[SR-Users] kamailio as SIP Agent
Sandeep Chakravarthi
ivschakravarthi at gmail.com
Tue Aug 11 16:48:16 CEST 2015
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
>
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