- sr-users - kamailio.org

Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # 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; if (is_method("INVITE") and (status=="200")) { start_recording(); } 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)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel

1 0

Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # 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; if (is_method("INVITE") and (status=="200")) { start_recording(); } 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)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel

1 0

rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # 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; } # 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)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

2 1

Is kamailio support load balancing for websockets
by SureshTummala 14 Mar '17

14 Mar '17

Hi, I want to use kamailio as a load balancer for web sockets. Is kamailio supports LB functionality for web-sockets? I can able to use kamailio as a proxy and can make calls with websockets. Now i want to add LB functionality. Setup: UA1 -->LB -->Proxy UA2 <--LB <--Proxy LB is not forwarding the packets from Proxy to UA Please help me Regards,Suresh Tummala

1 0

Several Asterisk on this same IP
by przeqpiciel 14 Mar '17

14 Mar '17

I would like to ask you how you deal with several asterisks and kamailio on that same IP address, I have installation where i route 5060 to internal server with kamailio and I would like to route RTP to asterisks, but is any way to get around a problem with RTP ports collisions?

6 19

how to unregister an remote registration at uac module
by José Lopes 13 Mar '17

13 Mar '17

Hello, I am using the module uac to do remote registration. But sometimes i need to do unregister, to control or debug the registration process. I tried using kamcmd with uac.reg_disable, but this doesn't unregister. There is any way to do it? Best regards, Jose Lopes

2 2

Segmentation fault - dialog RPC
by pars3c 13 Mar '17

13 Mar '17

Dear all, many times, when i call dialog_list on dialog module with jsonrpc, kamailio crash. The version i use is: 4.4.2 Thanks for any suggestion. #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 1666 if ((!value || (STR_EQ(*value, ph->value))) && ph->dlg) { The dump , with bt full is this : #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 profile = 0x7f2a5098fb40 ph = 0x0 i = 10 #1 0x00007f2a706eb6e4 in rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940) at dialog.c:1772 profile_name = {s = 0x1c78b00 "outbound", len = 8} value = {s = 0x0, len = 0} #2 0x00007f2a4fbe45c6 in jsonrpc_dispatch (msg=0x7fffb9fa9f10, s1=0x0, s2=0x0) at jsonrpc-s_mod.c:1091 rpce = 0x7f2a75609360 ctx = 0x7f2a4fdf9940 ret = 0 nj = 0x1c76af0 val = {s = 0x1c77b20 "dlg.profile_list", len = 16} __FUNCTION__ = "jsonrpc_dispatch" #3 0x000000000041f4ad in do_action (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1054 ret = -5 v = 4288256 dst = {send_sock = 0x1b9fa9b23, to = {s = {sa_family = 14080, sa_data = "Xu*\177\000\000\b7Xu*\177\000"}, sin = { sin_family = 14080, sin_port = 30040, sin_addr = {s_addr = 32554}, sin_zero = "\b7Xu*\177\000"}, sin6 = { sin6_family = 14080, sin6_port = 30040, sin6_flowinfo = 32554, sin6_addr = {__in6_u = { __u6_addr8 = "\b7Xu*\177\000\000S \205\000\060\000\000", __u6_addr16 = {14088, 30040, 32554, 0, 8275, 133, 48, 0}, __u6_addr32 = {1968715528, 32554, 8724563, 48}}}, sin6_scope_id = 26}}, id = 0, proto = 16 '\020', send_flags = { f = 155 '\233', blst_imask = 250 '\372'}} tmp = 0x7fffb9fab700 "\t" new_uri = 0x0 end = 0x0 crt = 0x7fffb9fa92d0 "" cmd = 0x7f2a754adcc0 len = 0 user = 0 uri = {user = {s = 0x12c <Address 0x12c out of bounds>, len = 0}, passwd = {s = 0x7fffb9fa92d0 "", len = 1969341184}, host = { s = 0x7f2a75475010 "\001", len = 4742114}, port = {s = 0x0, len = 0}, params = { s = 0x98206f7473 <Address 0x98206f7473 out of bounds>, len = 1969341256}, sip_params = { s = 0x7fffb9fa91e0 "`\222\372\271\377\177", len = 6538286}, headers = {s = 0x1c2b6f0 "\300\300\302\001", len = 0}, port_no = 50838, proto = 30049, type = 32554, flags = 0, transport = {s = 0x0, len = -940795965}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x6222286e00000000 <Address 0x6222286e00000000 out of bounds>, len = 577270895}, maddr = { s = 0x3a746e65696c6309 <Address 0x3a746e65696c6309 out of bounds>, len = 0}, method = {s = 0x75c5f0 "core", len = -1174760864}, lr = {s = 0x764661 "rval_destroy", len = 7718524}, r2 = {s = 0x7f2a7561c348 "\002", len = 1967607824}, gr = {s = 0x7fffb9fa91b8 "H\303au*\177", len = 1967607824}, transport_val = {s = 0x7f2a7561c300 "\300\004", len = 1969341256}, ttl_val = {s = 0x7f2a7557e200 "\020", len = -1174760316}, user_param_val = {s = 0x7f2a75583700 "\002", len = 1969341256}, maddr_val = {s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, method_val = { ---Type <return> to continue, or q <return> to quit--- s = 0x0, len = 0}, lr_val = {s = 0x7fffb9fa9260 "Ð\372\271\377\177", len = 5581478}, r2_val = {s = 0x0, len = 0}, gr_val = { s = 0x3a28d <Address 0x3a28d out of bounds>, len = 1969177728}} next_hop = {user = {s = 0x1c2b6f0 "\300\300\302\001", len = 1969330344}, passwd = { s = 0x69692c22 <Address 0x69692c22 out of bounds>, len = 29537760}, host = { s = 0x400000000 <Address 0x400000000 out of bounds>, len = -1174761304}, port = { s = 0x71657228 <Address 0x71657228 out of bounds>, len = 1969342102}, params = {s = 0x0, len = 1}, sip_params = {s = 0x0, len = -1174761328}, headers = {s = 0x1 <Address 0x1 out of bounds>, len = 2}, port_no = 40672, proto = 47610, type = 32767, flags = 3120206016, transport = {s = 0x7a6563697220616c <Address 0x7a6563697220616c out of bounds>, len = 1701736297}, ttl = { s = 0x7f2a75580848 "\001", len = -1174760828}, user_param = {s = 0x7fffb9fa9f10 "\b\005", len = -1174757840}, maddr = { s = 0x65722e2e223a7069 <Address 0x65722e2e223a7069 out of bounds>, len = 1702129517}, method = {s = 0x7f2a7561c696 "/sip", len = 29855024}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x500000004 <Address 0x500000004 out of bounds>, len = 4}, transport_val = {s = 0x400000004 <Address 0x400000004 out of bounds>, len = 6552774}, ttl_val = { s = 0x7fffb9fa9060 "", len = 1967607824}, user_param_val = {s = 0x1000001 <Address 0x1000001 out of bounds>, len = 1967607824}, maddr_val = {s = 0x7fffb9fa90b0 "\020PGu*\177", len = 1956545302}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14985000}, r2_val = { s = 0x165e80 <Address 0x165e80 out of bounds>, len = 1792216}, gr_val = {s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}} u = 0x7f2a75583f30 port = 0 dst_host = 0x7f2a75583700 i = 32767 flags = 0 avp = 0x7fffb9fa9e30 st = {flags = 151652617, id = 29193, name = {n = 0, s = {s = 0x6e65090a00000000 <Address 0x6e65090a00000000 out of bounds>, len = -1174761256}, re = 0x6e65090a00000000}, avp = 0x7fffb9fa9000} sct = 0x7fffb9fa94d0 sjt = 0x7fffb9fa9f10 rve = 0x416f00 mct = 0x7fffb9fab700 rv = 0x92922726f727265 rv1 = 0x0 c1 = {cache_type = 3120205776, val_type = 32767, c = {avp_val = {n = -940803561, s = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, re = 0x31c7ec7a17}, pval = {rs = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, ri = -1174757840, flags = 32767}}, i2s = "vp(s:inserver_erPor)\","} s = {s = 0x1c2b6f0 "\300\300\302\001", len = 0} srevp = {0x0, 0x0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, ---Type <return> to continue, or q <return> to quit--- select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #4 0x000000000042c218 in run_actions (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1549 t = 0x7f2a75584700 ret = -1 mod = 0x0 ms = 0 __FUNCTION__ = "run_actions" #5 0x000000000041f416 in do_action (h=0x7fffb9fa9e30, a=0x7f2a75585b10, msg=0x7fffb9fa9f10) at action.c:1045 ret = 1 v = 1 dst = {send_sock = 0x7fffb9fa9b50, to = {s = {sa_family = 64710, sa_data = "c\000\000\000\000\000\060\233\372\271\377\177\000"}, sin = {sin_family = 64710, sin_port = 99, sin_addr = {s_addr = 0}, sin_zero = "0\233\372\271\377\177\000"}, sin6 = { sin6_family = 64710, sin6_port = 99, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "0\233\372\271\377\177\000\000Ù¨bu*\177\000", __u6_addr16 = {39728, 47610, 32767, 0, 43225, 30050, 32554, 0}, __u6_addr32 = {3120208688, 32767, 1969400025, 32554}}}, sin6_scope_id = 366}}, id = 10, proto = -24 '\350', send_flags = {f = 234 '\352', blst_imask = 98 'b'}} tmp = 0x1467562aa20 <Address 0x1467562aa20 out of bounds> new_uri = 0x7a0a02 "parse_headers" end = 0x79db5c "core: parser/msg_parser.c" crt = 0x40 <Address 0x40 out of bounds> cmd = 0x7f2a75475010 len = 1967607824 user = 32554 uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x7fffb9fa9b70 "@\234\372\271\004", len = 6529985}, host = { s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, port = {s = 0x0, len = 0}, params = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6661320}, sip_params = {s = 0x7fffb9fa99f0 "", len = 1967607824}, headers = {s = 0x0, len = 1967607824}, port_no = 39488, proto = 47610, type = 32767, flags = 6552774, transport = { s = 0x7fffb9fa9a20 "", len = 1967607824}, ttl = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, user_param = {s = 0x7fffb9fa9a70 "\240\232\372\271\377\177", len = 1956545302}, maddr = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 0}, method = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986448}, lr = {s = 0x1658d8 <Address 0x1658d8 out of bounds>, len = 1790768}, r2 = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}, gr = {s = 0x12e <Address 0x12e out of bounds>, len = 1967607824}, transport_val = {s = 0x14600000000 <Address 0x14600000000 out of bounds>, len = 1969417808}, ttl_val = { s = 0x7fffb9fa9aa0 "\240\233\372\271\377\177", len = 4742114}, user_param_val = {s = 0x0, len = 1967607824}, maddr_val = { s = 0x0, len = 0}, method_val = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6534031}, lr_val = {s = 0x79d630 "core", len = 6552774}, r2_val = {s = 0x7fffb9fa9b00 "n\001", len = 1969417880}, gr_val = {s = 0x7f2a7562aa20 "", len = 1969399988}} next_hop = {user = {s = 0x7fffb9fa9820 "", len = 1967607824}, passwd = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, host = {s = 0x7fffb9fa9870 "\240\230", <incomplete sequence \372\271>, len = 1956545302}, port = { ---Type <return> to continue, or q <return> to quit--- s = 0x1c588e1 "\311\305\001", len = 0}, params = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986216}, sip_params = {s = 0x1659c0 <Address 0x1659c0 out of bounds>, len = 1969416936}, headers = {s = 0x7f2a7562aa20 "", len = 1969399993}, port_no = 301, proto = 0, type = ERROR_URI_T, flags = 0, transport = { s = 0x7f2a7562a8d9 "Content-Type: application/json-rpc\r\nCache-Control: no-cache\r\nPragma: no-cache\r\nUser-Agent: Java/1.8.0_121\r\nHost: \r\nAccept: text/html, image/gif, image/jpeg, *; q=.2, */*; q=.2\r\nConn"..., len = 0}, ttl = { s = 0xb9fa98a0 <Address 0xb9fa98a0 out of bounds>, len = 1969416936}, user_param = {s = 0x0, len = 1967607824}, maddr = { s = 0x0, len = 0}, method = {s = 0x7fffb9fa99a0 "\240\233\372\271\377\177", len = 6534031}, lr = {s = 0x79d630 "core", len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x7fffb9fa9e7c "\260\f\222\017", len = -1174757736}, transport_val = { s = 0x79d630 "core", len = -1174757704}, ttl_val = {s = 0x7a0a10 "get_hdr_field", len = 7986012}, user_param_val = { s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, maddr_val = {s = 0x79d630 "core", len = 1969416864}, method_val = { s = 0x7a0a10 "get_hdr_field", len = 7986012}, lr_val = {s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, r2_val = { s = 0x7f2a75475010 "\001", len = 1969416864}, gr_val = {s = 0x7f2a7562eaa0 <incomplete sequence \350>, len = 1969399988}} u = 0x7f2a7562ee50 port = 0 dst_host = 0x80067e961 i = 0 flags = 0 avp = 0x7f2a75475010 st = {flags = 3120207856, id = 32767, name = {n = 1967607824, s = {s = 0x7f2a75475010 "\001", len = 1969239958}, re = 0x7f2a75475010}, avp = 0x7f2a75475010} sct = 0x4b9fa9c40 sjt = 0x79d630 rve = 0x7f2a75582f20 mct = 0x7fffb9fab700 rv = 0x63fcc6 rv1 = 0x0 c1 = {cache_type = 43, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 7718384}, re = 0x0}, pval = {rs = {s = 0x0, len = 7718384}, ri = 26, flags = 0}}, i2s = "\t\000\000\000\000\000\000\000huTu*\177\000\000\000\000\000\000\000"} s = {s = 0x7fffb9fa99f0 "", len = 1969239512} srevp = {0x3000000018, 0x7fffb9fa9ab0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action"

2 2

unable to receive response from freeswitch
by ysuryas . 13 Mar '17

13 Mar '17

I have configured kamailio (192.168.56.102) and freeswitch(192.168.56.103) in separate virtualboxes when making a call, the invite goes to proxy and it is forwarded to freeswitch but response from freeswitch is "unreachable port" attached the wireshark trace: I don't why 10.0.2.15 is making the Invite to freeswitch, i was expecting 192.168.56.102 to do that. UA -> 192.168.56.102 -> 192.168.56.103 Any suggestions on this? [image: Inline image 1] Thanks, Surya

2 1

rtpproxy doesn't record after start_recording
by Marko Tirs 11 Mar '17

11 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # 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; } # 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)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

1 0

Dispatcher Module - Hava limit?
by Rodrigo Moreira 11 Mar '17

11 Mar '17

Hello, I am experiencing the following problem: my dispatcher module is losing many calls when the scenario for balancing is about 175 CPS or more. Is there a limit that the module can do? Where do I find this information? Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069

3 6

Multiple parallel branches, one invalid
by Nathan Ward 10 Mar '17

10 Mar '17

Hi, I have a scenario I’m not sure of the best way to solve. I use alias_db and have multiple destinations for a single number, and have append_branches set to 1. I then use lookup_branches to get details from the location table for each branch. I then do t_load_contacts and t_next_contacts, then t_relay to parallel fork the calls to the destination, and all the phones ring at once. This all works great if all the destinations from the dbaliases table are online. However, if one of them is offline, lookup_branches says "Not found in usrloc”. This branch is still there, and the call is attempted. In my environment, this then does a DNS lookup, can’t resolve the domain, and the branch fails. This is the first branch that is tried, and when this branch has an error like this, it causes t_relay() to fail - I check for the output of t_relay() and reply with an error - per the default config that Kamailio ships with. I guess what I need to figure out, is if there’s a way to have t_relay fail only if all branches fail? Or, is there a more elegant way to handle this? I should add, I only allow a single registration per subscriber - so, lookup_branches only returns one branch per alias from alias_db_lookup. -- Nathan Ward

2 1

404 not here for BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/ Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95: F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+ rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. 3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487;branch=z9hG4bK-524287-1--- 1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060;rport=5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

2 1

sip6 BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95:F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;branch=z9hG4bK-524287-1---1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060 ;rport=5060;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

3 2

Bounty for topos fix
by Trevor Peirce 09 Mar '17

09 Mar '17

Hello, I would like to offer a $100 USD bounty if a developer is able to provide a fix for the two topos problems I have experienced. I will add another $50 if this can be solved within 48 hours of the time stamp of this message. The bug report is at: https://github.com/kamailio/kamailio/issues/1005 You may find the most relevant information in my last few updates. Please refer to mar1-nat.txt and mar1-reinvite.txt for the most recent SIP captures, and the post below that for example database records. Steps to reproduce the NAT problem can be found at https://github.com/kamailio/kamailio/issues/1005#issuecomment-283255720 Some fine print to clarify the offer: Payment is contingent on resolving both described use-cases. Payment will be sent via your choice of PayPal, cheque drawn on a US bank, or cheque drawn on a Canadian bank. Payment will be issued once the patch is accepted by the Kamailio project, although the bonus for a quick resolution simply requires a suitable patch to be submitted to the bug tracker within 48 hours from now. Thanks! -- Trevor Peirce AcroVoice Solutions Inc

1 1

Changes in LCR Target List doesn't takes place immediately
by Nitesh Lohchab 09 Mar '17

09 Mar '17

Hello, I believe LCR Target list is stored in the database and any changes made here should affect the behaviour of the calls imedialtely. I have changed one of my Target List Rule , where I changed my gateway Id to another gateway. But the calls are still being routed via previous gateway. Is there something I can do to make the changes in production hours. Sometimes one of carriers delivers bad call quality I would like to switch the carriers. -- Regards, Nitesh Lohchab

2 1

rtpengine sending rtp to wrong endpoint after reinvite
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, One of our customers is using a SEMS box to place two outbound calls using our sip trunk. Once the first call is connected a second call is placed and when the second call answers their server sends a re-invite to switch audio ports so the rtp traffic doesn't flow through their server anymore but is routed inside our platform. Basically, they just switch SDP's of both calls. It seems like a random issue, and is not really reproducible, except for placing multiple calls and sometimes both parties can hear each other, other times they can't, because rtpengine fails (I think) to update the endpoint and keeps sending rtp back to their server for one of the call legs. We tried to reproduce the case using a freeswitch box and it worked every time. After the reinvite, the rtp remained within our platform. The signaling in both cases still goes through the freeswitch or sems for call control. Does anyone have experience with this case? Or seen the issue before where rtpengine keeps sending rtp to the original endpoint? Regards, Grant Bagdasarian CM

2 3

Access avp between sip transactions
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, Is it possible to access an avp variable from a different sip transaction? On an INVITE is set an avp, but on ReINVITE I would like to access this avp again and get the value. Currently the value is null, which makes sense. Not sure if this is possible and if it violates any rules inside the Kamailio engine? I guess I could use a htable, but I'm just wondering if there is a built-in way for doing this. Regards, Grant Bagdasarian CM

2 1

Kamailio Presence help required.
by SamyGo 08 Mar '17

08 Mar '17

Hi, I'm in need of making/tweak an existing Kamailio Presence setup which is giving some tough time. *Whats already working:* BLF dialog states changes are already sent across the users. SCA is working as well. *What isn't working:* When a User comes online then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state right then unless the monitored extensions make a call (BLF works) *Why is it not working:* As evident from wireshark traces, the user IP phones (Test sets: Polycoms, Yealink, CISCO, Grandstream) don't send our *Event: presence* rather only *Event:dialog* and Kamailio do not send NOTIFY out to everyone. Though yes there is an internally generated PUBLISH seen and handled properly upon registration state changes. Jitsi has been tested and Kamailio send out these registration state change info to jitsi, somce jitsi sends *Event:presence *in SUBSCRIBE. I need dialog based NOTIFY to be sent out on registration state-change. Need pointers and help on the topic, looking forward to some feedback. Regards, Sammy

4 8

TCP stack is not doing load distribution among children
by Surendra Pullaiah 08 Mar '17

08 Mar '17

Hi Daniel, Thanks for your suggestions. The approaches which you have suggested are good, event I thought before initiating this query (since am working with kamailio for last three years), but my requirement is to distribute on single connection. Since it requires code change, started working on it. Really thanks for understanding my concern. @juha: Dear juha, it is not broken TCP implementation, kamailio TCP stack well implemented. If you would like to build a server to handle HTTP and MSRP protocols at higher rate, here is the place. But for SIP we have options to handle either UDP/TCP. Thanks surendra

1 0

Kamailio push for Android VoIP Application and VoIP Contacts update
by sscc 08 Mar '17

08 Mar '17

good day, i have a running kamailio server having 200 users. I maintain the user ids in a vcf (vCard Format) form. i have developed an Android VoIP application. Android application is given to the user along with the contact vcf. The contact list get increased day by day and consequently the user ids in vcf (vCard Format) as well. the application is also updated. Aim is to update application and the contact list by sending push notification to user. (we don't want to place our application at play store). i want to do push notification from Kamilio server for application and contact list update. Is there any way to do push notification from server to users to update the app and VoIP contact list. regards sscc -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-push-for-Android-VoIP-Appl… Sent from the Users mailing list archive at Nabble.com.

