise that the UAS sends keepalive every 5 sec (I can see the ping-pong using ngrep). I set it to 5 sec to eliminate the possibility that the NAT closes the connection. Thanks, Mickael

1 0

call control for 4.2?
by Peter Villeneuve 18 Jan '15

18 Jan '15

Hi, I want to experiment with the following setup and was wondering which module(s) you all recommend. I have a working simple kamailio 4.2 setup where registered users can only call each other on the same domain (ie no outbound calling to PSTN or other domains). Now I want to explore the possibility of limiting call times for certain users but allowing unlimited call times for others (ie user 5000's calls are limited to 2 minutes while user 6000 has unlimited calling time). There are some tutorials available on the internet but they seem to be quite outdated. Given my simple scenario, should I be looking into using the callcontrol module and install the CDR tool from here http://callcontrol.ag-projects.com/ (I don't really need any complex billing scheme with various rates but rather a simple scheme were user x is either time limited or not). Before I dig in any further, I'd appreciate it of someone could tell me if this is the way to go or is there's a simpler solution. Are there any recent tutorials that I should be looking at? Cheers, Peter

3 4

UID-* modules
by Olle E. Johansson 18 Jan '15

18 Jan '15

hi! Frankly, I don't really know the difference between the uid* modules and the "regular" modules. I know the history, but would like to understand and maybe some text I can add to the README files of these modules to explain to other users what the difference is. /O

1 0

Load balancer - Path and Route header issue
by Mickael Marrache 18 Jan '15

18 Jan '15

Hi, I'm trying to use one Kamailio instance as load balancer for multiple Kamailio instances. The load balancer only performs NAT handling and dispatching of messages. The other instances handle registrations, routing, RTP proxy, authentication, etc. When a REGISTER request is received by the load balancer, a Path header is added to the request which is then forwarded to one of the registrars. The registration request is successfully handled and the Path header is saved in database. All subscribers register through the load balancer. When an INVITE is received from one subscriber by the load balancer, a Record-Route header is added by the load balancer which then forwards the INVITE to one of the other instances. The selected instance authenticates the subscriber, looks for the callee using the REGISTRAR module, adds a Record-Route header, adds a Route header according to the Path header associated with the callee, and forwards the INVITE to the load balancer. The load balancer performs loose routing (removing the Route header), adds another Record-Route header, and forwards the INVITE to the callee using the right transport address (indicated by the received parameter present in the Route header). The flow is right for the INVITE and its replies. However, when the caller sends the ACK, the route set is: load balancer, proxy, load balancer - without any information about how to reach the callee. Therefore, when the ACK reaches the "second" load balancer, the latter forwards it according to the RURI which is the Contact URI present in the 200 OK (which is not routable because of the NAT). I expected Kamailio to reuse the information that was present in the Path header to route the ACK correctly. What am I doing wrong? Thanks, Mickael

1 0

Kamailio with rtpproxy and asterisk in internal lan
by Willian Mazzardo - SYSSVOIP 17 Jan '15

17 Jan '15

Hi all. I want to create a Kamailio server who have dispatcher module, to loadbalance some asterisk servers who have my billing system. But, if I call from internet to my Public IP on Kamailio server, and then call goes to internal IP asterisk, who execute Echo App, I cant hear myself. If I call to my Public IP on Kamailio server, and call goes to EXTERNAL IP on Asterisk, everything works nice. I want this : Sip Client > Public IP Kamailio > Internal IP Kamailio > Internal IP Asterisk Server > Echo() I have rtpproxy running on same machine that Kamailio, and Im running with this command: rtpproxy -F -s udp:127.0.0.1:7722 -l PU.BL.IC.IP/172.16.1.1 -d DBUG:LOG_LOCAL0 My asterisk have 172.16.1.2 as internal IP. https://www.cloudshark.org/captures/15724d2d7e51 This is my tcpdump file captured. Thanks for any help. Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030

1 0

Using the iptel Switch for both home & office environments...
by H. KAIMARAH 15 Jan '15

15 Jan '15

Hello there, Kindly advise me on how i can download and install the iptel soft switch for both Windows and Linux platforms. I intend to install a hosted Sip VoiP service using your open source switch for both home and office environments. I have jst been checking out the documentation from the site but seams some documentation pages are failing to load (timing out). Kindly advise or share with me some of the download links for the application so i can make a successful installation. Sincere regards, -- ***/Happy New Year/*** *HENRY K. Kaimarah* /Cell:/ +256-754462010 /Now in Kampala/ - UGANDA.

2 2

REGISTER replication using t_replicate with ds_select_dst
by Spencer Thomason 15 Jan '15

15 Jan '15

Hello all, We are moving our registration and location services to dedicated instances geographically separate from our proxies and then using redirects to locate the registered user. The idea is to protect the proxies from a potential thundering herd of registrations if there was some network outage I’m using dispatcher to find a registrar that has a registration for the user, if a user is not locally registered a 503 is returned and the next server is contacted. Registrations are replicated to other servers when the client is not behind NAT. If the client is behind NAT, the registrar where the client initially registered performs keep-alives and NAT traversal and also acts as an outbound proxy instead of sending a redirect. I’m attempting to simplify the configuration by using a dispatcher group to classify incoming registrations and to select replication targets but the registrar that initially received the request is in the group as well. Is there an elegant way to chose all destinations except the local instance? I’m currently doing this with the hack below but any suggestions would be greatly appreciated. Also, if anyone has any comments or recommendations on the topology, please share! Thanks, Spencer if (isflagset(FLT_FROMREG)) return; subst('/^From:(.*)sip:.*@[a-zA-Z0-9.:]+(.*)$/From:\1sip:$au@DOMAIN\2/ig'); subst('/^To:(.*)sip:.*@[a-zA-Z0-9.:]+(.*)$/To:\1sip:$au@DOMAIN\2/ig’); $rd = "DOMAIN"; $var(local_uri) = "sip:" + $Ri + ":" + $Rp; ds_select_dst("10", "4"); if ($du != $var(local_uri)) { t_replicate($du); } while (ds_next_dst()) { if ($du != $var(local_uri)) { t_replicate($du); } }

1 0

Changing SDP with subst after rtpproxy_answer corrupts SDP
by Matthias van der Vlies 15 Jan '15

15 Jan '15

Dear all, I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :) Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk The issue is that the asterisk sends a reply 200 OK, with an updated version because it already sent SDP for 183 session progress. This can be patched in asterisk, but in my scenario I can unfortunately not do that. Thus trying to fix this on Kamailio. I am able to 'fix' this currently by performing a subst on the sdp owner variable: onreply_route[WITHSDP] { if (has_body("application/sdp")) { if(ds_is_from_list()) { rtpproxy_answer("wrei"); *if(subst("/^o=someowner ([0-9]+) ([0-9]+) IN IP4 (.*)$/o=someowner \1 \1 IN IP4 \3/")) {* xlog("L_INFO", "Fixed Asterisk incorrect version number in SDP"); } # tried the answer here as well, but that corrupts it even more } else { rtpproxy_answer("wrie"); } } exit; } However this corrupts the SDP: v=0 o=tismi 652858233 652858233 IN IP4 10.41.0.21*18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 57644 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes When I do not substitute the SDP looks perfectly fine and the external address shows as the IN IP4. But of course the version is incremented: v=0 o=tismi 1606876535 *1606876536* IN IP4 *18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 55410 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=sdpmangled:yes The ericsson is now accepting this (although it's corrupt, I know.... probably the ACME doing something funky with it), but it causes issues with another unknown piece of equipment that fails on parsing the session owner. I hope there is something wrong with my subst, but I'm afraid I can not do this from the on_reply route because SDP is only updated once it finishes? I know it's dangerous to alter the session version like this, so I made sure the Asterisk will never send a re-INVITE. Now I need a way to not corrupt the o= Kind regards, Matthias van der Vlies

3 3

websocket convert and forward to asterisk
by jaflong jaflong 15 Jan '15

15 Jan '15

Hi Everyone, Does anyone have a example of the config where I can get the following to work I want Kamailio to process websocket converting wss to tcp and srtp to rtp and forward to asterisk as tcp and rtp incoming call on websocket (wss rtp/savpf) --> kamailio/rtpproxy (protocol - convert wss to tcp) and (sdp - convert rtp/savpf to rtp/avp) and (audio - convert srtp to rtp) --> asterisk (recieve protocol: tcp, sdp: rtp/avp and audio:rtp) This application would only have incoming call from websocket to forward on to asterisk.

