<div dir="ltr">Thats because your configuration file is not sending packet out (RELAY) to MSC instead it is only doing a Loadbalancer / destination lookup in TOASTERISK route and comes out of it, processes the following routes in order<div><div style="font-size:12.8000001907349px"> route(SIPOUT);</div><div style="font-size:12.8000001907349px"><span style="font-size:12.8000001907349px"> route(PRESENCE);</span><br></div><div style="font-size:12.8000001907349px"><span style="font-size:12.8000001907349px"> route(REGISTRAR);</span><br></div><div style="font-size:12.8000001907349px"><span style="font-size:12.8000001907349px"> route(PSTN);</span><br></div><div style="font-size:12.8000001907349px"><span style="font-size:12.8000001907349px"> route(LOCATION);</span><br></div></div><div style="font-size:12.8000001907349px"><span style="font-size:12.8000001907349px"><br></span></div><div><span style="font-size:12.8000001907349px">Where finally in LOCATION route it tries to find the destination user 0730092190 online locally on Kamailio, which it can't find and says 404 Not Found.</span></div><div><span style="font-size:12.8000001907349px"><br></span></div><div><span style="font-size:12.8000001907349px">You should modify your TOASTERISK route as follow:</span></div><div><span style="font-size:12.8000001907349px"><br></span></div><div><div style="font-size:12.8000001907349px">route[TOASTERISK] {</div><div style="font-size:12.8000001907349px">if(ds_is_from_list("2")) {</div><span class="im" style="font-size:12.8000001907349px">#Call from Telco Should goto Asterisk pool in Loadbalanced mode</span><span class="im" style="font-size:12.8000001907349px"><div> if(!ds_select_dst("1", "4")) {</div><div> sl_send_reply("500", "Service Unavailable");</div><div> xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n");</div><div> exit;</div><div> }</div><div> <span style="background-color:rgb(255,229,153)">route(RELAY);</span></div></span><div style="font-size:12.8000001907349px">}if(ds_is_from_list("1")) {</div><span class="im" style="font-size:12.8000001907349px">#Call from Asterisk servers pool, send it to telco using LoadBalancer</span><span class="im" style="font-size:12.8000001907349px"><div> if(!ds_select_dst("2", "4")) {</div><div> sl_send_reply("500", "Service Unavailable");</div><div> xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n");</div><div> exit;</div><div> }</div><div> <span style="font-size:12.8000001907349px;background-color:rgb(255,229,153)">route(RELAY);</span></div><div> }</div><div><br></div><div>}</div></span></div><div><span style="font-size:12.8000001907349px"><br></span></div><div><br></div><div><span style="font-size:12.8000001907349px">This will immediately route the packet out towards the new $du after the loadbalancer function ds_select_dst(...)</span></div><div><span style="font-size:12.8000001907349px"><br></span></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Aug 11, 2015 at 10:48 AM, Sandeep Chakravarthi <span dir="ltr"><<a href="mailto:ivschakravarthi@gmail.com" target="_blank">ivschakravarthi@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div>Kamailio is sending 404 Response and its not MSC.</div><div>If you see the pcap file , Kamailio has to forward the SIP invite packet to MSC which it got from Asterisk server. But it is not happening.</div><div>I am attaching the pcap one more time for your reference.</div><div><br></div><div>In my pcap, below are the server details</div><div><br></div><div>172.22.14.12 - Kamailio server</div><div>172.22.14.17 - Asterisk server</div><div>172.22.0.68 - MSC</div><div><br></div><div><br></div><div>Regards,</div><div>Sandeep</div></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Warm Regards,<br>Sandeep Chakravarthi.</div></div></div><div><div class="h5">
<br><div class="gmail_quote">On Tue, Aug 11, 2015 at 7:10 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Sandeep,<div>what is the problem here ? Kamailio just sends a 404 on its own or is really sending calls to MSC and MSC is replying with 404 ?</div><div><br></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi <span dir="ltr"><<a href="mailto:ivschakravarthi@gmail.com" target="_blank">ivschakravarthi@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi ,<div>Sorry for the delayed reply.</div><div>I have configured my Asterisk and kamailio server, but when i initiate one outbound call from my asterisk server to kamailio server, kamailio server is initiating the call to MSC.</div><div>Please find the attached pcap details for your reference.</div><div>Below is my kamailio debug log and kamailio.cfg file.</div><div>Please check the pcap and below cfg file and log file and let me know whether to change anything in cfg file or not.