1 0

TCP stack is not doing load distribution among children for SIP
by Surendra Pullaiah 08 Mar '17

08 Mar '17

Dear Daniel, A small concern in kamailio, existing TCP stack is not doing load distribution among children for SIP messages. For example UE-->kamailio1 (TCP listen (4 children)) ----->kamailio2 (TCP Listen 4 children) As per stack if UE opens a connection with kamailio1 based on free children it is sharing the FD, so no problem towards from UE to kamailio1, then need requirement to forward same sip message to kamailio2, here kamailio1 is opening connection with kamailio2 and then forwarding. Suppose if we forward two requests to kamailio2, only one children is getting used among four. With this design suppose if we want to run our kamailio as two sip servers we may not achieve good results when performance matters. Please suggest something on this to move further. Regards surendra

3 5

new keepalive module ?
by Guillaume Bour 07 Mar '17

07 Mar '17

Hi all, I would like to implement destinations keepalive in drouting module (as it is done in dispatcher). But instead of duplicating what's implemented in dispatcher, I think it would be more clever to create a new module dedicated to pinging destinations, and to plug drouting, dispatcher and other modules to this new one What's you opinion about that ? Cheers, Guillaume

1 0

Kamailio core dumped
by Alexey V. Panfilov 07 Mar '17

07 Mar '17

Hi there, I've kamailio 4.2.3 (i386/freebsd) which works perfectly with 200-300 cps for years. Now I need migrate to freebsd amd64. I've installed kamailio 4.4.5 (amd64/freebsd), copied kamailio.cfg, made a test call - it was all right. But when I start work load (200-300 cps), kamailio core dumped and exited. In logs (skipped "<script>:" having): ======================= Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x809937000, 976); Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x809937000, 976) called from core: msg_translator.c: build_req_buf_from_sip_req(2154), module: core; Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [msg_translator.c:2159]: build_req_buf_from_sip_req(): out of memory Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x809937000, 864); Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x809937000, 864) called from tm: t_reply.c: relay_reply(1885), module: tm; Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: tm [t_reply.c:1887]: relay_reply(): ERROR: relay_reply: cannot alloc reply shmem Mar 1 12:39:37 sip7 /usr/local/sbin/kamailio[32038]: : <core> [mem/f_malloc.c:598]: fm_free(): BUG: fm_free: bad pointer 0x61743b3e6d6f632e (out of memory block!), called from tm: h_table.c: free_cell_helper(226) - aborting Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: ALERT: <core> [main.c:740]: handle_sigs(): child process 32034 exited by a signal 10 Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32044]: INFO: <core> [main.c:810]: sig_usr(): signal 15 received ======================= I've tried to start with -m 512 - nothing changes. Hardware: Xeon(R) CPU E5-2660, 16Gb RAM, 1Tb HDD. Software: OS: FreeBSD 10.3-STABLE (r313760) amd64 version: kamailio 4.4.5 (amd64/freebsd) f98162 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, select, kqueue. id: f98162 compiled on 12:25:48 Mar 1 2017 with clang 3.4 bt is here: http://tmp.lehis.ru/kam_gdb.txt I can't enable debug symbols for gdb (tried gmake mode=debug, which added EXTRA_DEBUG to flags). How can I do it? I need in direction for resolve this issue, please, help. Thanks. -- Best regards, Alexey V. Panfilov

2 3

Kamailio 4.2 crashed
by Igor Potjevlesch 07 Mar '17

07 Mar '17

Hello, I got two coredumps: (gdb) bt full #0 0x000000000061ab8e in qm_insert_free (qm=0x7f0f7fca7000, frag=0x7f0f8f6f6268) at mem/q_malloc.c:180 f = 0x7f0f8f6c45e8 prev = 0x0 hash = 5 And (gdb) bt full #0 0x000000000061b935 in qm_detach_free (qm=0x7f0f7fca7000, frag=0x7f0f8f6d70b8) at mem/q_malloc.c:271 prev = 0x7f0f7fca7248 next = 0x7f0f8f6c45e8 What are necessary to diagnose how this issue occurs? Thank you for your support. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 3

SCTP support for CDP
by Sergey Zyrianov 07 Mar '17

07 Mar '17

Hello, Can I enable use of SCTP for diameter connections by CDP module? Thanks, Sergey.

3 6

Installation of Siremi 4.3.0 on Centos 7.3 ?
by Lucas Viallon 07 Mar '17

07 Mar '17

Hi i want install Siremis 4.3.0 on a Centos 7.3 with Php 7.1 when i want access to the web site, i have this error in httpd logs: Mon Mar 06 18:35:06.833569 2017] [php7:warn] [pid 22018] [client 172.16.1.150:58375] PHP Warning: require_once(SMARTY_CORE_DIRcore.assemble_plugin_filepath.php): failed to open stream: No such file or directory in /var/www/siremis-4.3.0/openbiz/others/Smarty/libs/Smarty.class.php on line 1368 [Mon Mar 06 18:35:06.833641 2017] [php7:error] [pid 22018] [client 172.16.1.150:58375] PHP Fatal error: require_once(): Failed opening required 'SMARTY_CORE_DIRcore.assemble_plugin_filepath.php' (include_path='.:/usr/share/pear:/usr/share/php:/var/www/siremis-4.3.0/openbiz/others/') in /var/www/siremis-4.3.0/openbiz/others/Smarty/libs/Smarty.class.php on line 1368 Anyone know this errors ? regards Lucas

2 1

Error in kamailio (Need for help)
by Ghanshyam Variya 06 Mar '17

06 Mar '17

Mar 2 11:55:33 Ubuntu kamailio: INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Mar 2 11:55:33 Ubuntu kamailio: INFO: <core> [core/tcp_main.c:4665]: init_tcp(): using epoll_lt as the io watch method (auto detected) Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: sipcapture [sipcapture.c:454]: parse_table_names(): INFO: table name:sip_capture Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: <core> [db.c:203]: db_bind_mod(): Module db_mysql not found. Missing loadmodule? Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: sipcapture [sipcapture.c:526]: capture_mode_set_params(): parsing capture_mode: could not bind db funcs for url:[mysql://kamailio:kamailiorw@localhost/kamailio] Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: sipcapture [sipcapture.c:637]: capture_mode_init(): capture mode parsing failed Mar 2 11:55:33 Ubuntu kamailio[6479]: WARNING: sipcapture [sipcapture.c:769]: mod_init(): Default capture mode configuration failed. Suppose sip_capture calls will use other defined capture modes. Mar 2 11:55:33 Ubuntu kamailio[6479]: : <core> [core/mem/q_malloc.c:145]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(0, 0)[0x7fea6ae64ae0:0x7fea6ae64b18]! Memory allocator was called from sipcapture: sipcapture.c:831. Fragment marked by sipcapture: sipcapture.c:788. Mar 2 11:55:33 Ubuntu kamailio: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe This type of error are put's my kamailio so can you tell me why ? Thank you, Ghanshyam variya

3 2

Re: [SR-Users] Unknown $dlg(h_entry) and $dlg(h_id) in dialog
by Никитенко Виталий 06 Mar '17

06 Mar '17

Hi, very thanks! Before use dlg_get i lost 3-4 hour calls every day. Now one day - fly is good! > Hello, > > can you try using dlg_get(...) before the xlog? > > Cheers, > Daniel > > On 27/02/2017 08:33, Никитенко Виталий wrote: >> Hi i have kamailio 4.2, dialog module with parametr >> modparam("dialog", "dlg_flag", 5) >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "profiles_with_value", "caller") >> modparam("dialog", "dlg_match_mode", 1) >> modparam("dialog", "default_timeout", 3600) >> modparam("dialog", "timeout_noreset", 1) >> modparam("dialog", "send_bye", 1) >> modparam("dialog", "timeout_avp", "$avp(i:4242)") >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 0) >> >> in event_route[tm:local-request] i use >> if(is_known_dlg()) { >> xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I >> } >> >> And when call drop by timeout, sometimes in log i see >> /usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null> >> >> But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334" >> >> Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime. >> Different between unsuccessful and successful call i dont see >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > ------------------------------

2 1

Topoh module & call leg/transaction does not exist
by Olli Attila 06 Mar '17

06 Mar '17

Hello, I recently started testing topology hiding module on Kamailio and encountered a problem where B party hangs up, Kamailio can't identify the ongoing transaction to which the BYE should be related to and instead of tearing the call down Kamailio just sends back "481 Call leg/Transaction does not exist" to B party. On Kamailio config I have only enabled the topoh.so -module with default options and configured this to enable masking: modparam("topoh", "mask_callid", 1) Here is what the INVITE to B party looks like and the BYE message coming from the B party: http://pastebin.com/raw/f63zUmid When debugging on Kamailio end i get: http://pastebin.com/raw/ZefGAz30 Is this a Kamailio issue or a softphone issue on B party end? Cheers, --Olli

3 6

kamailio re.subst question.
by José Seabra 02 Mar '17

02 Mar '17

Hello There, I'm trying using the following regexp on re.subst function: (.*)Calling-Name: ?([\W|0-9A-Za-z_]{5,500} ?[<>:@.|0-9A-Za-z_]{5,500})(.*)$ And the string used for match is the following: #015#012Calling-Name-Status: available#015#012Calling-Name: "josé" < sip:5555555552@10.10.10.10>#015#012Presentation-Indicator: allowed#015#012 The Kamailio re.subst function is constructed as the following: $sht(cnam=>$ci::calling_name)=$(var(rb){re.subst,/(.*)Calling-Name: ?([\W|0-9A-Za-z_]{5,500} ?[<>:@.|0-9A-Za-z_]{5,500})(.*)$/\2/s}); The issue here is that this re.subst function doesn't match the accents, I have tested this regex on regex101.com and it does match. My question is how i can get the same result with re.subst? Anyone can help? Thank you Regards -- José Seabra

2 5

Dimensioning of shared memory for carrierroute
by Igor Potjevlesch 02 Mar '17

02 Mar '17

Hello, I'd like to dimension shared memory to use "kamctl cr reload" without issue. Currently, my table "carrierroute" has 204k entries and if I do "kamctl cr reload", the reload fails. Just after, I have thousands of these errors: Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: <core> [sip_msg_clone.c:507]: sip_msg_shm_clone(): cannot allocate memory Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: tm [t_lookup.c:1332]: new_t(): ERROR: new_t: out of mem: Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: tm [t_lookup.c:1472]: t_newtran(): ERROR: t_newtran: new_t failed After a reboot, Kamailio back to normal. Actually, I start Kamailio with -m 704 and -M 128. Thank you for your advises. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 2

Help with KEEP-ALIVE method
by Sergey Basov 02 Mar '17

02 Mar '17

Hi All. Today I have problem with connection from 1 of the clients. Their PBX sends KEEP-ALIVE after some time after REGISTER. I have next error in kamailio log Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR: <core> [tcp_read.c:1354]: tcp_read_req(): bad request, state=7, error=4 buf:#012KEEP-ALIVE sip:1.2.3.4 SIP/2.0#015#012#015#012#012parsed :#012KEEP-ALIVE sip:1.2.3.4 SIP/2.0 After which my kamailio closes TCP session.... This connection is done with sip TLS. Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

3 4

Lost of connection with MySQL?
by Igor Potjevlesch 02 Mar '17

02 Mar '17

Hello, Two times in the last week, Kamailio reports MySQL errors like these: /usr/local/sbin/kamailio[19225]: ERROR: db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on '192.168.2.6' (4) (2003) /usr/local/sbin/kamailio[19225]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query /usr/local/sbin/kamailio[19225]: ERROR: acc [acc.c:504]: acc_db_request(): failed to insert into database These errors are repeated many times and cause Kamailio to hang and don't reply to SIP requests. A reboot of Kamailio solves the issue. Anyone has already had this issue and had solved it? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 17

CentOS 6 RPMs don't provide utils.so nor http_client.so
by Trevor Peirce 01 Mar '17

01 Mar '17

Hello, Is there any reason the CentOS 6 RPMs don't build utils.so or http_client.so? Looking back to 4.3 it appears it was being built: kamailio-utils-4.3.6-1.1.x86_64.rpm However it is no longer available in 4.4 nor 5.0 RPMs for either CentOS 6 or 7. Thanks, -- Trevor Peirce AcroVoice Solutions Inc

2 5

Update Kamailio tables
by Andy Chen 01 Mar '17

01 Mar '17

Hi all, What is the quickest way to update Kamailio tables? I wanted to check out the SIPTRACE module and I noticed the table is not there. Thanks. -- Andy Chen

2 2

Kamailio -can we change 478 to 503 when dns fails?
by José Seabra 01 Mar '17

01 Mar '17

Hello there, Some times dns server fails and my kamailio server start replying "Unresolvable destination (478/SL)". The propose of my email is to know if is there any possibility to change this reply (478) to for example one of 5XX replies. I have modparam("sl", "default_code", 503) but in situations where dns fails, kamailio doesn't reply 503. Thank you for your support Regards José Seabra

2 4

Figuring out why topos stops script from working
by Trevor Peirce 01 Mar '17

01 Mar '17

Hello, I'm having some trouble with topos not working well when mixed with nathelper (and maybe other cases, too). There is a bug that describes exactly the problem at https://github.com/kamailio/kamailio/issues/1005 Has anyone else had trouble with this module? Anything I'm missing? Thanks, -- Trevor Peirce AcroVoice Solutions Inc

3 2

Unknown $dlg(h_entry) and $dlg(h_id) in dialog
by Никитенко Виталий 01 Mar '17

01 Mar '17

Hi i have kamailio 4.2, dialog module with parametr modparam("dialog", "dlg_flag", 5) modparam("dialog", "enable_stats", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "timeout_noreset", 1) modparam("dialog", "send_bye", 1) modparam("dialog", "timeout_avp", "$avp(i:4242)") modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 0) in event_route[tm:local-request] i use if(is_known_dlg()) { xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I } And when call drop by timeout, sometimes in log i see /usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null> But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334" Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime. Different between unsuccessful and successful call i dont see

2 1

Scaling websockets over SIP
by Jade SZ 01 Mar '17

01 Mar '17

Hi Guys, I am using kamailio/rtpengine and webrtc over sip client for a/v calls and all works fine. Now I want to scale it to 2 or more than 2 kamailio servers. As per search on mailing list I could: 1) Websocket Outbound/Edge Proxy 2) Each UA client to register with its Regards, Jade SZ

1 2

pushing alarms/counters other than with SNMPstats
by Valentin Christoph 28 Feb '17

28 Feb '17

Hi all I'm looking for ways to push stats/alarms from kamailio to some external destination other than using the SNMPstats module. Any ideas? Thanks Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

5 5

sip over websockets sip uri
by Jade SZ 28 Feb '17

28 Feb '17

Hi Team, Question: Just wanted to clarify regarding SIP URI in webrtc (over sip) connection. e.g. if we have a scenario where Kamailio is hosted with websocket support. Websocket URI is used to send packets to to wss/ws address and the SIP URI goes with it. I have tested it with any SIP URI including any domain and it works if I authenticate that domain, regardless that domain is real or not. // client side SIP URI config var configuration = { uri : 'sip:alice@example.com', }; Can we use any central domain for sip uri similar to realm concept and send traffic to multiple kamailio servers through websocket LB. dummy.domain - for ws/wss SIP URI kamailio-a.domain.com kamailio-b.domain.com Thanks in advance. Regards, Jade

2 4

Newbie starting - looking for docs / tutorials
by Antony Stone 28 Feb '17

28 Feb '17

Hi. I'm just starting with Kamailio - I've been to a few presentations about it, at Fosdem and KamailioWorld, but never used it so far. I'm pretty familiar with Asterisk, and I want to use them in combination. My first use case is to handle SIP registrations (both as a client, registering to a remote SIP server, and as a server, accepting registrations from handsets) in Kamailio instead of Asterisk, and still be able to route calls to/from the registered accounts through the Asterisk dialplan. I suspect this is a pretty common starting point for people who've used Asterisk and want to add Kamailio to the setup? So far I've been unsuccessful finding any Howto or Tutorial explaining how to do this - I can't help feeling I must simply be looking in the wrong place or in the wrong way. Searching for topics like "Asterisk Kamailio Getting Started" I find: http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf but with 12 chapters and 85 pages I'm not sure which bits are the essential parts to do what I need. I've also found http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- asterisk-11.3.0-astdb but this seems to focus on a database backend, which I guess I'll want to use in due course, but doesn't seem the simplest place to start just to learn how Asterisk & Kamailio work together? So, please can someone suggest a beginner's guide to introducing Kamailio into an Asterisk setup, keeping things as simple as possible to start with, so I can learn about Kamailio one bit at a time (since there do seem to be a lot of bits to it) from a familiar environment of using it with Asterisk? Thanks in advance, Antony. -- What do you get when you cross a joke with a rhetorical question? Please reply to the list; please *don't* CC me.

2 3

multi domain kamailio
by przeqpiciel 28 Feb '17

28 Feb '17

Let's suppose that i have two machines with installed asterisk and one with kamailio. I would like have routing based on sip domain. for domain sip.domain1.com send sip signalling to asterisk#1 server and for sip.domain2.com send to asterisk#2 server. What i should use? Domain module or what ? Tomasz

3 4

Kamailio v5.0.0 Released - new major version is out
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Kamailio v5.0.0 is out – it comes with 6 new modules and a consistent set of improvements touching more than 50 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to restructuring the source code tree to meet more modern patterns; introducing the KEMI framework that allows writing SIP routing blocks in other embedded interpreters such as Lua, JavaScript or Python; removal of obsoleted MI control mechanism, being replaced by RPC. Enjoy Kamailio v5.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2017 - http://www.kamailioworld.com

1 0

problem with avpops and mongodb modules
by hdssdsdsdsfsdf hdssdsdsdsfsdf 27 Feb '17

27 Feb '17

1 1

Releasing Kamailio v5.0.0
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Hello, I am considering to release Kamailio v5.0.0 at the beginning of next week, depending how things evolve in the next days, either on Monday, Feb 27, or Tuesday, Feb 28. I haven't noticed any new issues specific for the new features added for 5.0 lately, so it looks pretty good so far. If you are aware of issues not reported to the bug tracker, register them as soon as possible at: - https://github.com/kamailio/kamailio/issues There are still some parts of documentation that have to be updated, I am planning to work on them during the next days, but help from others are very appreciated, especially details of what you had to do changes while upgrading from 4.4 to 5.0. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

3 6

Kamailio 4.3.x location_attrs update issue
by Mridul B 24 Feb '17

24 Feb '17

Hi, We've noticed a case where the *location_attrs* table isn't updated when there's an unexpected shutdown of a SIP device (here there'll be no REGISTER message sent with expires=0). We're using Kamailio v4.3.x When the SIP device shuts down ungracefully. The location table is updated using DELETE query This is done by the timer which deletes the expired contacts, by default the timer_interval is set as 60. Here, the logs of auto-removal of expired contacts entries by the timer > Feb 17 17:16:13 localhost kamailio[23273]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:14' AND expires<>'1970-01-01 00:00:00'] > Feb 17 17:16:14 localhost kamailio[23272]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:15' AND expires<>'1970-01-01 00:00:00'] The delete operation is only on location table, shouldn't we have the same implementation on the location_attrs table based upon ruid or username? We're getting duplicate entries in the location_attrs table of the same sip user after it registers. But when the SIP device un-registers, the entries in the location as well as in the location_attrs table is deleted Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: usrloc [ucontact.c:1725]: > uldb_delete_attrs(): trying to delete location attributes > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location_attrs where username='mridul' > AND ruid='uloc-58a72243-5ace-2'] > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location where username='mridul' AND > contact='sip:mridul@127.0.0.1:37881;rinstance=b841a8c916e47dfa'] I think the usrloc module should have the implementation of deleting the entries of expired sip users based on its ruid/username in the location_attrs table also. Thanks for the help. Best regards, MRIDUL

2 3

want to see active users SIP users
by Mandar Nagarkar 24 Feb '17

24 Feb '17

Hello Everyone, I want to see active users SIP users for this purpose I checked the location table of kamailio database it does have active users entries however as per my observation there is 45 secs to 1 min delay before the entry actually added to this table i.e. suppose if I am registering the user through pjspi,jssip or any client I cannot see entry for that user in location table. I have to wait up-to 1 min to see the user in location table. Is there any way we can Identify if user is online without delayed entry? Regards, Mandar nagarkar The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 1

Fetching proxy-autheticate header from 407 Response
by vivek＠advaitamtech.com 23 Feb '17

23 Feb '17

Hi All, Am not using the authentication in the kamailio. So I will be receiving the "407 Proxy Authentication Required" response from the third party server for the first INVITE. I want to fetch the Proxy-Authenticate header present in the 407 response. I browsed and tried using the pseudo-variables "$adu/$aa/$ar". It dint work. Please let me know if there is a way to fetch this header. Thanks, Vivek.