1 0

Fosdem 2015
by Alexandr Dubovikov 15 Jan '15

15 Jan '15

Hello everybody, what is about Fosdem this year ? Who is going there ? Wbr, Alexandr

2 1

Kamailio with a third party operator
by Jyaim Jyaim 15 Jan '15

15 Jan '15

Hello all, I cannot find documentation on how to configure Kamailio to interwork with another operator's IMS Core. I think the s-cscf should be in charge of detecting a user is not known locally, in order to route requests toward other operator. Do you know where I should look to get infos on this topic ? Thank you!

1 0

Redirect Server Including Path/Recieved Information
by Asgaroth 15 Jan '15

15 Jan '15

Hi All, I am attempting to setup a standalone redirect server which will lookup contact info and redirect to appropriate outbound proxy. The problem I am having is that the registrar is storing the path and recieved information, however, when I perform a lookup and reply with 302, the recieved and path information is not included. Are there module parameters in the registrar module that will include these contact parameters (recieved) or create the appropriate headers (path/route) in the 302 response, or, is this something I need to do manually. Currently I have the following lookup code: route { t_check_trans(); if ( method == "INVITE" ) { xlog("route[MAIN] : $rm : ruri=$ru"); xlog("route[MAIN] : $rm : lookup=sip:$rU@registered.domain"); if ( lookup("location", "sip:$rU@registered.domain") ) { send_reply("302", "Moved Temporarily"); exit; } } } I had a look at the path_mode parameter but this looks like it only takes affect for REGISTER methods. The INVITE that comes in supports the PATH header but I dont see this being passed back. In fact, I'm not entirely sure if the path header is a supported header in the 302 response message. I had a quick google and I cant seem to easily find what headers are suported in the 302 message, and where I need to put the path/recieved information. I presume I can add the recieved info as a parameter to the contact header, but how would i specify the outbound proxy to use, would that be in a route or a path header in the 302 response? I can see that the $du pseudo variable is set with the appropriate outbound path and received information, now I just need to include this information in the 302 response. Any suggestions/comments to assist in how I get this info into the 302 message would be greatly appreciated. Thanks in advance.

4 12

Crash in usrloc, get_all_db_ucontacts
by Kristian F. Høgh 14 Jan '15

14 Jan '15

Hi, On an ancient kamailio 3.1, we occasionally experience crash in the function get_all_db_ucontacts in usrloc. A backtrace reveals the following: #0 0x00002b505f2fb4ff in get_all_db_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:176 176 if (VAL_NULL(ROW_VALUES(row)+4) || p1==0 || p1[0]==0){ The lines around line 176 (which look similar to trunk, except p and p1 are str today) 159 for(i = 0; i < RES_ROW_N(res); i++) { 160 row = RES_ROWS(res) + i; 161 162 /* received */ 163 p = (char*)VAL_STRING(ROW_VALUES(row)); 164 if ( VAL_NULL(ROW_VALUES(row)) || p==0 || p[0]==0 ) { 165 /* contact */ 166 p = (char*)VAL_STRING(ROW_VALUES(row)+1); 167 if (VAL_NULL(ROW_VALUES(row)+1) || p==0 || p[0]==0) { 168 LM_ERR("empty contact -> skipping\n"); 169 continue; 170 } 171 } 172 p_len = strlen(p); 173 174 /* path */ 175 p1 = (char*)VAL_STRING(ROW_VALUES(row)+4); 176 if (VAL_NULL(ROW_VALUES(row)+4) || p1==0 || p1[0]==0){ 177 p1 = NULL; 178 p1_len = 0; 179 } else { 180 p1_len = strlen(p1); 181 } The client from frame #2 (bt full below), send a REGISTER, which was replied "200 OK" at the same second as the crash. Is it a timing issue between REGISTER and nathelper ping? I have a trace and the core. Regards, Kristian Høgh Uni-tel bt full #0 0x00002b505f2fb4ff in get_all_db_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:176 sock = 0x0 dbflags = 11088 host = {s = 0x375ef8e "PROXY_IP:5060", len = 13} i = <value optimized out> row = 0xa13488 dom = 0x2b5062723278 p1 = 0x2 <Address 0x2 out of bounds> now_s = "'2015-01-13 12:58:51'\177\000\000)" port = 11088 p_len = 73 cp = <value optimized out> needed = 16 res = 0xa2ed38 p = 0xcfcdffaf <Address 0xcfcdffaf out of bounds> now_len = 21 proto = 1642847400 p1_len = 0 shortage = 0 query_buf = "select received, contact, socket, cflags, path from location where expires > '2015-01-13 12:58:51' and cflags & 64 = 64 and id % 30 = 1", '\000' <repeats 376 times> query_str = {s = 0x2b505f515520 "select received, contact, socket, cflags, path from location where expires > '2015-01-13 12:58:51' and cflags & 64 = 64 and id % 30 = 1", len = 135} #1 get_all_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:398 No locals. #2 0x00002b5061eb6b1a in nh_timer (ticks=<value optimized out>, timer_idx=<value optimized out>) at nathelper.c:1623 rval = <value optimized out> buf = 0xadd270 cp = <value optimized out> c = {s = 0x0, len = 0} opt = {s = 0x0, len = 0} path = {s = 0x0, len = 0} curi = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0xadd836 "CLIENT_IP:62252\250u\220", len = 14}, port = {s = 0xadd845 "62252\250u\220", len = 5}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = { s = 0x7fff110134b0 "", len = 566862356}, port_no = 50336, proto = 70, type = ERROR_URI_T, flags = 65536, transport = {s = 0x8cbfa0 "", len = 5441948}, ttl = {s = 0x7fff1101352f "", len = 4448297}, user_param = {s = 0x375ef8e "PROXY_IP:5060", len = 13}, maddr = {s = 0x9204b0 "sip3", len = 10677560}, method = {s = 0x0, len = 0}, lr = {s = 0xadd84a "\250u\220", len = 0}, r2 = {s = 0xa2ebc0 "\250\272\252", len = 285291296}, transport_val = {s = 0x40 <Address 0x40 out of bounds>, len = 335544320}, ttl_val = { s = 0x3221c302d0 "H\307\300\017", len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}} he = <value optimized out> send_sock = <value optimized out> flags = <value optimized out> proto = <value optimized out> dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {in6_u = {u6_addr8 = '\000' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} iteration = 1 __FUNCTION__ = "nh_timer" #3 0x0000000000521c54 in fork_dummy_timer (child_id=<value optimized out>, desc=<value optimized out>, make_sock=<value optimized out>, f=0x2b5061eb6a70 <nh_timer>, param=0x0, interval=1) at timer_proc.c:71 pid = <value optimized out> #4 0x00002b5061eb42ee in child_init (rank=<value optimized out>) at nathelper.c:673 i = 58055190 #5 0x00000000004db1d0 in init_mod_child (m=0x917dc0, rank=0) at sr_module.c:829 No locals. #6 0x00000000004db1a9 in init_mod_child (m=0x918420, rank=0) at sr_module.c:826 No locals. #7 0x00000000004db1a9 in init_mod_child (m=0x918aa0, rank=0) at sr_module.c:826 No locals.

2 10

One way audio & video - Kamailio + rtpproxy ( NAT call )
by Chirag Ajmera 14 Jan '15

14 Jan '15

Kamailio - 4.2.2 ( SIP server ) Rtpproxy - Git Compiled ( miconda patched version ) Issue: Remote NAT Call Bria Rmt Iphone SIP Extn (3G) ----> Kamailio Server -----> Desktop Bria Client ( Wifi ) Audio and Video packets are sent from iPhone to desktop client .. but nothing otherway Also, being totally new to kamailio I can't figure out logging SIP messages ( INVITE etc ) .. Kindly if anyone can help Thank You Best, Chirag A.

