Re: [SR-Users] kamailio as SIP Agent

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 < ivschakravarthi(a)gmail.com&gt; 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 &lt;govoiper(a)gmail.com&gt; 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 < >> ivschakravarthi(a)gmail.com&gt; 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 &lt;govoiper(a)gmail.com&gt; 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 < >>>> ivschakravarthi(a)gmail.com&gt; 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 &lt;govoiper(a)gmail.com&gt; >>>>> 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 < >>>>>> ivschakravarthi(a)gmail.com&gt; 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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users