3 4

How to release call without having to use fromTag and toTag, and resume a suspended call
by Cassandra Yurekli 23 Feb '17

23 Feb '17

To whom it may concern, There are several points I don’t quite understand regarding a couple of kamailio modules. I have a main application which controls calls by sending command messages via a socket (for example reject this sip invites). I communicate with the socket client via evapi module and using evapi_asynch_relay() in order to be sure that communication are suspended and kamailio waiting an order for doing something with them (Accepte, reject, release, ...). I want to release a call, using a command “release this call”. I’m actually doing this with dialog module (using get_dialog(callid,fromTag,toTag) and dlg_bye(‘all’)) but is there anything else (like another module) which could release a call without necessarily having to use fromTag and toTag to get the context ? I didn’t find it on modules documentations. Also after an INVITE I’m waiting for the main app to accept or reject the call. For accepting and continuing the transaction I found “t_continue(tindex,tlabel,route)” on tmx module, but at this moment $T(id_index) and $T(id_label) are null, is it normal or I did something wrong ? How can I simply resume the suspended call ? Thanks; Best regards,

2 1

Need help with tcpops
by Abdoul Osséni 22 Feb '17

22 Feb '17

Hi, My call flow is: Softphone --> TLS ---> Kamailio --> UDP --> ASTERISK --> PSTN I want to use tcpops module between the softphone and kamailio. http://www.kamailio.org/docs/modules/4.4.x/modules/tcpops - enable tcpops : no problem - disable tcpops : two cases - cancel or bye from softphone: no problem. - cancel or bye to softphone via kamailio: how disable tcpops? I can't use $avp(bye_conid) because it is asterisk thant sending the sip message. Can we use $avp(caller_conid)? In my case, the softphone is always at the origin of the call establishment. Regards Abdoul.

2 1

Automated integration tests for individual modules
by A Lai 22 Feb '17

22 Feb '17

Hi, I just started working with the Kamailio code base. I am wondering if there are some common strategies in having automated tests on proper functioning of modules when loaded into cfg.

2 1

problem with avpops and mongodb modules
by hsdsdssdfsdf dsasd 22 Feb '17

22 Feb '17

2 1

How to handle CANCEL if corresponding INVITE transaction is missing
by Venkatesh Macha 22 Feb '17

22 Feb '17

Hi all, I am using Kamailio 4.4.5, It is working fine. But in one case. If i make a video call between the two users and cancels the call then Call is not disconnecting on the callee side. Please look the following scenario. 1000 made one video call to 2000 2000 is ringing. Now 1000 disconnected the call by sending CANCEL. But 2000 is not disconnecting. This is happening because Kamailio is unable to find the corresponding INVITE transaction for the CANCEL so it is simply dropping the requests. Why Kamailio unable to find the transaction. And is there any possibility to handle this case and send the CANCEL to callee side. I tried with forwad() and someother things but it is just generating the loops. I am also attaching the ngrep trace. Thank you in Advance. CANCEL_not_working_nabble.CANCEL_not_working_nabble <http://sip-router.1086192.n5.nabble.com/file/n156243/CANCEL_not_working_nab…> -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-handle-CANCEL-if-correspondi… Sent from the Users mailing list archive at Nabble.com.

2 2

kamailio memory leak in private memory
by david 22 Feb '17

22 Feb '17

Hello all We are observing an issue in one kamailio proxy we use. The consecuence of an unknow cause is that the private memory of some UDP receiver processes in where we listen from a particular IP, seems to be draining along the time while processing traffic. Quite "slowly" but maybe aroung 200K in 3-4hours, depending on the moment of the day. This is quite weird for us, because it seemed like it was caused by some particular traffic profile, since we have several IP listening (8 children each) and we only see the issue in the UDP receiver processes of some of those IPs. Besides we have moved the kamailio service listening on the same IP to several different physical servers, and the behaviour is the same. We look at it ussing pkg.stats kamcmd command and for instance if we track one of the process affected pid: 23784 rank: 53 used: 2838616 free: 13005408 real_used: 3771808 pid: 23784 rank: 53 used: 2842584 free: 12999280 real_used: 3777936 This 2 outputs were taken with 10min diff between each other. But we dont know where or how to look at it. We have been these last days trying to search for patterns to allocate some particular traffic source causing something weird, but no luck. We have checked all the SIP INVITE messages looking for anything weird at any field but no luck, besides we sometimes see some bad headers with blank spaces in the uri or thing like that but those cases we have an error and we dont think it can cause a memory leak. Shared memory seems to be stable. We also have checked the config we are using and we have been discarding possible issues since we dont think the $var variables we are using could cause that issue (but you never know....) We tried to debug the kamailio on one affected IP compiling it with DBG_SYS_MALLOC , and doing the dump of the pkg mem status when the kamailio was restarted and no traffic was processed, and also when it had traffic processed. in this last case, we removed traffic, waited for 20min and made the dump i attach you both files, we have the debug logs, but they are very big to be passed here, and we dont know what to look for :) could you please give us some guidance or clue to at least start to searching for something that can cause this issue? thanks alot and regards david

3 6

parse_addr_spec(): unexpected char [^M] in status 6:
by Abdoul Osséni 21 Feb '17

21 Feb '17

Hello, Could you help us to fix this error? <core> [parser/parse_addr_spec.c:662]: parse_addr_spec(): unexpected char [^M] in status 6: Regards Abdoul.

3 5

Transaction-persistent data and uac_req_send()
by Alex Balashov 21 Feb '17

21 Feb '17

Hello, I am using uac_req_send() to send INVITEs to a redirect server. However, upon receiving the redirect, I need access to some state information that was last available at the time of sending the request. Does uac_req_send() allow me to persist AVPs like a normal TM transaction? If not, what are my options, other than building my own state using htable or what have you? I would really rather not maintain—and be responsible for garbage-collecting—externally constructed state like that if possible. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 10

Kamailio 4.4 with 2 Asterisk 13
by Graham Brown 20 Feb '17

20 Feb '17

I'm trying to load balance 2 Asterisk servers with Kamailio offloading the registration to Kamailio using the Asterisk real-time database tables. I have a problem with the dispatcher module, if there is a t_on_failure route it always goes into that, if I remove that then I would expect every call to round robin but I'm only ever hitting the same server from the same extension (seems like sticky session based on source ip). 1. how do I fix my failure route so I can reboot server 1 and server 2 will be used 2. how to I change the sticky session behaviour. Another issue I am facing with extensions registering to Asterisk are only avaliable on the server they routed to My assumption was if using the Asterisk real-time database I would get the contact on both servers. Maybe my architecture is incorrect, I'm open to suggestions. Currently Kamailio is not storing locations (assuming that is the extension contact) maybe that is a better solution, but I dont know how to configure Asterisk to forward everything to Kamailio then or integrate via the database or something. Thanks in advance for any help Regards Graham

2 1

Cleaning wiki accounts
by Daniel-Constantin Mierla 20 Feb '17

20 Feb '17

Hello, I noticed a lot of random usernames for the accounts created to edit the wiki. Apart of several which did some which did some edits to add spam, most of them are inactive. Not sure what would be the real purpose behind them, but I am going to remove anything that doesn't look as a community user. Shall I remove an account with random id, but for a good user, let me know, it can be readded/account recreated. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

memory: Free fragment not found
by Dmitri Savolainen 17 Feb '17

17 Feb '17

Hi. My kamailio v4.4.5 installation ( -m 512 -M 64 ) stop sip processing with repeatable error: Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:415]: > fm_search_defrag(): fm_search_defrag(0x7f39917fa010, 952); Free fragment > not found! > Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:498]: > fm_malloc(): fm_malloc(0x7f39917fa010, 952) called from core: rvalue.c: > rval_new_empty(224), module: > core; Free fragment not foun while it was enough free shared memory according to kamcmd> core.shmmem m > { > total: 512 > free: 472 > used: 27 > real_used: 39 > max_used: 51m > fragments: 37035 > } whether it was about package memory or smth else? Thanks! -- Savolainen Dmitri

2 4

MSILO offline messages are coming in reverse order.
by Venkatesh Macha 17 Feb '17

17 Feb '17

Hi all, I am using MSILO module for the offline messaging feature. It is working good. But If I send more than one message to the offline user. Those messages are delivering in the reverse order. Please consider the following scenario. A sending B ( B is offline, so messages are stored in DB) hi hello how are you Now B comes online and registers. Kamailio sending the messages but App is getting them in reverse order. how are you hello hi Is there any possibility to tell Kamailio to send them in order. ? Thank you in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/MSILO-offline-messages-are-coming-i… Sent from the Users mailing list archive at Nabble.com.

2 2

Problem with dispatcher on Kamailio 4.2 and Debian 8
by przeqpiciel 17 Feb '17

17 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 4

Kamailio 4.2 on Debian 8 and dispatcher problem
by przeqpiciel 16 Feb '17

16 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 1

Re: [SR-Users] Timejump causes kamailio to hang/stall
by erik 16 Feb '17

16 Feb '17

Thanks Daniel, I will look into that. Indeed I found the logic that corrects the time ticks for any drifting, using the time of day. I guess that causes the registrations to expire. I am out of the office for a couple of days, but will get back to you. Best Regards, Erik Hello, On 15/02/2017 14:47, Erik Visser wrote: > After I have kamailio up and running, with UAs properly registrered and sessions routed successfully, my platform faces a timejump. > The platform starts up with date&time set to 1-1-2000 and after kamailio is up and running, the date&time jumps to the current date&time, causing two effects: > - I loose all my registrations (since they expired in 3600 seconds) > - kamailio does not respond to any SIP messages for a couple of minutes > > Is there a way to have kamailio use timeticks rather than relative time for the registrations? > Does anybody recognize the hanging behavior? Any suggestions? > kamailio is using unixtime for registrations, not relative time. But the timer callbacks are using relative intervals to know when to be executed (e.g., timer callbacks are defined to be executed like very 1 minute, or every 90 seconds). It may be that some of the timers try to catch up on missed intervals. It is strange that you say no sip traffic is processed. Can you simulate again the situation and get the backtrace with gdb from few sip workers? If your traffic is over udp, then do 'kamctl ps', get the pid of processes with index 1, 2 and 3, then use gdb to attach to those PIDs and grab the output of: bt full info locals list Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

remove offline message prefix of Offline message of MSILO module
by Venkatesh Macha 16 Feb '17

16 Feb '17

Hi all, I am implementing Offline messages functionality using the MSILO module. Everything went well. I am able to store and send those offline messages when the user comes online. But my present problem is, I don't want the offline message prefix. Please look at the following example. This is what Kamailio sending. [Offline message - Wed Feb 15 05:43:38 2017] Test2 I want only Test2 Is there any possibility to remove that offline message prefix ? Thank you in advance ! -- View this message in context: http://sip-router.1086192.n5.nabble.com/remove-offline-message-prefix-of-Of… Sent from the Users mailing list archive at Nabble.com.

2 2

Timejump causes kamailio to hang/stall
by Erik Visser 16 Feb '17

16 Feb '17

After I have kamailio up and running, with UAs properly registrered and sessions routed successfully, my platform faces a timejump. The platform starts up with date&time set to 1-1-2000 and after kamailio is up and running, the date&time jumps to the current date&time, causing two effects: - I loose all my registrations (since they expired in 3600 seconds) - kamailio does not respond to any SIP messages for a couple of minutes Is there a way to have kamailio use timeticks rather than relative time for the registrations? Does anybody recognize the hanging behavior? Any suggestions? Thanks! Erik ____________________ Erik Visser erik(a)vissershut.nl

2 1

IPv6 addresses in dispatcher.list file
by Joel Serrano | VOZELIA 16 Feb '17

16 Feb '17

Hi, What is the correct format for the dispatcher.list file using IPv6? I have the following file: $ cat /etc/kamailio/dispatcher.list # $Id$ # dispatcher destination sets # # line format # setit(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # proxies 4 sip:XX.XX.XX.XX:5060 4 sip:ZZ.ZZ.ZZ.ZZ:5060 6 sip:[X:X:X:X::X:1]:5060 6 sip:[Z:Z:Z:Z::Z:2]:5060 $ For some reason, Kamailio doesn't recognize the v6 endpoints: kamcmd> dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 4 TARGETS: { DEST: { URI: sip:XX.XX.XX.XX:5060 FLAGS: AP PRIORITY: 0 } DEST: { URI: sip:ZZ.ZZ.ZZ.ZZ:5060 FLAGS: AP PRIORITY: 0 } } } } } kamcmd> I have tried multiple combinations for the v6 format but I cannot get one to work. What am I doing wrong? Thanks, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

2 1

record_route() function
by Andy Chen 15 Feb '17

15 Feb '17

Hi all, I need some advise on how to use the record_route() function. So I make that call when the first method comes in and there are two Record-Route headers created. However, one of of the headers I want to add transport=tcp. What is the best way to do this? I've tried adding that string per documentation within the () and added it to both. Thanks. -- Andy Chen

3 10

Branch 5.0 created
by Daniel-Constantin Mierla 15 Feb '17

15 Feb '17

Hello, the GIT branch 5.0 has just been created, it will host the release series 5.0.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.0 origin/5.0 Hopefully before the end of the month the full release of 5.0.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.0. No new features can get in branch 5.0. Enhancements to documentation or helping tools are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

4 9

Dispatcher and "src based" routing algorithm
by Olli Attila 15 Feb '17

15 Feb '17

Hello, I am using kamailio 4.4.3 (x86_64/linux) 5a2195 and Dispatcher module to route calls towards PSTN (4 pstn gateways). The Dispatcher module is using round robin mechanism to select the pstn gateway. I have built a routing logic in a way that when call arrives to Kamailio from one of the PSTN gateways, and fails to route towards an external sip target, the call is marked as CFU call (a diverting call) and routed back to PSTN with different B number, but the actual challenge is that I would like the target PSTN gateway _not_ to be the same PSTN gw from where the call originally came in to Kamailio. This feature is requested because of the technical limitations of the pstn gateway software. So is there any mechanism in Dispatcher or elsewhere in Kamailio to do this? Basically I would need a flag or something to mark the pstn gateway when the call comes in and assure that when routing back to pstn, the same gateway would not be used at all, or it should be used as last target from the four gateways in Dispatcher. Cheers, -- Olli

1 0

Refreshing: info about private emails
by Daniel-Constantin Mierla 15 Feb '17

15 Feb '17

As I still receive a lot, I am re-posting the notification about private emails, either initiating a topic or sent directly after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon). I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get this instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

TOPOH MODULE
by Annus Fictus 14 Feb '17

14 Feb '17

Hello, I'm using TOPOH module but I have a problem with Call-ID. If I use: modparam("topoh", "mask_callid", 1) and send the INVITE to a media server, when media server send BYE to KAMAILIO, Kamailio answer like transaction not exist. In the parameter description, I read i have to use "lreq_callee_headers" dialog parameter to resolve this kind of issues, but I can't find this parameter on the module description. I'm using Kamailio 4.4.5 Thank's Regards

3 3

Remove
by Ryan Holbein 13 Feb '17

13 Feb '17

Can you please remove me

2 1

SIP-T conversion to SIP
by Martin Quevedo 13 Feb '17

13 Feb '17

hi all I need to interact with an MSC that talks SIP-T that is the MSC embeds ISUP in the SIP messages IAM, ANM, ACM and REL messages specifically but I have a SIP client which does not speak SIP-T is it possible that kamailio parses invites with IAM ISUP parameters like forwarding number and forwarding reason for incoming messages and embeds ACM, ANM and REL isup messages in OK, Session Progress and BYE regards, Martin ?

1 0

VUC in 2 hours - Allison Smith and Kamailio World
by Daniel-Constantin Mierla 10 Feb '17

10 Feb '17

Hello, short note for the people that might be interested on the topic - Allison Smith (aka, the voice of Asterisk) will be on VoIP Users Conference later today, in about 2 hours, among the topics of discussion being the Kamailio World 2017. I will also join, trying to squeeze some updates about upcoming Kamailio 5.0. More details about the online audio/video conference can be found at: - https://www.vuc.me/2017/vuc632-get-ready-for-kamailio-world-2017-with-allis… The session will start in about 2 hours (17:00UTC), anyone can connect via webrtc capable browser, SIP or PSTN dial in. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

python_exec and parameters
by gmele 10 Feb '17

10 Feb '17

Hello, in our Kamailio, we are using python_exec to execute script used to send push notifications to Google or Apple. Currenty, the method we call only accepts the message as parameter. We then extract from the message the Call Id, to URI and from URI. Here is the python method signature: *def SendPushNotification(self, msg)* To extract the toUri and fromUri (and remove the sip:), we use python regex : *uriPattern=re.compile(".*sips?:(.+@[^>;]+)")* This is working well 99% of the time, except sometimes (we don't know why) where we get the error: *logger_LM_ERR(): Exception occurred: Traceback (most recent call last): File "mypythonExample.py", line XXX, in SendPushNotification requestUri = self.ExtractURIFromHeader(msg.getHeader("to")) File "mypythonExample.py", line XXX, in ExtractURIFromHeader m= self.uriPattern.match(header); TypeError: an integer is required* We tried to analyze the cause of this error, but we didn't find any problem with the message itself (the message is complete, with the *to *and *from *correctly filled). So, to avoid using the python regex, we have decided to pass parameters to the python script (callId, from and to URI): *def SendPushRequest(self, msg, fromUri, toUri, callid)* But it seems python_exec doesn't support more than 2 parameters ($mb and another one). Is this exact? If this is the case, how can we pass several parameters from kamailio.cfg? Thx for your help. Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/python-exec-and-parameters-tp155803… Sent from the Users mailing list archive at Nabble.com.

3 8

Video and Slides: SIP and WebRTC Load Balancing and High Availability for FreeSWITCH, FOSDEM
by Giovanni Maruzzelli 10 Feb '17

10 Feb '17

Hello, here you can find both video and slides of my presentation at FOSDEM 2017, Bruxelles: https://fosdem.org/2017/schedule/event/freeswitch/ Enjoy! -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT Cell : +39-347-2665618

1 0

MSRP with Kamailio
by Olli Attila 10 Feb '17

10 Feb '17

Hello, I am running kamailio version 4.4.2. I just configured MSRP module to kamailio with example config taken from "http://www.kamailio.org/docs/modules/4.4.x/modules/msrp.html#msrp.usage" (Example 1.16. Event Route (using htable for MSRP connection tracking). After configuring msrp relay, the kamailio service has been restarted succesfully and no errors occurred. Blink is configured to use my server name with tls and port 5061 for MSRP. I get this error on Blink when i try to send files: https://p.dnaip.fi/X4vwoy40 Any suggestion what is going wrong? Cheers, -- Olli

3 12

Replace old registration record with new one while keeping single AOR per contact
by Aqs Younas 09 Feb '17

09 Feb '17

Greetings list, I want to keep single AOR per contact. If new regitration comes i want to replace the old record with new. Suppose User Alice registers from device A and after certain time it try to register from device B. I want Registration from device B to replace A. Any pointer is mush appreciated. Cheers, Aqs

3 6

Fwd: [sr-dev] git:master:c32a5f64: sipcapture: added $hep(src_ip) variable
by David Villasmil 09 Feb '17

09 Feb '17

Hello guys, i just installed kamailio from deb http://deb.kamailio.org/kamailio50-nightly jessie main deb-src http://deb.kamailio.org/kamailio50-nightly jessie main And i'm trying to use the newly exposed $hep(src_ip) by: if ( ds_is_from_list("10", "2", "sip:$hep(src_ip)" ) ) but i'm getting: ERROR: sipcapture [sipcapture.c:2763]: pv_parse_hep_name(): REQUEST, PRE, src_ip ERROR: sipcapture [sipcapture.c:2793]: pv_parse_hep_name(): unknown hep name src_ip ERROR: <core> [core/pvapi.c:877]: pv_parse_spec2(): pvar "hep" has an invalid name param [src_ip] ERROR: <core> [core/pvapi.c:1032]: pv_parse_spec2(): wrong char [)/41] in [$hep(src_ip)] at [11 (5)] ERROR: <core> [core/sr_module.c:1251]: fix_param(): bad PVE format: "sip:$hep(src_ip)" ERROR: <core> [core/mod_fix.c:539]: fixup_spve_null(): Cannot convert function parameter 1 to spve Any ideas? Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Fri, Dec 9, 2016 at 10:17 AM Subject: [sr-dev] git:master:c32a5f64: sipcapture: added $hep(src_ip) variable To: sr-dev(a)lists.sip-router.org Module: kamailio Branch: master Commit: c32a5f64de4ff731488d9f6bf64651117d26d46e URL: https://github.com/kamailio/kamailio/commit/ c32a5f64de4ff731488d9f6bf64651117d26d46e Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: 2016-12-09T10:16:31+01:00 sipcapture: added $hep(src_ip) variable - reworked patch from GH #879 --- Modified: src/modules/sipcapture/sipcapture.c --- Diff: https://github.com/kamailio/kamailio/commit/ c32a5f64de4ff731488d9f6bf64651117d26d46e.diff Patch: https://github.com/kamailio/kamailio/commit/ c32a5f64de4ff731488d9f6bf64651117d26d46e.patch --- diff --git a/src/modules/sipcapture/sipcapture.c b/src/modules/sipcapture/ sipcapture.c index 117db74..b5bbe7b 100644 --- a/src/modules/sipcapture/sipcapture.c +++ b/src/modules/sipcapture/sipcapture.c @@ -2835,6 +2835,11 @@ static int pv_parse_hep_name (pv_spec_p sp, str *in) else goto error; } break; + case 6: + { + if(!strncmp(in->s, "src_ip", 6)) sp->pvp.pvn.u.isname.name.n = 2; + else goto error; + } case 7: { if(!strncmp(in->s, "version", 7)) sp->pvp.pvn.u.isname.name.n = 0; @@ -2857,17 +2862,24 @@ static int pv_parse_hep_name (pv_spec_p sp, str *in) static int pv_get_hep(struct sip_msg *msg, pv_param_t *param, pv_value_t *res) { + static char sc_buf_ip[IP_ADDR_MAX_STR_SIZE+12]; + int sc_buf_ip_len; + if(param==NULL) return -1; switch(param->pvn.u.isname.name.n) { case 0: - return pv_get_uintval(msg, param, res, hep_version(msg)); - case 1: - return pv_get_uintval(msg, param, res, hep_version(msg)); + return pv_get_uintval(msg, param, res, hep_version(msg)); + case 1: + return pv_get_uintval(msg, param, res, hep_version(msg)); + case 2: + sc_buf_ip_len=ip_addr2sbuf(&msg->rcv.src_ip, sc_buf_ip, + sizeof(sc_buf_ip)-1); + sc_buf_ip[sc_buf_ip_len]=0; + return pv_get_strlval(msg, param, res, sc_buf_ip, sc_buf_ip_len); default: return hepv3_get_chunk(msg, msg->buf, msg->len, param->pvn.u.isname.name.n, param, res); } return 0; } - _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev ᐧ