1 0

Fwd: Need a support for Siremis and kamailio server
by Gaurav Rai 14 Jan '15

14 Jan '15

Hi I am using a siremis and *kamailio * hosted server with lylix.net and they give a link for getting a support about Siremis. I need a support to how to make OUT call on PSTN with registered subscriber on Softphone. Currently we have registered user on subscriber table with their location and can make a PEER to PEER audio call and text messaging also. To make OUT call on PSTN we had create a LCR gateway and LCR rule with Prefix but still can not make a OUT call getting a error 404 can u please help me out to make a PSTN call Waiting for your reply Thanks Gaurav Rai +919716238804

2 1

Making rtp streams appear unidirectional
by aft 14 Jan '15

14 Jan '15

Hi, Is it possible to make the rtp stream appear unidirectional? By that i mean, The rtp stream from client to proxy will go through one rtpproxy and proxy to client stream will go through another rtpproxy instance? If not, is it possible to mimic something like that by running rtpproxy in bridge mode where both IPs in bridge mode will be public IP? Thanks in advance cheers arif

2 4

Dispatcher 8 algorithm based on priority strange works
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

Daniel. I added 8 algorithm to our server and it works with 2 asterisk now but it works strange because: While works server with priority 1 - all ok. When this server goes down dispatcher choose next server with lowes priority. But when server with highest priority waking up dispatcher use server with lowes priority until this server not goes down. here id my config for dispatcher modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:kamailio1@10.0.1.12") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0) (!ds_select_dst("$var(setid)", "8")){ sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","{$rm} from [$fU@$si:$sp] But NO destinations available for $rd \n"); t_on_failure("DISPATCHER_ROLLOVER"); } if (!t_relay()) { sl_reply_error(); we use 4.3 master branch kamailio

2 3

Unexpected evaluation to false on a phonenumber in $avp
by Daniel Tryba 14 Jan '15

14 Jan '15

Not a really important question, just an observation that caused some headscratching on my first day after a nice vacation. I'm using an avp to make a redirect decision. Normally this avp contains Dutch telephone numbers (+31[0-9]{7,10}). With a simple: if($avp(dst_redirectnumber)) { #change $ru } this works flawless (normally). But a client decided, to forward to a Swiss number (+4141743xxxx) but with this number the if condition results in a false. A German number just works fine (+49) The avp is stuffed in an avp_db_query loop with $avp($avp(i:1))=$avp(i:2) as key/value pair from the table. I changed above condition to check if the string length of dst_redirectnumber is larger than 0. But I'm at a loss why the false occurs, first guess was an octal conversion (number only contains the digits 0-7) which results in a number "just" above 2^32. But changing a digit to 8 still fails. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

4 5

Dispatcher Failover algorithm
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

I as wrote before - we find dispatcher algorithm than can do mechanism something like this: Try call to fist server with max priority or weight. OIf this server unavailible then call second server with less weight and etc. Does anyone know what ling of algorithm we can use for this?

2 6

event on table location Duplicate entry for key 'ruid_idx'
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

Hello. We use 2 kamailio servers cluster and we have porblems with db. Database failed pecause of error: Could not execute Write_rows_v1 event on table production.location; Duplicate entry 'uloc-54aae947-86d-a67' for key 'ruid_idx', Error_code: 1062; handler error HA_ERR_FOUND_DUPP_KEY; the event's master log FIRST, end_log_pos 380, Internal MariaDB error code: 1062 But a location table no row 'ruid_idx' and no entry uloc-54aae947-86d-a67.

3 3

Kamailio with Asterisk 13 PjSip Realtime
by Chirag Desai 14 Jan '15

14 Jan '15

Hi, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripling out authentication and registration from asterisk and solely letting Kamailio handle it. In order to do this would I be correct in assuming I would have to use the asterisk database rather than the Kamailio database? I've compared the two and the table structures are very different. If I use the asterisk database I guess asterisk still needs to be responsible for handling authentication, registration and writing the contacts to the database. If I use the Kamailio database how would I dial an extension from asterisk, because as far as I can fell asterisk will have no idea who is registered or how to find them (contact details). Maybe I'm over thinking something simple, or maybe I'm not. Either way I would love your thoughts on how this could be done. Kind regards, C

1 0

pua_subscribe and force_send_socket trouble
by Mikko Lehto 13 Jan '15

13 Jan '15

Hi I am having strange behavior on local port assignment when proxying out locally generated request. Kamailio is 4.2.1 from git. I use MI pua_subscribe to create subscription to be handled by external presence server. In event_route[tm:local-request] I set $du to point back to proxy itself. When request hits initial request_route, I add some headers, change request URI and prepare failure_route for later uac_auth() processing. So far so good and TLS packets are flying to correct destination. Here is the trouble: when I observe outgoing TLS traffic with tcpdump, I can see that local source port is not following what I set with force_send_socket(). I tried to place force_send_socket() in request_route, branch_route, and tm:local-request, but it is always some random high port (>30000), never the intended one. I am trying to set it to same as my TLS listening socket. Is my usage somehow incorrect? What should I try next to make Kamailio use constant source port? Relevant config snippet below (IP address and domain part is obscured). --- mhomed=1 listen=udp:A.B.C.D:5041 listen=tls:A.B.C.D:5041 request_route { route(REQINIT); if ( blahblah... ) { $ru = "sip:" + $rU + "@domain.part.invalid;transport=tls"; $avp(uac_auth) = 0; route(PR_HDRS); route(PR_TRIGGERS); force_send_socket(A.B.C.D:5041); t_newtran(); route(RELAY); } } route[PR_HDRS] { remove_hf("User-Agent"); insert_hf("User-Agent: pua_subscribe\r\n","Call-ID"); } route[PR_TRIGGERS] { t_on_branch("PR_BRANCH"); t_on_failure("PR_FAILURE"); } branch_route[PR_BRANCH] { force_send_socket(A.B.C.D:5041); } failure_route[PR_FAILURE] { if ( $avp(uac_auth) == 0 && ($T_reply_code == 401 or $T_reply_code == 407) ) { uac_auth(); $avp(uac_auth) = 1; route(PR_TRIGGERS); route(RELAY); } } event_route [tm:local-request] { force_send_socket(A.B.C.D:5041); $du = "sip:A.B.C.D:5041"; } route[RELAY] { is from default config } --- I found these recent commits, bugs and threads somehow relating to force_send_socket and tm:local-request: http://sip-router.org/tracker/index.php?do=details&task_id=462 http://lists.sip-router.org/pipermail/sr-users/2014-August/084459.html dbd8ea9b1fa216e59d4c36e2eb4b671202824259 http://lists.sip-router.org/pipermail/sr-dev/2014-September/024984.html e404d123610b63ddd1c75d39667b373c40071eab http://lists.sip-router.org/pipermail/sr-dev/2014-September/024977.html -- Mikko Lehto

3 5

Kamailio 4.2.1 crashes on using exec_msg
by Tobias 13 Jan '15

13 Jan '15

Hi! We recently upgraded our Kamailio 4.1 to 4.2.1. With the newer version Kamailio crashes after just running a few minutes. After some debugging it looks as though the problem is in exec_msg (which is used in our config). After disabling this 4.2.1 seem to run just fine. Core file exists, here's the output: (gdb) backtrace #0 0x00000000005ebf0f in fm_extract_free (frag=0x7f053ea08d18, qm=0x7f053e88e010) at mem/f_malloc.c:206 #1 fm_malloc (qm=0x7f053e88e010, size=<optimized out>, file=file@entry=0x7f053cbedfd4 "exec: exec_hf.c", func=func@entry=0x7f053cbef378 "replace_env", line=line@entry=375) at mem/f_malloc.c:490 #2 0x00007f053cbe7953 in replace_env (list=0x7f053ea08868) at exec_hf.c:375 #3 0x00007f053cbe862e in set_env (msg=0x7f053e91d690) at exec_hf.c:547 #4 0x00007f053cbeb835 in w_exec_msg (msg=0x7f053e87c480, cmd=0x7f053ea5e168 "X֤>\005\177", foo=<optimized out>) at exec_mod.c:164 #5 0x00000000004274f7 in do_action (h=h@entry=0x7fff02e9cf90, a=a@entry=0x7f053ea5cfb8, msg=msg@entry=0x7f053e87c480) at action.c:1094 #6 0x00000000004261a9 in run_actions (h=h@entry=0x7fff02e9cf90, a=a@entry=0x7f053ea5cfb8, msg=msg@entry=0x7f053e87c480) at action.c:1583 #7 0x0000000000432980 in run_top_route (a=0x7f053ea5cfb8, msg=msg@entry=0x7f053e87c480, c=c@entry=0x0) at action.c:1669 #8 0x00007f053e610d2a in run_failure_handlers (t=t@entry=0x7f0534ecc770, rpl=0x7f053ea71040, code=486, extra_flags=extra_flags@entry=64) at t_reply.c:1051 #9 0x00007f053e612aa3 in t_should_relay_response (Trans=Trans@entry=0x7f0534ecc770, new_code=new_code@entry=486, branch=branch@entry=0, should_store=should_store@entry=0x7fff02e9d2a0, should_relay=should_relay@entry=0x7fff02e9d290, cancel_data=cancel_data@entry=0x7fff02e9d490, reply=reply@entry=0x7f053ea71040) at t_reply.c:1406 #10 0x00007f053e616126 in relay_reply (t=t@entry=0x7f0534ecc770, p_msg=p_msg@entry=0x7f053ea71040, branch=0, msg_status=msg_status@entry=486, cancel_data=cancel_data@entry=0x7fff02e9d490, do_put_on_wait=do_put_on_wait@entry=1) at t_reply.c:1809 #11 0x00007f053e61a9f3 in reply_received (p_msg=0x7f053ea71040) at t_reply.c:2493 #12 0x00000000004920a6 in do_forward_reply (msg=msg@entry=0x7f053ea71040, mode=mode@entry=0) at forward.c:783 #13 0x0000000000493637 in forward_reply (msg=msg@entry=0x7f053ea71040) at forward.c:885 #14 0x00000000004f5634 in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:275 #15 0x00000000005d929d in udp_rcv_loop () at udp_server.c:521 #16 0x00000000004a73f1 in main_loop () at main.c:1629 #17 0x0000000000425085 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2561 Kind regards,/Tobias