</div><div><br></div><div>++++++++++++++++++++++++++++++++++++++++++++++++</div><div><br></div><div><div><br></div><div>request_route {</div><div><br></div><div> # per request initial checks</div><div> route(REQINIT);</div><div><br></div><div> # NAT detection</div><div> route(NATDETECT);</div><div><br></div><div> # CANCEL processing</div><div> if (is_method("CANCEL"))</div><div> {</div><div> if (t_check_trans()) {</div><div> route(RELAY);</div><div> }</div><div> exit;</div><div> }</div><div><br></div><div> # handle requests within SIP dialogs</div><div> route(WITHINDLG);</div><div><br></div><div> ### only initial requests (no To tag)</div><div><br></div><div> t_check_trans();</div><div><br></div><div> # authentication</div><div> route(AUTH);</div><div><br></div><div><br></div><div> # record routing for dialog forming requests (in case they are routed)</div><div> # - remove preloaded route headers</div><div> remove_hf("Route");</div><div> if (is_method("INVITE|SUBSCRIBE"))</div><div> record_route();</div><div><br></div><div> # account only INVITEs</div><div> if (is_method("INVITE"))</div><div> {</div><div> setflag(FLT_ACC); # do accounting</div><div> }</div><div> route(TOASTERISK);</div><div><br></div><div> # dispatch requests to foreign domains</div><div> route(SIPOUT);</div><div><br></div><div> ### requests for my local domains</div><div><br></div><div> # handle presence related requests</div><div> route(PRESENCE);</div><div><br></div><div> # handle registrations</div><div> route(REGISTRAR);</div><div><br></div><div> if ($rU==$null)</div><div> {</div><div> # request with no Username in RURI</div><div> sl_send_reply("484","Address Incomplete");</div><div> exit;</div><div> }</div><div><br></div><div> # dispatch destinations to PSTN</div><div> route(PSTN);</div><div><span style="white-space:pre-wrap"> </span># user location service</div><div> route(LOCATION);</div><div>}</div><div><br></div><div>route[TOASTERISK] {</div><div>if(ds_is_from_list("2")) {</div><span><div>#Call from Telco Should goto Asterisk pool in Loadbalanced mode</div></span><span><div> if(!ds_select_dst("1", "4")) {</div><div> sl_send_reply("500", "Service Unavailable");</div><div> xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n");</div><div> exit;</div><div> }</div></span><div>}if(ds_is_from_list("1")) {</div><span><div>#Call from Asterisk servers pool, send it to telco using LoadBalancer</div></span><span><div> if(!ds_select_dst("2", "4")) {</div><div> sl_send_reply("500", "Service Unavailable");</div><div> xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n");</div><div> exit;</div><div> }</div><div> }</div><div><br></div><div>}</div></span><div>+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++</div></div><div><br></div><div>Debug log</div><div><br></div><div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request:</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <INVITE></div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <<a href="mailto:sip%3A0730092190@172.22.14.12" target="_blank">sip:0730092190@172.22.14.12</a>></div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0></div><div> 8(1186) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK3c5fb091>; state=16</div><div> 8(1186) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via</div><div> 8(1186) DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg...</div><div> 8(1186) DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts...</div><div> 8(1186) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70</div><div> 8(1186) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=10</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [31]; uri=[<a href="mailto:sip%3A0730092190@172.22.14.12" target="_blank">sip:0730092190@172.22.14.12</a>]</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<<a href="mailto:sip%3A0730092190@172.22.14.12" target="_blank">sip:0730092190@172.22.14.12</a>></div><div>]</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <102> <INVITE></div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=327</div><div> 8(1186) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header</div><div> 8(1186) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=as4decf975</div><div> 8(1186) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29</div><div> 8(1186) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1</div><div> 8(1186) DEBUG: siputils [checks.c:103]: has_totag(): no totag</div><div> 8(1186) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffff</div><div> 8(1186) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=3888, isACK=0</div><div> 8(1186) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed</div><div> 8(1186) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found</div><div> 8(1186) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil)</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==9 && [172.22.14.17] == [127.0.0.1]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [172.22.14.17] == [172.22.14.12]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==9 && [172.22.14.17] == [127.0.0.1]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [172.22.14.17] == [172.22.14.12]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [forward.c:450]: check_self(): check_self: host != me</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==9 && [172.22.14.17] == [127.0.0.1]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [172.22.14.17] == [172.22.14.12]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==9 && [172.22.14.17] == [127.0.0.1]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [172.22.14.17] == [172.22.14.12]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [forward.c:450]: check_self(): check_self: host != me</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==9 && [172.22.14.12] == [127.0.0.1]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [172.22.14.12] == [172.22.14.12]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: dispatcher [dispatch.c:1629]: ds_select_dst(): set [2]</div><div> 8(1186) DEBUG: dispatcher [dispatch.c:1731]: ds_select_dst(): alg hash [0]</div><div> 8(1186) DEBUG: dispatcher [dispatch.c:1772]: ds_select_dst(): selected [4-2/0] <sip:<a href="http://172.28.0.68:5060" target="_blank">172.28.0.68:5060</a>></div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==9 && [172.22.14.12] == [127.0.0.1]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 12==12 && [172.22.14.12] == [172.22.14.12]</div><div> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060</div><div> 8(1186) DEBUG: registrar [lookup.c:158]: lookup(): '0730092190' Not found in usrloc</div><div> 8(1186) DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=2 , global msg id=2 , T on entrance=(nil)</div><div> 8(1186) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=3888, isACK=0</div><div> 8(1186) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed</div><div> 8(1186) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found</div><div> 8(1186) DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG: trans=0xb5d3f20c, callback type 1, id 0 entered</div><div> 8(1186) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: 3d26b7732e22874c5837c971c8ec76cd</div><div> 8(1186) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=0xb5d3f20c</div><div> 8(1186) DEBUG: tm [t_lookup.c:1144]: t_check_msg(): DEBUG: t_check_msg: T already found!</div><div> 8(1186) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(172.22.14.17, 172.22.14.17, 0)</div><div> 8(1186) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called</div><div> 8(1186) DEBUG: tm [t_reply.c:1653]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset</div><div> 8(1186) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb5d3f20c, callback type 512, id 0 entered</div><div> 8(1186) DEBUG: acc [acc_logic.c:571]: tmcb_func(): acc callback called for t(0xb5d3f20c) event type 512, reply code 404</div><div> 8(1186) DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent out. buf=0xb7bb8030: <b>SIP/2.0 404 Not Foun.</b>.., shmem=0xb5d40cdc: SIP/2.0 404 Not Foun</div><div> 8(1186) DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: _reply_light: finished</div><div> 8(1186) DEBUG: sl [sl.c:288]: send_reply(): reply in stateful mode (tm)</div><div><br></div><div>++++++++++++++++++++++++++++++++++++++++++++++++++++++++++</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></div><div><br></div></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Warm Regards,<br>Sandeep Chakravarthi.</div></div></div><div><div>
<br><div class="gmail_quote">On Thu, Jul 30, 2015 at 6:35 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Below is output from the dispatcher table, Set-2 is a pool of asterisk servers to be Load balanced, and Set-1 is the Telco IP.</div><div><span style="font-family:monospace,monospace"><br></span></div><div><span style="background-color:rgb(238,238,238)"><span style="font-family:monospace,monospace">KAMSBC01:~# kamctl dispatcher dump</span><br></span></div><div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)">SET_NO:: 2</font></div><div><font face="monospace, monospace"><b style="background-color:rgb(238,238,238)">SET:: 2 </b></font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> URI:: sip:<a href="http://192.168.0.150:5050" target="_blank">192.168.0.150:5050</a> flags=AP priority=1 attrs=</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> URI:: sip:<a href="http://192.168.0.151:5060" target="_blank">192.168.0.151:5060</a> flags=AP priority=1 attrs=<br></font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> URI:: sip:<a href="http://192.168.0.152:5070" target="_blank">192.168.0.152:5070</a> flags=AP priority=1 attrs=<br></font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> URI:: sip:<a href="http://192.168.0.153:5080" target="_blank">192.168.0.153:5080</a> flags=AP priority=1 attrs=<br></font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> URI:: sip:<a href="http://192.168.0.155:5090" target="_blank">192.168.0.155:5090</a> flags=AP priority=1 attrs=<br></font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"><b>SET:: 1</b></font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> URI:: sip:124.311.201.600:5060 flags=AP priority=1 attrs=</font></div></div><div><br></div><div>Now in my kamailio.cfg in relevant route</div><div><br></div><div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)">if(<a href="http://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#dispatcher.f.ds_is_from_list" target="_blank">ds_is_from_list</a>("1")) {</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)" size="1">#Call from Telco Should goto Asterisk pool in Loadbalanced mode</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> if(!ds_select_dst("2", "4")) {</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> sl_send_reply("500", "Service Unavailable");</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n");</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> exit;</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> }</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)">} else if (ds_is_from_list("2")) {</font></div></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)" size="1">#Call from Asterisk servers pool, send it to telco using LoadBalancer</font></div><div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> if(!ds_select_dst("1", "4")) {</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> sl_send_reply("500", "Service Unavailable");</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n");</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> exit;</font></div><div><font face="monospace, monospace" style="background-color:rgb(238,238,238)"> }</font></div></div><div><span style="background-color:rgb(238,238,238)"><font face="monospace, monospace">}</font><br></span></div><div><br></div><div><br></div><div>So if your Telco has more than 1 IP you can do Load balancing.</div><div><br></div><div>I hope this solves your problem.</div><div><br></div><div><br></div><div>Best Regards,</div><div>Sammy</div><div><br></div><div><br></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jul 30, 2015 at 3:17 AM, Sandeep Chakravarthi <span dir="ltr"><<a href="mailto:ivschakravarthi@gmail.com" target="_blank">ivschakravarthi@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>Can you share the sample code to differentiate the both telco IP and our server IP?</div><div><br></div><div>.</div><div><br></div><div><br></div></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Warm Regards,<br>Sandeep Chakravarthi.</div></div></div><div><div>
