[SR-Users] kamailio as SIP Agent
SamyGo
govoiper at gmail.com
Tue Aug 11 15:40:27 CEST 2015
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
>
>
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