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
>
>
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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
