Hi, Can you please tell me under which route i need keep those values. Whether in the RELAY route or ASTERISK rotue?
Regards, Sandeep Warm Regards, Sandeep Chakravarthi. On Tue, Aug 11, 2015 at 11:03 PM, SamyGo <govoi...@gmail.com> wrote: > 1 - Take a look at the Kamailio transformations and psuedo-variable page. > change the $td to the IP of the MSC; modify the $ru as $rU + "@ > 172.22.12.100:5060" where this is IP of MSC side. > 2 - Wireshark guys could've said it SIP-3 - point is it doesnt matter at > this point since you know your MSC is replying back and talking to you. > > > > On Tue, Aug 11, 2015 at 1:16 PM, Sandeep Chakravarthi < > ivschakravar...@gmail.com> wrote: > >> Yes, You are right and done the changes as you suggested. >> >> Kamailio server is forwarding the call to MSC. But two issues are there. >> 1 .In the INVITE packet which is being sent from kamailio server to MSC, >> it is coming Request-Line: INVITE sip:0730092190@*172.22.14.12* >> That is my kamailio server IP and it should be MSC IP(172.28.0.68) and >> as of now call is failing as MSC is sending 404 error. >> 2. Other issue is , in the pcap file it is coming SIP/SDP as protocol and >> it is not coming SIP-I. >> >> Please find the latest attached pcap. >> >> Regards, >> Sandeep >> >> >> Warm Regards, >> Sandeep Chakravarthi. >> >> On Tue, Aug 11, 2015 at 9:47 PM, SamyGo <govoi...@gmail.com> wrote: >> >>> Thats because your configuration file is not sending packet out (RELAY) >>> to MSC instead it is only doing a Loadbalancer / destination lookup in >>> TOASTERISK route and comes out of it, processes the following routes in >>> order >>> route(SIPOUT); >>> route(PRESENCE); >>> route(REGISTRAR); >>> route(PSTN); >>> route(LOCATION); >>> >>> Where finally in LOCATION route it tries to find the destination user >>> 0730092190 online locally on Kamailio, which it can't find and says 404 Not >>> Found. >>> >>> You should modify your TOASTERISK route as follow: >>> >>> route[TOASTERISK] { >>> if(ds_is_from_list("2")) { >>> #Call from Telco Should goto Asterisk pool in Loadbalanced mode >>> if(!ds_select_dst("1", "4")) { >>> sl_send_reply("500", "Service Unavailable"); >>> xlog("L_INFO","[$fU@$si:$sp]{$rm} No >>> destinations available for $rd \n"); >>> exit; >>> } >>> route(RELAY); >>> }if(ds_is_from_list("1")) { >>> #Call from Asterisk servers pool, send it to telco using LoadBalancer >>> if(!ds_select_dst("2", "4")) { >>> sl_send_reply("500", "Service Unavailable"); >>> xlog("L_INFO","[$fU@$si:$sp]{$rm} No >>> destinations available for $rd \n"); >>> exit; >>> } >>> route(RELAY); >>> } >>> >>> } >>> >>> >>> This will immediately route the packet out towards the new $du after the >>> loadbalancer function ds_select_dst(...) >>> >>> >>> On Tue, Aug 11, 2015 at 10:48 AM, Sandeep Chakravarthi < >>> ivschakravar...@gmail.com> wrote: >>> >>>> Hi, >>>> Kamailio is sending 404 Response and its not MSC. >>>> 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. >>>> I am attaching the pcap one more time for your reference. >>>> >>>> In my pcap, below are the server details >>>> >>>> 172.22.14.12 - Kamailio server >>>> 172.22.14.17 - Asterisk server >>>> 172.22.0.68 - MSC >>>> >>>> >>>> Regards, >>>> Sandeep >>>> >>>> Warm Regards, >>>> Sandeep Chakravarthi. >>>> >>>> On Tue, Aug 11, 2015 at 7:10 PM, SamyGo <govoi...@gmail.com> wrote: >>>> >>>>> Hi Sandeep, >>>>> 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 ? >>>>> >>>>> >>>>> On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi < >>>>> ivschakravar...