1 0

rtpengine key file
by Kjeld Flarup 09 Feb '17

09 Feb '17

Hello I have build rtpengine on a Raspberry PI 3 running Jessie, but it fails on startup pi@raspberrypi:/usr/local/src/rtpengine $ rtpengine --log-level=7 --log-stderr [1486630112.767604] CRIT: Fatal error: Bad command line: Key file does not start with a group Most likely I'm missing something which is found on regular PC Linux distributions. But what? Googling the error message reveals almost nothing and nothing related to rtpengine Any hints will be appreciated. -- Med venlig hilsen / Best regards Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef Viptel ApS, Hammershusvej 16C, DK-7400 Herning Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk

2 2

LCR module HowTo (Spanish)
by Campus VoToVoice 09 Feb '17

09 Feb '17

Hello, My small contribution to Kamailio's documentation: https://www.voztovoice.org/?q=node/1491 If you find any error, please write me. Regards

2 1

Branching v5.0
by Daniel-Constantin Mierla 09 Feb '17

09 Feb '17

Hello, following discussion on mailing lists and at Fosdem over the last weekend, I am considering to create the git branch 5.0 later this week, on Thursday, Feb 9, 2017. Afterwards, the master branch can take commits with new features. The branch 5.0 will be tested for 1-2 more weeks and if all is fine, the v5.0.0 can be released. Should anyone have other suggestions, jump into discussion. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

nathelper module fix_nated_register(); question
by JR Richardson 09 Feb '17

09 Feb '17

Hi All, Just confirming when using nathelper, the fix_nated_register function is only used to write corrected contact into to $avp then to database, this function does not actually change the contact in the SIP message when passing through the proxy, correct? fix_nated_contact is used to actually change the SIP message contact when passing through the proxy. I'm using kamailio 4.2. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 2

dlg_set_timeout bye to websockets client
by Switch168 09 Feb '17

09 Feb '17

Hello, Just wondering if it's possible to intercept the bye message that dlg_set_timeout sends out to do some nat_helper function to fix it so the bye can be sent to random.invalid uris. I know there is the edge proxy outbound module but wondering if I can do it without. Regular bye between 2 clients i was able intercept and fix with nat helper but I dont know how to fix the byes that is sent out from dlg_set_timeout function. Thanks in advance! Andrew

3 7

Re: [SR-Users] kamailio with cassandra
by Никитенко Виталий 09 Feb '17

09 Feb '17

Hi. >> Can you provide more details for what you need it,maybe we can provide some hints? I want to store the data of registered users (location) in the database cassandra (it would be great also dialog stored in cassandra). >>Can you provide access to a test system/vm where to try to fix it Yes, I can it, will prepare today the VM for access to public network and send to email data access. Thanks for the help. > Hello, > > unfortunately the module was not maintained for a while. Can you provide > access to a test system/vm where to try to fix it, because I don't use > it and have no platform where to test? > > There is another module, ndb_cassandra, depending of what you want to > do, it might help. Can you provide more details for what you need it, > maybe we can provide some hints? > > Cheers, > Daniel > > On 07/02/2017 10:55, Никитенко Виталий wrote: >> Hi! >> I want start kamailio with db_cassandra. do as it written in https://www.kamailio.org/docs/modules/4.2.x/modules/db_cassandra.html. Then trying kamctl start, failed with error "table version does not exist!". >> From CQLSH table is view >> cqlsh:kamailio> Describe tables; >> version location >> >> cqlsh:kamailio> select * from version; >> table_name | table_version >> ------------+--------------- >> version | 1 >> >> Tell me please wherein may be trouble? Full log below >> >> ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/location >> ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/version >> INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob >> INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available >> INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 >> INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled >> INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.allow >> INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio//permissions.allow) not found => empty rule set >> INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.deny >> INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio//permissions.deny) not found => empty rule set >> INFO: misc_radius [misc_radius.c:164]: mod_init(): initializing... >> ERROR: db_cassandra [dbcassa_base.cpp:449]: cassa_translate_query(): table version does not exist! >> ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster >> ERROR: <core> [db.c:401]: db_table_version(): error in db_query >> ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table location >> ERROR: usrloc [dlist.c:640]: register_udomain(): error during table version check. >> ERROR: registrar [reg_mod.c:554]: domain_fixup(): failed to register domain >> ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:982 >> ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:987 >> ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:992 >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > ------------------------------

1 0

Change Priority of Dispatcher
by Rodrigo Moreira 08 Feb '17

08 Feb '17

Hello, Help me with a question. Is it possible to change the prohirity of a Kamailio Dispatcher target at run time, without changing the destination entry in the database? If so, which module should I use? Have you some link for support? Thank you very much. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

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

08 Feb '17

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

8 18

Diversion Header - relay messed up
by Roman Dissauer 08 Feb '17

08 Feb '17

When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582.4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ...

3 3

Second coredump in a month
by Jon Bonilla (Manwe) 08 Feb '17

08 Feb '17

Hi all A server running version 4.4.4+0~20161223011227 has segfaulted for the second time this month. I'm attaching a bt in case someone can explain it to me. cheers, Jon

2 12

kamailio with cassandra
by Никитенко Виталий 08 Feb '17

08 Feb '17

Hi! I want start kamailio with db_cassandra. do as it written in https://www.kamailio.org/docs/modules/4.2.x/modules/db_cassandra.html. Then trying kamctl start, failed with error "table version does not exist!". >From CQLSH table is view cqlsh:kamailio> Describe tables; version location cqlsh:kamailio> select * from version; table_name | table_version ------------+--------------- version | 1 Tell me please wherein may be trouble? Full log below ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/location ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/version INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.allow INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio//permissions.allow) not found => empty rule set INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.deny INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio//permissions.deny) not found => empty rule set INFO: misc_radius [misc_radius.c:164]: mod_init(): initializing... ERROR: db_cassandra [dbcassa_base.cpp:449]: cassa_translate_query(): table version does not exist! ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster ERROR: <core> [db.c:401]: db_table_version(): error in db_query ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table location ERROR: usrloc [dlist.c:640]: register_udomain(): error during table version check. ERROR: registrar [reg_mod.c:554]: domain_fixup(): failed to register domain ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:982 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:987 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:992

2 1

Kamailio + Asterisk with multidomain
by Tomas Zanet 07 Feb '17

07 Feb '17

Hello, thanks to this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I successfully installed Asterisk 11.6 and Kamailio 4.4.4 on the same machine. Everything works fine. Right now I would like to extend this scenario adding MULTIDOMAIN support, which is not enabled by default Do you see any problems / technical limitations to add multidomain support in this scenario? Before enhancing Kamailio cfg file and Asterisk configuration I would like to know: is this possible? As far as I know Kamailio works fine with multidomain support but I don’t know if Asterisk or both processes can support it. Thanks in advance Regards, T. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 1

DMQ Cluster - hosts loose network connectivity
by José Seabra 07 Feb '17

07 Feb '17

Hello there, My DMQ cluster has 4 nodes and by some reason 2 of them lost the network connectivity for long time(~ 4 hours), after the network of these 2 nodes come back they didn't get connected on DMQ cluster automatically, i had to restart kamailio to get them again on DMQ list. My doubts here are: - Is this an expected behavior of DMQ module? - Is there any way of put them again on DMQ bus without need restart kamailio? Thank you Best Regards José Seabra

3 5

Kamailio parse failed message
by Hai Bui Duc Ha 07 Feb '17

07 Feb '17

Hi all, I have problem when make call with my Android mobile use PJSIP library. Scenario: my client -> Kamailio -> Freeswitch (media server) -> another client (soft phone on Windows) my client: + use Bluestack + Capture via Wireshark + use Wifi Issue: The call will be drop after ~ 30 second. I see the error on Kamailio: *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<p:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0#015#012Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias#015#012Max-Forwards: 70#015#012From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj#015#012Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB#015#012CSeq: 29055 ACK#015#012Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>#015#012Content-Length: 0#015#012#015#012>* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [receive.c:129]: receive_msg(): core parsing of SIP message failed (49.156.54.54:50785/2 <http://49.156.54.54:50785/2>)* Seem to the server error when parse (on INVITE SDP) *a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16* (on new ACK message) *ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0* *Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias* *Max-Forwards: 70* *From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* *To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj* *Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB* *CSeq: 29055 ACK* *Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>* *Content-Length: 0* I think the SIP message is fragmented but when resume package is not correct. Do you have any advice ? Thank you for watching ! Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 11

FOSDEM 2017
by Daniel-Constantin Mierla 07 Feb '17

07 Feb '17

Hello, another edition of FOSDEM is approaching, about 2 weeks left: - https://fosdem.org/2017/ Couple of devs and many Kamailio friends will be around. Like past year, there is a Realtime Communications devroom, this edition on Sunday. Very interesting presentation for our ecosystem: Olle giving a presentation about IPv4/IPv6, Inaki introducing its SFU media server project, Jose talking about JsSIP, Daniel debating about fundraising FreeRTC, Lorenzo with Janus SIP-WebRTC gateway, the other Lorenzo with Homer Sipcature, Saul with Jitsi, Dan with cgrates, Matt with Asterisk, Giovanni with FreeSwitch... Schedule at: - https://fosdem.org/2017/schedule/track/real_time_communications/ On Sunday morning, part of Lua devroom, I will present about using Lua for building RTC services with Kamailio: - https://fosdem.org/2017/schedule/track/lua/ - https://fosdem.org/2017/schedule/event/luartcserviceskamailio/ During the past editions we organized a dinner on Saturday evening. Shall we attempt to do it again in advance this year? Or should we do it on spot based on the weather and mood at the end of Saturday? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

7 13

kamailio proxy behind firewall
by JBF 07 Feb '17

07 Feb '17

Hello, we have the following Configuration for our kamailio installation (we are using TLS and not udp) (1) F5 Firewall (configured as message fowarding), opening a TLS server on the outside (2) SIP proxy, with a TLS server accessed by the F5 . The SIP proxy doesnt see the F5 TLS server (3) SIP registrar REGISTER works find We have the following issue on INVITE: A sends an INVITE to B. The Registrar patches the R-URI with the content of location, which contains the publi ip of the Device (because the device used stun) we force the routing from registrar to proxy by using t_relay (SIP_PROXY_IP) /The proxy tries to route to this R-URI, which is not visible/ I am not sure how to fix that: Record Route is for a true sip proxy, but the Firewall does not have an server facing the SIP proxy: the sip proxy needs to find the proper client socket opened at register to route the INVITE We have arranged for the Firewall to add its own Via, but if i understand correctly, this is used for replies, and here we are dealing with a request forwarding, and t_relay uses the r-ruri to route requests. IT might be why REGISTER works correctly (ie the 200 OK is routed correctly from proxy to firewall) I could arrange for the location table to contain the private ip and port of the firewall connection (through the use of the received/rport info inserted in the Via by the proxy ) That would mean, however that the contact of the user will contain the private interface of the F5 which i found weird. How do you think i should proceed ? any advices are welcome Thank you -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-proxy-behind-firewall-tp15… Sent from the Users mailing list archive at Nabble.com.

2 3

RTPEngine disable/enable crash kamailio
by Sergey Basov 07 Feb '17

07 Feb '17

Hi All. I ave just try to temporary disable 1 of 2 RTPEngines. It was disabled successfully all media dialogs was finished. But when I have try to enable it I have kamailio crashed with error: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 Does anyone have such issue? kamailio 4.4.5 on RHEL 7.2 x86_64 rtpengine 4.4.0 on RHEL 7.2 x86_64 # kamctl fifo nh_show_rtpp all which: no gdb in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/) udp:10.1.23.24:2223:: set=0 index:: 0 disabled:: 0 weight:: 1 recheck_ticks:: 0 udp:10.1.23.25:2223:: set=0 index:: 1 disabled:: 0 weight:: 1 recheck_ticks:: 0 after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 0 it was successfully disabled with media going on after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 1 I got kamailio crash with error CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 sorry I does not have coredump, it is production system. my config regarding rtpengine: modparam("rtpengine", "rtpengine_allow_op", 1) modparam("rtpengine", "rtpengine_sock", "udp:10.1.23.24:2223 udp:10.1.23.25:2223") Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

5 12

execute config route by ctp command?
by Juha Heinanen 07 Feb '17

07 Feb '17

Is there a way to execute a named config route, such as an event_route, by an rpc command? -- Juha

3 4

Re: [SR-Users] Spurious dialog timeouts
by Pranathi Venkatayogi 07 Feb '17

07 Feb '17

Any clue on this? Or to work around this, can someone shed light on how to differentiate when “Dialog:end” is called due to actual call drop - Bye vs some timeout. I need to do some special logic only on call drop. Relevant snippet from the logs are: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ -----Original Message----- From: Pranathi Venkatayogi Sent: Friday, February 03, 2017 2:29 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: Spurious dialog timeouts Hi, I have two kamailio proxies between the customer and agent. Cust -> proxy1 -> proxy2 -> agent. The dialogs are successfully established and remain undeleted within the customer / agent client. However in Proxy2 I am seeing spurious dialog timeout. I am attaching kamailio log trace below. Also attached is the siptrace on the agent/customer side. Question: How is this dlg_ontimeout() generated despite dialog being successfully established and undelete? How to avoid it. Kamailio (proxy2) log: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' 7728 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7729 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7730 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7731 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7732 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7733 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent5@translation.sms-test.cyracom.com. Marking free 7734 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7735 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7736 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '04d924dc160d4e0c9366d4699f579793' and tags '9b107772c82148c0a09182dc86804c51' '44b0a200438a42f8a6ac931ed72f795a' Thanks for your input, Pranathi

1 0

SIP message order
by Andy Chen 06 Feb '17

06 Feb '17

Hello sr-users, First of all, I'm new to this forum so any help would be greatly appreciated as my Kamailio knowledge is somewhat limited. Thanks for your patience. So to the question at hand. If the originator of the SIP method sends SIP messages out of order, does the Kamailio put them back in order before relaying it to the destination device? I'm seeing where the 200 OK was sent first before the SIP UPDATE within the same CSeq number. The reason I'm asking is because I'm seeing this behavior now with UDP transport protocol and I'm trying to justify why we don't need to switch to TCP to fix this issue. Thanks. --Andy

2 1

Kamailio 4.2.3-3.1 Segfaulting - Need debuginfo RPM for kamailio-4.2.3-3.1.x86_64
by Peter DiSanto 06 Feb '17

06 Feb '17

Our Kamailio 4.2.3 recently began seg-faulting in production. I ran a bt full, but without the debuginfo I presume it isn't much use. We cannot locate the original RPMs for this version to be able to obtain the debuginfo RPM. We are running the kamailio-4.2.3-3.1.x86_64 RPM for CentOS6. Anybody have it or can help us find the debug, or offer other advice?

2 1

loose_route ACK and BYE
by Ben Hood 06 Feb '17

06 Feb '17

Hi, I've fixed a loose_route problem with a broken gateway. Essentially gateway sends ACKs and BYEs with an unroutable contact header (i.e. it sticks the address of the proxy into the header, rather than the destination UA address). This is all on the same network, so there is not NAT. So by caching the contact URI to for replies back to the gateway, I can populate $ru to forward the ACK or BYE back to the correct UA. However, what I've noticed is that for BYEs, I also need to set the $du variable in addition to $ru. But this is required for ACKs. This is not a problem per se, but I'm interesting to know why - could anybody explain? Cheers, Ben

2 1

Os on virtual machine
by Diego Nadares 04 Feb '17

04 Feb '17

Hi guys, We are using vmware to run kamailio. The thing is that they give us support only if we install Red Hat. Did you test kamailio on RH? Any issues or things to have in mind? What linux dist do you recommend? Thanks! Diego.

4 4

Fosdem 2017 dinner
by Daniel-Constantin Mierla 04 Feb '17

04 Feb '17

Hello, the dinner at Fosdem 2017 is planned to start at 20:00, on Saturday, Feb 4, 2017. The address is: Le Pickwick Avenue Adolphe Buyl 79 1050 Ixelles, Belgium * https://en.yelp.be/biz/le-pickwick-ixelles The place is nearby Fosdem location. If you announced the intention to participate at the dinner before Feb 1, 2017, your seat is reserved. If you did it afterwards, please contact me during RTC Devroom to confirm the availability of the seats. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Spurious dialog timeouts
by Pranathi Venkatayogi 03 Feb '17

03 Feb '17

Hi, I have two kamailio proxies between the customer and agent. Cust -> proxy1 -> proxy2 -> agent. The dialogs are successfully established and remain undeleted within the customer / agent client. However in Proxy2 I am seeing spurious dialog timeout. I am attaching kamailio log trace below. Also attached is the siptrace on the agent/customer side. Question: How is this dlg_ontimeout() generated despite dialog being successfully established and undelete? How to avoid it. Kamailio (proxy2) log: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' 7728 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7729 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7730 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7731 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7732 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7733 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent5@translation.sms-test.cyracom.com. Marking free 7734 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7735 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7736 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '04d924dc160d4e0c9366d4699f579793' and tags '9b107772c82148c0a09182dc86804c51' '44b0a200438a42f8a6ac931ed72f795a' Thanks for your input, Pranathi

1 0

FreeSWITCH at FOSDEM, SIP and WebRTC Load Balancing and High Availability
by Giovanni Maruzzelli 03 Feb '17

03 Feb '17

See you all, in person or in the intertube, at: https://fosdem.org/2017/schedule/event/freeswitch/ sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT

1 0

Websocket TLS Issue
by Jade SZ 03 Feb '17

03 Feb '17

Hi Guys, I am trying to setup the following flow: Browser >> WSS >> HA Proxy >>> WSS >> Kamailio But getting TLS errors in Kamailio logs: *[29634]: ERROR: <core> [tcp_read.c:1321]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f68ebe872b0 r: 0x7f68ebe87330* *[29631]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number* Browser <-----wss---->Kamailio works fine with same certs. Both HA Proxy and Kamilio are installed on separate servers, hosting on same port with different domain. Kamailio tls.conf has method = TLSv1 *@HA Proxy:* openssl s_client -connect HA-PROXY-DOMAIN:*10443* SSL-Session: Protocol : TLSv1.2 *@Kamailio :* openssl s_client -connect KAMAILIO-DOMAIN:*10443* SSL-Session: Protocol : TLSv1 So I made HA Proxy to be on TLSv1 "ssl-default-bind-options force-tlsv10" But still I get the same TLS error in Kamailio. *HA Proxy config looks like:* *frontend public* * bind *:10443 ssl crt /etc/haproxy/certs/cert.pem* * acl is_websocket hdr_end(host) -i m1.some-domain.com <http://m1.some-domain.com>* * use_backend wss if is_websocket* * default_backend wss* *backend wss* * timeout server 600s* * server ws1 k1.some-domain.com:10443 <http://k1.some-domain.com:10443>* * server ws1 k2.some-domain.com:10443 <http://k2.some-domain.com:10443>* Need some direction, thanks in advance. Regards, Jade

4 4

id column i tables
by Kristian F. Høgh 02 Feb '17

02 Feb '17

Hi list, We provision our kamailio database from an external system. In the future we would like to provision from two different systems. To prevent the two systems from touching each others data, we would like to create views from the two tables. provision system 1 would write to table1 provision system 2 would write to table2 kamailio will read a view over the two tables. The tables we will create views over are: - address - aliases - dispatcher - uri - usr_preferences Now to my question :-) Does kamailio use the id field in any of the tables above? The id's are auto-incremented, and we might get conflicts between table1 and table2 *Kristian Høgh* Telefon: 4422 8822 support(a)uni-tel.dk[1] Gydevang 19 | 3450 Allerød www.uni-tel.dk [2] -------- [1] mailto:support@uni-tel.dk [2] http://www.uni-tel.dk

2 2

Disabling DNS SRV records checks
by Ahmed Fouad 01 Feb '17

01 Feb '17

Hi, I'm running kamailio version 4.1 and having the following DNS related configuration use_dns_failover = off dns_try_naptr = no dns_udp_pref = -1 dns_try_ipv6 = off dns_retr_time=1 dns_retr_no=1 dns_servers_no=1 dns_use_search_list=no But still am seeing SRV record checks for my domains which I like to completely disable these checks and only use A records only. x.x.x.x.57828 > 8.8.8.8.53: 7417+ SRV? _sip._udp.subdomain.com 8.8.8.8.53 > x.x.x.x.57828: 7417 NXDomain 0/1/0 (129) I can't find a DNS option to disable SRV checks. Can you please advise ?