2 3

TLS capture with sip_trace()
by Mikko Lehto 13 Jan '15

13 Jan '15

Hi I am getting incorrect source port to Homer web while tracking outgoing request from my proxy to remote SIP server. Juha Heinanen <jh(a)tutpro.com> wrote in another thread: > in case of tcp (and tls) the source port is always a random one. > only the destination port can be predetermined. Interface capture (tcpdump) shows exactly what Juha wrote i.e. random source port for TLS connection. I always get local TLS listener address as Homer source_port, clearly not true when comparing to output of tcpdump. I am calling sip_trace() from branch_route and onsend_route. Is it even possbile to use siptrace with TLS so that it records correct TCP (random) source port to Homer? I tried briefly printing some structure variables in siptrace/siptrace.c but it seems that real source port is not available in struct dest_info at that context. -- Mikko

2 1

siputils: E164 check is broken
by Seudin Kasumovic 12 Jan '15

12 Jan '15

Hi, The fix below is wrong, condition must be '<' instead of '<='. https://github.com/kamailio/kamailio/commit/fbcfa198a2c89868fa021f6a6377442… and backported in 4.2 https://github.com/kamailio/kamailio/commit/2d1032b3c8a1184357615b37da99dfd… Regards, -- Seudin Kasumovic

2 2

dispatcher html-rpc
by Yuriy Gorlichenko 12 Jan '15

12 Jan '15

Hello. How I must use this function for dynamic reload dispatcher without restarting me server?

3 4

Maxmind GeoIP2 implementation.
by Sergey Okhapkin 12 Jan '15

12 Jan '15

Is anybody working on implementation of geoip2 module for kamailio? The advantage of the new API is a seamless support of both ipv4 and ipv6 addresses. If there is no work in progress, I'm goild to take the duty to implement module geoip2 based on the existing geoip module. I don't think it worth to extend geoip module to support both maxmind database formats.

2 2

Problem with tag field with using a custom table for the permissions module.
by Jan Hazenberg 09 Jan '15

09 Jan '15

Hi, I'm trying to use a custom table for the permissions module and run into a issue with the tag field, all other fields seem to work fine. Here is my config: modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "peer_tag_avp", "$avp(permissions_tag)") modparam("permissions", "trusted_table", "trunks") modparam("permissions", "source_col", "ip") modparam("permissions", "proto_col", "transport") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "tag_col", "trunk_uuid") When i start kamailio i see the following error in the log: Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'trunk_uuid' in 'field list' Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: permissions [address.c:112]: reload_address_table(): failed to query database Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: CRITICAL: permissions [address.c:363]: init_addresses(): reload of address table failed Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: permissions [permissions.c:631]: mod_init(): failed to initialize the allow_address function Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module permissions (/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so) Jan 6 21:49:58 sip kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe If i change the tag_col to tag it works fine, both fields are varchar(64). And the field trunk_uuid exists in the trunks table. I'm running kamailio 4.2.1 Thanks, Jan Hazenberg

2 4

Camarillo 4.2 as outbound proxy with TLS support
by Maqbul Khan - Reform InfoTech 09 Jan '15

09 Jan '15

Hello, I just installed Kamailio for the first time 4.2 on Debian Wheezy. Everything went alright, I created two users, both can register and make calls, but with no audio. So the problems are: 1. No audio 2. I want to setup Kamailio as Outbound Proxy 3. I wish to have this proxy with TLS support. Please suggest solution Maq Khan

3 3

dialog timeout reinvite
by Kelvin Chua 09 Jan '15

09 Jan '15

Happy new year everyone! :) I have stumbled into a case where a switch would send a reinvite to my kamailio server every 15 minutes. I am setting dialog timeout using avp for prepaid purposes. now i think it resets the timer upon processing of the reinvite. what is the best way to work around this?

3 5

listening to specific interface with ip
by Jijo 09 Jan '15

09 Jan '15

HI All. I know that kamailio support listen on an interface with port (listen = eth0:5060). Does it support listen on an interface with ip, port, transport? For example listen=eth0: udp:10.10.10.10:5060 Thanks Jijo

2 1

Can you use the sr.modf() method / function to call append_branch?
by J Lee 08 Jan '15

08 Jan '15

I would like to know if there's a way to do parallel forking via lua code. So specifically, I'd like to be able to do something like this: seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); append_branch("sip:d@example.com"); t_relay(); except via lua, because the list of branches is dynamic. I've tried the following: sr.modf("append_branch","sip:10.x.x.x") but the error message I'm getting is: /usr/sbin/kamailio[7437]: ERROR: app_lua [app_lua_sr.c:177]: lua_sr_modf(): function 'append_branch' is not available Just wondering if I "should" be able to do this. I've been reading http://www.kamailio.org/wiki/embeddedapi/devel/lua#srtm_package but haven't seen any references to append_branch. The other alternative I'm looking at is to just use lua to generate the list of destinations and pass back a hashtable of sorts to the cfg file. Then from there I will try to loop through the hashtable and do the append_branch command for each entry in the table. If you have other suggestions, I'd appreciate it. Thanks.

2 1

Kamailio v4.2.2 Released
by Daniel-Constantin Mierla 08 Jan '15

08 Jan '15

Hello, Kamailio SIP Server v4.2.2 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.1. 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 v4.2.2. For more details about version 4.2.2 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/01/kamailio-v4-2-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning to release v4.2.2
by Daniel-Constantin Mierla 08 Jan '15

08 Jan '15

Hello, I am considering to release next patch update in 4.2 series next week, most probably on Thursday, January 8, 2015. Given the winter holidays, if there are issues not being taking care of, send a reminder to mailing list or, even better (less chances to be forgotten), open a case on Github issue tracker: - https://github.com/kamailio/kamailio/issues Wishing everyone a great 2015! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 7

dispatcher weight dont work
by Yuriy Gorlichenko 08 Jan '15

08 Jan '15

Hello I use dipatcher algorithm that works with weight. I added 2 Asterisks and try to call its with my kam, but this still works like 4 algorithm. Weight does not work. How I must configure dispatchr for working with weight? My configuration now is modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.1.1") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0)

2 3

Dialog update timeout
by Pars3c 07 Jan '15

07 Jan '15

Hi, I wish to update dialog timeout when i receive the 200ok in the on_reply route I try to make it with the dlg_set_timeout but it give an error "bogus dialog..." How can i make it? Thanks

4 12

3rd party REGISTER body
by Jyaim Jyaim 07 Jan '15

07 Jan '15

Hello, I'm wondering if I can configure Kamailio to get in 3rd party REGISTER request's body the initial REGISTER request received from user ? Thanks in advance and best regards.