@gmail.com> wrote: >>>>> >>>>>> Hi , >>>>>> Sorry for the delayed reply. >>>>>> 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. >>>>>> Please find the attached pcap details for your reference. >>>>>> Below is my kamailio debug log and kamailio.cfg file. >>>>>> Please check the pcap and below cfg file and log file and let me know >>>>>> whether to change anything in cfg file or not. >>>>>> >>>>>> ++++++++++++++++++++++++++++++++++++++++++++++++ >>>>>> >>>>>> >>>>>> request_route { >>>>>> >>>>>> # per request initial checks >>>>>> route(REQINIT); >>>>>> >>>>>> # NAT detection >>>>>> route(NATDETECT); >>>>>> >>>>>> # CANCEL processing >>>>>> if (is_method("CANCEL")) >>>>>> { >>>>>> if (t_check_trans()) { >>>>>> route(RELAY); >>>>>> } >>>>>> exit; >>>>>> } >>>>>> >>>>>> # handle requests within SIP dialogs >>>>>> route(WITHINDLG); >>>>>> >>>>>> ### only initial requests (no To tag) >>>>>> >>>>>> t_check_trans(); >>>>>> >>>>>> # authentication >>>>>> route(AUTH); >>>>>> >>>>>> >>>>>> # record routing for dialog forming requests (in case they >>>>>> are routed) >>>>>> # - remove preloaded route headers >>>>>> remove_hf("Route"); >>>>>> if (is_method("INVITE|SUBSCRIBE")) >>>>>> record_route(); >>>>>> >>>>>> # account only INVITEs >>>>>> if (is_method("INVITE")) >>>>>> { >>>>>> setflag(FLT_ACC); # do accounting >>>>>> } >>>>>> route(TOASTERISK); >>>>>> >>>>>> # dispatch requests to foreign domains >>>>>> route(SIPOUT); >>>>>> >>>>>> ### requests for my local domains >>>>>> >>>>>> # handle presence related requests >>>>>> route(PRESENCE); >>>>>> >>>>>> # handle registrations >>>>>> route(REGISTRAR); >>>>>> >>>>>> if ($rU==$null) >>>>>> { >>>>>> # request with no Username in RURI >>>>>> sl_send_reply("484","Address Incomplete"); >>>>>> exit; >>>>>> } >>>>>> >>>>>> # dispatch destinations to PSTN >>>>>> route(PSTN); >>>>>> # user location service >>>>>> route(LOCATION); >>>>>> } >>>>>> >>>>>> route[TOASTERISK] { >>>>>> if(ds_is_from_list("2")) { >>>>>> #Call from Telco Should goto Asterisk pool in Loadbalanced mode >>>>>> if(!ds_select_dst("1", "4")) { >>>>>> sl_send_reply("500", "Service Unavailable"); >>>>>> xlog("L_INFO","[$fU@$si:$sp]{$rm} No >>>>>> destinations available for $rd \n"); >>>>>> exit; >>>>>> } >>>>>> }if(ds_is_from_list("1")) { >>>>>> #Call from Asterisk servers pool, send it to telco using LoadBalancer >>>>>> if(!ds_select_dst("2", "4")) { >>>>>> sl_send_reply("500", "Service Unavailable"); >>>>>> xlog("L_INFO","[$fU@$si:$sp]{$rm} No >>>>>> destinations available for $rd \n"); >>>>>> exit; >>>>>> } >>>>>> } >>>>>> >>>>>> } >>>>>> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >>>>>> >>>>>> Debug log >>>>>> >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP >>>>>> Request: >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): >>>>>> method: <INVITE> >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: >>>>>> <sip:0730092190@172.22.14.12> >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): >>>>>> version: <SIP/2.0> >>>>>> 8(1186) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): >>>>>> Found param type 232, <branch> = <z9hG4bK3c5fb091>; state=16 >>>>>> 8(1186) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of >>>>>> header reached, state=5 >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): >>>>>> parse_headers: Via found, flags=2 >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): >>>>>> parse_headers: this is the first via >>>>>> 8(1186) DEBUG: <core> [receive.c:152]: receive_msg(): After >>>>>> parse_msg... >>>>>> 8(1186) DEBUG: <core> [receive.c:193]: receive_msg(): preparing to >>>>>> run routing scripts... >>>>>> 8(1186) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = >>>>>> 70 >>>>>> 8(1186) DEBUG: <core> [parser/parse_addr_spec.c:898]: >>>>>> parse_addr_spec(): end of header reached, state=10 >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): >>>>>> DEBUG: get_hdr_field: <To> [31]; uri=[sip:0730092190@172.22.14.12] >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): >>>>>> DEBUG: to body [<sip:0730092190@172.22.14.12> >>>>>> ] >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): >>>>>> get_hdr_field: cseq <CSeq>: <102> <INVITE> >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): >>>>>> DEBUG: get_hdr_body : content_length=327 >>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): >>>>>> found end of header >>>>>> 8(1186) DEBUG: <core> [parser/parse_addr_spec.c:176]: >>>>>> parse_to_param(): DEBUG: add_param: tag=as4decf975 >>>>>> 8(1186) DEBUG: <core> [parser/parse_addr_spec.c:898]: >>>>>> parse_addr_spec(): end of header reached, state=29 >>>>>> 8(1186) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity >>>>>> checks result: 1 >>>>>> 8(1186) DEBUG: siputils [checks.c:103]: has_totag(): no totag >>>>>> 8(1186) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: >>>>>> t_check_msg: msg id=2 global id=1 T start=0xffffffff >>>>>> 8(1186) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): >>>>>> t_lookup_request: start searching: hash=3888, isACK=0 >>>>>> 8(1186) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 >>>>>> transaction matching failed >>>>>> 8(1186) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: >>>>>> t_lookup_request: no transaction found >>>>>> 8(1186) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: >>>>>> t_check_msg: msg id=2 global id=2 T end=(nil) >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 8(1186) DEBUG: <core> [forward.c:450]: check_self(): check_self: >>>>>> host != me >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 8(1186) DEBUG: <core> [forward.c:450]: check_self(): check_self: >>>>>> host != me >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 8(1186) DEBUG: dispatcher [dispatch.c:1629]: ds_select_dst(): set [2] >>>>>> 8(1186) DEBUG: dispatcher [dispatch.c:1731]: ds_select_dst(): alg >>>>>> hash [0] >>>>>> 8(1186) DEBUG: dispatcher [dispatch.c:1772]: ds_select_dst(): >>>>>> selected [4-2/0] <sip:172.28.0.68:5060> >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 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] >>>>>> 8(1186) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): >>>>>> grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 >>>>>> 8(1186) DEBUG: registrar [lookup.c:158]: lookup(): '0730092190' Not >>>>>> found in usrloc >>>>>> 8(1186) DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: >>>>>> msg id=2 , global msg id=2 , T on entrance=(nil) >>>>>> 8(1186) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): >>>>>> t_lookup_request: start searching: hash=3888, isACK=0 >>>>>> 8(1186) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 >>>>>> transaction matching failed >>>>>> 8(1186) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: >>>>>> t_lookup_request: no transaction found >>>>>> 8(1186) DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): >>>>>> DBG: trans=0xb5d3f20c, callback type 1, id 0 entered >>>>>> 8(1186) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 >>>>>> calculated: 3d26b7732e22874c5837c971c8ec76cd >>>>>> 8(1186) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: >>>>>> t_check_msg: msg id=2 global id=2 T start=0xb5d3f20c >>>>>> 8(1186) DEBUG: tm [t_lookup.c:1144]: t_check_msg(): DEBUG: >>>>>> t_check_msg: T already found! >>>>>> 8(1186) DEBUG: <core> [msg_translator.c:205]: check_via_address(): >>>>>> check_via_address(172.22.14.17, 