- sr-users - kamailio.org

How to get modified SDP by RTPProxy
by Albert Petit 31 May '17

31 May '17

Hi list, We are using RTPProxy as media proxy. When doing a SIPcall we would need to know the final ports used by RTPProxy. Hence I would need to access the rewritten SDP of the 200 OK send by our application server. Unfortunately we are not achieving to get that rewritten SDP. I tried calling sdp_print("1") using module sdpops just after calling rtpproxy_offer or rtpproxy_answer in kamailio script. (According the doc those methods do: "Rewrites SDP body to ensure that media is passed through an RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK".) However the sdp we print just after calling that is still having the original ports set by the AS but not the ports used in the rtpproxy or that we will send to the client in the rewritten SDP What are we doing wrong? Any hint how we can get the rewritten SDP in the 200 OK answer we send to caller side?

2 1

Should a proxy change Contact or From domains?
by Phil Lavin 31 May '17

31 May '17

Hi all, Theoretical question about the role of a Kamailio proxy which is also a registration server for UAs. Assuming a B2BUA on IP 1.2.3.4 behind the Kamailio proxy. Kamailio receives an INVITE from this B2BUA with From and Contact as something like sip:4321@1.2.3.4 When Kamailio routes the INVITE to that UA, is it Kamailio’s responsibility to change the domain part of the From and/or Contact to itself (e.g. sip:4321@proxy.mytld.com)? The question has arisen from Polycom missed call “call back” behaviour. When issuing a call back from a Polycom, following a missed call, it sends the call directly to the full URI in the INVITE’s From (or maybe Contact… I’ve not fully tested yet). As such the call is sent to the B2BUA, which UAs do not have access to, and thus times out. I suspect here the answer is “no”, and the Polycom must be configured to force all outbound calls through the proxy… but thoughts welcome. Cheers Phil

2 1

Is it possible to replace Request-URI in the REGISTERs with the result of dispatcher?
by 刘鲲 31 May '17

31 May '17

Hello, I build three kamailioes, one acts as a pass-through proxy as well as load-balancing, the two registrars. The kamailio_1 decides to which registrar the REGISTER should forward by a dispatcher based on db_text. ----->kamailio_2(IP2, registrar) SoftPhone--->kamailio_1(IP1, pass-through proxy)--| | ----->kamailio_3(IP3, registrar) | dispatcher(db_text) I expect the kamailio_1 forward the REGISTERs to kamailio_2/kamailio_3, and kamailio_2/kamailio_3 respond with 200 OK. But in fact, the registrars send the REGISTER back to kamailio_1. I think the reason is that the host part of Request-URI in REGISTER from SoftPhone is IP1, and kamailio_1 does not rewrite it as a pass-through proxy, so kamailio_2/kamailio_3 will find the host in Request-URI is not mine, thus route the REGISTER back to kamailio_1 according Request-URI. The setting of SoftPhone cannot be changed, so is it possible to : 1) replace host part of Request-URI in the REGISTERs with the result of dispatcher in kamailio_1 , or 2) force kamailio_2/kamailio_3 to respond rather than forward? Thanks!

2 1

push notification
by volga629＠networklab.ca 30 May '17

30 May '17

Hello Everyone, Trying understand how push notification works based on Daniel example in 2014 Code in question: if(lua_runstring("do_push([[$hdr(X-VxTo)]], [[$tU]], [[$hdr(X-VxFrom)]], [[$fU]], [[$ci]])")<0) { send_reply("501", "No link to destination"); exit; } I understand how string should look like, but lua give this error ERROR: app_lua [app_lua_api.c:607]: app_lua_runstring(): lua loading state not initialized (call: do_push([[<null>]], [[4310]], [[<null>]], [[4311]], [[dd4cb9b2-bf21-1235-05b5-5254003e39bb]])) Is possible clarify how this part should work ? volga629

2 5

kamctl v5 issues sending rpc raw commands
by José Seabra 30 May '17

30 May '17

Hello there, I'm trying to use the following command: *kamctl rpc get_statistics usrloc::registered_users* and i'm getting the following errors: { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 12006 } ERROR: jsonrpcs [jsonrpcs_mod.c:1323]: jsonrpc_exec_ex(): method callback not found [get_statistics] Question here is: This rpc raw command isn't available on new kamctl v5? or i have to call this in a different way? Thank you Regards José Seabra

3 4

How to disable proxy function of kamailio
by sgy 30 May '17

30 May '17

Hi anyone, Kamailio server receive a register packet, but IP addr of request line is not same with addr of kamailio server ,then ther server will transfer it out. Now I want kamailio server process it as it receive a register itsself. So how do I close proxy function of kamailio server. For Example: Receive packet: ======================================== REGISTER sip:113.80.5.159:9198 SIP/2.0 kamailio server: 222.222.1.2 -- Best Regards, Shengy

3 2

Kamailio start failed
by sgy 29 May '17

29 May '17

Hi anyone, I can't start kamailio, Please help me Log: 1. ~# kamctl start database engine 'MYSQL' loaded Control engine 'RPCFIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio/kamailio.pid does not exist -- Kamailio start failed 2. :~# kamailio -M 8 -E -e -dd 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#24 #1/2: 3(3)/ (nil) 0(1685) DEBUG: <core> [core/sr_module.c:694]: find_mod_export_record(): find_export_record: found <t_is_canceled> in module tm [/usr/local/lib64/kamailio/modules/tm.so] 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#2 #0/2: 3(3)/ 0x1 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#2 #1/2: 3(3)/ 0x1 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #0/3: 22(16)/ 0x7f634f00f780 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #1/3: 8(8)/ 0x7f634f00feb0 0(1685) DEBUG: <core> [core/route_struct.c:129]: mk_action(): ACTION_#16 #2/3: 0(0)/ 0x7f6300000000 0(1685) DEBUG: <core> [core/route.c:129]: route_add(): mapping routing block (0xafc160)[MANAGE_FAILURE] to 1 ERROR: bad config file (1 errors) 0(1685) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- Best Regards, Shengy

3 7

Fwd: questions about FL_RPL flag in dmq_usrloc
by 张顺通 27 May '17

27 May '17

In dmq_usrloc, a ucontact_t with FL_RPL flag will not send it's modify to other kamailio. For example, I have one edge and three kamailio(for register, name A B C). And First SIP REGISTER send to edge, and edge forward to register kamailio A, then kamailio B & C have FL_RPL flag, kamailio A do not have FL_RPL flag. In this situation, If kamailio A is crash, the second SIP REGISTER(same call-id and tag) forward to kamailio B, kamailio B will not replicate reg info to C, because kamailio B have FL_RPL flag. What is the suggestion in this case? Thanks

3 6

mod.stats rpc call crashes
by Andy 26 May '17

26 May '17

Hi, I see kamailio crashing on a call to possibly (mod.stats rpc call), @function __strcmp_sse42 probably is referencing null, which is passed mname and pointer to mname. backtrace: thanks a lot. -- View this message in context: http://sip-router.1086192.n5.nabble.com/mod-stats-rpc-call-crashes-tp158967… Sent from the Users mailing list archive at Nabble.com.

2 4

question about cseq handling
by 张顺通 26 May '17

26 May '17

In urecord.c:get_ucontact_by_instance, REGISTER without instance-id will check cseq number(must be higher), REGISTER with instance-id will not check cseq number. what is the difference?why not check cseq in REGISTER with instance-id? Thanks

2 1

Some Errors
by Douglas Girard 25 May '17

25 May '17

Hello, I am using a fresh install of centos 7, homer (using install script) and kamailio (installed with homer script). After modifying kamailio.cfg file with correct DB username, password and Server IP, i am able to start and stop kamailio service. Now my issue is that I have begun to duplicate SIP messages from my PBX, but am receiving these and other errors from kamailio: May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Data too long for column 'msg' at row 1 (1406) May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query and: May 25 12:44:42 localhost.localdomain homer[14456]: ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP message failed ( 10.3.3.24:51553/1) May 25 12:45:12 localhost.localdomain homer[14460]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Any suggestions? Thanks!

1 0

Dialog module timeout behaviours
by Alex Balashov 25 May '17

25 May '17

Hello, I'm running 4.3.5:1b0c0a and have run into two distinct, bizarre issues with the dialog module that I am unable to explain. In all cases, I am using $dlg_ctx(send_bye) = 1, and passing in a dialog timeout of 7200 sec via AVP (timeout_avp modparam). 1) Local BYEs are sent after either ~60 or ~120 sec -- This happens on a very small number of answered calls for no apparent reason. I have spent days poring through the traces trying to figure out if there's anything eccentric about how the messages, but can't find anything out of the ordinary except (a) 1 retransmission of 200 OK to INVITE due to not receiving ACK fast enough due to end-to-end latency, and (b) consequently, retransmission of ACK. My understanding is that the dialog module does not prophylactically end calls as a response to strange or improper messaging, but only (a) when the dialog timeout is reached or (b) keepalive OPTION pings (if enabled) fail. Keepalive OPTIONS pings are not enabled in these scenarios, and are not sent in the traces. (2) Local BYEs sent in response to race conditions that result in strange dialog state transitions -- There are some calls that look something like: A. Caller INVITEs callee; B. Callee sends progress; C. Callee sends 200 OK answer and immediately sends BYE (e.g. 1 ms later); D. Kamailio receives these nearly simultaneously and forwards the BYE first; E. Caller sends 487 Request Terminated response (processing BYE as if it were CANCEL); F. Kamailio forwards 200 OK back to caller, who ignores it. G. Callee retransmits 200 OK through Kamailio. A few minutes later, Kamailio sends BYE in both directions even though dialog should have been removed from state tracking long ago because of the dispositive 487 reply. ... In particular, I'm curious as to whether there have been any fixes to deal with scenarios like #2 since 4.3.5. I wasn't able to spot any obvious ones in the commit history for the module. I'm also at a complete loss to explain #1. I have traces for all of these scenarios, but would prefer to send privately. Many thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 4

in-dialog contact update before dialog being confirmed
by Jayesh Nambiar 25 May '17

25 May '17

Hello, In a user-user based calling on mobile apps, when the call is connected and the user moves from a Wifi to Mobile data (network change); the device sends a Re-Invite with new contact information and subsequent messages go to appropriate changed contact address. But when the dialog is in say ringing state; and the caller moves from one network to another; what would be an appropriate SIP method to trigger the contact update. My problem is generating a Re-INVITE or an UPDATE before the call is connected is very difficult. What I tried is sending an in-dialog INFO message with new contact address; which gets relayed to the called party and the response for INFO is also properly received on the new contact address. But the subsequent response of the INVITE like a 486 or 200 goes to the original contact only. What would be the best way to update the contact address of a client if I'm maintaining all the dialogs in kamailio. Thanks, - Jayesh

3 4

Dispatcher - no destination sets
by Jean Cérien 24 May '17

24 May '17

Hello I am working with Kamaiio 5.0 on the dispatcher, with the default dispatcher.cfg file ( https://github.com/kamailio/kamailio/blob/master/src/modules/dispatcher/doc… ) I've added via mysql a gateway: mysql> select * from dispatcher \G; *************************** 1. row *************************** id: 1 setid: 1 destination: 192.168.2.20 flags: 0 priority: 0 attrs: description: local gw 1 row in set (0.00 sec) Kamctl sees it correctly: kamctl dispatcher show dispatcher gateways +----+-------+--------------+-------+----------+-------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+--------------+-------+----------+-------+-------------+ | 1 | 1 | 192.168.2.20 | 0 | 0 | | local gw | +----+-------+--------------+-------+----------+-------+-------------+ Here is the piece of cfg: if(ds_list_exist(1)) { xlog("L_INFO", "----------- data in set 1"); } else { xlog("L_INFO", "----------- NO data in set 1"); } if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } Yet, when an INVITE arrives, I get the following errors: DEBUG: dispatcher [dispatcher.c:876]: w_ds_list_exist(): --- Looking for dispatcher set 1 DEBUG: dispatcher [dispatch.c:1314]: ds_list_exist(): -- Looking for set 1 INFO: dispatcher [dispatch.c:1320]: ds_list_exist(): destination set [1] not found INFO: <script>: ----------- NO data in set 1 ERROR: dispatcher [dispatch.c:1679]: ds_select_dst_limit(): no destination sets same result wether I state the dataset as 1 or "1" - Mysql table column type is int(11). What could be wrong ? Thanks for the help J.