4 6

Hiding topology header From
by Kalala Alexander 07 Jan '15

07 Jan '15

2 2

Queue implementation in kamailio
by Ansuman Manna - Asergis Telecom 07 Jan '15

07 Jan '15

Hi, I am developing a system using Kamailio 4.2.1 as a front-end to a group of Asterisk servers. All of the usual functionality (registations, inbound, outbound, internal calls) are fine. We need to add call queueing to the system. I looked at the mohqueue module and read that we can use mohq_process() to send inbound calls to a suitably defined queue, based on the incoming URI. But think is that how can I get sip extension status in the particular queue? Any help would be appreciated. Thanks. Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

2 4

Hiding topology header From
by Kalala Alexander 07 Jan '15

07 Jan '15

1 1

Unregister is failing
by Anil Kumar 07 Jan '15

07 Jan '15

Hi All, I am running ims servers(pcscf,scscf,icscf and hss) as part of kamailio proxy. And I am trying to register and unregister the end-points, with help osip lib. But I see the below ERROR message. ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=908 a=25 n=t_check_status Please help me to fix the issue. Thanks and Regards, Anil

2 1

No audio when connect from public network but it works for lan users
by CK Lee 06 Jan '15

06 Jan '15

I am new to Kamailio after starting to play around for 3 weeks. Before Kamailio, I used asterisk and it works quite well in my network which has 2 public IP addresses and 10+ wired and wireless users. The asterisk listens to only one IP which my netgear router port forward the necessary sip and rdp ports to the asterisk. My interest is to use Kamailio as SIP proxy to sit in front of the asterisk. I follow the tutorial here - http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I use debian wheezy and kamailio is installed from the repository with version 4.2.1 Everything works for my lan subscribers. However, when I am outside the lan, I can register but do not the incoming packets at my sip client. What I observe is two strange behaviours 1. in asterisk, the kamailio peer has to use "sip.example.org or its public IP" in the host name field, otherwise it is not online 2. in looking at the log, I found the sip client peer is registered at my lan address i.e. 192.168.118.xx instead of a public IP when I directly login to asterisk. I add on rtproxy but not succeed with error as force_rtp_proxy_body: incorrect port 0 in reply from rtpproxy Below is my kamailio.cfg Help is appreciated. -------------------------------------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!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 ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_TLS #!define ADDR_IPV4 192.168.118.30 # *** 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://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://user:password@localhost/asterisk" #!endif #!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.example.org" /* 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:192.168.118.30:5060 listen=tls:192.168.118.30:5061 /* 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 #!ifdef WITH_XCAPSRV tcp_accept_no_cl=yes #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # 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 ####### 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 #!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 #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.118.30" desc "Asterisk IP Address" asterisk.bindport = "15066" desc "Asterisk Port" kamailio.bindip = "192.168.118.30" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/i386-linux-gnu/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 "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "outbound.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_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- modparam("ctl", "user", "kamailio") # ----- 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) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- 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", 3600) # 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", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- 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 # ----- speedial 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:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") # ----- 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 #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!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 { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { 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 # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # 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 } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } 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(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # 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 #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!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); } } # 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] { # 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; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($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 $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 4

Kamailio Active / Active HA.
by Mahmoud Ramadan Ali 05 Jan '15

05 Jan '15

Hi Dears, I have successfully configured Kamailio HA using Heartbeat and Pacemaker so if one of the two servers should go down the other server will own the virtual IP address and take over. But i have two questions: 1.This model is considered to be Active / Passive redundancy so one server will process the SIP signaling until it goes down and i'm wondering is there is any way two achieve Active / Active redundancy and if so how the signaling will be handled on the servers so they can be aware of the transactions and dialogs traversing the servers ? does the signalling will be replicated or synchronized between the servers or what ?! 2.What about the servers DB and how they should be designed in a cluster mode ? i want to replicate all the DBs of the server to get consistent user registration using the subscriber table for example. Thanks in advance and Best regards.

7 14

Kamailio fails to generate core dump
by Alex Balashov 03 Jan '15

03 Jan '15

Hi, I recently had a simultaneous crash on three production instances of Kamailio. Unfortunately, I have not been able to get to the bottom of the issue because Kamailio refused to produce a core dump: Dec 30 15:37:29 xxxxxx /usr/local/sbin/kamailio[13743]: ALERT: <core> [main.c:784]: handle_sigs(): child process 13763 exited by a signal 11 Dec 30 15:37:29 xxxxxx /usr/local/sbin/kamailio[13743]: ALERT: <core> [main.c:787]: handle_sigs(): core was not generated I do not have disable_core_dump=yes (default is no), and checked ulimits for limits on core file size. Kamailio was running as root, and there was no limit. What are the other reasons why Kamailio may not generate a core dump in such a case? 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/

3 5

Double attempt to record-route, RR param not added
by ron.kamailio＠mcleodnet.com 03 Jan '15

03 Jan '15

I have been away from Kamailio for a while and starting working on a project where it would be a good fit. I am having a problem when I try to add a Record-Route header with a tag. I see a message in the log indicating that a Double attempt to record-route was made, and the tag that I specified (;rtpproxy=yes), does not get included. I have checked-over the scripts defined in the kamailio.cfg, and I am very sure that record_route() does not get called more than once. Also, the log only shows record_route being execute a single time. Can anyone suggest how I can troubleshoot this problem further? 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=206 a=57 n=set_adv_addr 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=207 a=58 n=set_adv_port 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=208 a=25 n=record_route_preset 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=209 a=62 n=force_send_socket 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=210 a=2 n=exit 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=235 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=231 a=41 n=isflagset 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=232 a=25 n=add_rr_param 5(32087) DEBUG: rr [record.c:899]: add_rr_param(): rr_param_buf=<;rtpproxy=yes> 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=235 a=25 n=xlog 5(32087) ERROR: <script>: new ru: sip:19999009595@192.168.100.52:5080 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=236 a=24 n=record_route 5(32087) ERROR: rr [rr_mod.c:262]: w_record_route(): Double attempt to record-route 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=237 a=5 n=route 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=247 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=244 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=246 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=245 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=245 a=25 n=t_on_branch 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=250 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=247 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=249 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=248 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=248 a=25 n=t_on_reply 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=254 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=250 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=252 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=251 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=251 a=25 n=t_on_failure 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=257 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=254 a=24 n=t_relay The complete log file, script files, and SIP trace can be found here: http://www.siplabs.net/tracebin/record-route/kamailio-log.txt http://www.siplabs.net/tracebin/record-route/kamailio.cfg http://www.siplabs.net/tracebin/record-route/bgcf-local.cfg http://www.siplabs.net/tracebin/record-route/sip-trace.txt Thanks and Happy New Year, Ron

2 3

presence - Error deleting subscription
by Jan Gaida 03 Jan '15

03 Jan '15

Hello, We are upgrading our servers to Kamailio 4.2 and we see a lot of error messages "Failed to delete subscription from memory". Scenario: This happens every time there is a Notify Timeout (408) and Kamailio tries to delete the subscription. Consequences: I don't know whether the delete fails because the entry did never exist, was already deleted or because the comparison (to_tag, callid or from_tag) is incorrect. It would be nice to at least have some details in the log message. I have not noticed any consequences to general functionality. I suspect the commit (74d73953327ad7702922b27ba37c71e3283659c3) to be responsible for this error because it introduced some extra checks on deletion. Does this happen to anybody else? What may be configured incorrectly on our system? Kind regards *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

2 4

t_suspend()/t_continue() problem
by Dmytro Bogovych 02 Jan '15

02 Jan '15

Greetings. I'm trying to adopt kamailio to handle incoming calls / generate push notifications for softphone running on Windows Phone 8. Starting point was this publication http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… I adopted the script from publication a bit - now notification URI is extracted from REGISTER messages + INVITE transactions are not resuming when unregistering happens. However it does not work for me yet. I see the suspicious output in kamailio log when softphone registers after incoming push notification. Please ignore ERROR: level name - it is just dirty logging call, not real problem. Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: ERROR: <script>: resuming trasaction [40951:63820139] dbogovych (40951:63820139) Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: tm [t_lookup.c:1635]: t_lookup_ident(): DEBUG: t_lookup_ident: transaction found Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:181]: lookup(): contact for [dbogovych] found by address Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90211984 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91061218 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91229062 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90750343 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 87612750 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91841578 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Can this "script writer didn't release transaction" be the problem? Thank you :)

