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 > >
Trace1-Aug10th.pcap
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_______________________________________________ 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