3 5

Kamailio 4.4 crashing
by Pete Kelly 24 May '17

24 May '17

Hi I have Kamailio 4.4 proxying a "normal" call and am finding that it is crashing consistently when it tries to proxy the 200OK from a BYE request when dealing with a certain endpoint. The SIP looks very normal to me and I've managed to obtain a trace, full debug logs and a core dump of the issue. I can provide full copies of the files but the interesting parts from the logfile are here: https://pastebin.com/QkwVMxJj bt output from the coredump is here: https://pastebin.com/auNRaeaU Can anyone please advise?

2 1

can kamailio+rtpproxy traversal any kind of nat
by 赵国杰 24 May '17

24 May '17

Hi everyone, Right now I'm running kamailio and rtpproxy on a server behind nat with a public ip. I enabled the advertise address. If i registered two clients via my home wifi or 3g/4g network, then everything's fine. However, if i start the two clients via my office wifi, two clients can calll each other but with no audio. Obviously, the rtp media session has trouble traversal the office LAN nat. I have no idea what went wrong. Anyone can point me the right direction? Sincerely, jesse

1 0

tsilo module enhancement
by Jayesh Nambiar 24 May '17

24 May '17

Hello, I see the tsilo module works only with registered contacts using location table. Is there a possibility that the ts_append function can actually take the destination URI as an argument and append branch to respective destination? Basically I want to avoid maintaining location table. In a typical mobile use case, where the incoming call is suspended and a push notification is generated to the called party; the called party generates a REGISTER request with expires header as 0; kamailio just extracts the contact header and sends the call. Problem comes in when the same user receives a PUSH on another device; I cannot figure a neat way to append the same call on the new device. Is there a better alternative to achieve this without looking up the location table. Thanks for any help. - Jayesh

2 2

R-URI for ACK
by David Wilson 24 May '17

24 May '17

Hello All, I’m trying to interop with a provider whose ACK during an inbound call seems to be misrouted. During the INVITE, OK, ACK handshake, my understanding (which matches the examples in RFC 3665) is that the R-URI in the ACK should be the Contact specified in the 200 OK. I have also found multiple discussions that imply that this is necessary. However, I can’t find a convincing normative reference for this... Is there a reference (ideally in RFC 3261) that makes this clear? Cheers, Dave.

1 0

Kamailio and XMLRPC commands with POST
by Olli Attila 24 May '17

24 May '17

Hi, I'm using kamailio 5.0.0 and trying to use core commands through xmlrpc interface but the requests are failing towards Kamailio. I have compiled and enabled the xmlrpc module in config and I have this on my kamailio.cfg: # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif I have tried to test this by using curl and calling "core.uptime" method in xml body (core.uptime works when executed through kamcmd commandline tool). The curl is called from the same machine that I have Kamailio installed on: "curl -X POST -H 'Host: 127.0.0.1:5060' -H 'User-Agent: xmlrpcclient' -H 'Content-Type: text/xml' -d '<?xml version='1.0'?><methodCall><methodName>core.uptime</methodName><params></params></methodCall>' http://127.0.0.1:5060" This fails and I only get: "curl: (7) Failed to connect to 127.0.0.1 port 5060: Connection refused" network traffic shows me this.. firewall is disabled completely when doing the test: 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 74 55630→5060 [SYN] Seq=0 Win=43690 Len=0 MSS=65495 SACK_PERM=1 TSval=1347930477 TSecr=0 WS=64 2 0.000032 127.0.0.1 -> 127.0.0.1 TCP 54 5060→55630 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 Can anyone help me to get to the right direction with this? Cheers, --Olli

5 10

rtpengine opening second RTP port
by John Petrini 23 May '17

23 May '17

Hello List, I'm having an issue with rtpengine changing the source port of the RTP stream after receiving a 183 from the carrier. Though RTP engine continues to send RTP to the proper carrier IP and port the carrier drops the inbound audio once the source port changes. Can anyone tell me what might trigger this type of behavior from rtpengine and if it's expected? Thank You, ___ John Petrini

3 3

Missing accounting events in ACC table
by Muhammad Shahzad 23 May '17

23 May '17

Hi, About 3 weeks ago i upgraded one of my production server with latest stable kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about missing CDR events in ACC table. I observe following problems, 1. There are only BYE records in acc table, no record for INVITE or ACK. 2. In kamailio logs when ACK is received against 200 OK response for INVITE, i see following errors, -- ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse >From uri ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org' to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484' -- Of course all these errors are bogus, I have checked all headers in ACK (not just FROM header), they all seem perfectly fine and valid. 3. Then the dialog times out, -- WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags '1D3ECD34F5731AB845BA3064AC95BB2D' '7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4' -- 4. Any further sequential requests complain about "unable to find dialog", e.g. -- NOTICE: <script>: Sequencial 'BYE' request received from caller ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [ sip:00xxxxxxxxxx@sip.domain.com] WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog for BYE with route param '5ae1.d595' [7845:22877] -- 5. However the acc record for BYE is written to db and log file, -- NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain= sip.domain.com ;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy -- The same config was working fine with older version 4.2.0-97cab8. The kamailio config i am using is pretty much standard, -- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 ... modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # log to db modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "DBURL") ... request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); }; exit; }; #### only initial requests (no To tag) #### t_check_trans(); .... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog setflag(FLT_ACC); # do accounting setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(dlg_timeout) = 60; dlg_manage(); .... } -- Any ideas why its happening? Since it is 3 weeks old so may be problem has already been spotted and fixed by someone else. Otherwise let me know how can i provide more info to help fix this issue. Thank you.

3 11

How to make OutBound calls with kamailio
by sharad tyagi 23 May '17

23 May '17

Hi I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I have add and registered two IP phones with kamailio these phone able to call each other. How can i make outbound call with that I am using a asterisk server as PSTN Gateway. i am not famalier with kamailio script. Regards Sharad Tyagi SparkTG info Pvt Ltd

4 3

Siremis v4.4.0 Released
by Daniel-Constantin Mierla 23 May '17

23 May '17

Hello, a short note to announce that Siremis v4.4.0 has been released - the web management application for Kamailio. It is supposed to mark the compatibility with Kamailio v4.x series, its future version will target Kamailio v5.0 compatibility. - http://siremis.asipto.com/2017/05/18/siremis-v4-4-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

pua_dialoginfo avps and entity
by Dmitri Savolainen 22 May '17

22 May '17

hi all! i'm trying to use pua_dialoginfo module (current master) with modparam("pua_dialoginfo", "use_pubruri_avps", 1) so for $avp(puburis_caller) = "sip:111111@mydomain"; i have PUBLISH with xml "entity" from "From:" field, not from my avp. PUBLISH sip:111111@mydomain SIP/2.0 Via: SIP/2.0/UDP 213.170.66.150:5071 ;branch=z9hG4bK9a08.f20b7367000000000000000000000000.0 To: <sip:111111@mydomain> From: <sip:111111@mydomain>;tag=b5f1e2ebdfeaa37cea72edd75fe37385-0dfc CSeq: 10 PUBLISH Call-ID: 6d71159d7ab5b9ff-23010@mydomain Content-Length: 260 User-Agent: Dev_Kamailio_UA Max-Forwards: 70 Event: dialog Expires: 41 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:003@mydomain"> <dialog id="yciKcJRNdO1HRjFg0qOoPUe4V3IhbqIW" direction="initiator"> <state>Trying</state> </dialog> </dialog-info> is it ok? according to doc: > If use_pubruri_avps is enabled, PUBLISH-requests reporting > dialog-information about the caller (entity=caller) are sent using the > value of the specified avp as R-Uri. > -- Savolainen Dmitri

1 0

Lookup using the last N digits
by Emanuele Gambaro 22 May '17

22 May '17

Hello everyone I need to do a lookup for an existing user using the last N digit of its phone number. For example, if a subscriber has a phone number equal to "+391234567890" I would like to call him using the full phone number or also the last 8 digits "34567890". There is a simple way to do this using kamailio 5.0.1? -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

Siremis upgrade and features
by javier falbo 21 May '17

21 May '17

Hi, I notice Siremis is not usable for me as it is very limited in the user management interface. I will suggest the developpers to include a USER management GUI visual interfase, as nowadays i use command line to do the following by putty tool: a) "Create Users" kamctl add <user> <password> b) "Delete Users" kamctl rm <user> c) "Show users in database" kamctl db show subscriber d) "Show online users" kamctl online Adding this on Siremis will help a lot for daily use as a Manager Account. Someone knows the Siremis author to add this suggestion? or someone has already coded a VISUAL GUI interfase for the above and could share it? Regards, Javier

4 7

Siremis searching from htable
by Olli Attila 21 May '17

21 May '17

Hello, Is there any way to make partial search from all htable entries with for example: %partofkeyname%? I tried to put that on search box but I get no results. If I use the exact full name of the key in the search i get the result. The search is here: Sip admin services - > server services -> htable list -> search I would like that the search function could return all of the keys that match the searched pattern but I have not succeed in making that kind of search. Cheers, Olli

2 2

How to debug Kamailio
by Remsik,Robert 19 May '17

19 May '17

Hello! I've got a base install of Kamailio and I'm trying to have it forward packets generated by sipsak to a different proxy (a migration of an existing ruleset by opensips). I can get the Kamailio process started but I can't figure out why Kamailio isn't forwarding the packet and I haven't been able to get debugging working to determine what's going on inside Kamailio to fix it. Can anyone please help me figure out how to debug this? Thank you in advance, Robert Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

3 4

Repeat of packets
by Jean Cérien 18 May '17

18 May '17

Hello I am getting started with Kamailio (or restarted, used it briefly years ago), with the final objective to do load balancing. For the time being, I am just trying to have one asterisk and one kamailio, on the same box. I have setup a box with an asterisk 11.3, and kamailio 4.4. I've taken the config file from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . My idea is that asterisk runs on port 5080, while kamailio on port 5060. Client interacts with Kamailio on port 5060. It almost works... Registration is fine, but when I send an invite, it is properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK message gets repeated multiple times and asterisk issues its infamous 'Retransmission timeout reached ...' - as if Kamailio wasnt processing it. See below ngrep traces between asterisk and kamailio Any ideas where to look ? Thanks J. # U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080 INVITE sip:102@192.168.2.228 SIP/2.0..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922 4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102@ 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact: <sip:iper@(null)>..Content-Type: application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject: Phone call..Content-Length: 437....v=0..o=199 2799 2990 IN IP4 192. 168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1; usedtx=1..a=rtpmap:111 s peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078 RTP/AVP 103 99..a=rtpmap:103 VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3.. # U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as57c98c4b..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;rec eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip :102@192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H..Supported: replaces, timer..Contact: <sip:102@192.168.2.228:5080>..Content-Length: 0.... # U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as57c98c4b..To: < sip:199@192.168.2.228:506 0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as30d9cfb4..To: < sip:199@192.168.2.228:506 0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. #