<br><div class="gmail_quote">On Tue, Jul 14, 2015 at 10:55 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Sure but if you look into the dispatcher module there is a field called 'setid' or groupid. Use it wisely to differentiate between the Load Balanced asterisk pool and the Telco IP. <div>The dispatcher module is exactly what you should use. You can find out if incoming source IP belongs to a particular set in dispatcher table thus you can tell if call is coming from Telco or from your Asterisks.</div><div>You can select the dispatcher set for load balancing but if we only have one IP in there then it gets all the load.</div><div><br></div><div>BR,</div><div>Sammy</div><div><br></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Jul 14, 2015 at 1:21 PM, Sandeep Chakravarthi <span dir="ltr"><<a href="mailto:ivschakravarthi@gmail.com" target="_blank">ivschakravarthi@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div>Thanks for the immediate reply.</div><div><br></div><div>You are right ,using the dispatcher module , i am able to send the OPTIONS packet to MSC Telco.</div><div><br></div><div>But as i describer in my earlier mail, i am using the same dispatcher module to establish the sip trunk between my My Kamailio server and my Asterisk server.</div><div><br></div><div>There is a table in the database with the name dispatcher.</div><div>Now, in that table i have 2 records</div><div>one is my Telco SIP IP and the other is Asterisk PBX IP.</div><div><br></div><div>But as per my understanding from the google, dispatcher module is used for load balancing between the servers</div><div><br></div><div>Telco SIP server will be sending the calls to Kamailio and Kamailio has to distribute completely to Asterisk server instead of distributing the calls between Telco SIP IP and Asterisk.</div><div><br></div><div><br></div><div>Please help with it.</div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br clear="all"><div><div><div dir="ltr">Warm Regards,<br>Sandeep Chakravarthi.</div></div></div><div><div>
<br><div class="gmail_quote">On Tue, Jul 14, 2015 at 10:28 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div>You're right about using IP Auth in Kamailio. You'll need to use the permissions module. However I believe permissions module wont send the OPTIONS to the MSC SIP Server. For this you may alternatively use the "dispatcher" module.</div><div><br></div><div>Take a look at the sample kamailio.cfg here: <a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a></div><div><br></div><div>Follow the tag WITH_IPAUTH and I'm sure you'll be able to implement it easily.</div><div><br></div><div>BR,</div><div>Sammy</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Jul 14, 2015 at 12:51 PM, Sandeep Chakravarthi <span dir="ltr"><<a href="mailto:ivschakravarthi@gmail.com" target="_blank">ivschakravarthi@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><br>
<div>Hi,</div><div>We have a requirement with one of our telco</div><div>We are using asterisk in our servers and we are planning to implement SIP-I protocol and we choosed kamailio for it.</div><div><br></div><div>In Kamailio website, i came to know that kamailio will be supporting both SIP-I and SIP-T protocols</div><div><br></div><div>Below is what we need and pls confirm whether it is possible or not?</div><div><br></div><div>Asterisk PBX <-------> Kamailio <--------> Telco MSC</div><div><br></div><div><br></div><div>Telco will be forwarding the calls to kamailio on sip-i protocol and kamailio server has to forward the calls to our Asterisk server by converting sip-i to standard sip protocol</div><div><br></div><div>Similiarly Asterisk will be initiating sip call to kamailio server and kamailio server should convert it into SIP-I and should forward the call to Telco MSC</div><div><br></div><div><br></div><div>1. I am able to establish the SIP trunk [sending OPTIONS from asterisk and kamailio acknowledges with 200 OK] between Asterisk and Kamailio using dispatcher module in kamailio and sip.conf in asterisk.</div><div><br></div><div>How to establish the SIP trunk between kamailio and telco MSC?</div><div>[Generally MSC will act as SIP server and kamalio should send OPTIONS packet and MSC will acknowledges with 200 OK]<br></div><div><br></div><div><br></div><div>My telco MSC has only provided me the MSC SIP IP and there were no username/passwords provided.</div><div>Means i need to use IP based authentication for the SIP Trunk establishment.</div><div><br></div><div>In Kamailio how to achieve it?</div><div><br></div><div>Please help and any suggestions/feedback will be highly appreciated and thankful</div><div><br></div><div><br></div><div>Regards,</div><div>Sandeep</div></div>
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