2 3

could not make ul_rm_contact work
by Aqs Younas 01 Feb '17

01 Feb '17

Greetings list, I am trying to remove a specific contact from memory but always resulted in vain. root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: 2416 Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 . /kamctl fifo ul_rm_contact location 1040(a)191.168.0.101 sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Tried another command root@debian:/usr/local/kamailio/sbin# ./kamctl fifo ul_rm location 1040(a)192.168.0.101 root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: deleted Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 And this. . /kamctl ul rm 1040(a)192.168.0.1 sip:1040@192.168.0.102:36619 ;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Can someone help me know to remove a contact from memory? Any pointer is much appreciated. Best Regards.

2 3

sdp_with_transport
by M. Salman 01 Feb '17

01 Feb '17

Hi Guys, Just wanted to clarify the following case: what should be result of sdp_with_transport("RTP/SAVPF") on line: m=audio 10231 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 I am having a weird behavior for two different versions of Kamailio. I hope I am doing something wrong. kamailio 4.3.2 => sdp_with_transport("RTP/SAVPF") = true kamailio 4.4.5 => sdp_with_transport("RTP/SAVPF") = false -- Regards M. Salman VoIP Professional

3 2

Re: [SR-Users] sr-users Digest, Vol 140, Issue 30
by Andrew Miller 01 Feb '17

01 Feb '17

Is it too late to request a place for the meal at FOSDEM? I would love to come. Andy Miller On 30 Jan 2017 11:00, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Diversion Header - relay messed up (Roman Dissauer) > 2. Re: dlg_set_timeout bye to websockets client (Switch168) > 3. Presence Module do not load db_url (Vu. Minh Cao) > 4. Re: FOSDEM 2017 (Giacomo Vacca) > 5. Re: Replace old registration record with new one while > keeping single AOR per contact (Federico Cabiddu) > 6. Re: Replace old registration record with new one while > keeping single AOR per contact (Aqs Younas) > 7. Re: mhomed opened sockets (Diego Nadares) > 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) > 9. Re: FOSDEM 2017 (Ivan Todorov) > 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 29 Jan 2017 18:42:23 +0100 > From: Roman Dissauer <roman(a)dissauer.net> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Diversion Header - relay messed up > Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> > Content-Type: text/plain; charset=utf-8 > > When I get an INVITE with Diversion Header the Request is forwarded > without Diversion Header and the Request User is taken from Diversion User. > Problem is that on the Destination Host I cannot get original Request User > what is the intended destination! > Is this intended behaviour? How can I change this behaviour? > > I already tried to delete the Diversion Header on request_route but this > didn’t change the behaviour. > > Thanks in andvance for your help! > Roman > > Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as > Carrier IP > > Incoming INVITE: > > INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. > Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 19. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; > reason=unconditional; counter=1. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 260. > Content-Disposition: session; handling=required. > Content-Type: application/sdp. > ... > > > Relayed INVITE: > > INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. > Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. > Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. > Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. > 4fc0216dbecafde29127db502993222c.0. > Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch= > z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 18. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 274. > Content-Disposition: session; handling=required. > ... > > > > > > > > ------------------------------ > > Message: 2 > Date: Sun, 29 Jan 2017 18:21:37 +0000 > From: Switch168 <team(a)switch168.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com > Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client > Message-ID: > <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh > 7z+jRAEg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Daniel, > > > For anyone else who might run into this issue I found > https://github.com/kamailio/kamailio/issues/85 to be related to mine. And > its a good a starting point. > > Cheers > Andrew > > > > > On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > > > HI Daniel, > > > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > > modparam("dialog", "timeout_noreset", 1) > > > > So regular bye's by useragent i can succesfully relay to this > > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > > style of uri by using this snippet below > > > > # Handle requests within SIP dialogs > > route[WITHINDLG] { > > if(is_method("BYE")) { > > xlog("DEBUG: Received BYE"); > > route(NATDETECT); > > loose_route(); > > dlg_manage(); > > t_check_trans(); > > handle_ruri_alias(); > > rtpengine_delete(); > > route(RELAY); > > exit; > > } > > ... > > > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives > out > > error cannot resolve > > the random.uri and I don't know how to fix the bye before it get sent > out. > > > > Thanks > > Andrew > > > > > > > > > > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > > miconda(a)gmail.com> wrote: > > > > Hello, > > > > isn't the dialog module setting the right value there? What function are > > you using to update the contact? > > > > Cheers, > > Daniel > > > > On 27/01/2017 06:01, Andrew Tan wrote: > > > > Hello, > > > > Just wondering if it's possible to intercept the bye message that > > dlg_set_timeout sends out to do some nat_helper function to fix it so the > > bye can be sent to random.invalid uris. > > > > I know there is the edge proxy outbound module but wondering if I can do > > it without. > > > > Regular bye between 2 clients i was able intercept and fix with nat > helper > > but I dont know how to fix the byes that is sent out from dlg_set_timeout > > function. > > > > Thanks in advance! > > Andrew > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > listsr-users@lists.sip-router.orghttp://lists.sip-router. > org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > > Daniel-Constantin Mierlawww.twitter.com/miconda -- > www.linkedin.com/in/miconda > > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - > www.asipto.com > > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > >

1 0

Re: [SR-Users] FOSDEM 2017
by Hugh Waite 01 Feb '17

01 Feb '17

Hi all, Please count me and one other in for the meal on Saturday. See you in Brussels! Hugh Waite On 30 Jan 2017 11:00 a.m., <sr-users-request(a)lists.sip-router.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Diversion Header - relay messed up (Roman Dissauer) 2. Re: dlg_set_timeout bye to websockets client (Switch168) 3. Presence Module do not load db_url (Vu. Minh Cao) 4. Re: FOSDEM 2017 (Giacomo Vacca) 5. Re: Replace old registration record with new one while keeping single AOR per contact (Federico Cabiddu) 6. Re: Replace old registration record with new one while keeping single AOR per contact (Aqs Younas) 7. Re: mhomed opened sockets (Diego Nadares) 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) 9. Re: FOSDEM 2017 (Ivan Todorov) 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Sun, 29 Jan 2017 18:42:23 +0100 From: Roman Dissauer <roman(a)dissauer.net> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Diversion Header - relay messed up Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> Content-Type: text/plain; charset=utf-8 When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. 4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ... ------------------------------ Message: 2 Date: Sun, 29 Jan 2017 18:21:37 +0000 From: Switch168 <team(a)switch168.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client Message-ID: <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh7z+jRAEg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Daniel, For anyone else who might run into this issue I found https://github.com/kamailio/kamailio/issues/85 to be related to mine. And its a good a starting point. Cheers Andrew On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > HI Daniel, > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > modparam("dialog", "timeout_noreset", 1) > > So regular bye's by useragent i can succesfully relay to this > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > style of uri by using this snippet below > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if(is_method("BYE")) { > xlog("DEBUG: Received BYE"); > route(NATDETECT); > loose_route(); > dlg_manage(); > t_check_trans(); > handle_ruri_alias(); > rtpengine_delete(); > route(RELAY); > exit; > } > ... > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives out > error cannot resolve > the random.uri and I don't know how to fix the bye before it get sent out. > > Thanks > Andrew > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > > Hello, > > isn't the dialog module setting the right value there? What function are > you using to update the contact? > > Cheers, > Daniel > > On 27/01/2017 06:01, Andrew Tan wrote: > > Hello, > > Just wondering if it's possible to intercept the bye message that > dlg_set_timeout sends out to do some nat_helper function to fix it so the > bye can be sent to random.invalid uris. > > I know there is the edge proxy outbound module but wondering if I can do > it without. > > Regular bye between 2 clients i was able intercept and fix with nat helper > but I dont know how to fix the byes that is sent out from dlg_set_timeout > function. > > Thanks in advance! > Andrew > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router. org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >

1 0

any public databases or schemas for dial plans?
by Daniel Pocock 31 Jan '17

31 Jan '17

Hi, Are there any public databases of dial plan information? Are there any schemas (e.g. XML / XSD) useful for describing arbitrary dial plans from different carriers around the world? The type of things that I'm interested in: - being able to translate any local number to E.164 (if possible) - knowing the expected length (or range of lengths) for different numbers - being able to recognize numbers that can't be translated to E.164 (e.g. the emergency numbers) - being able to classify specific types of number that usually have a carrier-specific or country-specific meaning (e.g. classifying numbers as directory assistance, customer service, emergency) I'm familiar with Google's libphonenumber, although it is not a database, rather, it is a library that has dial plan logic in Java. It supports many of the things I want for numbers that can be translated to E.164 and country-level dial plans, but not local dial plans (e.g. dialing a London number without the 020 area code). Regards, Daniel

2 2

Using DNS for Service Discovery with Kamailio
by Casco, Alessio 31 Jan '17

31 Jan '17

Hello Guys! I'd like to be able to use SRV records inside dispatcher groups, so I can automatise the discovery of new backends on my network using DNS I've tried with the following formats into the dispatcher.list file: $group _sip._udp.sip-voice_backends $group sip-voice_backends and none looks to be working, running the kamailio_ctl dispatcher.list won't show me the group set with SRV records. ( I also try NAPTR records, with no luck as well.) * Does the dispatcher module support SRV or NAPTR records? (maybe my format is not correct) * If not, do you have any other idea on what I could do to achieve what I'd like to do? Thanks Alessio

3 7

RFC: Kamailio tutorials/cookbooks/docs - wiki or markdown
by Daniel-Constantin Mierla 31 Jan '17

31 Jan '17

Hello, I want to start a discussion about some of the kamailio tutorials from the wiki, mainly those related to core cookbook, variables and transformations. They are now in dokuwiki at: - https://www.kamailio.org/wiki/#cookbooks The wiki served us pretty well so far, however the contributions there are not that active. From time to time we still get spammers, even we added restriction that only registered and authenticated users can write. I am thinking that maybe people don't like creating yet another account just add some example or rephrase for clarity. On the other hand, there are over 400 registered users, many of them I expect to be either made because if misunderstanding the purpose or by spammers that guessed the (sip) captcha (plenty of random usernames). Besides that I think the wiki adds overhead when releasing a new version as each time we have to copy and paste old version content to a new set of pages and update the version number. With the above in mind, I wonder if won't be better to store those tutorials in markdown format and use mkdocs.org or gitbook (https://github.com/GitbookIO/gitbook) tools to generate the html and host it in kamailio.org. The wiki will still be used to index them. Among the benefits I see: * store in github.com (a new repo, like: kamailio-docs) and all devs can contribute directly, other users can make pull requests * create branches for each major version of kamailio and backport across them with git whenever is applicable * the edit can be done directly via github.com website (similar to the wiki right now) and even read it directly from github in html transformed from markdown by github * one can get them offline by cloning the git repo, use them or enhance when offline Many if the tutorials will stay in the wiki, but I think those that are related to documenting per kamailio version should be migrated to gihub storage + markdown. If there is a strong pro opinion for this change, I can try to convert one of the cookbooks very soon so people can feel better the difference. If we get to a consensus, then I think we can have those tutorials in the new format for v5.0. Should anyone have comments, other suggestions or improvements, let's put them on debate. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 7

serial forking calls using TSILO
by Tomas Zanet 30 Jan '17

30 Jan '17

Hello, I'm using Kamailio 4.4.4 with TSILO module in order to support Push Notifications used by our voip app on Apple ios10. So far, everything works fine: Kamailio can get an incoming call, suspend it, then send push notification with an external script, receive a new app registration and then call it after resuming the invite. To do so, thanks to Mr, Cabiddu, I used all the functions described here http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… Right now, I would like to support serial forking calls using push notification because it was already supported with legacy voip. To do so, I used failure_route function where, after getting the call destinations and setting them into an avp, I set the new SIP request uri $ru. Here is my failure_route: failure_route[MANAGE_FAILURE] route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) t_reply("404","Not found"); exit; #!endif if (is_avp_set("$avp(group_members_db)")) $ru ="sip:"+$avp(group_members_db)+"@"+$fd; $avp(group_members_db) = $null; route(LOCATION); exit; I am troubleshooting this scenario and this is what I see: 1) Kamailio receives incoming call 2) Suspend it 3) Send Push notification to account1 4) Kamailio receives account1 registration -> INVRESUME route is now called 5) Then call account1 6) Nobody answers the call 7) failure_route[MANAGE_FAILURE] is now called -> set $ru with next sipaccount: account2 8) suspend invite 9) send push notification to account2 10) account2 sends its sip registration -> INVRESUME route is now called 11) Kamailio calls account1 instead of account2 -> this is my issue On step 10: despite account2 is registered, I checked it using "kamctl ul show", the second call is forwarded all'account1 and not to account2. I also checked the TSILO logs and everything seems ok. Here is the logs: First call: suspended transaction [53945:1648394094] asterisk => account1 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account1 53945:1648394094) second call: suspended transaction [53945:1648394094] asterisk => account2 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account2 53945:1648394094) In order to find out the issue I put some xlogs, printing $ru value: what I see is the $ru value is set correctly on failure_route but as soon as the t_continue is called, the ru overwritten back to account1 Can anyone address me to find out the solution? Thanks in advance T. Tomas Zanet Software Design Department tzanet(a)came.com

3 7

Re: [SR-Users] FOSDEM 2017
by DanB 30 Jan '17

30 Jan '17

Hey Daniel, if anything to be reserved, please count us also in. Will come back regarding number of seats but just to be aware that we are interested to join the dinner. Thanks! DanB

8 7

mhomed opened sockets
by Diego Nadares 30 Jan '17

30 Jan '17

Hi Guys, We have two network interfaces. One public and one private. We had enabled mhomed and works great BUT if we do a netstat -naup | grep 5060 we see that some sockets keeps opened also when the call is finished. The source port in netstat it's not 5060 so I suppose that kamailio use another one for something else. Is this the right behavior? How long does this sockets remain opened? Are we missing something in our cfg? 6 2017-01-25 15:52:04.986651 172.16.213.38 172.16.200.159 SIP/SDP 146 Request: INVITE sip:xxx@172.16.200.159:5060 34 2017-01-25 15:52:19.636895 172.16.213.38 172.16.200.159 SIP 716 Request: ACK sip:xxx@172.16.200.159:5060 | 37 2017-01-25 15:54:38.947831 172.16.208.111 172.16.213.38 SIP 736 Request: BYE sip:127.0.0.8:5060;line=sr-987jhlk* | 40 2017-01-25 15:54:38.950899 172.16.213.38 172.16.200.159 SIP 660 Request: BYE sip:172.16.208.111:5060 | [root@dwrfsd01 kamailio]# netstat -naup | grep 5060 udp 0 0 172.16.213.38:*56086* <http://172.16.213.38:56086> 172.16.200.159:5060 ESTABLISHED 17015/kamailio udp 0 0 172.16.213.38:5060 0.0.0.0:* 17006/kamailio udp 0 0 172.16.213.38:*41971* <http://172.16.213.38:41971> 172.16.200.159:5060 ESTABLISHED 17051/kamailio udp 0 0 172.16.213.38:*54651* <http://172.16.213.38:54651> 172.16.200.159:5060 ESTABLISHED 17008/kamailio Thanks in advance! Diego.

2 3

Presence Module do not load db_url
by Vu. Minh Cao 29 Jan '17

29 Jan '17

Hi, I got error when loading presense module in Kamailio 4.4: [Configure] loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:kamailiorw@127.0.0.1/kamailio") [Error] ERROR: presence [presence.c:608]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) Which is really strange since according to the source code this error just occurs if “db_url” is not set. I already checked my database also, I can access mysql with about username/password. Yours sincerely, — Vu

1 0

rtpengine - Key file error
by Serhat Guler 29 Jan '17

29 Jan '17

Hello, I am trying to setup my Kamailio environment on a new Debian system. Everything went well, except I started facing a problem with the rtpengine. It did install it fine and I can view the program parameters via the help menu; however, when I run it with this command "rtpengine --interface 192.168.0.66 --listen-ng=127.0.0.1:22222 -m 30000 -M 35000 -L 2" or any other commands, it gives an error as "CRIT: Fatal error: Bad command line: Key file does not have group 'rtpengine'". Where is this key file it is complaining about ? I couldn't find anything neither in the doc nor anywhere else. Cheers, Serhat

2 2

missing memory after tls cleanup
by Andy 28 Jan '17

28 Jan '17

Hi, the tls total memory before and after a tls connection is cleaned up does not match, during test there were 10 tls connection created for which I see tls_h_close and tls_h_tcpconn_clean were called exactly 10 times for the same connection ids, bit before and after the tls shm memory does not match, using command: kamcmd mod.stats tls shm could there be a leak? please suggest thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/missing-memory-after-tls-cleanup-tp… Sent from the Users mailing list archive at Nabble.com.

2 4

Ansible Kamailio role
by Alberto Llamas 27 Jan '17

27 Jan '17

Hello Kamailians, I just wanted to share an Ansible role for the installation and compilation of a Kamailio server. It may required improvements and I will be more than happy to heard them. *https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/ <https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/>* Cheers, -- Alberto Llamas Telecommunications Engineer dCAA|dCAP|KPAC|SSCA

4 4

Re: [SR-Users] FOSDEM 2017
by DanB 26 Jan '17

26 Jan '17

Hey Daniel, Would be there 2 persons for sure, so please count us also in. Thanks, DanB

2 1

parallel media forking during ringing phase
by Tomas Zanet 26 Jan '17

26 Jan '17

Hi all, I'm writing to the community because I need a hint from Kamailio and voip gurus: the goal is adding parallel forking call with early-media to our PABX where there's a Kamailio instance running. I mean media streams should be sent to all call destinations before answering the call order, for instance, to see the incoming video. As far as I know, from signaling side should not be a problem because, using Kamailio, there’s append_branch function which forks many branches to all users. The problem is related to media streams because there’s no way to fork media to all users after 180 + SDP or 183, right? Obviously all users must negotiate the same media because media transcoding is basically impossible, especially for video streams. I think there are two ways to do that: 1) using Kamailio and Asterisk. Asterisk has Queue function with ringall strategy, but I’m not sure if Asterisk supports multi domain scenarios. Have ever used Asterisk with Kamailio in multi domain scenarios? 2) enhancing rttpproxy-ng. I don't really know the effort to achieve that. In your opinion, what’s the easiest way to achieve that? Thanks, Best Regards. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 2

How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image001.jpg@01D27631.42773770] From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Ryan Wagoner Sent: Tuesday, January 24, 2017 8:26 AM To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Asterisk Proxy Multiple Devices / BLF Issues I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { 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(); } Thanks, Ryan

2 3

kamctl cr reload got 500 failed to re-built tree
by Igor Potjevlesch 25 Jan '17

25 Jan '17

Hello, With more than 173k entriers into carrierroute, now I'm doing "kamctl cr reload", I got the following reply: 500 failed to re-built tree, see log What could going wrong? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 4

Issue with From address being modified in ACK when UAC module used version 4.1
by Jonathan Hunter 25 Jan '17

25 Jan '17

Hi Guys, running kamailio 4.1.4 and using uac_replace_from, I am seeing a strange issue with the proxying of an ACK message back from a carrier to freeswitch on the ingress path into a network. So its just a normal call inbound, where on outbound leg we modify the From address, on the inbound leg all remains the same. Now after the ingress side receives the 200ok, it sends an ACK as below; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+44792498881474@192.168.24.8:5080;user=phone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Route: <sip:192.168.24.8;lr=on;ftag=carrier.peering.telecom.im+3+863d20a3+1b9801d2;vsf=AAAAAAAAAAYNBgAPAAgLAgZ3UTMACgBXFUsVFwwcDkAKTwMZGh0YAA8KDkEEQ1NYC0NDXgdOFRpSHg1vbmU-> Content-Length: 0 Max-Forwards: 68 However kamailio changes the From address; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 109.73.69.165:5060;branch=z9hG4bKc1ce.47974fc3da2b669a78f2dcc9a057a127.0 Via: SIP/2.0/UDP 4.4.4.4:5060;rport=5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+441444680332@es132y$}-9>.8n?~9,*%(;zyk393;7e&C^NRone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Content-Length: 0 Max-Forwards: 67 Causing FreeSWITCH to not recognise the request, and therefore not send an ACK. There are no rules set against the ACK processing. Has anyone seen this before? We dont know when it started happening which doesnt help, I will look to setup debug on test environment but just wondered if this is an issue thats been seen before? Many thanks in advance. Jon

2 5

Event when acc is written?
by Tobias 25 Jan '17

25 Jan '17

Hi! I'm wondering if there is a way for the acc-module to notify/send an event to Kamailio (so I can pick it up from the config, similar to event_route) once the acc-module has performed a write? As an example I'm using acc and db-mysql to store acc data in MySQL, but instead of having a service polling the MySQL for new acc's written I'd like to make Kamailio send an event once an acc-row has been written. I realize this could most likely be done by using MySQL triggers "ON INSERT", but I'm wondering if there is another (reliable) way to do this directly from within Kamailio? I haven't tested it, but I would guess that if I tried to use for example event_route[dialog:end] there could be times where the event would hit Kamailio before the acc-module have written to missed_calls? Thanks! /Tobias

4 11

Executing perl script with command line parameters.
by Venkatesh Macha 25 Jan '17

25 Jan '17

Hi all, I am trying to execute external Perl script from the Kamailio config file. My script accepts few command line parameters but if I try to pass command line parameters like below, It is not working. exec_msg("/root/test.pl 10000 '$var(env)' '$fU'"); # i tried with exec_avp also How to execute the external Perl scripts with few arguments. Thank you in Advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Executing-perl-script-with-command-… Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

In the case below though TLS session is established between proxy1 and proxy2, invite is fwded on UDP. 1. How to force Kamailio to use existing tls session to communicate? (proxy1) 2. What is the command to look at if existing tls sessions are present from proxy1 a. kamcmd tls.list does it, but I need one which works from within config. From: Pranathi Venkatayogi Sent: Tuesday, January 24, 2017 11:02 AM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: How does Kamailio decide which protocol to use when fwding to another proxy? Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image002.jpg@01D27654.0A627200]

2 1

kamdbctl error.
by Venkatesh Macha 25 Jan '17

25 Jan '17

Hi all, I am using debian jessie, I just installed kamailio 4.4 from source and trying to create kamailio database using kamdbctl but when i try to run the kamdbctl create i am getting following charset error. /usr/local/sbin/kamdbctl create MySQL password for root: INFO: creating database kamailio ... ERROR 1115 (42000) at line 1: Unknown character set: 'ISO' Mysql Version : 5.5.52 Thank you in advance -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamdbctl-error-tp155288.html Sent from the Users mailing list archive at Nabble.com.