3 5

IMS: Including contents in the body of third-party REGIISTER request.
by Haggerty, Paul (phaggert) 18 May '17

18 May '17

It looks like the Kamailio IMS modules don't support including the original REGISTER request in third party registrations as defined in 3GPP TS 24.229. Can someone confirm this? Should this be a tracked defect or enhancement? In the meantime, is there any workaround available? Our application server needs to get the contents from the original REGISTER message from the UE. Thanks, Paul

2 4

Publish/notify for group kamailio
by Nguyen Tran Nhan 18 May '17

18 May '17

Hi, Is there any way to send Publish/Subscribe to group and Notify for all group (conference)? I have tried to send Subscribe to conference URI (SEMS) but I got method not implemented error. Any idea? Thanks, Nhan

1 0

kamcmd cli autocomplete fetaure stopped working after update from 4.3.4 to 4, 4.4 and 4.4.5
by Nuno Reis 18 May '17

18 May '17

Hi guys. I've just noticed that I can't use autocomplete under kamcmd cli anymore, after upgrading from 4.3.4 to latest versions. Should I open a github issue for this? -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

2 5

no audio with kamailio & rtpproxy bebind NAT
by 赵国杰 18 May '17

18 May '17

Hi everyone, I'm new to kamailio and rtpproxy and the problem's been bothering me for days. Any help would highly appreciated. My network topology is as follows: kamailio5.0 and rtpproxy2.0 are running on the same server. The server is behind NAT, its private ip is 10.240.0.62, public ip is <PUBLICIP> the server has 5060 tcp&udp, 35000-65000 udp port enabled. There're 2 sip clients(test && test1) running in another LAN(175.100.202.xxx) . Situation: Both clients registered successfully and they can call each other. But neither of them has audio.(If i kill rtpproxy, audio is ok). My kamailio cfg(only posted the part that i change), #!KAMAILIO #!define WITH_MYSQL #!define WITH_DEBUG #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PSTN listen=udp:10.240.0.62:5060 advertise <PUBLICIP>:5060 modparam("rtpproxy", "rtpproxy_sock", "udp:10.240.0.62:64000") rtpproxy cmd: rtpproxy -A <PUBLICIP> -F -l 10.240.0.62 -m 35000 -M 65000 -s udp:*:64000 -d INFO I used ngrep -d any -qt -W byline port 5060 to capture the sip msg, they are as follows: U 2017/05/18 05:56:29.320152 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.321103 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---9e44840a531df765;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="<PUBLICIP>", nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8". Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.384015 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 ACK. Content-Length: 0. . U 2017/05/18 05:56:29.388008 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Proxy-Authorization: Digest username="test",realm="<PUBLICIP>",nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8",uri="sip:test1@<PUBLICIP>",response="e7c42929b9a62de980651327b3a6999c",algorithm=MD5. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.389271 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.389935 10.240.0.62:5060 -> 175.100.202.254:53006 INVITE sip:test1@175.100.202.254:53006;ob SIP/2.0. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 349. . v=0. o=- 1500767650 1 IN IP4 <PUBLICIP>. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 46862 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=nortpproxy:yes. U 2017/05/18 05:56:29.782323 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>. CSeq: 2 INVITE. Content-Length: 0. . U 2017/05/18 05:56:29.846192 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:29.846709 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:32.768386 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 233. . v=0. o=- 3704075788 3704075789 IN IP4 10.60.208.171. s=pjmedia. c=IN IP4 10.60.208.171. t=0 0. m=audio 4008 RTP/AVP 0 101. c=IN IP4 10.60.208.171. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2017/05/18 05:56:32.769284 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 200 OK. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 255. . v=0. o=- 3704075788 3704075789 IN IP4 <PUBLICIP>. s=pjmedia. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 42584 RTP/AVP 0 101. c=IN IP4 <PUBLICIP>. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=nortpproxy:yes. U 2017/05/18 05:56:32.836731 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1 SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 70. Route: <sip:<PUBLICIP>;lr;nat=yes>. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:32.837266 10.240.0.62:5060 -> 175.100.202.254:53006 ACK sip:test1@175.100.202.254:53006;ob SIP/2.0. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.c283a6093e5f4143adec82663a0ff1d1.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:33.025120 10.240.0.62:5060 -> 175.100.202.254:53006 .... U 2017/05/18 05:56:38.025707 10.240.0.62:5060 -> 175.100.202.33:1490 .... U 2017/05/18 05:56:38.025731 10.240.0.62:5060 -> 175.100.202.33:10870 .... -- cheers, Jesse

1 0

Kamailio 4.4.X bug fixes
by José Seabra 17 May '17

17 May '17

Hello there, I apology for the following question but i didn't find this information on the kamailio site. When the development cycle in terms of bug fixes ends on version 4.4.x? Thank you Regards José

2 2

PV Setting Question
by Duarte Rocha 17 May '17

17 May '17

Good Afternoon, I'm having trouble with creation and value setting of some pseudo-variables. In the first cases, I used the pvapic.c function pv_get_sintval() like this : pv_get_sintval(msg, param, res, number_int_value); Using pv_get_sintval() i can assign the value to the PV and print it in the configuration file without any problem. However, if i need to assign a char* to the PV and use pv_get_strzval(msg, param, res, number_char_value);, it doesn't reach the configuration file. I made some debugs inside pv_get_strzval() in order to see the values of "res->rs.s" and res->rs.len and both values are fine. I just can't access them in the configuration file. Thanks in advance for your help! Cheers, Duarte Rocha

2 1

dmq_t_replicate
by Jay Patel 17 May '17

17 May '17

Hello Guys, Has anyone used dmq_t_replicate function to replicate INVITE? Documenation says it is useful for REGISTER , but does not mention anything about INVITE. I am trying to do something like this Box 1 ) Runs kamailio stateless ( no dialog module ) and calls dmq_t_replicate to box 2. Box 2 ) Runs kamailio with dialog module , This box generates CDR and tracks call count Goal is to restart kamailio on Box 1 without loosing CDR. Any input is appreciated.!! Regards, Jay

3 2

How to gateway MSRP messages into XMPP
by Pranathi Venkatayogi 16 May '17

16 May '17

I read about XMPP module in Kamailio, but it looks like it works with "MESSAGE" packets (SIMPLE). How do I gateway MSRP packets to XMPP in Kamailio. Do we have module support? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Failing in dropping duplicated 200 OK on onreply_route()
by Stefano Bertuola 16 May '17

16 May '17

Hi. I am trying to drop duplicated 200 OK on onreply_route(), but Kamailio still delivering them: if (is_method("INVITE") && t_check_status("200") && sdp_content()) { xlog("L_INFO", "200 OK Received\n"); if (!isflagset(20)) { xlog("L_INFO", "200 OK Relayed\n"); setflag(20); } else { xlog("L_INFO", "200 OK Dropped\n"); drop; } } >From logs, I can see the script is executed, but still 200 OK is relayed: INFO: <script>: 200 OK Received INFO: <script>: 200 OK Dropped DEBUG: tm [t_reply.c:1234]: t_should_relay_response(): ->>>>>>>>> T_code=200, new_code=200 DEBUG: tm [t_reply.c:1239]: t_should_relay_response(): DBG: t_should_relay_response: 200 INV after final sent DEBUG: tm [t_reply.c:1752]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 [...] How can I proceed to get the desired behavior? Br. Stefano

2 2

Why is the To URI the default in save()?
by Daniel Tryba 16 May '17

16 May '17

The save function from the registrar module uses the To header to disect and store the username for the location table according to observations and documentation http://www.kamailio.org/docs/modules/stable/modules/registrar.html#registra… After troubleshooting a ticket from an enduser unable to receive calls where all looked fine but the username used for authentication wasn't showing up in the location database. Finally I found the REGISTER was added to the location database, but not with the user its username, instead it was using the username (phonenumber) specified in the To header. Till now I always assumed that the username in the location table would be the username used during authentication(*). This opens the door to hijacking incoming calls to other users on the same kamailio registrar if one knows/guesses other usernames and use those in the To header. This realisation is kind of shocking to me. The solution is simple (if authentication is required): save("location", "0x00", "sip:$au@$rd"); *: which kind of answers my question in the subject, what else can be used if there is no authentication required?

3 3

✔I have finally found that stuff!
by poker poker 16 May '17

16 May '17

Greetings! You are going to be surprised a lot! I've finally found the stuff I told you about some time ago, just take a look http://www.realhopgar.com/wp-includes/pomo/rest.php?4c4d Thx, poker poker From: sr-users [mailto:sr-users@kamailio.org] Sent: Tuesday, May 16, 2017 2:49 AM To: pokerino76(a)libero.it Subject: Are you kidding? "Planets?" I squint my eyes and frown. "Who are you?" "I am Prince Eri. You have proven an excellent tactician and master manipulator of public perception." The (wo)man? looked at me with what I could only describe as hunger. Still rather confused I ask the only thing I could. "How do you figure that? I work a dead end job looking after two kids that aren't even mine." I scoff off what they said. "Simple. Remember when three months ago you received an email for a game in beta testing?" Wait, the game where I had to enact policies of keeping the public opinion of my army positive whilst simultaneously attempting to take over the known universe. "Wait don't tell me...." "Yes, it was real. Did you not think it odd how you were only able to take over two planets in three months? You were silently giving us our orders." The alien smiles? I don't know. "Why?" A sinking feeling enters the pit of his stomach. "Boredom mostly, we were mostly tired of steam rolling every planet we saw. So I devised the limitation of following some random person from Earth's orders. Surprisingly, the galactic opinion of us improved. We relied less and less on genocide. They now view us as a benevolent empire seeking to spread freedom and liberty. Thank you for that." I stood there dumbstruck. This isn't real, this is a dream. "Either join us and gain riches your entire world could never conceive of, or when we inevitably come after this planet, watch it all burn down." The choice already made for me I accept. But if what they say about me is true, then I will teach them something that never occurred to their alien minds. Humanity. Sent from Mail for Windows 10

1 0

Kamailio 5.0.1 install fails on Xenial
by Justin Clacherty 15 May '17

15 May '17

Hi, I’ve just set up a fresh install of Ubuntu 16.04.2 and tried to install Kamailio 5.0.1 using the apt repository at deb.kamailio.org <http://deb.kamailio.org/> but it fails to install during the configuration stage. Has anyone else seen this issue and are there any workarounds? Cheers, Justin. Setting up kamailio (5.0.1+xenial) ... Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. invoke-rc.d: initscript kamailio, action "start" failed. ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Sun 2017-05-14 23:54:15 AEST; 6ms ago Process: 1881 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <maxfwd> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <usrloc> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: bad config file (46 errors) May 14 23:54:15 km01 systemd[1]: kamailio.service: Control process exited, code=exited status=255 May 14 23:54:15 km01 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. May 14 23:54:15 km01 systemd[1]: kamailio.service: Unit entered failed state. May 14 23:54:15 km01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Hint: Some lines were ellipsized, use -l to show in full. dpkg: error processing package kamailio (--configure): subprocess installed post-installation script returned error exit status 1

2 2

SCTP Support for Rx/Dx interfaces
by Fred Schulz 15 May '17

15 May '17

Hi everybody, as I can read in the docs is the support for SCTP possible via Module just for SIP. Is there a possibility to add SCTP support to Rx/Dx interfaces at ICSCF? As read in the specs SCTP is mandatory for that interface… Thanks Mit freundlichen Grüßen fred schulz e-mail: fschulz(a)blackned.de<mailto:fschulz@blackned.de> tel.: +49 8331 9959-6-00 fax: +49 8331 9959-6-09 [cid:image001.png@01D2CD6A.382B1260] blackned gmbh · www.blackned.de<http://www.blackned.de/> niederlassung memmingen: zugspitzstr. 1 · 87751 heimertingen niederlassung münchen :friedrichshafener strasse 3 · 82205 gilching zentrale: dorfstrasse 3 · 88416 erlenmoos geschäftsführer: timo haas · josef stadler · hrb ulm 724147 Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und löschen Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and delete this e-mail. Any unauthorized copying, disclosure or distribution of the contents in this e-mail is strictly forbidden.