1 2

Happy New Year!
by Daniel-Constantin Mierla 01 Jan '15

01 Jan '15

A very dynamic 2014 for Kamailio has reached its end! Thank you everyone for contributing to it! Looking forward to 2015, a lot of new features in Kamailio and new contributors! I wish a healthy and prosperous year to all Kamailio friends, hoping to meet many of you at Kamailio World Conference and other events around the globe! Happy New Year! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 3

Making media packets associated with calls appear unidirectional using rtpproxy and kamailio
by aft 31 Dec '14

31 Dec '14

Hi, I'm trying use two instances of rtpproxy listening to two different public IPs, to handle incoming and outgoing legs. That means i want to implement this network topology: SipClient--------->media from client to proxy--->rtpproxy1 Sipclient<----------media from proxy to client<---rtpproxy2 Where rtpproxy1 listening to IP1 and rtpproxy2 listening to IP2 in kamailio cfg : modparam("rtpproxy", "rtpproxy_sock", " udp:127.0.0.1:22222 udp: 127.0.0.1:22223") if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } rtpproxy_offer("IP1"); } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } rtpproxy_answer("IP2"); } But its not working. Thanks in advance

1 0

kamailio 4.2.1 snmpstats module depends on libsnmp15
by René Bormann 31 Dec '14

31 Dec '14

hello guys, just wanted to add the snmpstats modules to our kamailio 4.2.1 installation. but we are running into a libssnmp15 dependency. searching google showed me that on 14.04 the lib is called libsnmp30 now. does anybody know how to fix this? thanks, rené ------------------- root@kamailio1:~# apt-get install kamailio-snmpstats-modules Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio-snmpstats-modules : Depends: libsnmp15 (>= 5.4.3~dfsg) but it is not installable E: Unable to correct problems, you have held broken packages. root@kamailio1:~#

2 1

Dialog_end from event route
by Dan-Cristian Bogos 30 Dec '14

30 Dec '14

Hey Guys I need assistance for the following case: I receive dialog_end request from external component in evapi:message-received block and would like to somehow kill the dialog corresponding to h_id and h_entry I get in the message. Is there any way to do that without involving mi_*? Any tip appreciated! Ta, DanB.

3 2

about kamailio
by ThatOther Dev 30 Dec '14

30 Dec '14

is it possible with ypur server project to call to country mobile networks just from a pc, without special equipment?

2 1

ul.dump failing when more than 26 users are registered
by João Vitor Arruda 30 Dec '14

30 Dec '14

Hello, I´m doing a test with the ul.dump output on both Kamailio 4.1.5 or 4.2.1 and it works great if the number of registered users is 26 or lower. Starting with 27 registered users the "kamcmd ul.dump" command returns the error: "error: 500 - Internal error creating aor struct" In Kamailio 4.2.1 the new (and nice) "kamcmd stats.get_statistics usrloc:" returns the correct number of registered_users without this "26 limit". Its possible to check why ul.dump is failing this way? Thanks,

2 3

'from' user != auth username
by Dima T. 30 Dec '14

30 Dec '14

Hello, I need migrate users from the old proprietary softswitch but they have predefined from/to header which is differ from auth username. I read about flags in auth_check but I need strict pairs from user/auth username. How this authentication scheme could be implemented with Kamailio? -- Dmitry

2 1

kamailio configuration problem
by javier falbo 30 Dec '14

30 Dec '14

Hi, I would like some help from a kamailio expert. I have installed on a Debian7 64 kbytes VPS this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, Problem is that there is no voice in a zrtp connection, but i hear only the ring sound ok. Strange is that on my local wifi works all perfect including the voice, but with external ip no. Any help? I would like to hire a kamailio expert for fixing this and also adding some extras. Write me directly so we could get in contact. regards, javier

2 2

Re: [SR-Users] bug in allow_source_address?
by Henry Fernandes 29 Dec '14

29 Dec '14

Does anyone have feedback about this? -H From: Henry Fernandes <henry(a)usinternet.com> Date: Wednesday, December 17, 2014 at 3:39 PM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: bug in allow_source_address? I¹m using allow_source_address in my Kamailio config and have been for months/years. I¹ve had no issues with it until today. The issue is that allow_source_address("1²) returns ³1" (for $si=123.17.29.201 and $sp=5060) even though we don¹t have this IP address in the address¹ table for group 1. We have defined this IP address for group 2 but not group 1. select * from address order by mask; +----+-----+---------------+------+------+----------+ | id | grp | ip_addr | mask | port | tag | +----+-----+---------------+------+------+----------+ | 20 | 1 | 123.17.0.0 | 23 | 5061 | Testing | | 22 | 1 | 123.17.0.0 | 23 | 5062 | Testing | | 23 | 1 | 123.17.0.0 | 23 | 5063 | Testing | | 24 | 1 | 123.17.0.0 | 23 | 5064 | Testing | | 1 | 1 | 123.17.0.0 | 23 | 5060 | | | 4 | 1 | 20.5.176.0 | 24 | 5060 | | | 21 | 1 | 123.17.1.235 | 32 | 5060 | Override | | 46 | 2 | 123.17.29.201 | 32 | 5060 | customer | +----+-----+---------------+------+------+----------+ Can anyone explain this? I¹m wondering if there¹s a bug in how allow_source_address does its net mask calculations. -H

2 2

about destination blacklisting
by Juha Heinanen 29 Dec '14

29 Dec '14

i made a test where domain name in request uri hostpart has two udp srv records, one with higher priority than the other. when i turn off the higher priority uac, there is about 5 sec delay before the lower priority uac starts to ring. if i make another call within a small number of seconds from the previous one, my sip proxy remembers that the higher priority uac is dead and the lower priority one starts to ring immediately. but if i wait about five seconds or more, my sip proxy again first tries the higher priority uac and there is again the about 5 second delay before the lower priority one starts to ring. i have in config use_dst_blacklist=on dst_blacklist_expire=20 from the docs i get the impression that my sip proxy should remember 20 seconds that the higher priority uac is dead, but it only remembers it about 5 seconds. what could be the explanation? -- juha

1 0

RTPProxy metod UPDATE
by Kalala Alexander 29 Dec '14

29 Dec '14

1 0

What's happening? Follow us on more channels than e-mail!
by Olle E. Johansson 29 Dec '14

29 Dec '14

Friends, The Kamailio project has great mailing lists for discussions, but did you know that we also spread news in various other channels? If you use these channels, please follow us there and spread the word about Kamailio! Facebook https://www.facebook.com/Kamailio Twitter @kamailioproject Google plus has both a Kamailio page and a discussion group - Page https://plus.google.com/u/0/b/114984922071338017631/114984922071338017631/p… - Group https://plus.google.com/communities/104535904708990386927 In addition to the news from the web site there are a few additional tips and tricks coming on these channels. See you there! /O

1 0

SIPURI sending private IP address
by Cibin Paul 28 Dec '14

28 Dec '14

Hi, I have a kamailio 4.1 as a gateway and registrar behind asterisk servers. On asterisk, I save the IP address of the originating call parsing the SIPURI or SIPCHANNEL(recvip). In both cases, I am receiving the private ip of the user agent registered with kamailio. Do I need to change anything in kamailio to receive the actual IP address. Please suggest. Regards Cibin

3 5

Meddling with password during authentication
by Olli Heiskanen 27 Dec '14

27 Dec '14

Hello all, During authentication, is there any way to affect the password user is sending? I do suspect not as it is a clear security matter, but won't hurt to ask. I use auth_db module with calculate_ha1 parameter set to 1. For reasons in integrating Kamailio into my system architecture there is a need to store a password in some other format than for example md5('555:domain.com:password)') while not allowing any passwords to be stored as plaintext. For example: md5('555:domain.com:md5('password')') but this would require me to hash the password before authentication, in Kamailio script as I can't do it in the clients. Reason for this question is to have my users in a separate database, and these users could have 0-n sip peers assigned to them, and have users authenticate to my software and the sip peers using the same password. cheers, Olli

3 3

Authorization Issue kamailio 4.2
by Ahmed Salem 27 Dec '14

27 Dec '14

Hi, I'm using kamailio 4.2 some of my clients are getting 401 unauthorized after a couple of days of usage without any problems, their usernames and passwords are still in the database server unchanged. Any idea what's going on? Thanks Ahmed

3 2

Merry Christmas!
by Daniel-Constantin Mierla 25 Dec '14

25 Dec '14

Another year getting slowly to the end, an excellent opportunity to thank to all old and new friends of Kamailio project, from contributing code to using it, helping on mailing lists and other places online, or advertising it across the world! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 3

Media trouble with kamailio/rtpengine
by Marc Soda 24 Dec '14

24 Dec '14

I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser ---------> kamailio/rtpengine <---------> asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 Thanks.