2 5

Asterisk Proxy Multiple Devices / BLF Issues
by Ryan Wagoner 24 Jan '17

24 Jan '17

I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { 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(); } Thanks, Ryan

1 0

(no subject)
by jay binks 24 Jan '17

24 Jan '17

Hi Guys, Im using a keepalived solution between 2 kamailio boxes, with net.ipv4.ip_nonlocal_bind =1 . On the "non master" I pretty much constantly get these logs in syslog. What Im looking for is a way to suppress these because they are expected in my configuration. is there any way to tell kamailio to not log this... ( or at least, not at CRIT and ERROR ) *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed* *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7fc0732d5ab0,380,0,xxx.xxx.xxx.xxx:5060,16): Invalid argument(22)* *Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: CRITICAL: <core> [udp_server.c:581]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net* -- Sincerely Jay

4 9

How to determine correct port to set in record-route header
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, I am using Kamailio behind NAT, unable to figure how to make it put “public ip” in Record-route header, I am manually inserting the hard-coded header myself as below. However now I am having trouble choosing the right port number in all scenarios. $var(dstPort) = 5061; if (dst_port==5060) { $var(dstPort) = 5060; } insert_hf("Record-Route: <sip:MY_PUBLICIP_ADDR:$var(dstPort);transport=tls;lr;nat=yes>\r\n", "Record-Route"); In one scenario I see the conflicting port numbers in “via” header vs “Record route”. The ACK is being set to port 5060 based on Record route header and is not being received by the callee. 2017-01-23 14:52:49.970233 [blink.exe 2780]: SENDING: Packet 11, +0:00:36.411867 10.0.27.108:58217 -(SIP over TLS)-> 172.31.211.31:5061 SIP/2.0 180 Ringing Via: SIP/2.0/TLS 63.149.103.72:5061;received=172.31.211.31;branch=z9hG4bK4799.7067bd2a48063748a4c353fa408eaefa.0 Via: SIP/2.0/UDP 10.0.16.52;rport=5060;branch=z9hG4bK4799.935a27c6ef5ad112225964cbe7c1be44.0;i=2 Via: SIP/2.0/TLS 10.11.200.12:51793;rport=51793;received=10.11.200.12;branch=z9hG4bKPj49f6aeca3f9b4155ab4c5304b544aa4d;alias Record-Route: <sip:63.149.103.72:5060;transport=tls;lr;nat=yes> Record-Route: <sip:10.0.16.52:5061;transport=tls;lr;nat=yes> Call-ID: 09ce10efa6a946bf9445ccc21857393e From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=95082caabecc42548c2fec5ccd29e5de To: <sip:spanish@translation.sms-test.cyracom.com>;tag=ec365cc8489b48a7bc5725d21b7d97a1 CSeq: 17422 INVITE Server: Blink 3.0.0 (Windows) Contact: <sip:20745891@10.0.27.108:58216;transport=tls> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 Questions: 1. How can I let in-built record-route function automatically choose “public ip” so I can get rid of my manual insertion altogether. 2. If no option, what is the right way to choose the port on which packet is being received, so it is same as what is on “VIA”. 3. Any other pointers to improve the logic here?

1 0

NAPTR/SRV lookups fail when use_dns_failover is enabled.
by Jeremy Gadd 23 Jan '17

23 Jan '17

Hello, everyone! I'm trying to enable dns_srv_lb but I'm stuck just before I can get there. If I don't enable use_dns_failover, lookups work and calls route but always to the first host. We need to load balance, so that doesn't work very well. If I enable use_dns_failover, the lookups stop working completely and we get a dns lookup failure like so: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve " redacted.hostname.here.com" :unresolvable A or AAAA request (-7) This works: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on # use_dns_failover=on This breaks it: use_dns_cache=on dns_use_search_list=no dns_try_naptr=on use_dns_failover=on I have checked to see that use_dns_failover is compiled in: version: kamailio 4.4.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2

2 4

"Named Telephone Event" (I think) problem
by Pete Kelly 23 Jan '17

23 Jan '17

Hi I am running Kamailio 4.4 and have come across a strange SBC which is sending DTMF within the SDP body of a NOTIFY with a Content-Type audio/telephone-event. Has anybody come across this before? The content itself comes through on a tcpdump/ngrep trace as four dots "...." and Kamailio puts an empty string into the logs if I try to log the contents of $rb, however Wireshark manages to decode it as something readable "\005\000\a\320" and usable. (essentially the digit 5) Is there any way I can get access to this decoded version within Kamailio at all? Even if I have to pass it out to an external program? Many thanks for any help anyone can offer! Pete

2 1

Best way to log incoming/outgoing sip messages?
by Pranathi Venkatayogi 23 Jan '17

23 Jan '17

Hi, What is the best way to get a complete dump of incoming/outgoing SIP messages of Kamailio server? I encountered “SIPTrace” module, but it writes to database. Is there a way we can write to syslog instead? Thanks Pranathi

5 4

Kamailio IMS Secure Websocket closes connection
by Serhat Guler 22 Jan '17

22 Jan '17

Dear all, I am trying to make a call between 2 sipml5 client over WSS. Everything works fine and as expected when using WS. While using WSS, the clients are registered successfully, but when I make a call, the called party is disconnected from the network and the caller gets "User offline" message. There a wireshark capture here: https://drive.google.com/file/d/0B6LMw8kmoAMYMGhrYXNUVXJySEk/view?usp=shari… The call is made from Alice to Bob, the port Nrs: 4060: PCSCF, 5060:ICSCF, and 6060 SCSCF. WWS is running on 4443. The error message that is shown in Firefox console: "The connection to wss://192.168.0.11:4443/ was interrupted while the page was loading." Thanks in advance, Serhat

1 0

Get ip from contact on redirect
by Diego Nadares 20 Jan '17

20 Jan '17

Hi Guys, Anybody knows how to get the ip of the new contact of the redirect? I need it to update one xavp variable that contains the INVITE destination. 6 2017-01-20 12:44:55.138473 172.16.213.38 *172.16.208.121* SIP/SDP 85 Request: INVITE sip:01152780776@172.16.208.121:5060 | 9 2017-01-20 12:44:55.142296 172.16.208.121 172.16.213.38 SIP 661 Status: 302 Moved temporarily | 10 2017-01-20 12:44:55.156268 172.16.213.38 172.16.208.121 SIP 423 Request: ACK sip:01152780776@172.16.208.121:5060 | 16 2017-01-20 12:44:55.293349 172.16.213.38 *172.16.208.111* SIP/SDP 95 Request: INVITE sip:1152780776@172.16.208.111:5060;user=phone | My cfg: # Manage failure routing cases failure_route[MANAGE_FAILURE] { .... if (t_check_status("302")) { xlog("Got a 302 - redirecting"); get_redirects("*:1"); * $xavp(ip_b[$avp(current_route)]=>name) = "172.16.208.111";#Hardcoded for testing* xlog("Got a 302 - redirecting $hdr(Contact)"); route(RELAY); #t_relay(); } Thanks in advance. Diego.

2 3

Siremis Multi domain
by Annus Fictus 20 Jan '17

20 Jan '17

Hi, I think Siremis not work correctly in a multi domain scenario if the extension number are the same for two domains. I explain a little more. I have two extension: 1 200(a)sip1.domain.org 2 200(a)sip2.domain.org On the Siremis I create the user 200 with the SIPuser role. When I login with this user, and go to subscriber data i can see bots extensions. I think is relate to this select on kamailio database: .....SELECT T0.id, T0.username, T0.domain, T0.password, T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 WHERE T0.username='200' LIMIT 10 Any hint? Regards

2 7

UAC authentication
by anfecora 20 Jan '17

20 Jan '17

can someone please guide me where to find the usage of auc module, thanks in advance. I am registering the uac in one end then i want to forward that registration to an external registrar, if i use $du some domain, some of the ipv6 info is going to be forwarded, i need to make it look more like an a b2bua than a outbound proxy therefore i need to use the auc module(also i need to cover all ipv6 info since i am proxying from ipv4 to ipv6). i am able to register the client to kamailio, then kamailio send the registration to the external registrar, but when the registrar sends back the challenge uac does not reply, any ideas how can achieve this. uac(linphone etc) --registration--->kamailio(registrar module)------register(uac)--->SBC sbc----invite---->kamailio(lookup)---------invite---->uac(linphone etc) this is the code i am using. # Handle SIP registrations route[REGISTRAR] { xlog("REGISTERING "); if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if(af==INET6) setbflag(FLB_IPV6); xlog("IT IS IPV6"); if (!save("location")) sl_reply_error(); route(REGSBCFWD); exit; } } route[REGSBCFWD]{ xlog("CREATING OTHER LEG TO THE SBC"); if(!is_method("REGISTER")) { return; } xlog("CREATING PACKAGE $si -- $fU-- $au--$du-- $tU --$ru"); $var(rip) = "sip.somedomain.com"; $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + "5060"; $uac_req(furi)="sip:" + $tU + "@" + $var(rip); $uac_req(turi)="sip:" + $tU + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $tU + "@" + "10.0.0.222" + ":" "5060" + ">\r\n"; if($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(); } REGISTER sip:sip.somedomain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK285f.13a7f0c0000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-ebaa. CSeq: 10 REGISTER. Call-ID: 357e21ae51d8416d-26605(a)10.0.0.222. Max-Forwards: 70. Content-Length: 0. User-Agent: kamailio (4.4.4 (x86_64/linux)). Contact: <sip:140524@10.0.0.222:5060>. Expires: 3600. . SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.0.0.222:5060 ;branch=z9hG4bK4b53.ad842757000000000000000000000000.0. To: <sip:14052423@sip.somedomain.com>. From: <sip:14052423@sip.somedomain.com >;tag=a82eaf71666096790d1397845fbd3254-37da. Call-ID: 357e21ae51d8416d-26603(a)10.0.0.222. CSeq: 10 REGISTER. WWW-Authenticate: Digest realm="sip.somedomain.com", nonce="f5831435-3e69-4b78-ab6d-dcb0897e477b", algorithm=MD5, qop="auth". User-Agent: somedomain.com Host:some proxy. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Content-Length: 0. .

2 3

Return body/string from Perl library
by David Cunningham 20 Jan '17

20 Jan '17

Hello, We are using Kamailio's perl module and want our custom library to set the body on a SIP message. Is it possible to set a SIP message body from within Perl, or can we return a string to the Kamailio configuration to then use with set_body()? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019

2 3

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 19 Jan '17

19 Jan '17

root@homer02:~# netstat -altp Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 0 0 homer02.xxxxx.xxx:sip *:* LISTEN 16018/kamailio tcp 0 0 homer02.xxxxx.xxx:5066 *:* LISTEN 16018/kamailio Thanks. JR > If you use netstat, what is in the recv queue for tcp packets on sip ports? > > netstat -altp > > Cheers, > Daniel > > > On 18/01/2017 16:43, JR Richardson wrote: >> Yes, this is a sipcapture node. I'm listening on a switch port that is >> set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the >> mirror port and the Ethernet port of the host node. Just don't see any >> TCP packets process in kamailio, debug 3. UDP packets are processed as >> expected. >> >> My config is using port mirror for the capture parameters: >> >> modparam("sipcapture", "capture_on", 1) >> modparam("sipcapture", "hep_capture_on", 0) >> modparam("sipcapture", "raw_ipip_capture_on", 0) >> modparam("sipcapture", "raw_moni_capture_on", 1) >> >> modparam("sipcapture", "raw_sock_children", 4) >> modparam("sipcapture", "raw_interface", "eth1") >> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >> modparam("sipcapture", "promiscious_on", 1) >> modparam("sipcapture", "raw_moni_bpf_on", 1) >> >> Is there a method I could diagnose if the SIP TCP Packets are getting >> from the kernel network process and the kamailio process? >> >> # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid >> >> Or pipe to kamailio local unix socket? >> >> I don't know, I'm just guessing. >> >> Thanks. >> >> JR >> >> >>> Somehow is not clear for me how you have the configuration there ... >>> before commenting further, this needs to be clarified. >>> >>> The node you presented the config is a sipcapture instance, right? What >>> is sending traffic to it? Is another kamailio with siptrace module? Or >>> the sipcature agent? Or you have a port mirroring in the router? >>> >>> Cheers, >>> Daniel >>> >>> >>> On 17/01/2017 16:37, JR Richardson wrote: >>>>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>>>> using raw socket capture, I think this may be a sipcapture issue, >>>>>> debuging kamailio shows normal startup and processing of UDP SIP >>>>>> packets, but does not show any activity with TCP packets. >>>>> I never used HOMER sofar but when I saw your first message my thoughts >>>>> was that this can't work in a simple way since for TCP you need to >>>>> complete a 4 way handshake before you can start to send data. >>>>> >>>> Interesting. Are you referring to handshaking on the network stack or >>>> SIP TCP TLS handshaking? I guess I can see it two ways. >>>> >>>> 1) if your talking about TCP/IP handshake, even though the SIP packet >>>> comes into the mirror port on the host node, the kernel processing the >>>> TCP packet is not establishing a valid connection due to no TCP >>>> handshake because its only a monitor port, no transmit back, then the >>>> kernel network stack does not pass the SIP TCP packet to the kamailio >>>> process for capture because it drops the packet due to no valid >>>> handshake? >>>> >>>> 2) the kernel network stack is passing the SIP TCP packet to the >>>> kamailio process, but since kamailio cannot handshake back it drops >>>> the packet and does not process through the sipcapture module. This >>>> kinda breaks the whole capture ability for homer with SIP TCP. Using >>>> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >>>> progress <--, request timeout <--, ack -->, etc... >>>> >>>> So how would I diagnose if the network stack is the culprit? Debugging >>>> kamailio is pretty straight forward, setup and listening for SIP TCP, >>>> but never see any processing of any TCP packets. >>>> >>>> Thanks. >>>> >>>> JR >

2 1

Phone number formats per country
by Daniel-Constantin Mierla 19 Jan '17

19 Jan '17

Hello, slightly off-topic, but related to voip -- does anyone have a link that can quickly share to a site with up to date details about phone numbering plans for as many country as possible? Ideally to include the toll and premium number prefixes, the split between mobile and fixed lines if it is the case. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

4 6

ds_is_from_list and NAPTR / SRV
by Björn Bylander 19 Jan '17

19 Jan '17

Hello, I’m trying to resolve an issue with the ds_is_from_list function from the dispatcher module. I’m calling it as follows: ds_is_from_list(”42”, "3", "sip:sip.foo.bar”) This makes Kamailio log an error: ERROR: dispatcher [dispatch.c:2327]: ds_is_addr_from_list(): could not resolve sip.foo.bar sip.foo.bar doesn’t have any A records but there are NAPTR and SRV records: $ host -t naptr sip.foo.bar sip.foo.bar has NAPTR record 20 0 "s" "SIP+D2T" "" _sip._tcp.sip.foo.bar. sip.foo.bar has NAPTR record 10 0 "s" "SIP+D2U" "" _sip._udp.sip.foo.bar. $ host -t srv _sip._tcp.sip.foo.bar _sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar. _sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar. $ host -t srv _sip._udp.sip.foo.bar _sip._udp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar. _sip._udp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar. $ host sip1.foo.bar. sip1.foo.bar has address 10.47.11.1 $ host sip2.foo.bar. sip2.foo.bar has address 10.47.11.2 The following DNS options are set: dns_srv_lb = yes dns_try_naptr = yes dns_naptr_ignore_rfc = no The Kamailio version is 4.2.8 (installed from the Debian binary package) on Debian Wheezy. Shouldn’t ds_is_from_list be able to resolve the NAPTR and SRV records? Can I change anything to be able to use ds_is_from_list with the specified domain? Thanks, Björn Bylander

2 1

Upgrade from 4.3 to 4.4
by Laura 18 Jan '17

18 Jan '17

Dear list, I'm planning the upgrade fromn 4.3.6 tree to 4.4.x version on Debian distribution. Can i run the apt-get update and upgrade without problem.. or is possible that i will found some error during upgrade ? Is the db schema upgraded or not from 4.3 up to 4.4 ? Regards Laura

2 1

Please explain : t_relay() vs t_relay_to_udp()
by Abdoul Osséni 18 Jan '17

18 Jan '17

hello all, I want to use kamailio like as a proxy SIP (outbound proxy) to transcode tls to udp like us: call flow is: Softphone --> tls --> Kamailio server (kamailio listen on 443 port) --> udp --> Asterisk server (asterisk listen on 5060 port) I know Kamailio can convert tls to udp. According tm documentation http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use t_relay_to_udp(). Do you confirm I must replace t_relay() function by t_relay_to_udp() function? Thank for your help. Regards Abdoul.

2 2

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 18 Jan '17

18 Jan '17

Yes, this is a sipcapture node. I'm listening on a switch port that is set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the mirror port and the Ethernet port of the host node. Just don't see any TCP packets process in kamailio, debug 3. UDP packets are processed as expected. My config is using port mirror for the capture parameters: modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_interface", "eth1") modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) Is there a method I could diagnose if the SIP TCP Packets are getting from the kernel network process and the kamailio process? # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid Or pipe to kamailio local unix socket? I don't know, I'm just guessing. Thanks. JR > Somehow is not clear for me how you have the configuration there ... > before commenting further, this needs to be clarified. > > The node you presented the config is a sipcapture instance, right? What > is sending traffic to it? Is another kamailio with siptrace module? Or > the sipcature agent? Or you have a port mirroring in the router? > > Cheers, > Daniel > > > On 17/01/2017 16:37, JR Richardson wrote: >>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >>>> using raw socket capture, I think this may be a sipcapture issue, >>>> debuging kamailio shows normal startup and processing of UDP SIP >>>> packets, but does not show any activity with TCP packets. >>> I never used HOMER sofar but when I saw your first message my thoughts >>> was that this can't work in a simple way since for TCP you need to >>> complete a 4 way handshake before you can start to send data. >>> >> Interesting. Are you referring to handshaking on the network stack or >> SIP TCP TLS handshaking? I guess I can see it two ways. >> >> 1) if your talking about TCP/IP handshake, even though the SIP packet >> comes into the mirror port on the host node, the kernel processing the >> TCP packet is not establishing a valid connection due to no TCP >> handshake because its only a monitor port, no transmit back, then the >> kernel network stack does not pass the SIP TCP packet to the kamailio >> process for capture because it drops the packet due to no valid >> handshake? >> >> 2) the kernel network stack is passing the SIP TCP packet to the >> kamailio process, but since kamailio cannot handshake back it drops >> the packet and does not process through the sipcapture module. This >> kinda breaks the whole capture ability for homer with SIP TCP. Using >> ngrep, I see all SIP TCP packets, invite -->, trying <--, session >> progress <--, request timeout <--, ack -->, etc... >> >> So how would I diagnose if the network stack is the culprit? Debugging >> kamailio is pretty straight forward, setup and listening for SIP TCP, >> but never see any processing of any TCP packets. >> >> Thanks. >> >> JR > -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Kamailio v4.4.5 Released
by Daniel-Constantin Mierla 18 Jan '17

18 Jan '17

Hello, Kamailio SIP Server v4.4.5 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.5. For more details about version 4.4.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/01/kamailio-v4-4-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning the release of v4.4.5
by Daniel-Constantin Mierla 18 Jan '17

18 Jan '17

Hello, I am considering to release v4.4.5 with latest fixes from branch 4.4 later this week, on Wednesday or Thursday (Jan 19 or 20). As usual, if anyone has fixes that were not backported yet, do it if you are a developer or write to sr-dev to be sure it is not missed. Sumbit also the new issues you are ware of and they are not on the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Record-Route Header Question
by Andres 17 Jan '17

17 Jan '17

Hi, I am troubleshooting some strange call setup problems using an Edgewater Networks SBC. A couple of Polycom phones sit behind the SBC and they connect to a SIP Server in the cloud. The SBC is in passthrough mode. What I see that I am not sure about is: INVITE from SIP Server to SBC IP (no Record-Route header...so far no surprise here) INVITE from SBC to Phone (Record-Route header inserted by SBC but with the IP of the SIP Server) I do not think the SBC is supposed to add a Record-Route header with the IP of the remote SIP Server. I think It should only be allowed to add Record-Route headers with its own IP. Does that sound right? The RFC lists the usage as: "20.30 Record-Route The Record-Route header field is inserted by proxies in a request to force future requests in the dialog to be routed through the proxy." It says nothing about being able to arbitrarily force future requests to go through a remote IP. Please advise. Thanks, -- Technical Support