2 2

forced tls over tcp
by David Villasmil 15 May '17

15 May '17

Hello guys, I'm listening on port 443 for tls and 5111 for tcp. Client registers on 443 and makes a call over tls which kamailio forwards over tcp. Because freeswitch doesn't support "sips" yet, when replies come back from freeswitch (which contact set as "sip:" i manually change that tp "sips:" for the client to not die with "SIPS Required". Call is established and the client responds with ACK. At this point i need to change back the contact "sips:" to "sip:". This is the route that changes it: route[FIXCONTACT] { # This is freeswitch sending a message, so we change "sip:" to "sips:" xlog("L_ERR","[FIXCONTACT]: User-Agent is: $ua\n"); if( $ua =~ "^FreeSWITCH" ) { if( subst('/^(Contact:.*)sip:/\1sips:/') ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from freeSWITCH from sip to sips [$ct]!\n"); } # Else it is the client, when change it back from "sips:" to "sip:" } else { if( subst("/^(Contact:.*)sips:/\1sip:/") ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from CLIENT from sips to sip [$ct]!\n"); } } return; } When kamailio is preparing to send out the ACK via TCP I'm seeing: [forward.c:268]: get_send_socket2(): protocol/port mismatch And I really don't know why it's doing this... Help is appreciated! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

kamcmd domain.dump ERROR: reply too big
by Denys Pozniak 15 May '17

15 May '17

Hello! I have updated Kamailio to version 5.1.0-dev2 and started to obtain next error: *kamcmd> domain.dump* *ERROR: reply too big* Tried to increase ctl buffer size, but result is the same. *loadmodule "ctl.so"* *modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")* *modparam("ctl", "binrpc_max_body_size", 32000)* *modparam("ctl", "binrpc_struct_max_body_size", 3200)* Please point me where to find issue. *# uname -a* *Linux 4.4.0-64-generic #85-Ubuntu SMP Mon Feb 20 11:50:30 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux* *# cat /etc/lsb-release* *DISTRIB_ID=Ubuntu* *DISTRIB_RELEASE=16.04* *DISTRIB_CODENAME=xenial* *DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS"* BR, Denys

2 2

Kamailio + Sems with TURN server
by Nguyen Tran Nhan 15 May '17

15 May '17

Hi, I am trying to enable TURN with Kamailio + SEMS. Is there any link for guide? Thanks, Nhan

2 2

MSILO DUMPING SAME MESSAGE MULTIPLE TIMES
by sagar malam 15 May '17

15 May '17

Hello, I have been using Kamailio 4.4.In there i have used MSILO module to store messages when B party is not available and dump message when it gets registered. But when user gets registered,MSILO is dumping same messages multiple times. On further observation i found that MSILO stops dumping message when 200 OK for any one of the multiple message requests arrives to server.Hence i tried configuring "check_time" parameter to higher value but it did not help. I am am using same example script as provided in below link : http://kamailio.org/docs/modules/4.4.x/modules/msilo.html Except tm module parameters. Please help me on this issue. Thanks in advance.

2 1

IP Address used for outgoing Packets?
by Keith Whyte 15 May '17

15 May '17

Hi All, tl;dr - Kamailio is sending traffic to the Voip server with a source IP of the private (vpn) interface. For some weeks, I've been experimenting with Kamailio v 4.4.4, trying different options, including running (or not) a vpn interface on the kamailio server with and without NAT/rtpproxy. I have not as yet fully "installed" kamailo with database etc, I have just been playing with the config file, mostly running with -D For most of the configurations I tried, I was routing SIP to/from the VPN via another box on the network, and I tried to get RTP streams running without using rtpproxy, sometimes it worked, but I have two many double/triple NAT problems with some of the freeswitch nodes. Now I have decided to run a VPN interface directly on the kamailio server, for simplicity. Essentially what I want to do is have kamailio/rtpproxy bridge between the VPN and the VOIP provider, so it should be a pretty standard NAT config. My setup looks like this: VPN CLUSTER of Freeswitch nodes <-10.23.0.1-> KAMAILIO Server <-192.168.1.1 - gateway - PUBLIC_IP-> Voip Provider The Kamailio Server has two interfaces, one VPN interface and one facing the Public Internet. It is behind a firewall, and so has a private IP address assigned on the interface, but the is dedicated public IP that forwards all relevant ports, and traffic seen on the public side originates from this address. I have two listen directives, listen=udp:192.168.1.1:5060 advertise PUBLIC_IP:5060 listen=udp:10.23.0.1:5060 I expected Kamailio to receive SIP INVITE on the 10.23.0.1 address and forward it to the voip provider via the default route (the 192.168.1.1 address), but it is sending the packet out to the voip provider with a source address of 10.23.0.1 !! Can anyone tell me how I can configure this? I cannot find anything obvious in the manual. Thanks!

3 2

SIREMIS 4.3.0 and Kamilio 5.1.0-dev1
by Emil Morganti 12 May '17

12 May '17

Hi everybody, I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to the second step i get an error: ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry '127.0.0.1' for key 'domain_idx' Some solution?

2 1

TLS to TCP
by David Villasmil 12 May '17

12 May '17

Hello Guys, I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp, but AFTER the call is established, kamailio hangs the call with "SIPS required"... Has this happened to anyone? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 3

t_drop_replies not working with t_suspend in failure route
by Huber Andreas 12 May '17

12 May '17

Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We'd like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas

2 1

Re: [SR-Users] Can rtpproxy stream audio inside a call? (Maxim Sobolev)
by Yufei Tao 12 May '17

12 May '17

Hi, Thank you very much for the answers! I'll have a play. Cheers, Yufei Message: 9 Date: Mon, 8 May 2017 01:01:51 -0700 From: Maxim Sobolev <sobomax(a)sippysoft.com> To: sr-users(a)lists.kamailio.org, Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call? Message-ID: <CAH7qZfs0tQTwa-aCwvHHhAS_hajCRNaKBLnjXBv+O0txJNd4Mw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Daniel is right. Technically speaking it can be done at any point after the media session has been established, but you need some kind of trigger to start/stop it from your routing script. Such as re-INVITE for example. -Max On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: > Hello, > > it can play rtp files encoded in the format expected by the audio codec. I > used only when there was a re-invite (like putting the call on hold). > > Cheers, > Daniel > > On 04.05.17 17:29, Yufei Tao wrote: > > Hi, > > Just a quick question: if I use the Kamailio rtpproxy module with the Sippy > RTPproxy, can I stream audio within a call? Or is it only possible before > call audio is established? > > For example can I mix/playback audio file into an existing established > call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() > somehow? If it is possible, does it need to be triggered by an in-dialog > message from either UAC or UAS? > > Cheers, > Yufei > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps:// lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda <http://mierlawww.twitter.com/miconda> -- www.linkedin.com/in/miconda > Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Config Kamailio with SEMS
by Nguyen Tran Nhan 11 May '17

11 May '17

Dear Daniel, I am working on Push To Talk PoC. I intent to use Kamailio as SIP signaling to work with SEMS for handling mixer conference audio (media server). The reason to use SEMS is performance is better in comparing with FreeSwitch or Asterisk. There are some guide to config Kamailio with Freeswitch or Asterisk but no for SEMS. If you have any information or guide line, please help!

4 4

Warning "constant value in if(...)"
by Ginhoux, Patrick 11 May '17

11 May '17

Hi, I discovered that when kamailio (version 5.0.1) starts, if reports a waring on a 'if' command : May 11 11:26:20 vm-vse02-siprouter2 kamailio: WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 978, column 6-13: constant value in if(...) The line 978 (in yellow) is part of the following routing below in my Kamailio.cfg : route[TO_MMM] { $var(mbxDN)=$null; # directory number containing mailbox number $var(uDigit)=$null; # application digits $var(i:NodePath)=$null; $var(TechNumber)=$null; $avp(nodeInx)=$null; # strip prefix from user in URI request user field $var(TechNumber) = $(rU{re.subst,/^(\+33|\+262|\+377|\+590|\+594|\+596|0)//}); # strip suffix from user in URI request user field - suffix headed by anything other than digit or '+' $var(TechNumber) = $(var(TechNumber){re.subst,/[^\+0-9].*//}); if (VM_TARGET_TECHNICAL_NUMBER!=$null) { $avp(vm_prefix) = VM_TARGET_TECHNICAL_NUMBER; } else { $avp(vm_prefix) = $(var(TechNumber){s.substr,0,5}); The same code with kamailio 3.3 doesn't report this warning. Also the same code exists in another part of the routine but don't report a warning. I have no idea of the impact of such warning. Also I'm looking for help to understand the problem. Thanks in advance for the answers. Cordialement Patrick GINHOUX

1 0

URGENT! stateless dispatcher BYE issue
by Satish Patel 11 May '17

11 May '17

We have 1 stateless dispatcher and 3 proxy behind that dispatcher, when Proxy send BYE message then dispatcher bouncing back that BYE to Proxy and they playing ping pong. I think i am missing something.. How BYE get routed? in dispatcher i am not using loose_route() function also should i need to use that but again we running dispatcher in stateless. Here i put SIP trace: https://pastebin.com/vp9JCdNP Dispatcher config, Very simple.. route{ remove_hf("Route"); if (method == "INVITE") { record_route(); } if ( !ds_select_dst("1", "0") ) { xlog("L_ERR", "Unable to route \n"); sl_send_reply("500","Unable to route"); break; } forward(uri:host, uri:port);

2 2

Conference Factory URI
by Nguyen Tran Nhan 11 May '17

11 May '17

Hi, I want to create conference without interact with IVR, so, trying to use Conference Factory URI. https://tools.ietf.org/html/rfc4579#section-3.2 I am looking for documentation but it seems very poor documentation. Could you please help to give more document or opensource for Conference Factory URI. SEMS does not support Conference Factory URI I guess? Thanks, Nhan