2 4

Dispatcher weight nor working
by Yuriy Gorlichenko 24 Dec '14

24 Dec '14

Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db instanse for dispatcher and at attrs column for our backend servers set WEIGHT=40 and 60 for first and second server, but packets sended only at first server ignoring weight

1 0

Re: [SR-Users] Mediaproxy do not work well with Kamailio SIP server issue
by Alfred C.M. Lu 24 Dec '14

24 Dec '14

|---------+-----------------------------> | | dong-ming.li(a)mail.| | | foxconn.com | | | | | | 2014/12/24 上午 | | | 11:54 | |---------+-----------------------------> >--------------------------------------------------------------------------------------------------------------------------------------------------| | | | To: alfred.lu(a)foxconn.com | | cc: | | Subject: Mediaproxy do not work well with Kamailio SIP server issue | >--------------------------------------------------------------------------------------------------------------------------------------------------| Secret Level: |------------------------| | ( ) 1.Non-Confidential | | ( ) 2.Confidential | | ( ) 3.Secret | | ( ) 4.Top Secret | |------------------------| Priority Level: |--------------------| | ( ) 1.Normal | | ( ) 2.Urgent | | ( ) 3.Extra Urgent | |--------------------| ------------------------------------------------------------------------- Hi, sir, We try to setup Mediaproxy on CentOS 6.6 with kamailio-4.2.1 SIP server. We use Linphone android version as SIP client to test. attached kamailio.cfg.txt is our kamailio-4.2.1 SIP server config file. pls help to rename kamailio.cfg.txt to kamailio.cfg. kamailio-4.2.1 SIP server is run on public IP server,and Linphone SIP client is run behind NAT. The problem is that only callee can hear caller's rtp audio data. caller can not hear callee's audio data. And the call will auto disconnect after about 30s time out. Below's our modification regarding the kamailio.cfg file,pls help to check if this is OK or not ? #foxconn gallice add start #### MediaProxy module loadmodule "mediaproxy.so" modparam("mediaproxy", "disable", 0) modparam("mediaproxy", "ice_candidate", "high-priority") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") modparam("mediaproxy", "mediaproxy_timeout", 500) modparam("mediaproxy", "signaling_ip_avp", "$avp(nat_ip)") modparam("mediaproxy", "media_relay_avp", "$avp(media_relay)") #foxconn gallice add end # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting use_media_proxy(); #foxconn gallice add } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { #foxconn gallice add #route(NATMANAGE); #foxconn gallice delete use_media_proxy(); #foxconn gallice add } #foxconn gallice add } For mediaproxy,we can see the media related processes are already run. 31212 ? SL 0:01 python ./media-dispatcher start 31218 ? SL 0:16 python ./media-relay start ***************************************************** dong-ming / 李東明 Foxconn Electronics Inc. HH Precision Ind. Co., Ltd. CNSBG NSDI TIPBU R&D Dept. LH Office 地址:深圳市龍華新區龍華東環二路二號富士康科技集團分機 : +86-0755-2812-9588 #26109 Mobile : 13510123670 Mail : dong-ming.li(a)mail.foxconn.com, ***************************************************** 溫馨提醒﹕入廠人員須遵守富士康出入管制,請勿攜帶手提電腦、各種攝像、U盤、MP3 等存儲設備本電子郵件及附件所載信息均為保密信息，受合同保護或依法不得洩漏。其內容僅供指定收件人按限定範圍或特殊目的使用。未經授權者收到此信息均無權閱讀、使用、複製、洩漏或散佈。若您因為誤傳而收到本郵件或者非本郵件之指定收件人，請即刻回覆郵件或致電Super Notes郵件客服熱線 560-104，並永久刪除此郵件及其附件和銷毀所有複印件。謝謝您的合作！ This e-mail message together with any attachments thereto (if any) is confidential, protected under an enforceable non-disclosure agreement, intended only for the use of the named recipient(s) above and may contain information that is privileged, belonging to professional work products or exempt from disclosure under applicable laws.Any unauthorized review, use, copying, disclosure, or distribution of any information contained in or attached to this transmission is STRICTLY PROHIBITED and may be against the laws. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender by e-mail or telephone at Super Notes support hotline 560-104 and delete this e-mail message and any attached documentation from your computer. Receipt by anyone other than the intended recipient(s) is not a waiver of any attorney-client or work product privilege. Thank you! mail from ip-->192.168.2.2 mail from pc-->MICROSOF-700F48 Version: Super Notes 1.6.4.5B (See attached file: kamailio.cfg.txt)

1 0

TLS enable false.
by Thanh Truong 23 Dec '14

23 Dec '14

Hi all, I have tried several configure TLS in kamailio but no luck. Please give me some suggestion that I can make it work correctly. This is my configure in TLS module. modparam("tls", "tls_method", "SSLv23") modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem") modparam("tls", "certificate", "/usr/local/etc/kamailio/ca/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) I am only getting issue with verify_certifiate = 1, it i let it to 0, my client can register correctly. When I set this flag, i got message from server as: Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading I Cant add any pem file into client, i have used Blink phone but no luck. Thank all in advance. ThanhTruong.

3 3

[BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)
by Andrey Utkin 23 Dec '14

23 Dec '14

I hope the subject line says it all. I need to set up a backend to establish calls between browser clients and usual SIP clients, and the abovementioned applications are of the highest interest. (Of course, calls between same UAs must work, too.) If Linphone is not going to work due to its internal issues, then other established open source mobile apps for iOS and Android will be fine. Currently I have configured latest kamailio+rtpengine with configs from here https://github.com/caruizdiaz/kamailio-ws The calls pass this way currently: jssip -> android: no sound from phone to the browser, i see that jssip sends "sendonly" attribute for audio in INVITE's SDP. Audio from browser to phone, and both video streams appear immediately, everything is fine with them. android -> jssip: video from browser to the phone appears in 1-2 _minutes_ after the call is answered. All other media streams are fine. The above results are the same in such browsers [IP-] [ ] www-client/firefox-bin-31.3.0:0 [IP-] [ ~] www-client/google-chrome-unstable-41.0.2243.0_p1:0 With sipml currently i have no stable results, so it's hard to describe what happens. Please contact me if you have configs to make the needed things work, or if you have experience of such things working stable, and can configure it quickly. -- Andrey Utkin

2 4

websocket ws_handle_handshake() error
by jaflong jaflong 23 Dec '14

23 Dec '14

Hi list, I am getting this error "ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection" Any idea why? This is the debug log 12(24113) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... 12(24113) DEBUG: xhttp [xhttp_mod.c:358]: xhttp_handler(): new fake msg created (537 bytes): <GET /ws HTTP/1.1 Via: SIP/2.0/TLS 10.1.1.1:45968 Host: 10.1.1.2:10443 User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:34.0) Gecko/20100101 Firefox/34.0 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: en-US,en;q=0.5 Accept-Encoding: gzip, deflate Sec-WebSocket-Version: 13 Origin: http://mydomain.co.uk Sec-WebSocket-Protocol: sip Sec-WebSocket-Key: GRbV+ovG+DKIBJtwb4pRiQ== Connection: keep-alive, Upgrade Pragma: no-cache Cache-Control: no-cache Upgrade: websocket > 12(24113) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: 12(24113) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <GET> 12(24113) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: </ws> 12(24113) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <HTTP/1.1> 12(24113) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 12(24113) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 12(24113) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (24113 (12)), reusing fd 12(24113) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... 12(24113) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fc08162ad68 n=421 fd=7 12(24113) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= �/�*�޹��!��S��zo�e�/D��V�Gs��s�� ^cd��S"Ŏ�K��<�c7��4lx�<Z��#�c(AJ�8V��d� ��4bF��Wu�(�w�a� f��n��s�KjQv�݈�q|Z��64{Wk��Z\ױ��M�,[�Arٵ��Q�_��u�d�Gza�f��I�fI��z��Š9=��Ħ3�ȀJPx�}��S?��eVe��\id��y�wP�r(kZs��Z�"��4`�N_MR��z�!��Eb}��+�\;�8�3��V��l��M�|�1�2��VVK�@%�a�B&�P��k 12(24113) INFO: <script>: Received HTTP request to /ws from [10.1.1.1:45968] with protocol tls 14(24115) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fc08162ad68, -1, fd -1 from 12 (24113) 14(24115) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7fc081626e20 12(24113) ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) ERROR: sl [../../forward.h:247]: msg_send(): tcp_send failed 12(24113) ERROR: websocket [ws_handshake.c:113]: ws_send_reply(): sending reply 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) 12(24113) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up 12(24113) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa31f40, 7, -1, 0x10) fd_no=2 called 12(24113) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7fc08162ad68, state -2, fd=7, id=3 12(24113) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7fc081626e20 14(24115) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7fc08162ad68, -2 from 2 9(24110) DEBUG: websocket [ws_conn.c:459]: wsconn_get_list(): wsconn_get_list