2 2

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 17 Jan '17

17 Jan '17

> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote: >> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, >> using raw socket capture, I think this may be a sipcapture issue, >> debuging kamailio shows normal startup and processing of UDP SIP >> packets, but does not show any activity with TCP packets. > > I never used HOMER sofar but when I saw your first message my thoughts > was that this can't work in a simple way since for TCP you need to > complete a 4 way handshake before you can start to send data. > Interesting. Are you referring to handshaking on the network stack or SIP TCP TLS handshaking? I guess I can see it two ways. 1) if your talking about TCP/IP handshake, even though the SIP packet comes into the mirror port on the host node, the kernel processing the TCP packet is not establishing a valid connection due to no TCP handshake because its only a monitor port, no transmit back, then the kernel network stack does not pass the SIP TCP packet to the kamailio process for capture because it drops the packet due to no valid handshake? 2) the kernel network stack is passing the SIP TCP packet to the kamailio process, but since kamailio cannot handshake back it drops the packet and does not process through the sipcapture module. This kinda breaks the whole capture ability for homer with SIP TCP. Using ngrep, I see all SIP TCP packets, invite -->, trying <--, session progress <--, request timeout <--, ack -->, etc... So how would I diagnose if the network stack is the culprit? Debugging kamailio is pretty straight forward, setup and listening for SIP TCP, but never see any processing of any TCP packets. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

Using the auth_ephemeral module
by Hemanshu Vadehra 17 Jan '17

17 Jan '17

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

5 8

usrloc module with dbmode=2 does not insert records to database
by Vik Killa 17 Jan '17

17 Jan '17

Hello, we've noticed that the usrloc module does not "sync" all the records from memory into the database. I use a bash script to generate in-memory AoRs ( http://paste.debian.net/plain/908521) then i perform kamcmd ul.flush and no records are inserted. We have tried various usrloc parameters but none seem to work Here is our basic setup # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "timer_interval", 120) modparam("usrloc", "timer_procs", 4) We are using postgresql. are we missing something? Thanks /V

2 11

Info: github template for reporting issues
by Daniel-Constantin Mierla 17 Jan '17

17 Jan '17

Hello, yesterday I added a template for opening new issues on github.com bug tracker. You can see it in action at: - https://github.com/kamailio/kamailio/issues/new Its bare text can be seen at: - https://raw.githubusercontent.com/kamailio/kamailio/master/.github/ISSUE_TE… More about the issue templates for github can be found at: - https://help.github.com/articles/creating-an-issue-template-for-your-reposi… Beside informing about this addition, I would like to get also feedback about it. Do you find it useful or you think it's going to be a hassle? If it should be kept, is anything that you think it should be added or removed? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

LCR module dont_strip_or_tag_flag
by Annus Fictus 17 Jan '17

17 Jan '17

Hello, I'm trying LCR module and dont_strip_or_tag_flag parameter. If I use this line in the module configuration: modparam("lcr", "dont_strip_or_tag_flag", 10) and then restart kamailio, i receive this error: ERROR: <core> [modparam.c:141]: set_mod_param_regex(): parameter <dont_strip_or_tag_flag> of type <2> not found in module <lcr> I'm using 4.4.4 version Any hint? Thank you Regards

3 3

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 16 Jan '17

16 Jan '17

Hey Daniel, Yes, I'm familiar with the methods sipcapture uses, I don't use HEP, using raw socket capture, I think this may be a sipcapture issue, debuging kamailio shows normal startup and processing of UDP SIP packets, but does not show any activity with TCP packets. I have enabled/disabled the different capture methods and no TCP SIP packets are processed. Also, I didn't mention before I'm testing on 2 different instances, one is a Homer 3.5 (Kamailio 4.1.2) and another is Homer 5.0 (Kamailio 4.4.0). I'll post to the Homer list and see what they say. Thanks. JR > is it this about a sipcature node? If yes, then it receives the packets > over HEP, meaning that is not going to the normal processing path. > Kamailio detects it is not a SIP packet, executes some callbacks for > non-sip packets registered by modules (in this case sipcapture). The > module detecting it's a packet for it, can execute the request_route > once it builds a local sip packet. > > To troubleshoot further, run with debug=3 in kamailio.cfg and see what > you get in the logs. > > Cheers, > Daniel > > On 13/01/2017 16:33, JR Richardson wrote: >> Iptables is not blocking, but it was worth a check. >> >> Thanks. >> >> JR >> >> >> I assume you have ruled out firewall? It's something that can nab even >> experienced people: >> >> # iptables -Ln >> >> -- Alex >> >> On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote: >> >>> Hi All, >>> >>> Just enabled SIP TCP on a homer capture server, I can see the SIP TCP >>> Sessions on the server with ngrep, just like all the UDP traffic. I >>> have Kamailio listening on TCP ports but its not capturing any TCP >>> traffic. >>> >>> kamailio.cfg: >>> >>> #disable_tcp=yes >>> listen=tcp:10.99.99.99:5060 #monitor port >>> listen=udp:10.99.99.99:5060 #monitor port >>> >>> loadmodule "pv.so" >>> loadmodule "db_mysql.so" >>> loadmodule "sipcapture.so" >>> loadmodule "textops.so" >>> loadmodule "rtimer.so" >>> loadmodule "xlog.so" >>> loadmodule "sqlops.so" >>> loadmodule "htable.so" >>> loadmodule "sl.so" >>> loadmodule "siputils.so" >>> >>> >>> modparam("sipcapture", "capture_on", 1) >>> modparam("sipcapture", "hep_capture_on", 0) >>> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") >>> modparam("sipcapture", "raw_interface", "eth1") >>> modparam("sipcapture", "raw_ipip_capture_on", 0) >>> modparam("sipcapture", "table_name", "sip_capture") >>> modparam("sipcapture", "raw_sock_children", 4) >>> modparam("sipcapture", "db_insert_mode", 0) >>> modparam("sipcapture", "raw_moni_capture_on", 1) >>> modparam("sipcapture", "promiscious_on", 1) >>> modparam("sipcapture", "raw_moni_bpf_on", 1) >>> modparam("sipcapture", "capture_node", "homer02") >>> modparam("sipcapture", "authorization_column", "authorization") >>> >>> >>> ## logging all INVITES top of the [route] block >>> if (is_method("INVITE|REGISTER")) { >>> xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" >>> from \"$si\"\n"); >>> >>> Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. >>> >>> root@homer02:~# netstat -al >>> Active Internet connections (servers and established) >>> Proto Recv-Q Send-Q Local Address Foreign Address State >>> tcp 0 0 homer02.me.com:sip *:* LISTEN >>> >>> >>> I don't think this is a homer issue because logging invites is prior >>> to any homer processing. I'm thinking this is something simple I'm >>> overlooking, any help is much appreciated. >>>

2 1

Remove/Append headers in 200 canceling and 200 ok to BYE
by Diego Nadares 16 Jan '17

16 Jan '17

Hi List, I'm trying to add and remove some headers in 200 canceling and 200 ok to BYEs with no success. I don't know in which route to add my remove_hf and append_hf. It seems that I can't catch it on reply either on request route. I tried adding it in event_route[tm:local-response] { xlog("L_ERR", "TM LOCAL REPONSE!"); route(MANAGE_HEADERS); } This is my route to manage hdrs: route[MANAGE_HEADERS]{ xlog("L_ERR", "MANAGE HDRS !"); keep_hf("TH"); # for topoh append("AHDR1:asdf\r\n"); return; } I suppose SERVER HDR should be removed but: U 2017/01/12 12:45:41.566733 172.16.213.38:5060 -> 172.16.50.41:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 172.16.50.41:5060;branch=z9hG4bKgg8ekd3080braugkm700.1. CSeq: 102 CANCEL. From: <sip:xx@172.16.50.41>;tag=SD7b10b01-as5ba5e269. To: <sip:xxx@172.16.213.38>;tag=4a86e96addfb27a943bc86e1872032b6-0a8b. Call-ID: SD7b10b01-6a7d1089793bda79481f25cf88c023ed-c540dl1. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. Any help will be very appreciated. Thanks in advance! Diego

3 9

Re: [SR-Users] Kamailio not processing SIP TCP
by JR Richardson 16 Jan '17

16 Jan '17

Iptables is not blocking, but it was worth a check. Thanks. JR I assume you have ruled out firewall? It's something that can nab even experienced people: # iptables -Ln -- Alex On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote: > Hi All, > > Just enabled SIP TCP on a homer capture server, I can see the SIP TCP > Sessions on the server with ngrep, just like all the UDP traffic. I > have Kamailio listening on TCP ports but its not capturing any TCP > traffic. > > kamailio.cfg: > > #disable_tcp=yes > listen=tcp:10.99.99.99:5060 #monitor port > listen=udp:10.99.99.99:5060 #monitor port > > loadmodule "pv.so" > loadmodule "db_mysql.so" > loadmodule "sipcapture.so" > loadmodule "textops.so" > loadmodule "rtimer.so" > loadmodule "xlog.so" > loadmodule "sqlops.so" > loadmodule "htable.so" > loadmodule "sl.so" > loadmodule "siputils.so" > > > modparam("sipcapture", "capture_on", 1) > modparam("sipcapture", "hep_capture_on", 0) > modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") > modparam("sipcapture", "raw_interface", "eth1") > modparam("sipcapture", "raw_ipip_capture_on", 0) > modparam("sipcapture", "table_name", "sip_capture") > modparam("sipcapture", "raw_sock_children", 4) > modparam("sipcapture", "db_insert_mode", 0) > modparam("sipcapture", "raw_moni_capture_on", 1) > modparam("sipcapture", "promiscious_on", 1) > modparam("sipcapture", "raw_moni_bpf_on", 1) > modparam("sipcapture", "capture_node", "homer02") > modparam("sipcapture", "authorization_column", "authorization") > > > ## logging all INVITES top of the [route] block > if (is_method("INVITE|REGISTER")) { > xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" > from \"$si\"\n"); > > Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. > > root@homer02:~# netstat -al > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 homer02.me.com:sip *:* LISTEN > > > I don't think this is a homer issue because logging invites is prior > to any homer processing. I'm thinking this is something simple I'm > overlooking, any help is much appreciated. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > Chasing the Azeotrope > > _______________________________________________ > 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

Kamailio segfault on TM module
by José Seabra 16 Jan '17

16 Jan '17

Hi there, I have noticed a segfault on my kamailio server but as i have coredumps disabled and i don't know how i can reproduce it, i can't provide that kind of information. Anyway i don't know if anyone of you can help with the following information: segfault at 600 ip 00007f5cb97af1ea sp 00007ffed96b8d60 error 6 in tm.so[7f5cb9718000+11d000] ERROR: tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 404 reply when a final 200 was sent out ERROR: sl [sl.c:269]: send_reply(): failed to reply stateful (tm) WARNING: db_postgres [km_dbase.c:281]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] ALERT: <core> [main.c:739]: handle_sigs(): child process 2048 exited by a signal 11 CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 14 ALERT: <core> [main.c:742]: handle_sigs(): core was not generated INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received INFO: <core> [main.c:809]: sig_usr(): signal 15 received Database server was down during this segfault, but the segfault was on TM module not on database module. Thank you Regards José Seabra

2 1

dbtext module issue
by Satish Patel 16 Jan '17

16 Jan '17

I am trying use dbtext module instead of mysql and following is my configuration. We are using old SER version because of custom written module. Do you think following config looks good. # -- permissions params -- modparam("permissions", "db_url", "dbtext:///etc/ser/dbtext") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") In function if ( !allow_trusted() && !search ("^User-Agent:.*Some Perl client.*") ) { xlog("L_WARN", "Drop if User-Agent isn't Perl client"); drop; break; }; My dbtest contents $ cat /etc/ser/dbtext/trusted src_ip(str) proto(str) from_pattern(str) 10.0.2.20:any:.* When i run ser in fork=no mode i am seeing following ser -f ser.cfg Content of result table_version(int) 2 Content of [trusted] src_ip(str) proto(str) from_pattern(str) 10.0.2.20:any:.* Content of result proto(str) from_pattern(str) "any" ".* " Content of result proto(str) from_pattern(str) "any" ".* " it look like it read dbtext file and load data but when i use sipp to generate some call then i am seeing following result only, that means its not loading IP address from table, why its extracting two field but not src_ip ? Content of result proto(str) from_pattern(str) "any" ".* "

2 2

comma as param.count delimiter produces parse error
by Juha Heinanen 15 Jan '17

15 Jan '17

Parameter list transformation param.count produces parse error when delimiter is comma (,) character. >From document: {param.value,name[, delimiter]} 'delimiter' allows you to specify a single character to use as the parameter delimiter. For example, when parsing HTTP URL query strings use '&'. When I have in config statement: $var(count) = $(var(servers){param.count,,}); I get to syslog: Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: pv [pv_trans.c:2645]: tr_parse_paramlist(): invalid separator in transformation: count,,} Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1629]: tr_lookup(): error parsing [{param.count,,}] Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "var" Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) ERROR: <core> [core/pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(var(servers){param.count,,})] at (4) Jan 13 07:48:15 lohi sip-proxy[18000]: 0(18050) : <core> [core/cfg.y:3407]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 6426, column 19-48: Can't get from cache: $(var(servers){param.count,,}) Jan 13 07:48:15 lohi sip-proxy[18000]: ERROR: bad config file (1 errors) -- Juha

2 2

Kamailio not processing SIP TCP
by JR Richardson 12 Jan '17

12 Jan '17

Hi All, Just enabled SIP TCP on a homer capture server, I can see the SIP TCP Sessions on the server with ngrep, just like all the UDP traffic. I have Kamailio listening on TCP ports but its not capturing any TCP traffic. kamailio.cfg: #disable_tcp=yes listen=tcp:10.99.99.99:5060 #monitor port listen=udp:10.99.99.99:5060 #monitor port loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "rtimer.so" loadmodule "xlog.so" loadmodule "sqlops.so" loadmodule "htable.so" loadmodule "sl.so" loadmodule "siputils.so" modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070") modparam("sipcapture", "raw_interface", "eth1") modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "db_insert_mode", 0) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) modparam("sipcapture", "capture_node", "homer02") modparam("sipcapture", "authorization_column", "authorization") ## logging all INVITES top of the [route] block if (is_method("INVITE|REGISTER")) { xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\" from \"$si\"\n"); Logging reports all SIP UDP traffic to logs fine, but no TCP traffic. root@homer02:~# netstat -al Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 homer02.me.com:sip *:* LISTEN I don't think this is a homer issue because logging invites is prior to any homer processing. I'm thinking this is something simple I'm overlooking, any help is much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

2 1

C++ module
by Patrick 12 Jan '17

12 Jan '17

I added a C++ module to kamalio at some point, and now I am trying to integrate that module with another C++ internal library, but I'm getting all sorts of headaches when trying to compile. The latest problem is during the "make install" part of the build - it seems like the objects from the module are "infecting" the link line of the internal library it's dependent on. Here is the Makefile for the module. At the time, the only way I figured to link in the C++ object was adding them to the LIBS. Now, i think this interferes with the 'make install' dependencies. Any thoughs on how to do this properly would be great! Thanks *Module Makefile* # # db_cassandra module makefile # # # WARNING: do not run this directly, it should be run by the master Makefile include ../../Makefile.defs CXX=g++ LD=g++ auto_gen= NAME=db_arl.so ifeq ($(CROSS_COMPILE),) CURL_BUILDER=$(shell \ if pkg-config --exists libcurl; then \ echo 'pkg-config libcurl'; \ else \ which curl-config; \ fi) endif DEFS+=-DKAMAILIO_MOD_INTERFACE -DRTC_COMMON_TRACE *LIBS += db_arl_base.o jsoncpp.o* -lcurl -pthread CXXFLAGS=$(CFLAGS:-Wno-deprecated option=) CXXFLAGS+= -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing CXXFLAGS+= -I ${COL_LOCAL_BUILDS_ROOT}/ExternalLibs/boost/V1.60.0_27/ CXXFLAGS+= -I ./jsoncpp-dist/ SERLIBPATH=../../lib SER_LIBS+=$(SERLIBPATH)/srdb2/srdb2 SER_LIBS+=$(SERLIBPATH)/srdb1/srdb1 *SER_LIBS+=$(SERLIBPATH)/rtctrace/rtctrace* include ../../Makefile.modules jsoncpp.o: jsoncpp-dist/jsoncpp.cpp @echo COL_LOCAL_BUILDS_ROOT=${COL_LOCAL_BUILDS_ROOT} @echo "Compiling $<" $(CXX) $(CXXFLAGS) $(CFLAGS) $(C_DEFS) $(DEFS) -c $< -o $@ db_arl_base.o: db_arl_base.cpp db_arl_base.h @echo COL_LOCAL_BUILDS_ROOT=${COL_LOCAL_BUILDS_ROOT} @echo "Compiling $<" $(CXX) $(CXXFLAGS) $(CFLAGS) $(C_DEFS) $(DEFS) -c $< -o $@ db_arl.so: db_arl_base.o jsoncpp.o During 'make install' the objects in *red* have no business in the link line for the internal library /opt/rh/devtoolset-4/root/usr/bin/g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=opteron -Wall -I/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/include -DNAME='"kamailio"' -DVERSION='"4.2.4"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.7"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=4002004 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DF_MALLOC -DDBG_F_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -std=c++0x -I/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/include -c rtctrace.c -o rtctrace.o -MMD -MP /opt/rh/devtoolset-4/root/usr/bin/g++ -shared -m64 -Wl,-O2 -Wl,-E -Wl,-soname,librtctrace.so.1 rtctrace.o * db_arl_base.o jsoncpp.o* -lcurl -pthread -L/home/pshea/_localbuilds/RTC/Common/2.0.0_247//Applications_rhel6_x64_dts4_pic/lib -lrtccommon -L/home/pshea/_localbuilds/ExternalLibs/boost/V1.60.0_22/Server_Linux26_x64_dts4_pic/lib -lboost_system -lboost_thread -lpthread -o librtctrace.so.1.0 *g++: error: db_arl_base.o: No such file or directory* *g++: error: jsoncpp.o: No such file or directory*

2 7

Difference between HTTP clients (utils.so vs http_client.so)
by Joel Serrano | VOZELIA 12 Jan '17

12 Jan '17

Hi all, Currently I see there are 2 http clients: utils.so --> http_query() http_client.so --> http_client_query() Any advice on which to use? Any pros/cons of using one over the other? Thanks, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

3 4

RPC exact command to change voicemail IP or Port
by Annus Fictus 12 Jan '17

12 Jan '17

Hello list, I can't find the exact RPC command to change IP or port of voicemail server (module CFG_RPC): the default Kamailio configuration is: */#!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/* Thank you Regards

2 8

how to immediately close a tls connection
by Andy 12 Jan '17

12 Jan '17

Hi, trying to close a tls connection immediately, have ref to the connection (tcpconn) pointer please suggest, thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/how-to-immediately-close-a-tls-conn… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio and default paramenters to a specific gatewy
by Laura 11 Jan '17

11 Jan '17

Dear all, I'm using kamailio 4.3.6 and LCR modules.. i was wondering if it was possibile to add some special paramenter to all call sent to a specific gateway. I was try to use the " params" field on lcr_gw table but it's not working like i was think.. The issue is simple.. i want to add to field "Privacy: id" to any call i send over a specific gateway.. What is the best and easy way to do that ? Any idea is appreciate. Regards Laura

3 9

Diameter AVP handling from kamaiio.cfg
by Zoltán Miricz 10 Jan '17

10 Jan '17

Dear All, I'm wondering is it possible to catch Diameter Messages and their AVP values from kamailio.cfg just like for SIP messages? (Using pseudo variables) I looked into this function from the attached S-CSCF example. route[REG_MAR_REPLY] { #this is async so to know status we have to check the reply avp xlog("L_DBG","maa_return code is *$avp(s:maa_return_code)*\n"); Sadly it gives back only 1, -1, -2. (which is based on cxdx_mar.c the rc) What I mainly looking for is the EXPERIMENTAL_RESULT (experimental_rc) and some different AVP values besides. Thank you very much in advance! *Cheers,* *Zoltan*