2 4

permissions module db
by David Villasmil 11 May '17

11 May '17

Hello guys, I0ve freated a view taken from other tables to use with the permissions module, the result looks like this: And some values: 23 10 192.198.1.215 32 0 Training-215 Still, Kamailio complaints with: 0(5796) DEBUG: <core> [db_res.c:188]: db_allocate_rows(): allocate 48 bytes for rows at 0x7f69510651b8 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065250 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.215] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [Training-215] 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065358 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.216] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [PBX-Service] 0(5796) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 160 bytes for row values at 0x7f6951065460 0(5796) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [10] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [67.215.186.212] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [32] 0(5796) DEBUG: <core> [db_val.c:85]: db_str2val(): converting BIGINT [0] 0(5796) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [JuanPablo] 0(5796) DEBUG: permissions [address.c:142]: reload_address_table(): Number of rows in address table: 3 0(5796) DEBUG: permissions [address.c:169]: reload_address_table(): failure during checks of database value 3 (subnet size/CIDR) in address table 0(5796) ERROR: permissions [address.c:247]: reload_address_table(): database problem - invalid record 0(5796) DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 5 columns 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7f6951064790 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[1] at 0x7f6951064718 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[2] at 0x7f6951063950 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[3] at 0x7f6951064340 0(5796) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[4] at 0x7f6951064498 0(5796) DEBUG: <core> [db_res.c:96]: db_free_columns(): freeing result names at 0x7f6951064610 0(5796) DEBUG: <core> [db_res.c:101]: db_free_columns(): freeing result types at 0x7f6951064808 0(5796) DEBUG: <core> [db_res.c:52]: db_free_rows(): freeing 3 rows 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065250 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065358 0(5796) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at 0x7f6951065460 0(5796) DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at 0x7f69510651b8 0(5796) DEBUG: <core> [db_res.c:138]: db_free_result(): freeing result set at 0x7f6951065118 0(5796) CRITICAL: permissions [address.c:359]: init_addresses(): reload of address table failed 0(5796) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(5796) ERROR: permissions [permissions.c:614]: mod_init(): failed to initialize the allow_address function 0(5796) ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module permissions (/usr/local/kamailio5/lib64/kamailio/modules/permissions.so) ERROR: error while initializing modules Can anyone help? thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 4

Live streaming from Kamailio World Conference 2017
by Daniel-Constantin Mierla 10 May '17

10 May '17

Hello, if you haven't noticed from other social networking channels, there is a live stream from Kamailio World 2017, thanks to Pascom team: - http://live.pascom.net/ Enjoy! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

3 3

Kamailio ro user
by Донат Зеничев 10 May '17

10 May '17

Yes, I saw this part of kamailio doc, and even I added this part in my.cnf: [kamailio] socket = /path/to/mysql.sock user = kamailiouser password = kamailiopass default-character-set = utf8 (with valid password) it didn't help me (of course I restarted the mysqld). Both of the following syntax worked fine for me: # mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> # mysql --user=xxx --password=xx --protocol=socket <database> So that, I could get into a mysql console (with kamailioro user/pasword). I mentioned in a previous message, that Kamailio is installed from repository and is not compiled. But, kamailio-mysql-modules:amd64 is installed. So looks, like there is no problem with socket connection, because kamailio doesn't say that problem depends on mysql module, it said that problem consists of accessing for kamailioro user. >Hi, > >So, I have not been using Kamailio with MySQL before, but I do have >some basic MySQL knowledge. > >MySQL can be used with both socket and TCP, have you verified that >your setup is matching the server setup. > >Have you look at this section: >https://kamailio.org/docs/modules/4.3.x/modules/db_mysql.html#idp3037648 > >You can also test/check that socket/tcp is working by testing it like this: ># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> ># mysql --user=xxx --password=xx --protocol=socket <database> > > >On Fri, May 5, 2017 at 4:47 PM, Донат Зеничев <donat.zenichev at gmail.com> wrote: >> Hi. >> >> I have just written routing logic for my new sip-proxy server, but have a >> problem with start of kamailio. >> >> ./kamailio.cfg file has no mistakes - it's made well done. >> >> before to start coding with routing logic, of course I installed kamailio >> (ver 4.3.4) from repository. >> OS - ubuntu 16.04 >> >> and add next modules from repository: >> kamailio-mysql-modules:amd64 >> kamailio-perl-modules:amd64 >> kamailio-radius-modules:amd64 >> kamailio-tls-modules:amd64 >> kamailio-utils-modules:amd64 >> >> next step, I edited the ./kamctlrc file to contain proper user/password and >> mysql engine settings. >> And than, ran the kamdbctl create (without special parameters). >> >> A bit about my kamailio.cfg : >> fork=yes >> children=4 >> disable_tcp=no >> tcp_children=4 >> tcp_send_timeout=3 >> tcp_connect_timeout=5 >> tcp_connection_lifetime=3600 >> tcp_max_connections=4096 >> >> of course mpath is defined and proper modparams (depending on my ver) for >> modules are used. >> >> A bit about my kamctlrc: >> Ro and rw user are defined (and connections to needed db from these users is >> checked, as like it was real kamailio that tries to update some values in db >> - well done). >> >> Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) >> >> So that, now I'm trying to start the server as a daemon, I have the >> following row in systemctl: >> ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: >> Access denied for user 'kamailioro'@'localhost' (using password: YES) >> >> And of course all other modules can't run because they depend on the db >> connections. >> So looks like very simple problem, just need to edit mysql user table to >> have proper passwords. >> >> But I've refurbished ro and rw user several times, it didn't help me. >> Also tried to drop kamailio db with db users and run kamdbctl create again - >> nothing changed. >> >> I tried to find where kamailioro user can be defined as default, but didn't >> find any files (apart of kamctlrc in /etc/kamailio directory). >> /etc/default/kamailio file contains proper values and paths. >> >> Please advise something. >> >> >> >> -- >> -- >> BR, Donat Zenichev >> Wnet VoIP team >> Tel: +380(44) 5-900-808 >> http://wnet.ua >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users at lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > > >-- >Best regards, > >Glenn O. Larsen -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

3 3

Kamailio ro user
by Донат Зеничев 10 May '17

10 May '17

Guys, I'm so sorry for wasting your time. I've just dropped all kamailio mysql users, database and commented kamailioro user in kamctlrc config file and than ran a kamdbctl create command again, to refurbish all the structure. So that, it didn't create kamailioro user, like it was in previous time. And now it doesn't requesting the kamailioro user credentials at all. Works fine, daemon starts without problem. Thanks for reply! Question is now closed for me. >Yes, I saw this part of kamailio doc, and even I added this part in my.cnf: >[kamailio] >socket = /path/to/mysql.sock >user = kamailiouser >password = kamailiopass >default-character-set = utf8 > >(with valid password) it didn't help me (of course I restarted the mysqld). > >Both of the following syntax worked fine for me: ># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> ># mysql --user=xxx --password=xx --protocol=socket <database> > >So that, I could get into a mysql console (with kamailioro user/pasword). > >I mentioned in a previous message, that Kamailio is installed from repository and is not compiled. >But, kamailio-mysql-modules:amd64 is installed. > >So looks, like there is no problem with socket connection, because kamailio doesn't say that problem depends on mysql module, it said that problem consists of accessing for kamailioro user. > >>Hi, >> >>So, I have not been using Kamailio with MySQL before, but I do have >>some basic MySQL knowledge. >> >>MySQL can be used with both socket and TCP, have you verified that >>your setup is matching the server setup. >> >>Have you look at this section: >>https://kamailio.org/docs/modules/4.3.x/modules/db_mysql.html#idp3037648 >> >>You can also test/check that socket/tcp is working by testing it like this: >># mysql --user=xxx --password=xx --host=localhost --protocol=tcp <database> >># mysql --user=xxx --password=xx --protocol=socket <database> >> >> >>On Fri, May 5, 2017 at 4:47 PM, Донат Зеничев <donat.zenichev at gmail.com> wrote: >>> Hi. >>> >>> I have just written routing logic for my new sip-proxy server, but have a >>> problem with start of kamailio. >>> >>> ./kamailio.cfg file has no mistakes - it's made well done. >>> >>> before to start coding with routing logic, of course I installed kamailio >>> (ver 4.3.4) from repository. >>> OS - ubuntu 16.04 >>> >>> and add next modules from repository: >>> kamailio-mysql-modules:amd64 >>> kamailio-perl-modules:amd64 >>> kamailio-radius-modules:amd64 >>> kamailio-tls-modules:amd64 >>> kamailio-utils-modules:amd64 >>> >>> next step, I edited the ./kamctlrc file to contain proper user/password and >>> mysql engine settings. >>> And than, ran the kamdbctl create (without special parameters). >>> >>> A bit about my kamailio.cfg : >>> fork=yes >>> children=4 >>> disable_tcp=no >>> tcp_children=4 >>> tcp_send_timeout=3 >>> tcp_connect_timeout=5 >>> tcp_connection_lifetime=3600 >>> tcp_max_connections=4096 >>> >>> of course mpath is defined and proper modparams (depending on my ver) for >>> modules are used. >>> >>> A bit about my kamctlrc: >>> Ro and rw user are defined (and connections to needed db from these users is >>> checked, as like it was real kamailio that tries to update some values in db >>> - well done). >>> >>> Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) >>> >>> So that, now I'm trying to start the server as a daemon, I have the >>> following row in systemctl: >>> ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: >>> Access denied for user 'kamailioro'@'localhost' (using password: YES) >>> >>> And of course all other modules can't run because they depend on the db >>> connections. >>> So looks like very simple problem, just need to edit mysql user table to >>> have proper passwords. >>> >>> But I've refurbished ro and rw user several times, it didn't help me. >>> Also tried to drop kamailio db with db users and run kamdbctl create again - >>> nothing changed. >>> >>> I tried to find where kamailioro user can be defined as default, but didn't >>> find any files (apart of kamctlrc in /etc/kamailio directory). >>> /etc/default/kamailio file contains proper values and paths. >>> >>> Please advise something. >>> >>> >>> >>> -- >>> -- >>> BR, Donat Zenichev >>> Wnet VoIP team >>> Tel: +380(44) 5-900-808 >>> http://wnet.ua >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users at lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> >>-- >>Best regards, >> >>Glenn O. Larsen -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

1 0

dlg_refer caller fails to create callee dialog
by Pranathi Venkatayogi 10 May '17

10 May '17

Hi, I am calling dlg_refer after dialog is confirmed on both sides. This time the callee is a real client, so call was established end to end. But still I get the error "No contact available" when callee dialog is being created - see below. I am attaching the snippet of my config where I am calling dlg_refer as well.. Is there something wrong with my usage? Please advise. Thanks, Pranathi 4719 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO: <script>: Translation: Dialog confirmed 3 and agenturi: sip:agent1@devtranslation.sms-test.cyracom.com 4720 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=573 a=16 n=if 4721 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=567 a=26 n=xlog 4722 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: INFO: <script>: Translation: Referring dialog from sip:71594386@10.11.200.59:49376;alias=10.11.200.59~49376~1 4723 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: exec: *** cfgtrace:local_route=[dialog:start] c=[routingdev.cfg] l=568 a=26 n=dlg_refer 4724 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available 4725 May 8 22:27:31 devsip00 /sbin/kamailio[1125]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t event_route[dialog:start] { if(is_known_dlg()) { $var(agentUri) = $dlg_var("agenturi"); xlog("L_INFO", "Translation: Dialog confirmed $dlg(state) and agenturi: $var(agentUri)"); if ($(var(agentUri){uri.user}) != "") { xlog("L_INFO", "Translation: Referring dialog from $dlg(from_contact)"); dlg_refer("caller", "$var(agentUri)"); } else { xlog("L_INFO", "Translation: Dialog confirmed $dlg(state) but no agent info yet\n"); } } } Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 3

KEMI, Lua external modules and nonblicking operations
by Yuriy Gorlichenko 10 May '17

10 May '17

Hi all. I m trying to use KEMI with lua and want to make call http client lua module (external). I want to ask: is it will block kamailio event loop (or somethig like it in kamailio)? For more understanding what im talking about i will provide example with openresty (nginx + lua) It uses nginx cosockets and if i will use not cosockets based module in openresty it will block nginx eventloop because of using network operations in nginx worker context.