172.22.14.17, 0) >>>>>> 8(1186) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): >>>>>> WARNING:vqm_resize: resize(0) called >>>>>> 8(1186) DEBUG: tm [t_reply.c:1653]: cleanup_uac_timers(): DEBUG: >>>>>> cleanup_uac_timers: RETR/FR timers reset >>>>>> 8(1186) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): >>>>>> DBG: trans=0xb5d3f20c, callback type 512, id 0 entered >>>>>> 8(1186) DEBUG: acc [acc_logic.c:571]: tmcb_func(): acc callback >>>>>> called for t(0xb5d3f20c) event type 512, reply code 404 >>>>>> 8(1186) DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent >>>>>> out. buf=0xb7bb8030: *SIP/2.0 404 Not Foun.*.., shmem=0xb5d40cdc: >>>>>> SIP/2.0 404 Not Foun >>>>>> 8(1186) DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: >>>>>> _reply_light: finished >>>>>> 8(1186) DEBUG: sl [sl.c:288]: send_reply(): reply in stateful mode >>>>>> (tm) >>>>>> >>>>>> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Warm Regards, >>>>>> Sandeep Chakravarthi. >>>>>> >>>>>> On Thu, Jul 30, 2015 at 6:35 PM, SamyGo <govoi...@gmail.com> wrote: >>>>>> >>>>>>> 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. >>>>>>> >>>>>>> KAMSBC01:~# kamctl dispatcher dump >>>>>>> SET_NO:: 2 >>>>>>> *SET:: 2 * >>>>>>> URI:: sip:192.168.0.150:5050 flags=AP priority=1 attrs= >>>>>>> URI:: sip:192.168.0.151:5060 flags=AP priority=1 attrs= >>>>>>> URI:: sip:192.168.0.152:5070 flags=AP priority=1 attrs= >>>>>>> URI:: sip:192.168.0.153:5080 flags=AP priority=1 attrs= >>>>>>> URI:: sip:192.168.0.155:5090 flags=AP priority=1 attrs= >>>>>>> *SET:: 1* >>>>>>> URI:: sip:124.311.201.600:5060 flags=AP priority=1 attrs= >>>>>>> >>>>>>> Now in my kamailio.cfg in relevant route >>>>>>> >>>>>>> if(ds_is_from_list >>>>>>> <http://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#dispatcher.f.ds_is_from_list>("1")) >>>>>>> { >>>>>>> #Call from Telco Should goto Asterisk pool in Loadbalanced mode >>>>>>> if(!ds_select_dst("2", "4")) { >>>>>>> sl_send_reply("500", "Service Unavailable"); >>>>>>> xlog("L_INFO","[$fU@$si:$sp]{$rm} No >>>>>>> destinations available for $rd \n"); >>>>>>> exit; >>>>>>> } >>>>>>> } else if (ds_is_from_list("2")) { >>>>>>> #Call from Asterisk servers pool, send it to telco using LoadBalancer >>>>>>> if(!ds_select_dst("1", "4")) { >>>>>>> sl_send_reply("500", "Service Unavailable"); >>>>>>> xlog("L_INFO","[$fU@$si:$sp]{$rm} No >>>>>>> destinations available for $rd \n"); >>>>>>> exit; >>>>>>> } >>>>>>> } >>>>>>> >>>>>>> >>>>>>> So if your Telco has more than 1 IP you can do Load balancing. >>>>>>> >>>>>>> I hope this solves your problem. >>>>>>> >>>>>>> >>>>>>> Best Regards, >>>>>>> Sammy >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, Jul 30, 2015 at 3:17 AM, Sandeep Chakravarthi < >>>>>>> ivschakravar...@gmail.com> wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Can you share the sample code to differentiate the both telco IP >>>>>>>> and our server IP? >>>>>>>> >>>>>>>> . >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Warm Regards, >>>>>>>> Sandeep Chakravarthi. >>>>>>>> >>>>>>>> On Tue, Jul 14, 2015 at 10:55 PM, SamyGo <govoi...@gmail.com> >>>>>>>> wrote: >>>>>>>> >>>>>>>>> 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. >>>>>>>>> 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. >>>>>>>>> You can select the dispatcher set for load balancing but if we >>>>>>>>> only have one IP in there then it gets all the load. >>>>>>>>> >>>>>>>>> BR, >>>>>>>>> Sammy >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jul 14, 2015 at 1:21 PM, Sandeep Chakravarthi < >>>>>>>>> ivschakravar...