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
>
>
_______________________________________________
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