[SR-Users] kamailio as SIP Agent
Sandeep Chakravarthi
ivschakravarthi at gmail.com
Wed Aug 12 20:08:03 CEST 2015
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 <govoiper at 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 $rU + "@
> 172.22.12.100:5060" where this is IP of MSC side.
> 2 - Wireshark guys could've said it SIP-3 - point is it doesnt matter at
> this point since you know your MSC is replying back and talking to you.
>
>
>
> On Tue, Aug 11, 2015 at 1:16 PM, Sandeep Chakravarthi <
> ivschakravarthi at gmail.com> wrote:
>
>> Yes, You are right and done the changes as you suggested.
>>
>> Kamailio server is forwarding the call to MSC. But two issues are there.
>> 1 .In the INVITE packet which is being sent from kamailio server to MSC,
>> it is coming Request-Line: INVITE sip:0730092190@*172.22.14.12*
>> That is my kamailio server IP and it should be MSC IP(172.28.0.68) and
>> as of now call is failing as MSC is sending 404 error.
>> 2. Other issue is , in the pcap file it is coming SIP/SDP as protocol and
>> it is not coming SIP-I.
>>
>> Please find the latest attached pcap.
>>
>> Regards,
>> Sandeep
>>
>>
>> Warm Regards,
>> Sandeep Chakravarthi.
>>
>> On Tue, Aug 11, 2015 at 9:47 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Thats because your configuration file is not sending packet out (RELAY)
>>> to MSC instead it is only doing a Loadbalancer / destination lookup in
>>> TOASTERISK route and comes out of it, processes the following routes in
>>> order
>>> route(SIPOUT);
>>> route(PRESENCE);
>>> route(REGISTRAR);
>>> route(PSTN);
>>> route(LOCATION);
>>>
>>> Where finally in LOCATION route it tries to find the destination user
>>> 0730092190 online locally on Kamailio, which it can't find and says 404 Not
>>> Found.
>>>
>>> You should modify your TOASTERISK route as follow:
>>>
>>> route[TOASTERISK] {
>>> if(ds_is_from_list("2")) {
>>> #Call from Telco Should goto Asterisk pool in Loadbalanced mode
>>> if(!ds_select_dst("1", "4")) {
>>> sl_send_reply("500", "Service Unavailable");
>>> xlog("L_INFO","[$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 <
>>> ivschakravarthi at 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 <govoiper at 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 <
>>>>> ivschakravarthi at 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 at 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 at 172.22.14.12]
>>>>>> 8(1186) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field():
>>>>>> DEBUG: to body [<sip:0730092190 at 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 <govoiper at 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 <
>>>>>>> ivschakravarthi at 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 <govoiper at 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 <
>>>>>>>>> ivschakravarthi at 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 <govoiper at 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 <
>>>>>>>>>>> ivschakravarthi at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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