2 1

Re: [SR-Users] [OpenSIPS-Users] TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 09 May '17

09 May '17

On 9 May 2017 at 15:18, Bogdan-Andrei Iancu <bogdan(a)opensips.org> wrote: > Thank you Giovanni, that is a useful tool - we will document it in the > OpenSIPS TLS tutorial, so other can benefit ;) > > Glad about it! Be sure to get it from https://freeswitch.org/confluence/display/FREESWITCH/ Packet+Capture#PacketCapture-TLSwithsharka , is the latest version with a couple fixes. -giovanni > Many thanks, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > OpenSIPS Summit May 2017 Amsterdam > http://www.opensips.org/events/Summit-2017Amsterdam.html > > On 05/02/2017 05:52 PM, Giovanni Maruzzelli wrote: > > For a cut and paste ready version, that has the correct carriage returns > (mangled by mail), check it in FreeSWITCH documentation: > > https://freeswitch.org/confluence/display/FREESWITCH/ > Packet+Capture#PacketCapture-TLSwithsharka > > -giovanni > > On 2 May 2017 at 16:26, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > >> Hello fellows, >> >> after some experimentation with various tools, I come out with a little >> shell tool that maybe can be useful to you too. >> >> It can only work with non-forward secrecy ciphers, obviously, and only if >> is started before the client do the initial TLS handshake (eg, just restart >> the client). Forward secrecy cannot be decrypted after fact, so don't waste >> effort. >> >> An example of ciphers that can be decrypted are the "AES256-SHA" openssl >> cipher group. You can use ssldump to check what cipher is used by >> serverhello. >> >> Enjoy, make it better, and share it :) >> >> >> #!/bin/bash >> # brought to you by Giovanni Maruzzelli >> # >> SERVERIP="192.168.1.150" >> SERVERPORT="5061" >> PRIVKEY="/etc/certs/privkey.pem" >> STDERR2DEVNULL=" 2>/dev/null " >> REGEX="notyet" >> >> if [ -z "$1" ]; then >> REGEX="\\\.*" >> else >> REGEX="$1" >> fi >> FILTER="ssl.app_data and sip matches" >> FILTER2="$FILTER \"$REGEX\"" >> FILTER3="'$FILTER2'" >> ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e >> frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e >> sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d >> tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" >> $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u >> 's/^[0-9]*$/\n==&==============================/g'" >> >> echo "" >> echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" >> and try again" >> echo "" >> echo "NB: remember to quote and escape match patterns, using triple slash" >> echo " eg, for matching 1010(a)pbx.example.com, use \" >> 1010(a)pbx.example.com\"" >> echo " eg, for matching anything, use \"\\\\\\.*\"" >> echo " eg, for matching *98, use \"\\\\\\*98\"" >> echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" >> echo "" >> >> >> case "$1" in >> -help|--help|?) >> exit 0 >> ;; >> *) >> echo "THIS TIME WE'RE DOING:" >> echo "tshark $ARGUMENT" >> echo "" >> bash -c "tshark $ARGUMENT" >> ;; >> esac >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

1 0

Issue with INVITE to NATed client
by Iskren Hadzhinedev 08 May '17

08 May '17

Hi list! I'm using kamailio 4.2 for load-balancing and failover via dispatcher, but I'm having some NAT related issues and I was hoping that someone might point me in the right direction. My setup is the following: PSTN - PBX - kamailio - NAT - client Calls from the NATed client to PSTN and/or PBX features (e.g. voicemail) work just fine. However, when a call comes from PSTN to the client, kamailio sends the INVITE to the client's RFC1918 IP and I can't figure out how to send it to the correct destination. I tried loading the registrar and usrloc modules and used save("location") during REGISTER and lookup("location") just before loose_route(), but the end result was that kamailio replied to REGISTERs with any username/password with a 200 OK and the endpoints never registered with the PBX. I can attach the config file if that will help. Thanks for all input in advance! Regards, Iskren

3 5

Can rtpproxy stream audio inside a call?
by Yufei Tao 08 May '17

08 May '17

Hi, Just a quick question: if I use the Kamailio rtpproxy module with the Sippy RTPproxy, can I stream audio within a call? Or is it only possible before call audio is established? For example can I mix/playback audio file into an existing established call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() somehow? If it is possible, does it need to be triggered by an in-dialog message from either UAC or UAS? Cheers, Yufei

3 2

kamailio crashes when it receives CANCEL
by Andy 08 May '17

08 May '17

Hi, when one of the proxies receive CANCEL, and it also the tries to generate 408 because of frv_inv_timer timeout at the sametime the e2e_cancel tries to generate a response to CANCEL - 487 at this time kamailio crashes from build_res_buf_from_sip_req, << for(lump=msg->reply_lump;lump;lump=lump->next) { len += lump->text.len; >> is it because the 408 generated deletes the reply lump or how can it be prevented. thanks for looking -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-crashes-when-it-receives-C… Sent from the Users mailing list archive at Nabble.com.

3 5

serctl moni command output question
by Satish Patel 08 May '17

08 May '17

We have old ser running and in "serctl moni" command output in "Transaction Statistics" section "waiting counter" is growing it's almost millions sometime i do restart to get it clear so wanted to understand what is that counter for and why its keep growing? see following output after restart its 237 but it will grow by time.. #serctl moni [cycle #: 20; if constant make sure server lives and fifo is on] Server: Vivox Proxy (0.10.99+cvs20050913 (i386/linux)) Now: Fri May 5 21:20:10 2017 Up Since: Sun Apr 30 02:31:34 2017 Up time: 499716 [sec] Transaction Statistics Current: 73 (237 waiting) Total: 1015654 (0 local) Replied localy: 1748134 Deleted: 1015581 Completion status 6xx: 0, 5xx: 226, 4xx: 241347, 3xx: 0, 2xx: 848765

2 1

dispatcher load balancing behavior..
by Jignesh Gandhi 07 May '17

07 May '17

Hello, Following is an output from my sercmd dispatcher.list command. I have noticed that sometimes the FLAGS: value is set to IX ( which is ok, it just means DEST is de-registered ) No calls are routed to this DEST, but for some reason the load balancing such as hash over callid is affected. So If I had 4 DESTs and the first DEST had FLAGS:IX, no calls would be routed to it but the next destination would get more calls than the rest. So, in the example below DEST 132.24 would get more calls than DEST 132.25 [root@ser-kam-001 ~]# sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:176.22.132.23:5060 FLAGS: IX PRIORITY: 0 ATTRS: reg-time=1494197124 } DEST: { URI: sip:176.22.132.24:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1494197125 } DEST: { URI: sip:176.22.132.25:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1494197124 } } } Thanks, --Jignesh Gandhi

1 0

Kamailio ro user
by Донат Зеничев 05 May '17

05 May '17

Hi. I have just written routing logic for my new sip-proxy server, but have a problem with start of kamailio. ./kamailio.cfg file has no mistakes - it's made well done. before to start coding with routing logic, of course I installed kamailio (ver 4.3.4) from repository. OS - ubuntu 16.04 and add next modules from repository: kamailio-mysql-modules:amd64 kamailio-perl-modules:amd64 kamailio-radius-modules:amd64 kamailio-tls-modules:amd64 kamailio-utils-modules:amd64 next step, I edited the ./kamctlrc file to contain proper user/password and mysql engine settings. And than, ran the kamdbctl create (without special parameters). A bit about my kamailio.cfg : fork=yes children=4 disable_tcp=no tcp_children=4 tcp_send_timeout=3 tcp_connect_timeout=5 tcp_connection_lifetime=3600 tcp_max_connections=4096 of course mpath is defined and proper modparams (depending on my ver) for modules are used. A bit about my kamctlrc: Ro and rw user are defined (and connections to needed db from these users is checked, as like it was real kamailio that tries to update some values in db - well done). Also PID_FILE and FIFOPATH are defined (/var/run/kamailio dir. created) So that, now I'm trying to start the server as a daemon, I have the following row in systemctl: ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Access denied for user 'kamailioro'@'localhost' (using password: YES) And of course all other modules can't run because they depend on the db connections. So looks like very simple problem, just need to edit mysql user table to have proper passwords. But I've refurbished ro and rw user several times, it didn't help me. Also tried to drop kamailio db with db users and run kamdbctl create again - nothing changed. I tried to find where kamailioro user can be defined as default, but didn't find any files (apart of kamctlrc in /etc/kamailio directory). /etc/default/kamailio file contains proper values and paths. Please advise something. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

2 1

Test Message
by Remsik,Robert 05 May '17

05 May '17

Test Message to verify that I can receive it (and that I am actually subscribed to the list as I don't think the registration has worked so far). Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

2 3

Re: [SR-Users] How does dlg_bridge work?
by Pranathi Venkatayogi 05 May '17

05 May '17

When I use dlg_refer I get the following error: 2689 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: exec: *** cfgtrace:failure_route=[HTTP_ALLOCATE_RESPONSE] c=[routingdev.cfg] l=567 a=26 n=dlg_refer 2690 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:690]: dlg_lookup(): ref dlg 0x7f92d19835e8 with 1 -> 2 2691 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:692]: dlg_lookup(): dialog id=10603 found on entry 2058 2692 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:452]: build_new_dlg(): new dialog on hash 2058 2693 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available 2694 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t 2695 May 4 16:09:56 devsip00 /sbin/kamailio[24617]: DEBUG: dialog [dlg_hash.c:922]: dlg_unref_helper(): unref op on 0x7f92d19835e8 with 1 from dlg_hash.c:940 From: Pranathi Venkatayogi Sent: Thursday, May 04, 2017 5:28 AM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org> Subject: How does dlg_bridge work? I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 2

KEMI Python and msg.call_function()
by Samuel F. 05 May '17

05 May '17

Hi, Using KEMI Python and doing serial forking via: msg.call_function("t_load_contacts") msg.call_function("t_next_contacts") * I couldn't see that the functions are exported to KEMI yet. I'm not great with C so I haven't tried exporting them myself. I get the following in the log files: PYT {INVITE}: 5(11963) ERROR: *** cfgtrace:request_route=[] c=[] l=0 a=26 n=t_load_contacts PYT {INVITE}: 5(11963) ERROR: *** cfgtrace:request_route=[] c=[] l=0 a=26 n=t_next_contacts However, it seems to be working anyway. Is the ERROR because I use the msg.call_function() ? Is there any better way to do serial forking using KEMI? Thank you, Samuel F

2 2

UAC and TLS
by Marino Mileti 04 May '17

04 May '17

Hi guys, is there any way to allow that the UAC module will make a REGISTER to another Kamailio with the parameters configured in uacreg table using the TLS protocol instead of UDP? Marino

3 2

if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 04 May '17

04 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

3 3

How does dlg_bridge work?
by Pranathi Venkatayogi 04 May '17

04 May '17

I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio KEMI / Python and the usage of KSR.pv.get("$rm") vs msg.Method
by Samuel F. 04 May '17