@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Hi, >>>>>>>>>> Thanks for the immediate reply. >>>>>>>>>> >>>>>>>>>> You are right ,using the dispatcher module , i am able to send >>>>>>>>>> the OPTIONS packet to MSC Telco. >>>>>>>>>> >>>>>>>>>> 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. >>>>>>>>>> >>>>>>>>>> There is a table in the database with the name dispatcher. >>>>>>>>>> Now, in that table i have 2 records >>>>>>>>>> one is my Telco SIP IP and the other is Asterisk PBX IP. >>>>>>>>>> >>>>>>>>>> But as per my understanding from the google, dispatcher module is >>>>>>>>>> used for load balancing between the servers >>>>>>>>>> >>>>>>>>>> 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. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Please help with it. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Warm Regards, >>>>>>>>>> Sandeep Chakravarthi. >>>>>>>>>> >>>>>>>>>> On Tue, Jul 14, 2015 at 10:28 PM, SamyGo <govoi...@gmail.com> >>>>>>>>>> wrote: >>>>>>>>>> >>>>>>>>>>> Hi, >>>>>>>>>>> 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. >>>>>>>>>>> >>>>>>>>>>> Take a look at the sample kamailio.cfg here: >>>>>>>>>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb >>>>>>>>>>> >>>>>>>>>>> Follow the tag WITH_IPAUTH and I'm sure you'll be able to >>>>>>>>>>> implement it easily. >>>>>>>>>>> >>>>>>>>>>> BR, >>>>>>>>>>> Sammy >>>>>>>>>>> >>>>>>>>>>> On Tue, Jul 14, 2015 at 12:51 PM, Sandeep Chakravarthi < >>>>>>>>>>> ivschakravar...@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Hi, >>>>>>>>>>>> We have a requirement with one of our telco >>>>>>>>>>>> We are using asterisk in our servers and we are planning to >>>>>>>>>>>> implement SIP-I protocol and we choosed kamailio for it. >>>>>>>>>>>> >>>>>>>>>>>> In Kamailio website, i came to know that kamailio will be >>>>>>>>>>>> supporting both SIP-I and SIP-T protocols >>>>>>>>>>>> >>>>>>>>>>>> Below is what we need and pls confirm whether it is possible or >>>>>>>>>>>> not? >>>>>>>>>>>> >>>>>>>>>>>> Asterisk PBX <-------> Kamailio <--------> Telco MSC >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 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. >>>>>>>>>>>> >>>>>>>>>>>> How to establish the SIP trunk between kamailio and telco MSC? >>>>>>>>>>>> [Generally MSC will act as SIP server and kamalio should send >>>>>>>>>>>> OPTIONS packet and MSC will acknowledges with 200 OK] >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> My telco MSC has only provided me the MSC SIP IP and there were >>>>>>>>>>>> no username/passwords provided. >>>>>>>>>>>> Means i need to use IP based authentication for the SIP Trunk >>>>>>>>>>>> establishment. >>>>>>>>>>>> >>>>>>>>>>>> In Kamailio how to achieve it? >>>>>>>>>>>> >>>>>>>>>>>> Please help and any suggestions/feedback will be highly >>>>>>>>>>>> appreciated and thankful >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> Sandeep >>>>>>>>>>>> >>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>>>>>> mailing list >>>>>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>>>>> mailing list >>>>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>>>>>>>>> mailing list >>>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>>>> list >>>>>>>>> sr-users@lists.sip-router.org >>>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>>> list >>>>>>>> sr-users@lists.sip-router.org >>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> sr-users@lists.sip-router.org >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>> list >>>>>> sr-users@lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users@lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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