1 0

Rpc dialog variable
by Pars3c 22 Dec '14

22 Dec '14

Hi How can i do to see the dialog variables like $dlg_var with RPC ? It's possible? Thanks

1 0

ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for
by Will Ferrer 22 Dec '14

22 Dec '14

Hi All We recently discovered that our dispatcher boxes have this error constantly at the end of our calls. ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for (<callid goes here>) We aren't seeing any issues arising from it that we can notice but wanted to see if any one knew what it meant and what it might cause. Thanks much in advance. All the best Will Ferrer Switchsoft Inc

2 7

Kamailio not start if redis server is down
by pars3c 22 Dec '14

22 Dec '14

Hi, i'm trying to use the ndb_redis connector , but i've a problem when redis server is down and i try to restart kamailio. Kamailio in this case not start and release this log: 2(19590) ERROR: ndb_redis [redis_client.c:112]: redisc_init(): error communicating with redis server [srv1] (xxxxxx/0): Connection refused 2(19590) ERROR: ndb_redis [ndb_redis_mod.c:126]: child_init(): failed to initialize redis connections 2(19590) ERROR: <core> [sr_module.c:927]: init_mod_child(): init_mod_child(): Error while initializing module ndb_redis (//lib/kamailio/modules/ndb_redis.so) 30(19622) ERROR: <core> [tcp_main.c:4962]: tcp_init_children(): ERROR: tcp_main: fork failed: Connection refused 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 11 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 31 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19589 exited normally, status=255 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19590 exited normally, status=255

2 2

rtpengine stats
by Marc Soda 22 Dec '14

22 Dec '14

Does anyone know how a can get stats from rtpengine? I see the $rtpstat pseudo variable in Kamailio, but from the documentation it looks like that will only give me stats on a particular call. I'm looking for overall stats like concurrent calls, bandwidth, etc... Thanks, Marc

3 2

Multi-tenant SBC Kamailio
by Kalala Alexander 22 Dec '14

22 Dec '14

3 5

Re: [SR-Users] sr-users Digest, Vol 115, Issue 16
by Kees Meijs - SIGNET B.V. 22 Dec '14

22 Dec '14

Hi there, We haven't used gdb on Kamailio yet but some time soon we'll most likely will. Obviously I'll post our findings. Thanks again! Cheers, Kees On 17-12-14 12:00, sr-users-request(a)lists.sip-router.org wrote: > did you get a core dump file? If yes, send the full backtrace taken with > gdb.

2 1

kamailio dispatcher module stops processing REGISTER
by José Seabra 22 Dec '14

22 Dec '14

Hello there, I'm doing stress testing against kamailio (sip load balancer role) sending sip registrar, my stress test software is sending 50 registrations per second, after 30 seconds and 1000 registers processed the function ds_select_dst (I'm using algorithm 4) returns null for all registers sent from my software, and all registrar sip servers are available and in active mode, if I reduce the software to send 20 registrar's per second, the issue only happens after 60 seconds and kamailio (sip load balancer role) process 4000 registrers, after this, function ds_select_dst starts returning null Anyone has any idea why it is happening? My kamailio version is 4.2 Best Regards -- Cumprimentos José Seabra

3 9

The "Subscriber" table VS the "Location" table ?!
by Mahmoud Ramadan Ali 20 Dec '14

20 Dec '14

Hi Dears, I'm a little bit confused about the difference between the "Subscriber" table and the "Location" table.I read that the "Location" table is used for persistent user registration BUT i did NOT configured the "Location" table and can get persistent user registration with the "Subscriber" table ! So can anyone explain the difference please ? Best Regards and Thanks in Advance.

3 2

rtpengine returns 0.0.0.0 even with sendrecv in offer
by Juha Heinanen 19 Dec '14

19 Dec '14

i got mozilla to generate sdp with sendrecv, but still rtpengine does not replace 0.0.0.0 address on o and c lines. why? -- juha Dec 19 10:20:18 box /usr/bin/sip-proxy[5841]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1) Dec 19 10:20:18 box rtpengine[20171]: Got valid command from 127.0.0.1:39586: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:f5511cb8#015#012a=ice-pwd:1d062e6b56db4f738c7613bc97086194#015#012a=fingerprint:sha-256 45:CA:2E:A5:12:4E:80:0C:AD:9D:11:94:C5:2D:ED:15:55:FD:83:70:91:59:3D:57:3D:F9:A5:77:74:85:BD:E8#015#012m=audio 9 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8 ... Dec 19 10:20:18 box rtpengine[20171]: ... 000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=rtcp-mux#015#012a=candidate:0 1 UDP 2122252543 192.168.1.52 42190 typ host#015#012a=candidate:0 2 UDP 2122252542 192.168.1.52 54801 typ host#015#012a=candidate:1 1 UDP 1686110207 192.98.100.128 42190 typ srflx raddr 192.168.1.52 rport 42190#015#012a=candidate:1 2 UDP 1686110206 192.98.100.128 54801 typ srflx raddr 192.168.1.52 rport 54801#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "call-i ... Dec 19 10:20:18 box rtpengine[20171]: ... d": "lr92g1vkbkvmgdt9m1dg", "via-branch": "z9hG4bK2313203", "received-from": [ "IP4", "192.98.100.128" ], "from-tag": "d03ug77pi6", "command": "offer" } Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Creating new call Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports 8056..8057 for media relay Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports 8012..8013 for media relay Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Returning to SIP proxy: d3:sdp725:v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-lite#015#012m=audio 8056 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:8057#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8FwYLdKkGvVH4Ykc+kTgrFN1Cwz0KZVpT2JK7+g0#015#012a=setup:actpass#015#012a=fingerprint:sha-1 ED:DE:59:BA:F7 ... Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] ... :F8:78:9D:1D:78:42:AA:00:6C:88:85:A4:2B:36:42#015#012a=ice-ufrag:4dmKpmV3#015#012a=ice-pwd:75jY5mekvVULGZnnB9mJmHQoVa9i#015#012a=candidate:bVSyZZLCFRG94R1D 1 UDP 2130706431 192.26.111.29 8056 typ host#015#012a=candidate:bVSyZZLCFRG94R1D 2 UDP 2130706430 192.26.111.29 8057 typ host#015#0126:result2:oke

2 7

Initial REGISTER request/response in 3rd Party Register
by jyaim 19 Dec '14

19 Dec '14

Hello, I'm using Kamailio with a SIP Application Server, when a user registers on the IMS Core a 3rd Party REGISTER request is sent to the application server to start some logic. I'm trying to add in the 3rd party register request body the initial REGISTER request sent by user device, and the 200 OK response sent back by Kamailio. Do you think it is possible to configure Kamailio to get this behaviour ? Thank you!

1 0

Accounting: syslog vs db_flatfile
by DanB 19 Dec '14

19 Dec '14

Hey Guys, We are in the process to review the interoperability between Kamailio and CGRateS and was wondering what is the best/most recommended/maintained way to generate accounting from Kamailio: * Syslog * db_flatfile * other way maybe? Is CDR generation planned for files already, or just accounting events? Thanks in advance! DanB

6 8

SIP Fragments
by Marc Soda 19 Dec '14

19 Dec '14

I'm having a problem reassembling UDP packets on my Asterisk servers after passing through Kamailio (it appears to me an OS level issue, nothing to do with Kamailio). Is there a way, with Kamailio, to limit the size of a SIP message? I know I can just start removing headers, but that doesn't seem like a realistic solution. I see that Kamailio can compress the message body, but can Asterisk handle that? How do other people handle this? Thanks in advance, Marc

7 13

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