04 May '17

Hi, I see in the examples that KSR.pv.get("$rm") is frequently used rather than the more simple msg.Method. Why is this preferred? Kind regards, Samuel F

2 1

Freeswitch 403 Forbidden on invite
by Ravi sanyal 04 May '17

04 May '17

Hello, I'm new to using freeswitch. I'm using a client called SIPJS to communicate with freeswitch. The SIP message and it's response: INVITE sip:1001@104.130.11.232 SIP/2.0 Via: SIP/2.0/WS CODE;branch=BRANCH Max-Forwards: 70 To: <sip:1001@104.130.11.232> From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE Contact: <sip:k86b2npa@v9figramd1qs.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.7 Content-Type: application/sdp Content-Length: 759 v=0 o=- 84832118975142245 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE data a=msid-semantic: WMS m=application 49422 DTLS/SCTP 5000 c=IN IP4 111.69.178.179 a=candidate:1520298561 1 udp 2113937151 192.168.178.32 49422 typ host generation 0 network-cost 50 a=candidate:900384389 1 udp 2113939711 2406:e007:2f57:1:e951:8d6e:71bd:cff8 49423 typ host generation 0 network-cost 50 a=candidate:842163049 1 udp 1677729535 111.69.178.179 49422 typ srflx raddr 192.168.178.32 rport 49422 generation 0 network-cost 50 a=ice-ufrag:6Jk+ a=ice-pwd:XDsf8/lOfUpw8f3CQas5srLl a=fingerprint:sha-256 23:DF:44:BC:1B:A5:B3:77:E0:ED:54:F9:A1:EE:CB:A1:99:FC:3A:0F:A2:2B:60:07:9A:F9:2D:12:BB:5B:60:49 a=setup:actpass a=mid:data a=sctpmap:5000 webrtc-datachannel 1024 SIP/2.0 403 Forbidden Via: SIP/2.0/WS CODE;branch=BRANCH;received=111.69.178.179;rport=64431 From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb To: <sip:1001@104.130.11.232>;tag=gt0m2j3aa4m1K Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-380-c66a012~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 I thought to try get the invite to happen after subscribe events had taken place however the same messages are received back. This leads me to think that it's something to do with my freeswitch installation. Register messages are going through fine. that tells me that the users are being authenticated when talking to freeswitch. Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

1 0

How does dlg_bridge work?
by Pranathi Venkatayogi 03 May '17

03 May '17

I have TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. I am attaching Kamailio log / my config files for reference. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Unknown reason for 500 I'm terribly sorry, server error occurred
by Pranathi Venkatayogi 03 May '17

03 May '17

I am encountering "500 I'm terribly sorry, server error occurred" . However I don't see this message in the Kamailio.log or anything related to find the root cause. I am attaching herewith the packet capture and the entire Kamailio log and the config files. Can someone please shed some light how do I go about debugging this? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

rtpengine cpu usage
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm using rtpengine (compiled from source) and I see a lot of difference in cpu usage comparing with rtpproxy. Is this normal? With 600 calls: cpu *41.7%* rtpengine #rtpengine threads 4 # lscpu Architecture: x86_64 CPU op-mode(s): 32-bit, 64-bit Byte Order: Little Endian *CPU(s): 2* On-line CPU(s) list: 0,1 Thread(s) per core: 1 Core(s) per socket: 1 Socket(s): 2 NUMA node(s): 1 Vendor ID: GenuineIntel CPU family: 6 Model: 26 Model name: Intel(R) Xeon(R) CPU X7560 @ 2.27GHz Stepping: 4 CPU MHz: 2259.955 BogoMIPS: 4522.00 Hypervisor vendor: VMware Virtualization type: full L1d cache: 32K L1i cache: 32K L2 cache: 256K L3 cache: 24576K NUMA node0 CPU(s): 0,1 Calls established with a=rtpmap:0 PCMU/8000 Thanks in advance! Diego.

3 3

if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 03 May '17

03 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

1 0

Re: [SR-Users] topos and PRACK problem
by Sergey Basov 03 May '17

03 May '17

Hi, Daniel. Seems I found how to fix PRACK handling. It works for me. please lock at https://github.com/kamailio/kamailio/pull/1097 Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com 2017-04-27 13:04 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: > Yes, you are right. > > But now before 200 OK there empty field b_contact. > > May be you does not populate it from contact in 183 Progress? > > I see that this field is not empty only after 200 OK with a Contact field. > -- > Best regards, > Sergey Basov e-mail: sergey.v.basov(a)gmail.com > > > 2017-04-27 12:37 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> Hello, >> >> thanks for troubleshooting further. I haven't got the time to look at >> the source code, but I expect that the b-leg attributes (contact, record >> routes) to be set on 200ok for dialog (topos_d). I think for PRACK, the >> routing information should be stored and taken from transaction (topos_t). >> >> Cheers, >> Daniel >> >> On 27.04.17 11:29, Sergey Basov wrote: >>> Hi, Daniel. >>> >>> I just done one more test topos with re-invite which comes from caller >>> to callee, same direction as PRACK. >>> >>> And I found that, in case of prack, b_contact field is empty... >>> >>> Please find attached debug=3 part of re-invite message. >>> >>> Hope it helps. >>> -- >>> Best regards, >>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>> >>> >>> 2017-04-27 10:30 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>> Hi, Daniel >>>> >>>> Please look at attached part of debug=3 while receiving and parsing PRACK. >>>> At line 208 seems rr module does not find correct part of record_route >>>> which in DB consists from 2 parts. >>>> >>>> So at line 219 and later uac module cannot restore uris. >>>> >>>> Thank you. >>>> -- >>>> Best regards, >>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>> >>>> >>>> 2017-04-26 17:08 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>> Thanks for workaround. >>>>> >>>>> But I will wait for you solution ) >>>>> >>>>> I ready for testing ) >>>>> >>>>> Thank you Daniel for your work! >>>>> >>>>> -- >>>>> Best regards, >>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>> >>>>> >>>>> 2017-04-26 16:57 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>>>> Hello, >>>>>> >>>>>> >>>>>> On 26.04.17 14:53, Sergey Basov wrote: >>>>>>> Hi All. >>>>>>> >>>>>>> I have just try to test topos with GW which requires PRACK. >>>>>>> >>>>>>> As you can see UA at packet 21 send PRACK to topos contact, but after >>>>>>> topos, on other kamailio side in PRACK request line present not >>>>>>> kontact but record-route header. >>>>>>> >>>>>>> Can you fix it? >>>>>>> >>>>>>> >>>>>> probably needs to look into the code. If you need a quick workaround, >>>>>> try to remove Supported header from INVITE so the callee should no >>>>>> longer Require 100rel. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla >>>>>> www.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>>> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >>>>>> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>

2 9

acces $dlg_var in failure_route
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm trying to access dialog vars in failure_route with no success. Dialog exists but vars are null and I'can modify them either. Is there any restriction? if ($dlg_var(v1) == 0){ $dlg_var(v1) = "1"; if (is_known_dlg()){ xlog("L_NOTICE", "negative_reply $ci $T_rpl($si) $dlg_var(v2)"); $dlg_var(v2) = $T_rpl($si); } } Log Apr 26 12:36:26 drwsip02 /usr/local/sbin/kamailio[27223]: NOTICE: <script>: negative_reply 6daf5e8d-f47972c1(a)172.31.4.21 * <null>*

2 5

Oracle and package function
by Diego Nadares 03 May '17

03 May '17

Hi list, I compiled db_unixodbc and sqlops to connect with oracle db. I have a function that has dml inside. Because of this I can't just execute 'select pkg.function(a, b) from dual'. I'm trying with BEGIN / END. The row is inserted but always are log errors: Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:60]: db_unixodbc_get_columns(): no columns returned from the query Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:261]: db_unixodbc_convert_result(): getting column names failed Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [dbase.c:211]: db_unixodbc_store_result(): failed to convert result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [DECLARE ret CURSOR] My query: sql_query("cb","DECLARE ret CURSOR; BEGIN ret := pkg_partitioning.insert(True,'1'); END;"); I tried with other functions and no matter what type returns the oracle function; always logs errors. It's like unixodbc is waiting for a result with rows. Any hint? Thanks in advance, Diego

2 2

Kamailio 4.4 NAT issue, nathelper I can not get to work
by Jason Davies 03 May '17

03 May '17

Hello I have my servers behind a NAT, with servers that make a call with bypass media / direct media all works fine. But when my media servers need to process audio I'm having issues. Looking at the SDP my local IP of the server is in there. I changed my config to add the nethelper.so and add the below: # ----- NAT Helper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #modparam("usrloc", "nat_bflag", 6) #!endif In my route I added: if (search("User-Agent: FS Box") {fix_nated_sdp("3");};

1 0

TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 03 May '17

03 May '17

Hello fellows, after some experimentation with various tools, I come out with a little shell tool that maybe can be useful to you too. It can only work with non-forward secrecy ciphers, obviously, and only if is started before the client do the initial TLS handshake (eg, just restart the client). Forward secrecy cannot be decrypted after fact, so don't waste effort. An example of ciphers that can be decrypted are the "AES256-SHA" openssl cipher group. You can use ssldump to check what cipher is used by serverhello. Enjoy, make it better, and share it :) #!/bin/bash # brought to you by Giovanni Maruzzelli # SERVERIP="192.168.1.150" SERVERPORT="5061" PRIVKEY="/etc/certs/privkey.pem" STDERR2DEVNULL=" 2>/dev/null " REGEX="notyet" if [ -z "$1" ]; then REGEX="\\\.*" else REGEX="$1" fi FILTER="ssl.app_data and sip matches" FILTER2="$FILTER \"$REGEX\"" FILTER3="'$FILTER2'" ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u 's/^[0-9]*$/\n==&==============================/g'" echo "" echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and try again" echo "" echo "NB: remember to quote and escape match patterns, using triple slash" echo " eg, for matching 1010(a)pbx.example.com, use \"1010(a)pbx.example.com \"" echo " eg, for matching anything, use \"\\\\\\.*\"" echo " eg, for matching *98, use \"\\\\\\*98\"" echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" echo "" case "$1" in -help|--help|?) exit 0 ;; *) echo "THIS TIME WE'RE DOING:" echo "tshark $ARGUMENT" echo "" bash -c "tshark $ARGUMENT" ;; esac -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 3

Memory leak when reloading htable using db_cluster
by Kristian F. Høgh 03 May '17

03 May '17

Hi list, I use current git master. When I reload a htable using "kamcmd htable.reload htable1", the "ctl handler" process leaks 384 bytes of pkg memory If I use a direct mysql connection without db_cluster, the is no memory leak modparam("db_cluster", "connection", "con1=>mysql://user:pass@ip.addr/database") modparam("db_cluster", "connection", "con2=>mysql://user:pass@ip.addr2/database") modparam("db_cluster", "cluster", "cls1=>con1=9s9s;con2=8s8s") modparam("htable", "db_url", "cluster://cls1") # modparam("htable", "db_url", "mysql://user:pass@ip.addr/database" modparam("htable", "htable", "htable1=>size=8;autoexpire=0;dbtable=htable1;") 16(10429) ALERT: qm_status: 6513. N address=0x7f4e65e01720 frag=0x7f4e65e016e8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from db_cluster: dbcl_api.c: db_cluster_init(294) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6514. N address=0x7f4e65e01808 frag=0x7f4e65e017d0 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6515. N address=0x7f4e65e018f0 frag=0x7f4e65e018b8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Regards, Kristian Høgh Uni-tel A/S