2 5

Kamailio on Cluecon Weekly – Jan 11, 2017
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, for those interested in participating to a (video) chat tomorrow, Wed, Jan 11, I will be joining the ClueCon Weekly at 12:00CT (USA) to discuss about latest Kamailio, what's new and the plans for the near future. Integration with FreeSwitch is going to be among the topics. You can join the video conference call with a webrtc capable browser. You can also join only for audio using a SIP phone or dialing from PSTN or mobile networks. More specific details are available at: - https://www.kamailio.org/w/2017/01/kamailio-on-cluecon-weekly-jan-11-2017/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, are you using topoh module? A pcap with all sip messages part of the dialog, from the initial INVITE to the last BYE transaction would be important to have in order to investigate how the headers are changed. Cheers, Daniel On 28/12/2016 13:11, Nihar Ranjan Deb wrote: > > Hi All, > > > > I am a new user of kamailio. We have setup a kamilio server with RTP > proxy. Most of the features are working fine except BYE. > > > > > > Say User A and B are registered in kamailio server. A (caller) > connected to B(called), call established. > > > > 1. If Caller (A) disconnects the call, everything goes fine. B gets > disconnected and call releases. > > > > 2. But if Called party (B) disconnects the call, A never gets > disconnected. I checked trace as well, in this case caller (A) gets a > BYE with B's callid. So A rejects call with "481 call leg transaction > doesn't exists". > > Basically A party and B party has separate dialog established with > Kamilio. So they have two separate callids . All messages are > exchanged correctly. Only when B party disconnects call, kamailio > sends BYE to A party with B party callid. > > > > Please help me to resolve this issue. Do let me know if any more > information required. > > > > > > ------ INVITE-------------------------- > > > > INVITE sip:966268@*** SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport > > Max-Forwards: 70 > > Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> > > To: <sip:966268@***> > > From: "334031"<sip:334031@****>;tag=4406552c > > Call-ID: *82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc* > > CSeq: 2 INVITE > > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, > OPTIONS, MESSAGE > > Content-Type: application/sdp > > Proxy-Authorization: Digest > username="334031",realm="*****",nonce="****",uri="sip:966268@***",response="***",algorithm=MD5 > > Supported: replaces > > User-Agent: X-Lite release 4.9.6 stamp 82158 > > Content-Length: 334 > > > > v=0 > > o=- 13127393523589215 1 IN IP4 192.168.1.36 > > s=X-Lite release 4.9.6 stamp 82158 > > c=IN IP4 192.168.1.36 > > t=0 0 > > m=audio 57116 RTP/AVP 9 8 120 0 84 101 > > a=rtpmap:120 opus/48000/2 > > a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 > > a=rtpmap:84 speex/16000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > > > > > > > -------------------------BYE-------------------------------- > > > > > > BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > From: <sip:966268@***>;tag=1665200125 > > To: "334031" <sip:334031@***>;tag=4406552c > > Call-ID: > *!!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV*** > > CSeq: 3 BYE > > Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> > > Max-Forwards: 69 > > User-Agent: Yealink SIP-T20P 9.72.0.80 > > Content-Length: 0 > > > > > > -------------------------- 481--------------------- > > > > SIP/2.0 481 Call/Transaction Does Not Exist > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.164.228.12 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > To: "334031" <sip:334031@***>;tag=4406552c > > From: <sip:966268@***>;tag=1665200125 > > Call-ID: > !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV** > > CSeq: 3 BYE > > Accept-Language: en > > Content-Length: 0 > > > > > > > > > > Regards, > > Nihar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

[Error] Install Kamailio 4.4 in Mac 10.12
by Luis De Marchi 10 Jan '17

10 Jan '17

Hi, Sorry, my english, I'm using the translator. I'm having trouble installing Kamailio for Mac 10.12 Terminal: Luis:kamailio luisdemarchi$ sudo make all CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:41:10: fatal error: 'mysql.h' file not found #include <mysql.h> ^ 1 error generated. make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Luís De Marchi Diretor Administrativo +55 (41) 99815-6800 +55 (41) 3027-7669 www.snowmanlabs.com <http://www.snowmanlabs.com/>

4 4

t_precheck_trans
by Annus Fictus 10 Jan '17

10 Jan '17

Hi, where can I find documentation for this function present on the default script? In TM module not exist (Kamailio 4.4.4) Thank you Regards

3 3

use case
by Thufir Hawat 09 Jan '17

09 Jan '17

I'm ok with Linux and am dipping my toe into Asterisk by running an AWS EC2 instance. What is the most frequent usage for kamailio? Would it mainly be for SIP to SIP? So that the URI would be for a specific domain? thanks, Thufir

3 2

using kamailio to forward ipv6 to ipv4
by anfecora 09 Jan '17

09 Jan '17

Hi guys hope you have a great new year. I would appreciate if anyone can point me in to the right direction . I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. is that possible? thanks.

4 5

why a register error?
by Olvera, Victor 09 Jan '17

09 Jan '17

As I attempt to use X-lite I get a 504 error. On the CSCFs I get the following: pcscf 0(2066) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2066) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.1:55880) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) icscf 11(2150) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 0(2138) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2138) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.76:4060) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) 0(2138) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 0 scscf 11(2047) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 192.168.56.78 port 3868 11(2047) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 9(2045) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

2 1

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 09 Jan '17

09 Jan '17

Anyone any help on this. Regards, Nihar From: Nihar Ranjan Deb [mailto:nihar.ranjan@microtalkgroup.com] Sent: Wednesday, December 28, 2016 5:42 PM To: 'sr-users(a)lists.sip-router.org' Subject: BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists". Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

1 0

VERY IMPORTANT: deciding when to remove the MI code
by Daniel-Constantin Mierla 06 Jan '17

06 Jan '17

Hello, we started discussing about removing MI (so called management interface) for very long time, more or less since 2008. The RPC should remain the control interface, given its better structure for commands, parameters, etc ... MI is custom protocol using a line-oriented communication via fifo or socket file with kamailio (e.g., implemented mi_fifo and mi_datagram modules). RPC is the alternative, a more standardized concept, with better structured format. I think it's time to set a clear roadmap for doing the removal. Overall, it will be easier to maintain the code, right now being duplicated code for doing the same operation over MI or RPC, and MI shows its limitations (or complexity to deal with) for advanced needs (see the discussions about how to provide multi-line value parameters over MI). So, I want to know if there are many relying on MI directly and they still want to keep it, what would be the expected duration they need for upgrading their tools to work with RPC interface, other relevant aspects people have in favour of mi vs rpc. I am even willing to do the removal in time befire freezing the 5.0 branch. We will ensure a clean start of 5.x series. The main concern from my point of view is kamctl -- but I think we can preserve the compatibility for kamctl commands and parameters (so command line execution of kamctl will be the same), but the output might be different. That's because it should be easy to updated it to communicate with jsonrpc-s module, but then it will get json-formatted results. To summarize, two big questions to answer: a) Are you ok to remove the MI code/commands? b) If yes to a), are you ok to be done for v5.0? Not providing feedback will be considered as 'yes' for both questions, so **speak up if you want MI to be kept or delay it removal**. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

6 30

presentity tabla, faster update when drop TCP/TLS connection
by João Resende 05 Jan '17

05 Jan '17

I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes. Exists a way to define this? or it is a bug? or should be a new feature? I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.

3 9

Re: [SR-Users] using kamailio to forward ipv6 to ipv4
by anfecora 05 Jan '17

05 Jan '17

Yes i am planning on proxing media. The case is that client will only talk ipv6 but the registrar only have ipv4 connectivity. Therefore need to create a proxy to translate this dialogs. Thi thnking on using kamailio to receive the registration modify the pckets and forward registrations in ipv4. I was thinking in using the asterisk kamailio example but instead of asterisk behind will be an sbc. Therefore kamailio will listen in ipv6 and translate over to ipv4 including media of course via rtpproxy. On Jan 5, 2017 9:43 AM, "Frank Carmickle" <frank(a)carmickle.com> wrote: > On Jan 5, 2017, at 12:24 PM, anfecora <anfecora(a)gmail.com> wrote: > > Hi guys hope you have a great new year. > I would appreciate if anyone can point me in to the right direction . > > I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. A bit more information about the situation would be helpful. What exactly are you trying to do? > is that possible? 6 to 4 translation is possible, but until we understand the situation I’m not sure we can be that helpful. Are these phones? If so are they dual stack? Is the underlying network dual stack? There’s a lot of work to do to get this right in a Kamailio config. You’ll need to run rtpengine to proxy the media if the endpoints are not dual stack. Who keeps the mapping of the v6 to v4 and v4 to v6 translations? How is that mapping determined? I’d recommend ds-lite if you are in a position to alter the network and the phones/systems allow for it. HTH —FC > thanks. > > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

CDR - acc_cdrs
by Annus Fictus 05 Jan '17

05 Jan '17

Hello, I'm trying to save Kamailio CDR on acc_cdrs table without success. My relevant configuration look like: ----- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 ----- #!define WITH_ACCDB ---- loadmodule "dialog.so" loadmodule "acc.so" modparam("dialog", "dlg_flag", 0) ---- # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) 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) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$td") #!endif Any hint is really appreciate. Regards

2 4

unexpected char - status 6
by AL S 05 Jan '17

05 Jan '17

Hi, Kamailio is throwing error message from "parse_to.c" with status=6. The source code shows status 6 as URI_ENCLOSED. However, tcpdump of the call does not show any enclosed angle bracket is missing. (tcpdump pcap file is attached.) Kamailio log Jan 4 10:18:21 orl-ar-openser01 /usr/sbin/kamailio[2374]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK04001cd3>> . FROM and TO in INVITE received by Kamailio From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161>;tag=gK04001cd3 FROM and TO in INVITE forward by Kamailio for the same call From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161\000\000>;tag=gK04001cd3 PS: this error message is happening on all 4 kamailio servers (3.3.0) PS: call failure only happens in re INVITE (call stays active longer than 15 min) Your help would be appreciated. Thanks, AS

2 2

RTPProxy with VPN
by Rodrigo Moreira 05 Jan '17

05 Jan '17

Hello, I am having problems with RTPPROXY. I need it to work as a bridge. On one side will have a VPN tunnel called a tun0, on the other an eth0 interface. Is it possible for RTPPROXY to run correctly in this scenario? How can I test if RTPPROXY is running correctly? Is there any command? What would the configuration look like inside /etc/default/rtpproxy Please help me. Best regards -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Re: [SR-Users] Regarding usage of Redis Module in Kamailio core
by Vicente Hernando 05 Jan '17

05 Jan '17

Hello Hemanth, this minimal kamailio cfg file works for me to test ndb_redis module. #!KAMAILIO debug=4 log_stderror=yes children=4 fork=yes #disable_tcp=yes mpath="modules/" loadmodule "ndb_redis.so" # An already configured redis server is needed here modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=0") request_route { redis_cmd("srv1", "SET foo bar", "r"); exit; } Changing the parameters to suit your configuration, you can use it to debug your system. Regards, Vicente. On 01/05/2017 07:55 AM, hemanth(a)advaitamtech.com wrote: > > Hello Vicente, > > Thanks for your response, > > In response to your query, > > Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, > > I have also modified the kamailio.cfg file loadmodule and modparam list, > > I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. > > Thanks & Regards > Hemanth > > > Message: 8 > Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) > From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core > Message-ID: <1483521102.18025940(a)apps.rackspace.com> > Content-Type: text/plain;charset=UTF-8 > > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > > > ------------------------------ > > Message: 9 > Date: Wed, 4 Jan 2017 10:50:05 +0100 > From: Vicente Hernando <vhernando(a)systemonenoc.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio > core > Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello Hemanth, > > just a quick response, I would check if you compiled ndb_redis module > correctly (also if you have hiredis library dependency). > > Also I would check if you loaded ndb_redis module in your configuration > file. > > > Regards, > Vicente. > > On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: >> Hello, >> >> I am using Kamailio as a SIP proxy, >> >> I want to store Registered user information in Redis DB, >> >> I am able to configure REDIS DB, But i am not able to run the redis command from >> >> Kamailio core using the function >> >> "redis_cmd("srvN","SET key 1", "r")" >> >> I am getting undefined reference error as below >> >> "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" >> >> can anyone please help me to fix this issue >> >> Thanks & Regards >> Hemanth >> >> >> _______________________________________________ >> 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 > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ------------------------------ > > End of sr-users Digest, Vol 140, Issue 3 > **************************************** > > > >

1 0

Kamailio cancel branch only after receiving replies
by Aqs Younas 05 Jan '17

05 Jan '17

Greetings list, I am forking a call to multiple destinations and want to keep the only branch which sends quicker first 180/183 reply and cancel the remaining branches. Below is my related cfg snippet. modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "cancel_b_method", 2) ... route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); append_branch(); append_branch(); route(RELAY); exit; } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","Received $rs (IP:$si:$sp)\n"); if(status=~"18[03]"){ t_cancel_branches("others"); xlog("L_INFO","cancelled all other branches\n"); } route(NATMANAGE); } } ... But I see kamailio does not instantly send CANCEL to other branches after it has received 180/180 from any branch. Kamailio is sending CANCEL to other branches after they start sending 180/183 one by one. How can I cancel all other branches instantly? I am sure there is something wrongly configured in my configuration. Any suggestion is much appreciated. Best Regards.

3 3

location
by Slava Bendersky 05 Jan '17

05 Jan '17

Hello Everyone, Why save location will ignore on pv var when trying save on reply 200 OK. if(is_method("REGISTER") && fnmatch("$rs", "200")) { xlog("L_INFO", "OnReply: Registration is $rs OK saving --> $avp(rcvcontact) in db\n"); $var(oar) = "sip:" + $tU + "@" + $(avp(rcvcontact){uri.host}) + ":" + $(avp(rcvcontact){uri.port}); if(!save("location", "0x04", "$var(oar)")) { xlog("L_INFO", "Error saving the location\n"); } Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=619 a=26 n=xlog Dec 26 11:43:20 cavprx00 kamailio: 4(5546) INFO: <script>: OnReply: Registration is 200 OK saving --> sip:4300@client_pub_ip:31376;transport=tls in db Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=620 a=63 n=assign Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=624 a=16 n=if Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=621 a=27 n=save Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-1] (1 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-2] (2 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-2 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1004]: db_update_ucontact_ruid(): ruid:uloc-5860b463-10c5-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1133]: db_update_ucontact_ruid(): contact:sip:4300@10.18.130.27:5060;transport=udp ----> LAN side contact Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 10 chars, out: 10 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 21 chars, out: 21 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 33 chars, out: 33 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 40 chars, out: 40 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 20 chars, out: 20 chars Slava.

2 7

Kamailio v5.0 - development frozen
by Daniel-Constantin Mierla 05 Jan '17

05 Jan '17

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 5.0.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 5.0 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to reach a stable state before releasing 5.0.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Besides the new features, there were two major changes for 5.0: 1) source code tree restructuring - this should not affect the stability of the code, only installation scripts or packaging may still need tuning 2) mi (management/control interface) code has been removed. SIP routing code should not be affected by this change that much, but testing of RPC commands needs a special care. There are few RPC commands not ported yet from the MI code, they can be done during the testing period Moreover, help with updating the wiki page for migration from 4.4 to 5.0 as well as what is new in 5.0 is very appreciated. I will try post updates about them very soon. Many thanks to everyone involved in development of 5.0 and the early testers that played with master branch so far. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

unexpected char [#015]
by AL S 04 Jan '17

04 Jan '17

Hi, Kamailio is complaining about unexpected char [#015] in TO header of INVITE: Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK006cb08b>> . Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/msg_parser.c:180]: ERROR: get_hdr_field: bad to header Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:2057030997@] Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: tm [t_lookup.c:1118]: ERROR: reply cannot be parsed Here is the TO header for the same call in both text and hex format. I am unable to see any ^M char in this TO header. To: <sip:2057030997@72.31.12.161;tag=gK006cb08b 546f3a203c7369703a323035373033303939374037322e33312e31322e3136313e3b7461673d674b30303663623038620d0a Any help would be appreciated. AS,

2 2

IMPORTANT: MI code removed (or disabled)
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, as of now, no active MI reladed code should be in master branch -- the internal library kmi and the mi_datagram, mi_fifo, mi_xmlrpc, mi_rpc and pua_mi modules were removed. Couple of modules still have some mi code disabled with ifdefs, they are pending the port to RPC commands. These are: - carrierroute - ims_dialog - mohqueue - p_usrloc - pdb - rtpengine - userblacklist - utils They should be updated in the next few days. With MI code removed, the code is slimmer and the development effort is reduced -- in the recent past there were various issue rising due to usage of the old line-based MI commands. >From now on, the RPC interface has to be used for interacting with kamailio at runtime. The tools: - kamctl - was already migrated to use RPC via jsonrpcs module. - kamcmd - was developed to use only RPC interface via ctl module. - kamcli - is also working with RPC via jsonrpcs module Helping with testing the RPC commands is very appreciated -- open an issue on bug tracker whenever you discover a problem. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] sr-users Digest, Vol 140, Issue 3
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello Vicente, Thanks for your response, In response to your query, Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, I have also modified the kamailio.cfg file loadmodule and modparam list, I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. Thanks & Regards Hemanth Message: 8 Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <1483521102.18025940(a)apps.rackspace.com> Content-Type: text/plain;charset=UTF-8 Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth ------------------------------ Message: 9 Date: Wed, 4 Jan 2017 10:50:05 +0100 From: Vicente Hernando <vhernando(a)systemonenoc.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> Content-Type: text/plain; charset=utf-8; format=flowed Hello Hemanth, just a quick response, I would check if you compiled ndb_redis module correctly (also if you have hiredis library dependency). Also I would check if you loaded ndb_redis module in your configuration file. Regards, Vicente. On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > _______________________________________________ > 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 ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 140, Issue 3 ****************************************

1 0

Regarding usage of Redis Module in Kamailio core
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth

2 1

Update: removing MI code
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, based on lib linking in Makefiles, the following modules still use MI, categorized by their state from my point of view. Hopefully some other people will help with those listed in 2) to 6). 1) to be removed - mi_datagram - mi_fifo - mi_rpc - mi_xmlrpc - pua_mi 2) rpc commands implemented - dialog - others modules depend on some mi callback from it (qos and sst) - qos - this looks ready to wipe out the mi code, but didn't get the time not analyze properly as I don't use it usually 3) some rpc commands implemented and the rest should not be very complex - carrierroute - I implemented the dump rpc command, but it was reported that it has issues, so it needs follow up. 4) rpc commands have to be implemented, expecting not to be very complex - cplc - imc 5) expecting some degree of complexity, but they are important modules - rtpengine - rtpproxy 6) not familiar with the mi commands in these modules, so not able to assert what to expect - ims_dialog - some rpc commands are implemented, not sure if the rest of MI are used/usefull - mohqueue - it doesn't seem complex to implement rpc commands at first sight, but the indentation style didn't allowed a fast analyze on a quick look - p_usrloc - several mi commands - sst - uses some callback for MI from dialog module. qos has something similar with rpc alternative already implemented - userblacklist - several mi commands - utils - there are few mi commands related to some forwarding rules: https://www.kamailio.org/docs/modules/devel/modules/utils.html#idp21741924 The 1) to 5) should be done in a way or another, before of after freeze of 5.0.0. But 6) would require the devs of the modules (or the MI parts) to help if they want those commands via RPC. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Kamailio as loadbalancer + NAT pinning
by Vladyslav Zakhozhai 03 Jan '17

03 Jan '17

Hi, I want to use Kamailio as SIP loadbalancer and NAT helper in front of freeswitch boxes. Main tasks here are: 1. load balancing SIP requests to freeswitch boxes 2. fixing contact information during registrations and invites for nat'ed clients (kamailio is not registrar) 3. fixing sdp (or proxying rtp via rtpproxy) for nat'ed clients 4. security (check SIP messages, pike, etc) 5. outbound proxy for freeswitch boxes The main question here is about nat pinning. Acctording to module doc ( http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html) I need nathelper module and usrloc module. So I can proxy REGISTER requests to freeswitch boxes and in onreply_route check for REGISTER statuses: 1. if no success we use pike to limit bruteforce attacks 2. if success I need to save contact to usrloc db table But there is one issue. Multiple registration are allowed on freeswitch boxes and in replies (freeswitch -> kamailio) I see more than one Contact header. Is it an issue or kamailio will handle it without problems? Or I need to save Contact header from request to hash table and retreive it in onreply_route? But I do not know how to save it in usrloc db table then. I thin this is more sophisticated approach. I appreciate your advice. Thank you in advance. -- С уважением, Владислав Захожай

2 4

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 03 Jan '17

03 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

2 11

Rewrite From domain
by Gonzalo Gasca Meza 03 Jan '17

03 Jan '17

Hi all, I'm using Kamailio to forward calls between 2 Service Providers and I need to rewrite the From header "domain" URI. Example: From: "+18888888888" <sip:+18888888888@sip.sp1.com> to From: "+18888888888" <sip:+18888888888@*sip.sp2.com <http://sip.sp2.com>*> *Call flow:* Phone A --- > SP1 ---> sip ----> (kamailio) SP2 --(LOCATION)-> Phone B When Phone A calls SP2 PhoneB, it contains original sip domain from sp1. ( sip.sp1.com) hence user in SP2 can see call comes from SP1. I would like to rewrite the From domain field in this conditions: a) Calls comes from "sip.sp1.com" AND b) Call is being routed to PhoneB. Right now Im using the following code to find user and send call to B. #!ifdef WITH_ALIASDB # search in DB-based aliases xlog("L_INFO","alias_db_lookup: Call received. $rU\n"); if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif I found this in documentation: $fd - From URI domain if($hdr(From)=~"sip.sp1\.com") { ...} But not sure where is the best place to overwrite the From URI domain header. Thanks

3 7

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 02 Jan '17

02 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

1 0

Freezing for release v5.0.0
by Daniel-Constantin Mierla 02 Jan '17

02 Jan '17

Hello, I propose to freeze the development of v5.0.0 on January 4, 2017. Hopefully it is just in time for getting the frozen version in next Debian stable. Meanwhile, I hope we can get rid of mi code to have a proper fresh start with cleaned up code in 5.0.0. The big code tree restructuring was already done, config files in embedded languages is there as well (it needs more exports, but that's only about available time to fill some structures, not new feature development). If anyone wants to make other proposals, reply to mailing lists and let's discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Happy 2017!
by Daniel-Constantin Mierla 01 Jan '17

01 Jan '17

A fresh year ahead, let it be a great one for all Kamailio friends! Wishing a prosperous and safe 2017 everywhere! Happy New Year! -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users