2 8

SIP over WebSockets
by Nicolas Pace 02 May '17

02 May '17

Hi everyone, As I shared with you before, We are working on a mobile app to connect the Community Cellular Networks (that uses 2G Cellular Networks) with the rest of the world via SIP, and we are considering using Kamailio for this. We choosed to build a Progresive Web App (PWA) through WebSockets to reduce the friction involved in using a new tool. My question is, is there a reference config to use a web client through WebSockets? I don't have a clue on how to get from the stock config: https://gist.github.com/nicopace/9d44bf813fe309df69ff4cd0affe8324#file- _etc_kamailio_kamailio-conf to a fully functional WebSocket-enabled config. I've already read the WebSocket module documentation, but it is hard to understand as it assumes you already know how Kamailio works... too many moving parts :( I tried with https://github.com/caruizdiaz/kamailio-ws but it is giving me some errors: May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/modules.cfg, line 48, column 12-25: failed to load module May 02 13:37:13 ubuntu kamailio[2329]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command ws_handle_handshake (params 0) May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 143, column 41: unknown command, missing loadmodule? Which relates with: https://github.com/caruizdiaz/kamailio-ws/blob/master/modules.cfg#L48 and https://github.com/caruizdiaz/kamailio-ws/blob/master/kamailio-ws.cfg#L 143 What I'm trying to accomplish is to have a PWA that sends SIMPLE messages and CALLS to and from SIP Users (the 2G users are behind a FreeSwitch, so it could be done). Just to give you some context, we will also add Push Notifications as 'ringing' or 'meesage recieved' alert, and that would be all (we also don't know how to do it). Well... hope to hear from you guys! Thanks!

2 2

TCP read/write buffer size on per connection or per module basis
by M S 02 May '17

02 May '17

Hi, I have a kamailio v4.4.5 deployment that serves TCP clients for SIP and XCAP requests (without SIP presence). The kamailio's core TCP parameters allow us to set TCP read and write buffer sizes which are set globally for all TCP connections. However, i noticed that in general SIP over TCP request is much smaller (<8KB for audio + video SIP INVITE request) then XCAP requests (say to support up to 500 contacts we need TCP buffers =~ 64KB). So, if I set global TCP buffer size according to XCAP service needs (e.g. 64KB), they waste too much RAM for SIP connections (assuming max SIP packet is 8KB, there is 56KB buffer space wasted per connection). On the other hand, if i set global TCP buffer sizes to SIP service needs (e.g. 8KB) then it is insufficient for XCAP (only 70-80 contacts would fix in XCAP list). Therefore, i am looking for a way to allow TCP buffer size settings per connection OR per module basis. Is it possible? If so then how? Thank you.

2 2

Retrieving the custom header parameter value from contact header
by vivek＠advaitamtech.com 02 May '17

02 May '17

Hi, Am trying to extract a custom parameter from the contact header. Below it the example of contact header, Eg: Contact: <sip:fayazat@XXX.XXX.XXX.XXX:5060;tgrp=tg_0615_prsw1> I want to extract the parameter value of "tgrp" Tried using @hf_value.contact.p.tgrp . But it is giving me complete contact header. I want only "tgrp" paramter value "tg_0615_prsw1". Anyone , any sugeestions? Thanks, Vivek.

3 3

TCP port re-using when doing HTTP transfer
by Ramachandran, Agalya (Contractor) 02 May '17

02 May '17

Hi Giacomo, I am using http_async_client module to do http transfer. Currently am testing the HTTP service with a mock server. If I send the HTTP response immediately for the POST request, then the TCP connections are reused from kamailio. In the mock server if I send the response with a delay of 1 sec then, TCP port are opened and closed for every HTTP transfer. Is there a way or any configuration to limit the number of TCP port opening for the HTTP transfers? Please clarify me. Regards, Agalya

2 1

Kamailio IMS - Handling 2 SIP transports in an IMS client
by Haggerty, Paul (phaggert) 01 May '17

01 May '17

Hi, Our client registers with the IMS core using UDP but uses TCP when the packet size is over 1300 bytes. When the client sends the TCP reques, the PCSCF replies with 403 Forbidden "you need to reg with the SCSCF" We've traced this back to the code in the PCSCF and it looks like it is checking the via headers and the received ip/port combinations for a match. The actual contact header looks like it matches the registered client. Question is what do we need to do in the client to ensure that the PCSCF recognizes the request as one from a registered client? We've tried the config options is_registered_fallback2ip , ignore_contact_rxport_check and they don't seem to be used. I'm attaching a network trace and included here are some debug messages from the PCSCF. In the network trace the client is 10.8.0.6. PCSCF Trace messages ------------------------------------------- 34207) DEBUG: ims_registrar_pcscf [service_routes.c:208]: getContactP(): This is a request - using first via to find contact 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:216]: getContactP(): searching for contact with host:port:proto contact [2://10.8.0.6:49946] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 0 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) INFO: ims_registrar_pcscf [service_routes.c:258]: getContactP(): Contact not found based on Contact-header, trying IP/Port/Proto 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 1 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:263]: getContactP(): No entry in usrloc for 10.8.0.6:49946 (Proto 2) found! 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:300]: getContactP(): Asserted identity not set10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=975 a=26 n=xlog 10(34207) WARNING: <script>: 2.1 10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=976 a=26 n=send_reply Appreciate any help we can get, - Paul

1 0

handle_uri_alias persists through failure_route
by David Wilson 01 May '17

01 May '17

Hello All, We use registrar servers (call them A,B) for TLS connections from clients and a central routing proxy (P) that authenticates and manages all call routing. On INVITE, P uses lookup(“location”) and handle_uri_alias() to route the INVITE to A or B depending on the AOR from REGISTER. In failure_route I’m trying to send INVITE to a voicemail URI at host (V), using alias_db_lookup to retrieve it and sending with t_relay(). However, despite showing V as the host in the URI, the message is sent to A (or B) as if the handle_uri_alias is still in effect. I’ve tried calling revert_uri() before the alias_db_lookup(), but this doesn’t seem to help. Is there a way to reset the uri_alias? The failure_route (simplified) looks like this: failure_route[1] { if (method == "INVITE") { revert_uri(); if (alias_db_lookup("dbaliases")) { t_relay(); } } } I have also tried using t_relay(“$rd”, “$rp”) after confirming (by logging) that the uri contains the desired values, but kamailio failed to start, with this error logged: ERROR: tm [tm.c:672]: fixup_hostport2proxy(): TM module:fixup_hostport2proxy: bad port number <$rp> Any suggestions? Best regards, Dave.

3 3

Cannot get service to start
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi Apologies, I was a bit hasty in my previous email, I forgot to run the Chown command I still can’t get the service to start and I’m getting the below error. I’ve gone through all the steps multiple times, please help. /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceFailed to start kamailio.service: Unit kamailio.service failed to load: No such file or directory. failed! Kind regards Stephen tyers

2 1

Date Time with microsec
by Diego Nadares 28 Apr '17

28 Apr '17

Hi Guys, Is to convert or obtain $TV(s) - seconds (cached at first call per sip message) $TV(u) - microseconds (cached at first call per sip message) to datetime.microsecs? I'm trying to save 180/183 datetime (cached) like YYYY-MM-DD HH:mm:SS.microsecs $dlg_var(ring_time) = $timef(%Y-%m-%d %H:%M:%S); <--this is current time, not cached $dlg_var(ring_time_micro) = $TV(u); <-- this is cached us Thanks in advance! Diego.

2 3

htable Arrays and dbmode 1
by Sebastian Damm 28 Apr '17

28 Apr '17

Hi, I'm trying to use a htable array. For that, I have inserted some values in the database (in pseudo CSV): key_name; key_type; value_type; key_value '1234567'; 1; 0; 'foo' '1234567', 1, 0, 'bar' '1234567', 1, 0, 'baz' insert into htable (key_name , key_type , value_type , key_value ) values ('1234567', 1, 0, 'foo'),('1234567', 1, 0, 'bar'),('1234567', 1, 0, 'baz'); Now after just starting Kamailio, reading those values, from db and then stopping Kamailio again, the table content looks like this: key_name; key_type; value_type; key_value '1234567[2]'; 0; 0; 'baz' '1234567[0]', 0, 0, 'foo' '1234567[1]', 0, 0, 'bar' '1234567::size', 0, 1, 3 Is this how it is supposed to work? When re-reading those entries, the htable contents look the same as during the first start, but in the database it is a bit harder to read. And while I'm at it: Is there a way to regularly write back the current htable contents to the database? Or even to sync every write directly into the database? Thanks in advance Sebastian

1 0

Not able to replace route_uri in contact header uri
by Narayan P 28 Apr '17

28 Apr '17

Hi All, I am using pjsip client and kamailiio,both running on different servers.I want to replace the uri on which pjsip is running with the uri on which kamailio is running in contact header. I put some logs on route[REGISTRAR] and able to see the $route_uri as kamailio server uri($avp(s:fs) and $hdr(Contact) as pjsip server uri ($avp(s:contact).But I am not able to replace the uri with avp_subst() function. Can anybody help me how to replace the uri or with any any other function. Below I have put the snippet of config file. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } xlog("Narayan: before sending contact\n"); $avp(s:contact) = $hdr(Contact); $avp(s:fs) = $route_uri; xlog("Narayan: Forced socket is $avp(s:fs)\n"); xlog("Narayan: contact header is $avp(s:contact)\n"); avp_subst("$avp(s:contact)/avp(s:contac)/g", "/(.*)(a)(.*)/$route_uri/"); #avp_pushto("$ru/domain","$fd"); #rewritehostport("185.122.206.62:5060"); t_on_reply("MANAGE_REPLY"); t_relay(); #if (!save("location")) { # sl_reply_error(); #} exit; } Any suggestion will be highly appreciated. Thanks, Narayan

3 11

Problem with installing Kamailio 5
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi, I’ve hit a road block when following URL http://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ I’m very new to open source and linux so please bare with me. I’ve reached step Kamailio start and I get the below error. Please help, I’ve think I’ve followed all the proceeding commands and created the databases. root@kamailio:/var/run# /etc/init.d/kamailio start -bash: /etc/init.d/kamailio: Permission denied Many thanks Steve

1 0

An Potential Issue in Kamailio Installation Guide
by Edward Lu 28 Apr '17

28 Apr '17

Dear Sir, This is Edward, nice to meet you. When I follow the installation guide of Kamailio 5.0.x https://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ to install Kamailio, I met an issue, it report "unit kamailio.service not found" when I issue command /etc/init.d/kamailio start. After google, I found one step was missed: to copy kamailio.service to /etc/system.d/system. Except above issue, all others are all okay for my installation on Ubuntu 16.04. Thanks & Regards, Edward

1 0

How do we send "MSRP Message" from config
by Pranathi Venkatayogi 28 Apr '17

28 Apr '17

While implementing Kamailio to hold customer call, I would to send "Pls hold on looking for an agent" MSRP message from Kamailio config. Is it doable? >From the module, we can only send msrp_reply and msrp_relay... Please advise. Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

2 1

Re: [SR-Users] SIGCHLD causes termination, but it appears that no child process has actually stopped.
by Cody Herzog 27 Apr '17

27 Apr '17

Thanks for the quick reply. Pull request #1101 created: https://github.com/kamailio/kamailio/pull/1101

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users